update for 1.0.10

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

.gitattributes

@@ -1 +0,0 @@
-*.pnm -diff -text

.gitignore

@@ -3,19 +3,12 @@
 *.d
 *.def
 *.dll
-*.dylib
 *.exe
-*.exp
-*.gcda
-*.gcno
 *.h.c
-*.ilk
 *.lib
 *.pc
-*.pdb
 *.so
 *.so.*
-*.swp
 *.ver
 *-example
 *-test
@@ -27,63 +20,41 @@
 /ffprobe
 /ffserver
 /config.*
-/coverage.info
+/version.h
 /doc/*.1
-/doc/*.3
 /doc/*.html
 /doc/*.pod
-/doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/doxy/html/
-/doc/examples/avio_reading
 /doc/examples/decoding_encoding
-/doc/examples/demuxing_decoding
-/doc/examples/extract_mvs
-/doc/examples/filter_audio
+/doc/examples/demuxing
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
-/doc/examples/pc-uninstalled
-/doc/examples/remuxing
-/doc/examples/resampling_audio
 /doc/examples/scaling_video
-/doc/examples/transcode_aac
-/doc/examples/transcoding
 /doc/fate.txt
 /doc/print_options
-/lcov/
+/doxy/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
-/libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
-/tests/pixfmts.mak
 /tests/rotozoom
-/tests/test_copy.ffmeta
 /tests/tiny_psnr
-/tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
-/tools/crypto_bench
 /tools/cws2fws
-/tools/fourcc2pixfmt
-/tools/ffescape
 /tools/ffeval
-/tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
 /tools/trasher
-/tools/seek_print
-/tools/uncoded_frame
-/tools/zmqsend

CREDITS

@@ -1,6 +1,55 @@
-See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
-get the names of people who have contributed to FFmpeg.
+This file contains the names of some of the people who have contributed to
+FFmpeg. The names are sorted alphabetically by last name.  As this file is
+currently quite outdated and git serves as a much better tool for determining
+authorship, it remains here for historical reasons only.
 
-To check the log, you can type the command "git log" in the FFmpeg
-source directory, or browse the online repository at
-http://source.ffmpeg.org.
+Dénes Balatoni
+Michel Bardiaux
+Fabrice Bellard
+Patrice Bensoussan
+Alex Beregszaszi
+BERO
+Thilo Borgmann
+Mario Brito
+Ronald Bultje
+Alex Converse
+Maarten Daniels
+Reimar Doeffinger
+Tim Ferguson
+Brian Foley
+Arpad Gereoffy
+Philip Gladstone
+Vladimir Gneushev
+Roine Gustafsson
+David Hammerton
+Wolfgang Hesseler
+Marc Hoffman
+Falk Hueffner
+Aurélien Jacobs
+Steven Johnson
+Zdenek Kabelac
+Robin Kay
+Todd Kirby
+Nick Kurshev
+Benjamin Larsson
+Loïc Le Loarer
+Daniel Maas
+Mike Melanson
+Loren Merritt
+Jeff Muizelaar
+Michael Niedermayer
+François Revol
+Peter Ross
+Måns Rullgård
+Stefano Sabatini
+Roman Shaposhnik
+Oded Shimon
+Dieter Shirley
+Konstantin Shishkov
+Juan J. Sierralta
+Ewald Snel
+Sascha Sommer
+Leon van Stuivenberg
+Roberto Togni
+Lionel Ulmer
+Reynaldo Verdejo

Changelog

@@ -1,845 +1,10 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 2.5.11
-- MAINTAINERS: remove unmaintained releases
-- avcodec/jpeg2000dec: More completely check cdef
-- avutil/opt: check for and handle errors in av_opt_set_dict2()
-- avcodec/flacenc: fix calculation of bits required in case of custom sample rate
-- avformat: Document urls a bit
-- avformat/libquvi: Set default demuxer and protocol limitations
-- avformat/concat: Check protocol prefix
-- doc/demuxers: Document enable_drefs and use_absolute_path
-- avcodec/mjpegdec: Check for end for both bytes in unescaping
-- avcodec/mpegvideo_enc: Check for integer overflow in ff_mpv_reallocate_putbitbuffer()
-- avformat/avformat: Replace some references to filenames by urls
-- avcodec/wmaenc: Check ff_wma_init() for failure
-- avcodec/mpeg12enc: Move high resolution thread check to before initializing threads
-- avformat/avio: Limit url option parsing to the documented cases
-- avcodec/ass_split: Fix null pointer dereference in ff_ass_style_get()
-- mov: Add an option to toggle dref opening
-- avcodec/gif: Fix lzw buffer size
-- avcodec/put_bits: Assert buf_ptr in flush_put_bits()
-- avcodec/tiff: Check subsample & rps values more completely
-- swscale/swscale: Add some sanity checks for srcSlice* parameters
-- swscale/x86/rgb2rgb_template: Fix planar2x() for short width
-- swscale/swscale_unscaled: Fix odd height inputs for bayer_to_yv12_wrapper()
-- swscale/swscale_unscaled: Fix odd height inputs for bayer_to_rgb24_wrapper()
-- avcodec/aacenc: Check both channels for finiteness
-- swscale/swscale-test: Fix slice height in random reference data creation.
-- dca: fix misaligned access in avpriv_dca_convert_bitstream
-- brstm: fix missing closing brace
-- brstm: also allocate b->table in read_packet
-- brstm: make sure an ADPC chunk was read for adpcm_thp
-- vorbisdec: reject rangebits 0 with non-0 partitions
-- vorbisdec: reject channel mapping with less than two channels
-- ffmdec: reset packet_end in case of failure
-- avformat/ipmovie: put video decoding_map_size into packet and use it in decoder
-- avcodec/samidec: make sure to properly restore parsing context after a tag
-
-version 2.5.10
-- configure: bump copyright year to 2016
-- avformat/hls: Even stricter URL checks
-- avformat/hls: More strict url checks
-- swscale/utils: Detect and skip unneeded sws_setColorspaceDetails() calls
-- swscale/yuv2rgb: Increase YUV2RGB table headroom
-- swscale/yuv2rgb: Factor YUVRGB_TABLE_LUMA_HEADROOM out
-- avformat/hls: forbid all protocols except http(s) & file
-- avformat/aviobuf: Fix end check in put_str16()
-- avformat/asfenc: Check pts
-- avcodec/mpeg4video: Check time_incr
-- avcodec/wavpackenc: Check the number of channels
-- avcodec/wavpackenc: Headers are per channel
-- avcodec/dvdec: Fix "left shift of negative value -254"
-- avcodec/mjpegdec: Fix negative shift
-- avcodec/mss2: Check for repeat overflow
-- avformat: Add integer fps from 31 to 60 to get_std_framerate()
-- avcodec/mpegvideo_enc: Clip bits_per_raw_sample within valid range
-- avfilter/vf_scale: set proper out frame color range
-- avcodec/motion_est: Fix mv_penalty table size
-- avcodec/h264_slice: Fix integer overflow in implicit weight computation
-- swscale/utils: Use normal bilinear scaler if fast cannot be used due to tiny dimensions
-- avcodec/put_bits: Always check buffer end before writing
-- mjpegdec: extend check for incompatible values of s->rgb and s->ls
-- swscale/utils: Fix intermediate format for cascaded alpha downscaling
-- avcodec/h264_refs: Fix long_idx check
-- avfilter/vf_mpdecimate: Add missing emms_c()
-- avformat/mxfenc: Do not crash if there is no packet in the first stream
-- avformat/utils: estimate_timings_from_pts - increase retry counter, fixes invalid duration for ts files with hevc codec
-- avformat/matroskaenc: Check codecdelay before use
-- avutil/mathematics: Fix division by 0
-- x86/float_dsp: zero extend offset from ff_scalarproduct_float_sse
-- avcodec/mpeg4videodec: also for empty partitioned slices
-- nuv: sanitize negative fps rate
-- rawdec: only exempt BIT0 with need_copy from buffer sanity check
-- mlvdec: check that index_entries exist
-- nutdec: reject negative value_len in read_sm_data
-- xwddec: prevent overflow of lsize * avctx->height
-- nutdec: only copy the header if it exists
-- exr: fix out of bounds read in get_code
-- on2avc: limit number of bits to 30 in get_egolomb
-- sonic: make sure num_taps * channels is not larger than frame_size
-- opus_silk: fix typo causing overflow in silk_stabilize_lsf
-- ffm: reject invalid codec_id and codec_type
-- aaccoder: prevent crash of anmr coder
-- ffmdec: reject zero-sized chunks
-- swscale/x86/rgb2rgb_template: Fallback to mmx in interleaveBytes() if the alignment is insufficient for SSE*
-- swscale/x86/rgb2rgb_template: Do not crash on misaligend stride
-
-
-version 2.5.9
-- avcodec/hevc: Check max ctb addresses for WPP
-- avcodec/vp3: ensure header is parsed successfully before tables
-- avcodec/jpeg2000dec: Check bpno in decode_cblk()
-- avcodec/pgssubdec: Fix left shift of 255 by 24 places cannot be represented in type int
-- swscale/utils: Fix for runtime error: left shift of negative value -1
-- avcodec/hevc: Fix integer overflow of entry_point_offset
-- avcodec/dirac_parser: Check that there is a previous PU before accessing it
-- avcodec/dirac_parser: Add basic validity checks for next_pu_offset and prev_pu_offset
-- avcodec/dirac_parser: Fix potential overflows in pointer checks
-- avcodec/wmaprodec: Check bits per sample to be within the range not causing integer overflows
-- avcodec/wmaprodec: Fix overflow of cutoff
-- avformat/smacker: fix integer overflow with pts_inc
-- avcodec/vp3: Fix "runtime error: left shift of negative value"
-- mpegencts: Fix overflow in cbr mode period calculations
-- avutil/timecode: Fix fps check
-- avutil/mathematics: return INT64_MIN (=AV_NOPTS_VALUE) from av_rescale_rnd() for overflows
-- avcodec/apedec: Check length in long_filter_high_3800()
-- avcodec/vp3: always set pix_fmt in theora_decode_header()
-- avcodec/mpeg4videodec: Check available data before reading custom matrix
-- avutil/mathematics: Do not treat INT64_MIN as positive in av_rescale_rnd
-- avutil/integer: Fix av_mod_i() with negative dividend
-- avformat/dump: Fix integer overflow in av_dump_format()
-- avcodec/utils: Clear dimensions in ff_get_buffer() on failure
-- avcodec/utils: Use 64bit for aspect ratio calculation in avcodec_string()
-- avcodec/vp3: Clear context on reinitialization failure
-- avcodec/hevc: allocate entries unconditionally
-- avcodec/hevc_cabac: Fix multiple integer overflows
-- avcodec/jpeg2000dwt: Check ndeclevels before calling dwt_encode*()
-- avcodec/jpeg2000dwt: Check ndeclevels before calling dwt_decode*()
-- avcodec/hevc: Check entry_point_offsets
-- avcodec/cabac: Check initial cabac decoder state
-- avcodec/cabac_functions: Fix "left shift of negative value -31767"
-- avcodec/ffv1dec: Clear quant_table_count if its invalid
-- avcodec/ffv1dec: Print an error if the quant table count is invalid
-- doc/filters/drawtext: fix centering example
-- avcodec/h264_slice: Limit max_contexts when slice_context_count is initialized
-- avcodec/vp8: Do not use num_coeff_partitions in thread/buffer setup
-- rtmpcrypt: Do the xtea decryption in little endian mode
-- avformat/matroskadec: Check subtitle stream before dereferencing
-- avformat/utils: Do not init parser if probing is unfinished
-- avcodec/jpeg2000dec: Fix potential integer overflow with tile dimensions
-- avcodec/jpeg2000dec: Check SIZ dimensions to be within the supported range
-- avcodec/jpeg2000: Check comp coords to be within the supported size
-- avcodec/jpeg2000: Use av_image_check_size() in ff_jpeg2000_init_component()
-- avcodec/wmaprodec: Check for overread in decode_packet()
-- avcodec/smacker: Check that the data size is a multiple of a sample vector
-- avcodec/takdec: Skip last p2 sample (which is unused)
-- avcodec/dxtory: Fix input size check in dxtory_decode_v1_410()
-- avcodec/dxtory: Fix input size check in dxtory_decode_v1_420()
-- avcodec/error_resilience: avoid accessing previous or next frames tables beyond height
-- avcodec/dpx: Move need_align to act per line
-- avcodec/flashsv: Check size before updating it
-- avcodec/ivi: Check image dimensions
-- avcodec/utils: Better check for channels in av_get_audio_frame_duration()
-- avcodec/jpeg2000dec: Check for duplicate SIZ marker
-- avcodec/jpeg2000dec: Clip all tile coordinates
-- avcodec/microdvddec: Check for string end in 'P' case
-- avcodec/dirac_parser: Fix undefined memcpy() use
-- avformat/xmv: Discard remainder of packet on error
-- avformat/xmv: factor return check out of if/else
-- libavutil/channel_layout: Check strtol*() for failure
-- avcodec/ffv1dec: Check for 0 quant tables
-- avcodec/mjpegdec: Reinitialize IDCT on BPP changes
-- avcodec/mjpegdec: Check index in ljpeg_decode_yuv_scan() before using it
-- avutil/file_open: avoid file handle inheritance on Windows
-- opusdec: Don't run vector_fmul_scalar on zero length arrays
-- avcodec/ffv1: Initialize vlc_state on allocation
-- avcodec/ffv1dec: update progress in case of broken pointer chains
-- avcodec/ffv1dec: Clear slice coordinates if they are invalid or slice header decoding fails for other reasons
-- avformat/httpauth: Add space after commas in HTTP/RTSP auth header
-- avcodec/x86/sbrdsp: Fix using uninitialized upper 32bit of noise
-- avcodec/ffv1dec: Fix off by 1 error in quant_table_count check
-- avcodec/ffv1dec: Explicitly check read_quant_table() return value
-- avcodec/rangecoder: Check e
-- lavf/webvttenc: Require webvtt file to contain exactly one WebVTT stream.
-- avcodec/mjpegdec: Fix decoding RGBA RCT LJPEG
-- avfilter/af_asyncts: use llabs for int64_t
-- avcodec/g2meet: Also clear tile dimensions on header_fail
-- avcodec/g2meet: Fix potential overflow in tile dimensions check
-- avcodec/svq1dec: Check init_get_bits8() for failure
-- avcodec/tta: Check init_get_bits8() for failure
-- swresample/swresample: Fix integer overflow in seed calculation
-- avformat/mov: Fix integer overflow in FFABS
-- avutil/common: Add FFNABS()
-- avutil/common: Document FFABS() corner case
-- avformat/dump: Fix integer overflow in aspect ratio calculation
-- avcodec/truemotion1: Check for even width
-- avcodec/mpeg12dec: Set dimensions in mpeg1_decode_sequence() only in absence of errors
-- avcodec/libopusenc: Fix infinite loop on flushing after 0 input
-- avformat/hevc: Check num_long_term_ref_pics_sps to avoid potentially long loops
-- avformat/hevc: Fix parsing errors
-- ffmpeg: Use correct codec_id for av_parser_change() check
-- ffmpeg: Check av_parser_change() for failure
-- ffmpeg: Check for RAWVIDEO and do not relay only on AVFMT_RAWPICTURE
-- ffmpeg: check avpicture_fill() return value
-- avformat/mux: Update sidedata in ff_write_chained()
-- avcodec/flashsvenc: Correct max dimension in error message
-- avcodec/svq1enc: Check dimensions
-- avcodec/dcaenc: clear bitstream end
-- libavcodec/aacdec_template: Use init_get_bits8() in aac_decode_frame()
-- mxfdec: check edit_rate also for physical_track
-- mpegvideo: clear overread in clear_context
-- dvdsubdec: validate offset2 similar to offset1
-- avcodec/takdec: Use memove, avoid undefined memcpy() use
-- jvdec: avoid unsigned overflow in comparison
-- avcodec/mpeg12dec: Do not call show_bits() with invalid bits
-- riffdec: prevent negative bit rate
-- Merge commit 'd80811c94e068085aab797f9ba35790529126f85'
-- imc: use correct position for flcoeffs2 calculation
-- wavpack: limit extra_bits to 32 and use get_bits_long
-- wavpack: use get_bits_long to read up to 32 bits
-- nutdec: check maxpos in read_sm_data before returning success
-- s302m: fix arithmetic exception
-- avcodec/s302m: Only set the sample rate when some data is output
-- vp9: add support for resolution changes in inter frames.
-- alsdec: limit avctx->bits_per_raw_sample to 32
-- vp9: avoid infinite loop with broken files
-- videodsp: don't overread edges in vfix3 emu_edge.
-- avcodec/h264_mp4toannexb_bsf: Reorder operations in nal_size check
-- avformat/oggenc: Check segments_count for headers too
-- avformat/avidec: Workaround broken initial frame
-- hevc: properly handle no_rasl_output_flag when removing pictures from the DPB
-- hevc: fix wpp threading deadlock.
-- avcodec/ffv1: separate slice_count from max_slice_count
-- lavf/img2dec: Fix memory leak
-- avcodec/mp3: fix skipping zeros
-- avformat/srtdec: make sure we probe a number
-- avformat/srtdec: more lenient first line probing
-- doc: mention libavcodec can decode Opus natively
-- MAINTAINERS: Remove myself as leader
-
-
-version 2.5.8
-- snow: remove an obsolete av_assert2
-- huffyuvdec: validate image size
-- vc1dec: use get_bits_long and limit the read bits to 32
-- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
-- libshine: fix support for shine 3.0
-- avidec: check for valid bit_rate range
-- avformat/nut: support WavPack
-- avcodec/diracdec: Check slices malloc and propagate error code
-- avcodec/vp8: Check buffer size in vp8_decode_frame_header()
-- avcodec/vp8: Fix null pointer dereference in ff_vp8_decode_free()
-- avcodec/diracdec: Check for hpel_base allocation failure
-- avcodec/rv34: Clear pointers in ff_rv34_decode_init_thread_copy()
-- avfilter/af_aresample: Check ff_all_* for allocation failures
-- avcodec/pthread_frame: clear priv_data, avoid stale pointer in error case
-- swscale/utils: Clear pix buffers
-- avutil/fifo: Fix the case where func() returns less bytes than requested in av_fifo_generic_write()
-- avformat/mov: Fix deallocation when MOVStreamContext failed to allocate
-- ffmpeg: Fix crash with ost->last_frame allocation failure
-- ffmpeg: Fix cleanup with ost = NULL
-- avcodec/pthread_frame: check avctx on deallocation
-- avcodec/sanm: Reset sizes in destroy_buffers()
-- avcodec/alac: Clear pointers in allocate_buffers()
-- bytestream2: set the reader to the end when reading more than available
-- avcodec/utils: use a minimum 32pixel width in  avcodec_align_dimensions2() for H.264
-- avcodec/mpegvideo: Clear pointers in ff_mpv_common_init()
-- oggparsedirac: check return value of init_get_bits
-- wmalosslessdec: reset frame->nb_samples on packet loss
-- wmalosslessdec: avoid reading 0 bits with get_bits
-- avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2()
-- avcodec/aacsbr: Assert that bs_num_env is positive
-- avcodec/aacsbr: check that the element type matches before applying SBR
-- avcodec/h264_slice: Use w/h from the AVFrame instead of mb_w/h
-- vp9/update_prob: prevent out of bounds table read
-- avfilter/vf_transpose: Fix rounding error
-- avcodec/pngdec: Check values before updating context in decode_fctl_chunk()
-- avcodec/pngdec: Require a IHDR chunk before fctl
-- avcodec/pngdec: Only allow one IHDR chunk
-- wmavoice: limit wmavoice_decode_packet return value to packet size
-- swscale/swscale_unscaled: Fix rounding difference with RGBA output between little and big endian
-- ffmpeg: Do not use the data/size of a bitstream filter after failure
-- swscale/x86/rgb2rgb_template: fix signedness of v in shuffle_bytes_2103_{mmx,mmxext}
-- swscale/x86/rgb2rgb_template: add missing xmm clobbers
-- vda: unlock the pixel buffer base address.
-- swscale/rgb2rgb_template: Fix signedness of v in shuffle_bytes_2103_c()
-- swscale/rgb2rgb_template: Implement shuffle_bytes_0321_c and fix shuffle_bytes_2103_c on BE
-- swscale/rgb2rgb_template: Disable shuffle_bytes_2103_c on big endian
-- swr: Remember previously set int_sample_format from user
-- matroskadec: check audio sample rate
-- matroskadec: validate audio channels and bitdepth
-- avcodec/dpxenc: implement write16/32 as functions
-- postproc: fix unaligned access
-- ffmpeg: Free last_frame instead of just unref
-- avio: fix potential crashes when combining ffio_ensure_seekback + crc
-- h264: er: Copy from the previous reference only if compatible
-- sonic: set avctx->channels in sonic_decode_init
-- vp8: change mv_{min,max}.{x,y} type to int
-- vp9: change type of tile_size from unsigned to int64_t
-- arm: only enable setend on ARMv6
-- libopenjpegdec: check existence of image component data
-- mov: abort on EOF in ff_mov_read_chan
-- ffmpeg_opt: Check for localtime() failure
-- avformat: Fix bug in parse_rps for HEVC.
-- takdec: ensure chan2 is a valid channel index
-- avcodec/h264_slice: Use AVFrame diemensions for grayscale handling
-- avdevice/lavfi: do not rescale AV_NOPTS_VALUE in lavfi_read_packet()
-- libavutil/channel_layout: Correctly return layout when channel specification ends with a trailing 'c'.
-- avcodec/jpeg2000dec: Check that coords match before applying ICT
-- avformat/ffmdec: Check ffio_set_buf_size() return value
-- avcodec/adpcm: Check for overreads
-- avcodec/alsdec: Check for overread
-- avcodec/atrac3plusdec: consume only as many bytes as available
-- libavutil/softfloat: Fix av_normalize1_sf bias.
-- swresample/swresample: Cleanup on init failure.
-- Revert "avformat/rtpenc: check av_packet_get_side_data() return, fix null ptr dereference"
-- avformat/mxfenc: Accept MXF D-10 with 49.999840 Mbit/sec
-- swresample/dither: check memory allocation
-- libopenjpegenc: add NULL check for img before accessing it
-- swresample: Check the return value of resampler->init()
-- h264: Make sure reinit failures mark the context as not initialized
-- ffmpeg_opt: Set the video VBV parameters only for the video stream from -target
-- avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits()
-- avcodec/put_bits: Assert that there is enough space left in skip_put_bytes()
-- avcodec/mpegvideo_enc: Update the buffer size as more slices are merged
-- avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size()
-- avformat/wavdec: Increase dts packet threshold to fix more misdetections
-- avformat/wavdec: Increase probe_packets limit
-- avformat/swfdec: Do not error out on pixel format changes
-- avfilter/x86/vf_hqdn3d: Fix register types
-- avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer()
-- avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out
-- avformat/mov: Mark avio context of decompressed atoms as seekable
-- avcodec/hevc_ps: Only discard overread VPS if a previous is available
-- avcodec/x86/h264_weight: handle weight1=128
-- avcodec/exr: fix crash caused by merge
-
-
-version 2.5.7
-- avformat/nutdec: Fix recovery when immedeately after seeking a failure happens
-- nutdec: fix memleaks on error in nut_read_header
-- rtpenc_jpeg: handle case of picture dimensions not dividing by 8
-- avformat/mov: Fix parsing short loci
-- avcodec/shorten: Fix code depending on signed overflow behavior
-- avcodec/proresdec2: Reset slice_count on deallocation
-- ffmpeg_opt: Fix -timestamp parsing
-- hevc: make avcodec_decode_video2() fail if get_format() fails
-- avcodec/mpeg4audio: add some padding/alignment to MAX_PCE_SIZE
-- swr: fix alignment issue caused by 8ch sse functions
-- libswscale/x86/hscale_fast_bilinear_simd.c: Include BX in the clobber list on x86_64, because it isn't implicitly included when PIC is on.
-- aacdec: don't return frames without data
-- avformat/matroskadec: Cleanup error handling for bz2 & zlib
-- avformat/nutdec: Fix use of uinitialized value
-- tools/graph2dot: use larger data types than int for array/string sizes
-- id3v2: catch avio_read errors in check_tag
-- aacsbr: break infinite loop in sbr_hf_calc_npatches
-- diracdec: avoid overflow of bytes*8 in decode_lowdelay
-- diracdec: prevent overflow in data_unit_size check
-- avidec: avoid infinite loop due to negative ast->sample_size
-- pngdec: don't use AV_PIX_FMT_MONOBLACK for apng
-- avcodec/wavpack: Check L/R values before use to avoid harmless integer overflow and undefined behavior in fate
-- xcbgrab: Validate the capture area
-- xcbgrab: Do not assume the non shm image data is always available
-- avfilter/lavfutils: disable frame threads when decoding a single image
-- nutdec: fix illegal count check in decode_main_header
-- ffmpeg: remove incorrect network deinit
-- OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c
-- apedec: set s->samples only when init_frame_decoder succeeded
-- swscale/ppc/swscale_altivec.c: POWER LE support in yuv2planeX_8() delete macro GET_VF()
-- libvpxenc: only set noise reduction w/vp8
-- tests/fate-run: do not attempt to parse tiny_psnrs output if it failed
-- alac: reject rice_limit 0 if compression is used
-- alsdec: only adapt order for positive max_order
-- alsdec: check sample pointer range in revert_channel_correlation
-- tests: drop bc dependency
-- fate: Include branch information in the payload header
-
-
-version 2.5.6
-- avcodec/atrac3plusdsp: fix on stack alignment
-- ac3: validate end in ff_ac3_bit_alloc_calc_mask
-- aacpsy: avoid psy_band->threshold becoming NaN
-- aasc: return correct buffer size from aasc_decode_frame
-- msrledec: use signed pixel_ptr in msrle_decode_pal4
-- swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07)
-- swresample: Allow reinitialization without ever setting channel counts
-- avcodec/h264: Do not fail with randomly truncated VUIs
-- avcodec/h264_ps: Move truncation check from VUI to SPS
-- avcodec/h264: Be more tolerant to changing pps id between slices
-- avcodec/aacdec: Fix storing state before PCE decode
-- avcodec/h264: reset the counts in the correct context
-- avcodec/h264_slice: Do not reset mb_aff_frame per slice
-- avcodec/h264: finish previous slices before switching to single thread mode
-- avcodec/h264: Fix race between slices where one overwrites data from the next
-- avcodec/h264_refs: Do not set reference to things which do not exist
-- avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode
-- h264: avoid unnecessary calls to get_format
-- avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
-
-
-version 2.5.5:
-- vp9: make above buffer pointer 32-byte aligned.
-- avcodec/dnxhddec: Check that the frame is interlaced before using cur_field
-- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
-- avformat/mov: Check for string truncation in mov_open_dref()
-- avformat/mov: Use sizeof(filename) instead of a literal number
-- eac3dec: fix scaling
-- ac3_fixed: fix computation of spx_noise_blend
-- ac3_fixed: fix out-of-bound read
-- ac3dec_fixed: always use the USE_FIXED=1 variant of the AC3DecodeContext
-- avcodec/012v: redesign main loop
-- avcodec/012v: Check dimensions more completely
-- asfenc: fix leaking asf->index_ptr on error
-- avcodec/options_table: remove extradata_size from the AVOptions table
-- ffmdec: limit the backward seek to the last resync position
-- ffmdec: make sure the time base is valid
-- ffmdec: fix infinite loop at EOF
-- ffmdec: initialize f_cprv, f_stvi and f_stau
-- avformat/rm: limit packet size
-- avcodec/webp: validate the distance prefix code
-- avcodec/rv10: check size of s->mb_width * s->mb_height
-- eamad: check for out of bounds read
-- mdec: check for out of bounds read
-- arm: Suppress tags about used cpu arch and extensions
-- aic: Fix decoding files with odd dimensions
-- avcodec/tiff: move bpp check to after "end:"
-- mxfdec: Fix the error handling for when strftime fails
-- avcodec/opusdec: Fix delayed sample value
-- avcodec/opusdec: Clear out pointers per packet
-- avcodec/utils: Align YUV411 by as much as the other YUV variants
-- vp9: fix segmentation map retention with threading enabled.
-- webp: ensure that each transform is only used once
-- doc/protocols/tcp: fix units of listen_timeout option value, from microseconds to milliseconds
-- fix VP9 packet decoder returning 0 instead of the used data size
-- avformat/flvenc: check that the codec_tag fits in the available bits
-- avcodec/utils: use correct printf specifier in ff_set_sar
-- avutil/imgutils: correctly check for negative SAR components
-- swscale/utils: clear formatConvBuffer on allocation
-- avformat/bit: only accept the g729 codec and 1 channel
-- avformat/bit: check that pkt->size is 10 in write_packet
-- avformat/adxdec: check avctx->channels for invalid values
-- avformat/adxdec: set avctx->channels in adx_read_header
-- Fix buffer_size argument to init_put_bits() in multiple encoders.
-- mips/acelp_filters: fix incorrect register constraint
-- avcodec/hevc_ps: Sanity checks for some log2_* values
-- avcodec/zmbv: Check len before reading in decode_frame()
-- avcodec/h264: Only reinit quant tables if a new PPS is allowed
-- avcodec/snowdec: Fix ref value check
-- swscale/utils: More carefully merge and clear coefficients outside the input
-- avcodec/a64multienc: Assert that the Packet size does not grow
-- avcodec/a64multienc: simplify frame handling code
-- avcodec/a64multienc: fix use of uninitialized values in to_meta_with_crop
-- avcodec/a64multienc: initialize mc_meta_charset to zero
-- avcodec/a64multienc: don't set incorrect packet size
-- avcodec/a64multienc: use av_frame_ref instead of copying the frame
-- avcodec/x86/mlpdsp_init: Simplify mlp_filter_channel_x86()
-- h264: initialize H264Context.avctx in init_thread_copy
-- wtvdec: fix integer overflow resulting in errors with large files
-- avcodec/gif: fix off by one in column offsetting finding
-
-
-version 2.5.4:
-- avcodec/arm/videodsp_armv5te: Fix linking failure with shared libs
-- avcodec/mjpegdec: Skip blocks which are outside the visible area
-- avcodec/h264_slice: ignore SAR changes in slices after the first
-- avcodec/h264_slice: Check picture structure before setting the related fields
-- avcodec/h264_slice: Do not change frame_num after the first slice
-- avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT
-- avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT
-- avcodec/h264: Be more strict on rejecting pps/sps changes
-- avcodec/h264: Be more strict on rejecting pps_id changes
-- avcodec/h264_ps: More completely check the bit depths
-- avformat/thp: Check av_get_packet() for failure not only for partial output
-- swscale/utils: Limit filter shifting so as not to read from prior the array
-- avcodec/mpegvideo_motion: Fix gmc chroma dimensions
-- avcodec/mjpegdec: Check number of components for JPEG-LS
-- avcodec/mjpegdec: Check escape sequence validity
-- avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior
-- avformat/mpc8: fix broken pointer math
-- avformat/mpc8: fix hang with fuzzed file
-- avformat/tta: fix crash with corrupted files
-- avcodec/ppc/idctdsp.c: POWER LE support in idct_add_altivec()
-- swscale/input: fix rgba64 alpha non native
-- swscale/input: Fix alpha of YA16 input
-- libavcodec/ppc/mpegvideoencdsp.c: fix stack smashing in pix_norm1_altivec() and pix_sum_altivec()
-- avformat/rmdec: Check for overflow in ff_rm_read_mdpr_codecdata()
-- avformat/mpeg: do not count PES packets inside PES packets during probing
-- hevc: always clip luma_log2_weight_denom
-- rtpdec_h263_rfc2190: Clear the stored bits if discarding buffered data
-- aacenc: correctly check returned value
-- swscale: check memory allocations
-- opt: check memory allocation
-- avformat/utils: check for malloc failure
-- avcodec/flac_parser: fix handling EOF if no headers are found
-- avfilter/vf_framepack: Check and update frame_rate
-- vp8: improve memory allocation checks
-- configure: enable vsx together with altivec for ppc64el
-- avcodec/hevc: Fix handling of skipped_bytes() reallocation failures
-- qpeg: avoid pointless invalid memcpy()
-
-
-version 2.5.3:
-- vp9: fix parser return values in error case
-- ffmpeg: Clear error message array at init.
-- avcodec/dvdsubdec: fix accessing dangling pointers
-- avcodec/dvdsubdec: error on bitmaps with size 0
-- cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file()
-- mov: Fix negative size calculation in mov_read_default().
-- avformat/mov: fix integer overflow in mov_read_udta_string()
-- mov: Fix overflow and error handling in read_tfra().
-- mov: Avoid overflow with mov_metadata_raw()
-- avcodec/dvdsubdec: fix out of bounds accesses
-- avfilter/vf_sab: fix filtering tiny images
-- avformat/flvdec: Increase string array size
-- avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream
-- avformat/cdxl: Fix integer overflow of image_size
-- libavformat: Build hevc.o when building the RTP muxer
-
-version 2.5.2:
-- avcodec/indeo3: ensure offsets are non negative
-- avcodec/h264: Check *log2_weight_denom
-- avcodec/hevc_ps: Check diff_cu_qp_delta_depth
-- avcodec/h264: Clear delayed_pic on deallocation
-- avcodec/hevc: clear filter_slice_edges() on allocation
-- avcodec/dcadec: Check that the added xch channel isnt already there
-- avcodec/indeo3: use signed variables to avoid underflow
-- swscale: increase yuv2rgb table headroom
-- avformat/mov: fix integer overflow of size
-- avformat/mov: check atom nesting depth
-- avcodec/utvideodec: Fix handling of slice_height=0
-- avcodec/xface: correct the XFACE_MAX_* values
-- avcodec/vmdvideo: Check len before using it in method 3
-- configure: create the tests directory like the doc directory
-- mmvideo: check frame dimensions
-- jvdec: check frame dimensions
-
-version 2.5.1:
-- lavu/frame: fix malloc error path in av_frame_copy_props()
-- avformat/aviobuf: Check that avio_seek() target is non negative
-- swresample/soxr_resample: fix error handling
-- avformat/flvdec: fix potential use of uninitialized variables
-- avformat/crypto: fix key vs iv typo
-- configure: use use_pkg_config() instead of check_pkg_config() for libsmbclient
-- avcodec/ppc/vp3dsp_altivec: POWER LE support to vp3_idct_add_altivec()
-- avformat/matroskadec: fix handling of recursive SeekHead elements
-- doc/examples/filtering_video: fix frame rate
-- avcodec/mpegaudiodec_template: only allocate fdsp when its used
-- doc/examples/transcoding: check encoder before using it
-- update MAINTAINERS file
-- POWER LE support in put_vp8_epel_h_altivec_core() put_vp8_epel_v_altivec_core() put_vp8_pixels16_altivec()
-- POWER LE support in vc1_inv_trans_8x4_altivec()
-
-version 2.5:
-- HEVC/H.265 RTP payload format (draft v6) packetizer
-- SUP/PGS subtitle demuxer
-- ffprobe -show_pixel_formats option
-- CAST128 symmetric block cipher, ECB mode
-- STL subtitle demuxer and decoder
-- libutvideo YUV 4:2:2 10bit support
-- XCB-based screen-grabber
-- UDP-Lite support (RFC 3828)
-- xBR scaling filter
-- AVFoundation screen capturing support
-- ffserver supports codec private options
-- creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
-- WebP muxer with animated WebP support
-- zygoaudio decoding support
-- APNG decoder and demuxer
-- postproc visualization support
-
-
-version 2.4:
-- Icecast protocol
-- ported lenscorrection filter from frei0r filter
-- large optimizations in dctdnoiz to make it usable
-- ICY metadata are now requested by default with the HTTP protocol
-- support for using metadata in stream specifiers in fftools
-- LZMA compression support in TIFF decoder
-- support for H.261 RTP payload format (RFC 4587)
-- HEVC/H.265 RTP payload format (draft v6) depacketizer
-- added codecview filter to visualize information exported by some codecs
-- Matroska 3D support thorugh side data
-- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
-- silenceremove filter
-
-
-version 2.3:
-- AC3 fixed-point decoding
-- shuffleplanes filter
-- subfile protocol
-- Phantom Cine demuxer
-- replaygain data export
-- VP7 video decoder
-- Alias PIX image encoder and decoder
-- Improvements to the BRender PIX image decoder
-- Improvements to the XBM decoder
-- QTKit input device
-- improvements to OpenEXR image decoder
-- support decoding 16-bit RLE SGI images
-- GDI screen grabbing for Windows
-- alternative rendition support for HTTP Live Streaming
-- AVFoundation input device
-- Direct Stream Digital (DSD) decoder
-- Magic Lantern Video (MLV) demuxer
-- On2 AVC (Audio for Video) decoder
-- support for decoding through DXVA2 in ffmpeg
-- libbs2b-based stereo-to-binaural audio filter
-- libx264 reference frames count limiting depending on level
-- native Opus decoder
-- display matrix export and rotation API
-- WebVTT encoder
-- showcqt multimedia filter
-- zoompan filter
-- signalstats filter
-- hqx filter (hq2x, hq3x, hq4x)
-- flanger filter
-- Image format auto-detection
-- LRC demuxer and muxer
-- Samba protocol (via libsmbclient)
-- WebM DASH Manifest muxer
-- libfribidi support in drawtext
-
-
-version 2.2:
-
-- HNM version 4 demuxer and video decoder
-- Live HDS muxer
-- setsar/setdar filters now support variables in ratio expressions
-- elbg filter
-- string validation in ffprobe
-- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
-- complete Voxware MetaSound decoder
-- remove mp3_header_compress bitstream filter
-- Windows resource files for shared libraries
-- aeval filter
-- stereoscopic 3d metadata handling
-- WebP encoding via libwebp
-- ATRAC3+ decoder
-- VP8 in Ogg demuxing
-- side & metadata support in NUT
-- framepack filter
-- XYZ12 rawvideo support in NUT
-- Exif metadata support in WebP decoder
-- OpenGL device
-- Use metadata_header_padding to control padding in ID3 tags (currently used in
-  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
-- Mirillis FIC video decoder
-- Support DNx444
-- libx265 encoder
-- dejudder filter
-- Autodetect VDA like all other hardware accelerations
-- aliases and defaults for Ogg subtypes (opus, spx)
-
-
-version 2.1:
-
-- aecho filter
-- perspective filter ported from libmpcodecs
-- ffprobe -show_programs option
-- compand filter
-- RTMP seek support
-- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
-  even when used as an input option. Previous behavior can be restored with
-  the -noaccurate_seek option.
-- ffmpeg -t option can now be used for inputs, to limit the duration of
-  data read from an input file
-- incomplete Voxware MetaSound decoder
-- read EXIF metadata from JPEG
-- DVB teletext decoder
-- phase filter ported from libmpcodecs
-- w3fdif filter
-- Opus support in Matroska
-- FFV1 version 1.3 is stable and no longer experimental
-- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
-- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
-  more consistent with other muxers.
-- adelay filter
-- pullup filter ported from libmpcodecs
-- ffprobe -read_intervals option
-- Lossless and alpha support for WebP decoder
-- Error Resilient AAC syntax (ER AAC LC) decoding
-- Low Delay AAC (ER AAC LD) decoding
-- mux chapters in ASF files
-- SFTP protocol (via libssh)
-- libx264: add ability to encode in YUVJ422P and YUVJ444P
-- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
-- make decoding alpha optional for prores, ffv1 and vp6 by setting
-  the skip_alpha flag.
-- ladspa wrapper filter
-- native VP9 decoder
-- dpx parser
-- max_error_rate parameter in ffmpeg
-- PulseAudio output device
-- ReplayGain scanner
-- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
-- Linux framebuffer output device
-- HEVC decoder
-- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
-- mergeplanes filter
-
-
-version 2.0:
-
-- curves filter
-- reference-counting for AVFrame and AVPacket data
-- ffmpeg now fails when input options are used for output file
-  or vice versa
-- support for Monkey's Audio versions from 3.93
-- perms and aperms filters
-- audio filtering support in ffplay
-- 10% faster aac encoding on x86 and MIPS
-- sine audio filter source
-- WebP demuxing and decoding support
-- ffmpeg options -filter_script and -filter_complex_script, which allow a
-  filtergraph description to be read from a file
-- OpenCL support
-- audio phaser filter
-- separatefields filter
-- libquvi demuxer
-- uniform options syntax across all filters
-- telecine filter
-- interlace filter
-- smptehdbars source
-- inverse telecine filters (fieldmatch and decimate)
-- colorbalance filter
-- colorchannelmixer filter
-- The matroska demuxer can now output proper verbatim ASS packets. It will
-  become the default at the next libavformat major bump.
-- decent native animated GIF encoding
-- asetrate filter
-- interleave filter
-- timeline editing with filters
-- vidstabdetect and vidstabtransform filters for video stabilization using
-  the vid.stab library
-- astats filter
-- trim and atrim filters
-- ffmpeg -t and -ss (output-only) options are now sample-accurate when
-  transcoding audio
-- Matroska muxer can now put the index at the beginning of the file.
-- extractplanes filter
-- avectorscope filter
-- ADPCM DTK decoder
-- ADP demuxer
-- RSD demuxer
-- RedSpark demuxer
-- ADPCM IMA Radical decoder
-- zmq filters
-- DCT denoiser filter (dctdnoiz)
-- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
-- Apple Intermediate Codec decoder
-- Escape 130 video decoder
-- FTP protocol support
-- V4L2 output device
-- 3D LUT filter (lut3d)
-- SMPTE 302M audio encoder
-- support for slice multithreading in libavfilter
-- Hald CLUT support (generation and filtering)
-- VC-1 interlaced B-frame support
-- support for WavPack muxing (raw and in Matroska)
-- XVideo output device
-- vignette filter
-- True Audio (TTA) encoder
-- Go2Webinar decoder
-- mcdeint filter ported from libmpcodecs
-- sab filter ported from libmpcodecs
-- ffprobe -show_chapters option
-- WavPack encoding through libwavpack
-- rotate filter
-- spp filter ported from libmpcodecs
-- libgme support
-- psnr filter
-
-
-version 1.2:
-
-- VDPAU hardware acceleration through normal hwaccel
-- SRTP support
-- Error diffusion dither in Swscale
-- Chained Ogg support
-- Theora Midstream reconfiguration support
-- EVRC decoder
-- audio fade filter
-- filtering audio with unknown channel layout
-- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
-  and treble audio filter
-- improved showspectrum filter, with multichannel support and sox-like colors
-- histogram filter
-- tee muxer
-- il filter ported from libmpcodecs
-- support ID3v2 tags in ASF files
-- encrypted TTA stream decoding support
-- RF64 support in WAV muxer
-- noise filter ported from libmpcodecs
-- Subtitles character encoding conversion
-- blend filter
-- stereo3d filter ported from libmpcodecs
-
-
-version 1.1:
-
-- stream disposition information printing in ffprobe
-- filter for loudness analysis following EBU R128
-- Opus encoder using libopus
-- ffprobe -select_streams option
-- Pinnacle TARGA CineWave YUV16 decoder
-- TAK demuxer, decoder and parser
-- DTS-HD demuxer
-- remove -same_quant, it hasn't worked for years
-- FFM2 support
-- X-Face image encoder and decoder
-- 24-bit FLAC encoding
-- multi-channel ALAC encoding up to 7.1
-- metadata (INFO tag) support in WAV muxer
-- subtitles raw text decoder
-- support for building DLLs using MSVC
-- LVF demuxer
-- ffescape tool
-- metadata (info chunk) support in CAF muxer
-- field filter ported from libmpcodecs
-- AVR demuxer
-- geq filter ported from libmpcodecs
-- remove ffserver daemon mode
-- AST muxer/demuxer
-- new expansion syntax for drawtext
-- BRender PIX image decoder
-- ffprobe -show_entries option
-- ffprobe -sections option
-- ADPCM IMA Dialogic decoder
-- BRSTM demuxer
-- animated GIF decoder and demuxer
-- PVF demuxer
-- subtitles filter
-- IRCAM muxer/demuxer
-- Paris Audio File demuxer
-- Virtual concatenation demuxer
-- VobSub demuxer
-- JSON captions for TED talks decoding support
-- SOX Resampler support in libswresample
-- aselect filter
-- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
-- Silicon Graphics Motion Video Compressor 1 & 2 decoder
-- Silicon Graphics Movie demuxer
-- apad filter
-- Resolution & pixel format change support with multithreading for H.264
-- documentation split into per-component manuals
-- pp (postproc) filter ported from MPlayer
-- NIST Sphere demuxer
-- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
-- Sony Wave64 muxer
-- adobe and limelight publisher authentication in RTMP
-- data: URI scheme
-- support building on the Plan 9 operating system
-- kerndeint filter ported from MPlayer
-- histeq filter ported from VirtualDub
-- Megalux Frame demuxer
-- 012v decoder
-- Improved AVC Intra decoding support
+version next:
 
 
 version 1.0:
-
 - INI and flat output in ffprobe
 - Scene detection in libavfilter
 - Indeo Audio decoder
@@ -862,9 +27,7 @@ version 1.0:
 - RTMPE protocol support
 - RTMPTE protocol support
 - showwaves and showspectrum filter
-- LucasArts SMUSH SANM playback support
-- LucasArts SMUSH VIMA audio decoder (ADPCM)
-- LucasArts SMUSH demuxer
+- LucasArts SMUSH playback support
 - SAMI, RealText and SubViewer demuxers and decoders
 - Heart Of Darkness PAF playback support
 - iec61883 device
@@ -894,6 +57,8 @@ version 1.0:
 - volume measurement filter
 - Ut Video encoder
 - Microsoft Screen 2 decoder
+- Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP
+  instead of AV_CODEC_ID_TEXT
 - smartblur filter ported from MPlayer
 - CPiA decoder
 - decimate filter ported from MPlayer
@@ -904,7 +69,6 @@ version 1.0:
 - WebVTT demuxer and decoder (simple tags supported)
 - RTP packetization of JPEG
 - faststart option in the MOV/MP4 muxer
-- support for building with MSVC
 
 
 version 0.11:
@@ -948,7 +112,6 @@ version 0.11:
 - vorbis parser
 - png parser
 - audio mix filter
-- ffv1: support (draft) version 1.3
 
 
 version 0.10:
@@ -988,7 +151,6 @@ version 0.10:
 - ffwavesynth decoder
 - aviocat tool
 - ffeval tool
-- support encoding and decoding 4-channel SGI images
 
 
 version 0.9:
@@ -1289,7 +451,7 @@ version 0.6:
 - LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
 - WMA Pro decoder
 - Core Audio Format demuxer
-- ATRAC1 decoder
+- Atrac1 decoder
 - MD STUDIO audio demuxer
 - RF64 support in WAV demuxer
 - MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -1389,7 +551,7 @@ version 0.5:
 - MXF demuxer
 - VC-1/WMV3/WMV9 video decoder
 - MacIntel support
-- AviSynth support
+- AVISynth support
 - VMware video decoder
 - VP5 video decoder
 - VP6 video decoder
@@ -1417,7 +579,7 @@ version 0.5:
 - Interplay C93 demuxer and video decoder
 - Bethsoft VID demuxer and video decoder
 - CRYO APC demuxer
-- ATRAC3 decoder
+- Atrac3 decoder
 - V.Flash PTX decoder
 - RoQ muxer, RoQ audio encoder
 - Renderware TXD demuxer and decoder

Doxyfile

@@ -31,9 +31,9 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 2.5.11
+PROJECT_NUMBER         = 1.0.10
 
-# With the PROJECT_LOGO tag one can specify a logo or icon that is included
+# With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
 # pixels and the maximum width should not exceed 200 pixels. Doxygen will
 # copy the logo to the output directory.
@@ -44,7 +44,7 @@ PROJECT_LOGO           =
 # If a relative path is entered, it will be relative to the location
 # where doxygen was started. If left blank the current directory will be used.
 
-OUTPUT_DIRECTORY       = doc/doxy
+OUTPUT_DIRECTORY       = doxy
 
 # If the CREATE_SUBDIRS tag is set to YES, then doxygen will create
 # 4096 sub-directories (in 2 levels) under the output directory of each output
@@ -277,7 +277,7 @@ SUBGROUPING            = YES
 # be useful for C code in case the coding convention dictates that all compound
 # types are typedef'ed and only the typedef is referenced, never the tag name.
 
-TYPEDEF_HIDES_STRUCT   = YES
+TYPEDEF_HIDES_STRUCT   = NO
 
 # The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
 # determine which symbols to keep in memory and which to flush to disk.
@@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT   = YES
 # causing a significant performance penality.
 # If the system has enough physical memory increasing the cache will improve the
 # performance by keeping more symbols in memory. Note that the value works on
-# a logarithmic scale so increasing the size by one will roughly double the
+# a logarithmic scale so increasing the size by one will rougly double the
 # memory usage. The cache size is given by this formula:
 # 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
 # corresponding to a cache size of 2^16 = 65536 symbols
@@ -409,7 +409,7 @@ INLINE_INFO            = YES
 # alphabetically by member name. If set to NO the members will appear in
 # declaration order.
 
-SORT_MEMBER_DOCS       = NO
+SORT_MEMBER_DOCS       = YES
 
 # If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
 # brief documentation of file, namespace and class members alphabetically
@@ -489,6 +489,12 @@ MAX_INITIALIZER_LINES  = 30
 
 SHOW_USED_FILES        = YES
 
+# If the sources in your project are distributed over multiple directories
+# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy
+# in the documentation. The default is NO.
+
+SHOW_DIRECTORIES       = NO
+
 # Set the SHOW_FILES tag to NO to disable the generation of the Files page.
 # This will remove the Files entry from the Quick Index and from the
 # Folder Tree View (if specified). The default is YES.
@@ -639,14 +645,15 @@ EXCLUDE_SYMBOLS        =
 # directories that contain example code fragments that are included (see
 # the \include command).
 
-EXAMPLE_PATH           = doc/examples/
+EXAMPLE_PATH           = libavcodec/ \
+                         libavformat/
 
 # If the value of the EXAMPLE_PATH tag contains directories, you can use the
 # EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
 # and *.h) to filter out the source-files in the directories. If left
 # blank all files are included.
 
-EXAMPLE_PATTERNS       = *.c
+EXAMPLE_PATTERNS       = *-example.c
 
 # If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
 # searched for input files to be used with the \include or \dontinclude
@@ -709,7 +716,7 @@ INLINE_SOURCES         = NO
 # doxygen to hide any special comment blocks from generated source code
 # fragments. Normal C and C++ comments will always remain visible.
 
-STRIP_CODE_COMMENTS    = NO
+STRIP_CODE_COMMENTS    = YES
 
 # If the REFERENCED_BY_RELATION tag is set to YES
 # then for each documented function all documented
@@ -759,7 +766,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])
 
-COLS_IN_ALPHA_INDEX    = 5
+COLS_IN_ALPHA_INDEX    = 2
 
 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -793,13 +800,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.
 
-HTML_HEADER            =
+HTML_HEADER            = doc/doxy/header.html
 
 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.
 
-HTML_FOOTER            =
+HTML_FOOTER            = doc/doxy/footer.html
 
 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +815,7 @@ HTML_FOOTER            =
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!
 
-HTML_STYLESHEET        =
+HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
 
 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -818,7 +825,7 @@ HTML_STYLESHEET        =
 # 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
 # The allowed range is 0 to 359.
 
-#HTML_COLORSTYLE_HUE    = 120
+HTML_COLORSTYLE_HUE    = 120
 
 # The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of
 # the colors in the HTML output. For a value of 0 the output will use
@@ -841,6 +848,12 @@ HTML_COLORSTYLE_GAMMA  = 80
 
 HTML_TIMESTAMP         = YES
 
+# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
+# files or namespaces will be aligned in HTML using tables. If set to
+# NO a bullet list will be used.
+
+HTML_ALIGN_MEMBERS     = YES
+
 # If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML
 # documentation will contain sections that can be hidden and shown after the
 # page has loaded. For this to work a browser that supports
@@ -1021,6 +1034,11 @@ ENUM_VALUES_PER_LINE   = 4
 
 GENERATE_TREEVIEW      = NO
 
+# By enabling USE_INLINE_TREES, doxygen will generate the Groups, Directories,
+# and Class Hierarchy pages using a tree view instead of an ordered list.
+
+USE_INLINE_TREES       = NO
+
 # If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
 # used to set the initial width (in pixels) of the frame in which the tree
 # is shown.
@@ -1056,7 +1074,7 @@ FORMULA_TRANSPARENT    = YES
 # typically be disabled. For large projects the javascript based search engine
 # can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.
 
-SEARCHENGINE           = YES
+SEARCHENGINE           = NO
 
 # When the SERVER_BASED_SEARCH tag is enabled the search engine will be
 # implemented using a PHP enabled web server instead of at the web client
@@ -1356,11 +1374,14 @@ INCLUDE_FILE_PATTERNS  =
 # instead of the = operator.
 
 PREDEFINED             = "__attribute__(x)=" \
+                         "RENAME(x)=x ## _TMPL" \
+                         "DEF(x)=x ## _TMPL" \
+                         HAVE_AV_CONFIG_H \
+                         HAVE_MMX \
+                         HAVE_MMXEXT \
+                         HAVE_AMD3DNOW \
                          "DECLARE_ALIGNED(a,t,n)=t n" \
-                         "offsetof(x,y)=0x42" \
-                         av_alloc_size \
-                         AV_GCC_VERSION_AT_LEAST(x,y)=1 \
-                         __GNUC__=1 \
+                         "offsetof(x,y)=0x42"
 
 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
 # this tag can be used to specify a list of macro names that should be expanded.

INSTALL

@@ -0,0 +1,15 @@
+
+1) Type './configure' to create the configuration. A list of configure
+options is printed by running 'configure --help'.
+
+'configure' can be launched from a directory different from the FFmpeg
+sources to build the objects out of tree. To do this, use an absolute
+path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
+
+2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
+
+3) Type 'make install' to install all binaries and libraries you built.
+
+NOTICE
+
+ - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

INSTALL.md

@@ -1,17 +0,0 @@
-#Installing FFmpeg:
-
-1. Type `./configure` to create the configuration. A list of configure
-options is printed by running `configure --help`.
-
-    `configure` can be launched from a directory different from the FFmpeg
-sources to build the objects out of tree. To do this, use an absolute
-path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
-
-2. Then type `make` to build FFmpeg. GNU Make 3.81 or later is required.
-
-3. Type `make install` to install all binaries and libraries you built.
-
-NOTICE
-------
-
- - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

LICENSE

@@ -0,0 +1,81 @@
+FFmpeg:
+
+Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
+or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
+files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
+FFmpeg.
+
+Some optional parts of FFmpeg are licensed under the GNU General Public License
+version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
+these parts are used by default, you have to explicitly pass --enable-gpl to
+configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
+
+Specifically, the GPL parts of FFmpeg are
+
+- libpostproc
+- libmpcodecs
+- optional x86 optimizations in the files
+  libavcodec/x86/idct_mmx.c
+- libutvideo encoding/decoding wrappers in
+  libavcodec/libutvideo*.cpp
+- the X11 grabber in libavdevice/x11grab.c
+- the swresample test app in
+  libswresample/swresample-test.c
+- the texi2pod.pl tool
+- the following filters in libavfilter:
+    - vf_blackframe.c
+    - vf_boxblur.c
+    - vf_colormatrix.c
+    - vf_cropdetect.c
+    - vf_delogo.c
+    - vf_hqdn3d.c
+    - vf_mp.c
+    - vf_super2xsai.c
+    - vf_tinterlace.c
+    - vf_yadif.c
+    - vsrc_mptestsrc.c
+
+There are a handful of files under other licensing terms, namely:
+
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
+  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
+  licensing details. Specifically note that you must credit the IJG in the
+  documentation accompanying your program if you only distribute executables.
+  You must also indicate any changes including additions and deletions to
+  those three files in the documentation.
+
+Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
+the configure parameter --enable-version3 will activate this licensing option
+for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
+COPYING.GPLv3 to learn the exact legal terms that apply in this case.
+
+
+external libraries
+==================
+
+FFmpeg can be combined with a number of external libraries, which sometimes
+affect the licensing of binaries resulting from the combination.
+
+compatible libraries
+--------------------
+
+The libcdio, libx264, libxavs and libxvid libraries are under GPL. When
+combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
+passing --enable-gpl to configure.
+
+The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
+license is incompatible with the LGPL v2.1 and the GPL v2, but not with
+version 3 of those licenses. So to combine these libraries with FFmpeg, the
+license version needs to be upgraded by passing --enable-version3 to configure.
+
+incompatible libraries
+----------------------
+
+The Fraunhofer AAC library, FAAC and aacplus are under licenses which
+are incompatible with the GPLv2 and v3. We do not know for certain if their
+licenses are compatible with the LGPL.
+If you wish to enable these libraries, pass --enable-nonfree to configure.
+But note that if you enable any of these libraries the resulting binary will
+be under a complex license mix that is more restrictive than the LGPL and that
+may result in additional obligations. It is possible that these
+restrictions cause the resulting binary to be unredistributeable.

LICENSE.md

@@ -1,106 +0,0 @@
-#FFmpeg:
-
-Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
-or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
-files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
-FFmpeg.
-
-Some optional parts of FFmpeg are licensed under the GNU General Public License
-version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
-these parts are used by default, you have to explicitly pass --enable-gpl to
-configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
-
-Specifically, the GPL parts of FFmpeg are:
-
-- libpostproc
-- libmpcodecs
-- optional x86 optimizations in the files
-  libavcodec/x86/flac_dsp_gpl.asm
-  libavcodec/x86/idct_mmx.c
-- libutvideo encoding/decoding wrappers in
-  libavcodec/libutvideo*.cpp
-- the X11 grabber in libavdevice/x11grab.c
-- the swresample test app in
-  libswresample/swresample-test.c
-- the texi2pod.pl tool
-- the following filters in libavfilter:
-    - f_ebur128.c
-    - vf_blackframe.c
-    - vf_boxblur.c
-    - vf_colormatrix.c
-    - vf_cropdetect.c
-    - vf_decimate.c
-    - vf_delogo.c
-    - vf_geq.c
-    - vf_histeq.c
-    - vf_hqdn3d.c
-    - vf_interlace.c
-    - vf_kerndeint.c
-    - vf_mcdeint.c
-    - vf_mp.c
-    - vf_owdenoise.c
-    - vf_perspective.c
-    - vf_phase.c
-    - vf_pp.c
-    - vf_pullup.c
-    - vf_sab.c
-    - vf_smartblur.c
-    - vf_spp.c
-    - vf_stereo3d.c
-    - vf_super2xsai.c
-    - vf_tinterlace.c
-    - vsrc_mptestsrc.c
-
-Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter --enable-version3 will activate this licensing option
-for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-COPYING.GPLv3 to learn the exact legal terms that apply in this case.
-
-There are a handful of files under other licensing terms, namely:
-
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
-  licensing details. Specifically note that you must credit the IJG in the
-  documentation accompanying your program if you only distribute executables.
-  You must also indicate any changes including additions and deletions to
-  those three files in the documentation.
-  tests/reference.pnm is under the expat license
-
-
-external libraries
-==================
-
-FFmpeg can be combined with a number of external libraries, which sometimes
-affect the licensing of binaries resulting from the combination.
-
-compatible libraries
---------------------
-
-The following libraries are under GPL:
-    - frei0r
-    - libcdio
-    - libutvideo
-    - libvidstab
-    - libx264
-    - libx265
-    - libxavs
-    - libxvid
-When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing --enable-gpl to configure.
-
-The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
-license is incompatible with the LGPL v2.1 and the GPL v2, but not with
-version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing --enable-version3 to configure.
-
-incompatible libraries
-----------------------
-
-The Fraunhofer AAC library, FAAC and aacplus are under licenses which
-are incompatible with the GPLv2 and v3. We do not know for certain if their
-licenses are compatible with the LGPL.
-If you wish to enable these libraries, pass --enable-nonfree to configure.
-But note that if you enable any of these libraries the resulting binary will
-be under a complex license mix that is more restrictive than the LGPL and that
-may result in additional obligations. It is possible that these
-restrictions cause the resulting binary to be unredistributeable.

MAINTAINERS

@@ -7,13 +7,14 @@ FFmpeg code.
 Please try to keep entries where you are the maintainer up to date!
 
 Names in () mean that the maintainer currently has no time to maintain the code.
-A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
-patches and related discussions.
+A CC after the name means that the maintainer prefers to be CC-ed on patches
+and related discussions.
 
 
 Project Leader
 ==============
 
+Michael Niedermayer
   final design decisions
 
 
@@ -30,7 +31,7 @@ ffprobe:
   ffprobe.c                             Stefano Sabatini
 
 ffserver:
-  ffserver.c                            Reynaldo H. Verdejo Pinochet
+  ffserver.c, ffserver.h                Baptiste Coudurier
 
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -42,26 +43,16 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
-build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
+documentation                           Mike Melanson
+website                                 Robert Swain, Lou Logan
+build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
 
 
-Communication
-=============
-
-website                                 Deby Barbara Lepage
-fate.ffmpeg.org                         Timothy Gu
-Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
-Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan
-Launchpad                               Timothy Gu
-
-
 libavutil
 =========
 
@@ -71,24 +62,11 @@ Internal Interfaces:
   libavutil/common.h                    Michael Niedermayer
 
 Other:
-  bprint                                Nicolas George
-  bswap.h
-  des                                   Reimar Doeffinger
-  dynarray.h                            Nicolas George
-  eval.c, eval.h                        Michael Niedermayer
-  float_dsp                             Loren Merritt
-  hash                                  Reimar Doeffinger
   intfloat*                             Michael Niedermayer
-  integer.c, integer.h                  Michael Niedermayer
-  lzo                                   Reimar Doeffinger
-  mathematics.c, mathematics.h          Michael Niedermayer
-  mem.c, mem.h                          Michael Niedermayer
-  opencl.c, opencl.h                    Wei Gao
-  opt.c, opt.h                          Michael Niedermayer
   rational.c, rational.h                Michael Niedermayer
-  rc4                                   Reimar Doeffinger
-  ripemd.c, ripemd.h                    James Almer
-  timecode                              Clément Bœsch
+  mathematics.c, mathematics.h          Michael Niedermayer
+  integer.c, integer.h                  Michael Niedermayer
+  bswap.h
 
 
 libavcodec
@@ -99,6 +77,10 @@ Generic Parts:
     avcodec.h                           Michael Niedermayer
   utility code:
     utils.c                             Michael Niedermayer
+    mem.c                               Michael Niedermayer
+    opt.c, opt.h                        Michael Niedermayer
+  arithmetic expression evaluator:
+    eval.c                              Michael Niedermayer
   audio and video frame extraction:
     parser.c                            Michael Niedermayer
   bitstream reading:
@@ -129,9 +111,6 @@ Generic Parts:
     libpostproc/*                       Michael Niedermayer
   table generation:
     tableprint.c, tableprint.h          Reimar Doeffinger
-  fixed point FFT:
-    fft*                                Zeljko Lukac
-  Text Subtitles                        Clément Bœsch
 
 Codecs:
   4xm.c                                 Michael Niedermayer
@@ -145,16 +124,13 @@ Codecs:
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3*                               Benjamin Larsson
-  atrac3plus*                           Maxim Poliakovski
   bgmc.c, bgmc.h                        Thilo Borgmann
   bink.c                                Kostya Shishkov
   binkaudio.c                           Peter Ross
   bmp.c                                 Mans Rullgard, Kostya Shishkov
   cavs*                                 Stefan Gehrer
-  cdxl.c                                Paul B Mahol
   celp_filters.*                        Vitor Sessak
   cinepak.c                             Roberto Togni
-  cinepakenc.c                          Rl / Aetey G.T. AB
   cljr                                  Alex Beregszaszi
   cllc.c                                Derek Buitenhuis
   cook.c, cookdata.h                    Benjamin Larsson
@@ -164,24 +140,21 @@ Codecs:
   dca.c                                 Kostya Shishkov, Benjamin Larsson
   dnxhd*                                Baptiste Coudurier
   dpcm.c                                Mike Melanson
-  dv.c                                  Roman Shaposhnik
-  dvbsubdec.c                           Anshul Maheshwari
   dxa.c                                 Kostya Shishkov
+  dv.c                                  Roman Shaposhnik
   eacmv*, eaidct*, eat*                 Peter Ross
-  exif.c, exif.h                        Thilo Borgmann
-  ffv1*                                 Michael Niedermayer
+  ffv1.c                                Michael Niedermayer
   ffwavesynth.c                         Nicolas George
-  fic.c                                 Derek Buitenhuis
   flac*                                 Justin Ruggles
   flashsv*                              Benjamin Larsson
   flicvideo.c                           Mike Melanson
   g722.c                                Martin Storsjo
   g726.c                                Roman Shaposhnik
   gifdec.c                              Baptiste Coudurier
+  h264*                                 Loren Merritt, Michael Niedermayer
   h261*                                 Michael Niedermayer
   h263*                                 Michael Niedermayer
-  h264*                                 Loren Merritt, Michael Niedermayer
-  huffyuv*                              Michael Niedermayer, Christophe Gisquet
+  huffyuv.c                             Michael Niedermayer
   idcinvideo.c                          Mike Melanson
   imc*                                  Benjamin Larsson
   indeo2*                               Kostya Shishkov
@@ -189,14 +162,13 @@ Codecs:
   interplayvideo.c                      Mike Melanson
   ivi*                                  Kostya Shishkov
   jacosub*                              Clément Bœsch
-  jpeg2000*                             Nicolas Bertrand
   jpeg_ls.c                             Kostya Shishkov
   jvdec.c                               Peter Ross
   kmvc.c                                Kostya Shishkov
   lcl*.c                                Roberto Togni, Reimar Doeffinger
   libcelt_dec.c                         Nicolas George
-  libdirac*                             David Conrad
   libgsm.c                              Michel Bardiaux
+  libdirac*                             David Conrad
   libopenjpeg.c                         Jaikrishnan Menon
   libopenjpegenc.c                      Michael Bradshaw
   libschroedinger*                      David Conrad
@@ -204,28 +176,23 @@ Codecs:
   libtheoraenc.c                        David Conrad
   libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
-  libvpx*                               James Zern
-  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
-  libx265.c                             Derek Buitenhuis
   libxavs.c                             Stefan Gehrer
-  libzvbi-teletextdec.c                 Marton Balint
+  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
   loco.c                                Kostya Shishkov
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
   mimic.c                               Ramiro Polla
-  mjpeg*.c                              Michael Niedermayer
+  mjpeg.c                               Michael Niedermayer
   mlp*                                  Ramiro Polla
   mmvideo.c                             Peter Ross
   mpc*                                  Kostya Shishkov
   mpeg12.c, mpeg12data.h                Michael Niedermayer
   mpegvideo.c, mpegvideo.h              Michael Niedermayer
-  mqc*                                  Nicolas Bertrand
   msmpeg4.c, msmpeg4data.h              Michael Niedermayer
   msrle.c                               Mike Melanson
   msvideo1.c                            Mike Melanson
   nellymoserdec.c                       Benjamin Larsson
   nuv.c                                 Reimar Doeffinger
-  paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
   pgssubdec.c                           Reimar Doeffinger
   ptx.c                                 Ivo van Poorten
@@ -241,17 +208,15 @@ Codecs:
   rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
   rv10.c                                Michael Niedermayer
   rv3*                                  Kostya Shishkov
-  rv4*                                  Kostya Shishkov, Christophe Gisquet
+  rv4*                                  Kostya Shishkov
   s3tc*                                 Ivo van Poorten
   smacker.c                             Kostya Shishkov
   smc.c                                 Mike Melanson
-  smvjpegdec.c                          Ash Hughes
-  snow*                                 Michael Niedermayer, Loren Merritt
+  snow.c                                Michael Niedermayer, Loren Merritt
   sonic.c                               Alex Beregszaszi
   srt*                                  Aurelien Jacobs
   sunrast.c                             Ivo van Poorten
   svq3.c                                Michael Niedermayer
-  tak*                                  Paul B Mahol
   targa.c                               Kostya Shishkov
   tiff.c                                Kostya Shishkov
   truemotion1*                          Mike Melanson
@@ -259,24 +224,21 @@ Codecs:
   truespeech.c                          Kostya Shishkov
   tscc.c                                Kostya Shishkov
   tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
-  ttaenc.c                              Paul B Mahol
   txd.c                                 Ivo van Poorten
   ulti*                                 Kostya Shishkov
   v410*.c                               Derek Buitenhuis
   vb.c                                  Kostya Shishkov
   vble.c                                Derek Buitenhuis
-  vc1*                                  Kostya Shishkov, Christophe Gisquet
+  vc1*                                  Kostya Shishkov
   vcr1.c                                Michael Niedermayer
   vda_h264_dec.c                        Xidorn Quan
-  vima.c                                Paul B Mahol
   vmnc.c                                Kostya Shishkov
-  vorbisdec.c                           Denes Balatoni, David Conrad
-  vorbisenc.c                           Oded Shimon
+  vorbis_enc.c                          Oded Shimon
+  vorbis_dec.c                          Denes Balatoni, David Conrad
   vp3*                                  Mike Melanson
   vp5                                   Aurelien Jacobs
   vp6                                   Aurelien Jacobs
   vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
-  vp9                                   Ronald Bultje, Clément Bœsch
   vqavideo.c                            Mike Melanson
   wavpack.c                             Kostya Shishkov
   wmaprodec.c                           Sascha Sommer
@@ -284,11 +246,8 @@ Codecs:
   wmv2.c                                Michael Niedermayer
   wnv1.c                                Kostya Shishkov
   xan.c                                 Mike Melanson
-  xbm*                                  Paul B Mahol
-  xface                                 Stefano Sabatini
   xl.c                                  Kostya Shishkov
   xvmc.c                                Ivan Kalvachev
-  xwd*                                  Paul B Mahol
   zerocodec.c                           Derek Buitenhuis
   zmbv*                                 Kostya Shishkov
 
@@ -307,21 +266,11 @@ libavdevice
     libavdevice/avdevice.h
 
 
-  avfoundation.m                        Thilo Borgmann
-  decklink*                             Deti Fliegl
-  dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
-  fbdev_enc.c                           Lukasz Marek
-  gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
   iec61883.c                            Georg Lippitsch
-  lavfi                                 Stefano Sabatini
   libdc1394.c                           Roman Shaposhnik
-  opengl_enc.c                          Lukasz Marek
-  pulse_audio_enc.c                     Lukasz Marek
-  qtkit.m                               Thilo Borgmann
-  sdl                                   Stefano Sabatini
-  v4l2.c                                Giorgio Vazzana
+  v4l2.c                                Luca Abeni
   vfwcap.c                              Ramiro Polla
-  xv.c                                  Lukasz Marek
+
 
 libavfilter
 ===========
@@ -330,39 +279,10 @@ Generic parts:
   graphdump.c                           Nicolas George
 
 Filters:
-  af_adelay.c                           Paul B Mahol
-  af_aecho.c                            Paul B Mahol
-  af_afade.c                            Paul B Mahol
   af_amerge.c                           Nicolas George
-  af_aphaser.c                          Paul B Mahol
-  af_aresample.c                        Michael Niedermayer
-  af_astats.c                           Paul B Mahol
   af_astreamsync.c                      Nicolas George
   af_atempo.c                           Pavel Koshevoy
-  af_biquads.c                          Paul B Mahol
-  af_compand.c                          Paul B Mahol
-  af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
-  af_silenceremove.c                    Paul B Mahol
-  avf_avectorscope.c                    Paul B Mahol
-  avf_showcqt.c                         Muhammad Faiz
-  vf_blend.c                            Paul B Mahol
-  vf_colorbalance.c                     Paul B Mahol
-  vf_dejudder.c                         Nicholas Robbins
-  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
-  vf_drawbox.c/drawgrid                 Andrey Utkin
-  vf_extractplanes.c                    Paul B Mahol
-  vf_histogram.c                        Paul B Mahol
-  vf_hqx.c                              Clément Bœsch
-  vf_idet.c                             Pascal Massimino
-  vf_il.c                               Paul B Mahol
-  vf_lenscorrection.c                   Daniel Oberhoff
-  vf_mergeplanes.c                      Paul B Mahol
-  vf_psnr.c                             Paul B Mahol
-  vf_scale.c                            Michael Niedermayer
-  vf_separatefields.c                   Paul B Mahol
-  vf_stereo3d.c                         Paul B Mahol
-  vf_telecine.c                         Paul B Mahol
   vf_yadif.c                            Michael Niedermayer
 
 Sources:
@@ -381,80 +301,60 @@ Generic parts:
 Muxers/Demuxers:
   4xm.c                                 Mike Melanson
   adtsenc.c                             Robert Swain
-  afc.c                                 Paul B Mahol
-  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
-  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
+  aiff.c                                Baptiste Coudurier
   ape.c                                 Kostya Shishkov
-  apngdec.c                             Benoit Fouet
   ass*                                  Aurelien Jacobs
-  astdec.c                              Paul B Mahol
-  astenc.c                              James Almer
   avi*                                  Michael Niedermayer
-  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
-  avr.c                                 Paul B Mahol
   bink.c                                Peter Ross
-  brstm.c                               Paul B Mahol
   caf*                                  Peter Ross
-  cdxl.c                                Paul B Mahol
   crc.c                                 Michael Niedermayer
   daud.c                                Reimar Doeffinger
-  dtshddec.c                            Paul B Mahol
   dv.c                                  Roman Shaposhnik
   dxa.c                                 Kostya Shishkov
   electronicarts.c                      Peter Ross
-  epafdec.c                             Paul B Mahol
   ffm*                                  Baptiste Coudurier
   flac*                                 Justin Ruggles
   flic.c                                Mike Melanson
   flvdec.c, flvenc.c                    Michael Niedermayer
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Anssi Hannula
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
-  img2*.c                               Michael Niedermayer
   ipmovie.c                             Mike Melanson
-  ircam*                                Paul B Mahol
+  img2.c                                Michael Niedermayer
   iss.c                                 Stefan Gehrer
   jacosub*                              Clément Bœsch
   jvdec.c                               Peter Ross
   libmodplug.c                          Clément Bœsch
   libnut.c                              Oded Shimon
   lmlm4.c                               Ivo van Poorten
-  lvfdec.c                              Paul B Mahol
   lxfdec.c                              Tomas Härdin
   matroska.c                            Aurelien Jacobs
   matroskadec.c                         Aurelien Jacobs
   matroskaenc.c                         David Conrad
-  matroska subtitles (matroskaenc.c)    John Peebles
   metadata*                             Aurelien Jacobs
-  mgsts.c                               Paul B Mahol
   microdvd*                             Aurelien Jacobs
   mm.c                                  Peter Ross
   mov.c                                 Michael Niedermayer, Baptiste Coudurier
-  movenc.c                              Baptiste Coudurier, Matthieu Bouron
+  movenc.c                              Michael Niedermayer, Baptiste Coudurier
   mpc.c                                 Kostya Shishkov
   mpeg.c                                Michael Niedermayer
   mpegenc.c                             Michael Niedermayer
-  mpegts.c                              Marton Balint
-  mpegtsenc.c                           Baptiste Coudurier
+  mpegts*                               Baptiste Coudurier
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
   mxfdec.c                              Tomas Härdin
-  nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
-  nut*                                  Michael Niedermayer
+  nut.c                                 Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
   oggdec.c, oggdec.h                    David Conrad
   oggenc.c                              Baptiste Coudurier
   oggparse*.c                           David Conrad
   oma.c                                 Maxim Poliakovski
-  paf.c                                 Paul B Mahol
   psxstr.c                              Mike Melanson
   pva.c                                 Ivo van Poorten
-  pvfdec.c                              Paul B Mahol
   r3d.c                                 Baptiste Coudurier
   raw.c                                 Michael Niedermayer
   rdt.c                                 Ronald S. Bultje
@@ -462,56 +362,30 @@ Muxers/Demuxers:
   rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
   rtmp*                                 Kostya Shishkov
   rtp.c, rtpenc.c                       Martin Storsjo
-  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
-  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
   rtpdec_asf.*                          Ronald S. Bultje
   rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
   rtsp.c                                Luca Barbato
   sbgdec.c                              Nicolas George
   sdp.c                                 Martin Storsjo
   segafilm.c                            Mike Melanson
-  segment.c                             Stefano Sabatini
   siff.c                                Kostya Shishkov
   smacker.c                             Kostya Shishkov
-  smjpeg*                               Paul B Mahol
-  spdif*                                Anssi Hannula
   srtdec.c                              Aurelien Jacobs
   swf.c                                 Baptiste Coudurier
-  takdec.c                              Paul B Mahol
   tta.c                                 Alex Beregszaszi
   txd.c                                 Ivo van Poorten
   voc.c                                 Aurelien Jacobs
   wav.c                                 Michael Niedermayer
   wc3movie.c                            Mike Melanson
-  webm dash (matroskaenc.c)             Vignesh Venkatasubramanian
-  webvtt*                               Matthew J Heaney
   westwood.c                            Mike Melanson
   wtv.c                                 Peter Ross
   wv.c                                  Kostya Shishkov
-  wvenc.c                               Paul B Mahol
 
 Protocols:
   bluray.c                              Petri Hintukainen
-  ftp.c                                 Lukasz Marek
   http.c                                Ronald S. Bultje
-  libssh.c                              Lukasz Marek
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
-  icecast.c                             Marvin Scholz
-
-
-libswresample
-=============
-
-Generic parts:
-  audioconvert.c                        Michael Niedermayer
-  dither.c                              Michael Niedermayer
-  rematrix*.c                           Michael Niedermayer
-  swresample*.c                         Michael Niedermayer
-
-Resamplers:
-  resample*.c                           Michael Niedermayer
-  soxr_resample.c                       Rob Sykes
 
 
 Operating systems / CPU architectures
@@ -520,24 +394,23 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard, Nedeljko Babic
+MIPS                                    Mans Rullgard
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver
-Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
-OS/2                                    KO Myung-Hun
 
 
 Releases
 ========
 
-2.5                                     Michael Niedermayer
+1.2                                     Michael Niedermayer
+1.1                                     Michael Niedermayer
+1.0                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 
@@ -545,21 +418,17 @@ If you want to maintain an older release, please contact us
 GnuPG Fingerprints of maintainers and contributors
 ==================================================
 
-Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
-Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
 Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
-Clément Bœsch                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
+Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
-FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
-Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
@@ -568,14 +437,11 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
-Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
+Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
-Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
+Stefano Sabatini              9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
-Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
-Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
-Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
-Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
+Tomas Härdin                  D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD

Makefile

@@ -4,71 +4,62 @@ include config.mak
 vpath %.c    $(SRC_PATH)
 vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
-vpath %.m    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
-vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
-AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
-AVPROGS-$(CONFIG_FFPLAY)   += ffplay
-AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
-AVPROGS-$(CONFIG_FFSERVER) += ffserver
+PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_FFPLAY)   += ffplay
+PROGS-$(CONFIG_FFPROBE)  += ffprobe
+PROGS-$(CONFIG_FFSERVER) += ffserver
 
-AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-PROGS      += $(AVPROGS)
-
-AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
-ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
-
-OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
-OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
-OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
-OBJS-ffserver                 += ffserver_config.o
-
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
+PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+OBJS        = cmdutils.o
+OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher uncoded_frame
+TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
-# $(FFLIBS-yes) needs to be in linking order
-FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
-FFLIBS-$(CONFIG_AVFILTER)   += avfilter
-FFLIBS-$(CONFIG_AVFORMAT)   += avformat
-FFLIBS-$(CONFIG_AVCODEC)    += avcodec
+BASENAMES   = ffmpeg ffplay ffprobe ffserver
+ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+ALLMANPAGES = $(BASENAMES:%=%.1)
+
+FFLIBS-$(CONFIG_AVDEVICE) += avdevice
+FFLIBS-$(CONFIG_AVFILTER) += avfilter
+FFLIBS-$(CONFIG_AVFORMAT) += avformat
 FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
-FFLIBS-$(CONFIG_POSTPROC)   += postproc
-FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
-FFLIBS-$(CONFIG_SWSCALE)    += swscale
+FFLIBS-$(CONFIG_AVCODEC)  += avcodec
+FFLIBS-$(CONFIG_POSTPROC) += postproc
+FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
+FFLIBS-$(CONFIG_SWSCALE)  += swscale
 
 FFLIBS := avutil
 
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
+EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile
 
-SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h
 
 include $(SRC_PATH)/common.mak
 
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 
-all: $(AVPROGS)
+all: $(PROGS)
 
-$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
+$(PROGS): %$(EXESUF): %_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@
+
+$(TOOLS): %$(EXESUF): %.o
+	$(LD) $(LDFLAGS) $(LD_O) $< $(ELIBS)
 
 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
-tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 
 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -77,11 +68,12 @@ config.h: .config
 	@-tput sgr0 2>/dev/null
 
 SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
-               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
+               ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
+               ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS NEON-OBJS             \
+               MMI-OBJS ALTIVEC-OBJS VIS-OBJS                            \
+               MMX-OBJS YASM-OBJS                                        \
                MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               OBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -93,33 +85,24 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
--include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
 include $(SRC_PATH)/library.mak
 endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-include $(SRC_PATH)/doc/Makefile
-
 define DOPROG
-OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
-$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
+OBJS-$(1) += $(1).o
+$(1)$(PROGSSUF)_g$(EXESUF): $(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef
 
-$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
 
-ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
-
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
-
-%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) cmdutils.o $(FF_EXTRALIBS)
 
 OBJDIRS += tools
 
@@ -131,14 +114,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
 
-libavutil/ffversion.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+version.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
 	$(Q)touch .version
 
 # force version.sh to run whenever version might have changed
 -include .version
 
-ifdef AVPROGS
+ifdef PROGS
 install: install-progs install-data
 endif
 
@@ -149,7 +132,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs
 
-install-progs: install-progs-yes $(AVPROGS)
+install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
@@ -161,27 +144,38 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
 clean::
-	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
+	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
+	$(RM) $(TOOLS)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
+	$(RM) coverage.info
 	$(RM) -r coverage-html
-	$(RM) -rf coverage.info lcov
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
+# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
+coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
+	$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
+
+coverage-html: coverage.info
+	$(Q)mkdir -p $@
+	$(Q)genhtml -o $@ $<
+	$(Q)touch $@
+
 check: all alltools examples testprogs fate
 
+include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile
 
 $(sort $(OBJDIRS)):

README

@@ -0,0 +1,18 @@
+FFmpeg README
+-------------
+
+1) Documentation
+----------------
+
+* Read the documentation in the doc/ directory in git.
+  You can also view it online at http://ffmpeg.org/documentation.html
+
+2) Licensing
+------------
+
+* See the LICENSE file.
+
+3) Build and Install
+--------------------
+
+* See the INSTALL file.

README.md

@@ -1,40 +0,0 @@
-FFmpeg README
-=============
-
-FFmpeg is a collection of libraries and tools to process multimedia content
-such as audio, video, subtitles and related metadata.
-
-## Libraries
-
-* `libavcodec` provides implementation of a wider range of codecs.
-* `libavformat` implements streaming protocols, container formats and basic I/O access.
-* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
-* `libavdevice` provides an abstraction to access capture and playback devices.
-* `libswresample` implements audio mixing and resampling routines.
-* `libswscale` implements color conversion and scaling routines.
-
-## Tools
-
-* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
-  manipulate, convert and stream multimedia content.
-* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
-* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analisys tool to inspect
-  multimedia content.
-* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
-
-## Documentation
-
-The offline documentation is available in the **doc/** directory.
-
-The online documentation is available in the main [website](http://ffmpeg.org)
-and in the [wiki](http://trac.ffmpeg.org).
-
-### Examples
-
-Coding examples are available in the **doc/examples** directory.
-
-## License
-
-FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
-GPL. Please refer to the LICENSE file for detailed information.

RELEASE

@@ -1 +1 @@
-2.5.11
+1.0.10

RELEASE_NOTES

@@ -1,101 +0,0 @@
- ┌────────────────────────────────────────┐
- │ RELEASE NOTES for FFmpeg 2.5 "Bohr"    │
- └────────────────────────────────────────┘
-
-   The FFmpeg Project proudly presents FFmpeg 2.5 "Bohr", 2.5 months after the
-   release of 2.4.
-
-   The most important new features are AVFoundation screen-grabbing support,
-   animated WebP decoding support, and Animated PNG support. In addition, many
-   exciting features for video streaming are also implemented, including MPEG-
-   DASH fragmenting muxer, HEVC RTP payload muxer, and UDP Lite support.
-
-   As usual, if you have any question on this release or any FFmpeg related
-   topic, feel free to join us on the #ffmpeg IRC channel (on
-   irc.freenode.net).
-
-   ┌────────────────────────────┐
-   │ 🔨  API Information         │
-   └────────────────────────────┘
-
-     FFmpeg 2.5 includes the following library versions:
-
-       • libavutil      54.15.100
-       • libavcodec     56.13.100
-       • libavformat    56.15.102
-       • libavdevice    56. 3.100
-       • libavfilter     5. 2.103
-       • libswscale      3. 1.101
-       • libswresample   1. 1.100
-       • libpostproc    53. 3.100
-
-     Important API changes since 2.4:
-
-       • avpriv_dv_frame_profile2() has been deprecated
-
-
-     Please refer to the doc/APIchanges file for more information.
-
- ┌────────────────────────────┐
- │ ★  List of New Features    │
- └────────────────────────────┘
-
-   ┌────────────────────────────┐
-   │ ffprobe                    │
-   └────────────────────────────┘
-
-    • -show_pixel_formats option
-
-   ┌────────────────────────────┐
-   │ ffserver                   │
-   └────────────────────────────┘
-
-    • codec private options support
-
-   ┌────────────────────────────┐
-   │ libavcodec                 │
-   └────────────────────────────┘
-
-    • STL subtitle decoder
-    • libutvideo YUV 4:2:2 10bit support
-    • animated WebP decoding support
-    • zygoaudio decoding support
-    • APNG decoder
-
-   ┌────────────────────────────┐
-   │ libavdevice                │
-   └────────────────────────────┘
-
-    • XCB-based screen-grabber
-    • AVFoundation screen capturing support
-
-   ┌────────────────────────────┐
-   │ libavformat                │
-   └────────────────────────────┘
-
-    • HEVC/H.265 RTP payload format (draft v6) packetizer
-    • SUP/PGS subtitle demuxer
-    • STL subtitle demuxer
-    • UDP-Lite support (RFC 3828)
-    • MPEG-DASH segmenting muxer, which allows creating DASH compatible
-      fragmented MP4
-    • WebP muxer
-    • APNG demuxer
-
-   ┌────────────────────────────┐
-   │ libavfilter                │
-   └────────────────────────────┘
-
-    • xBR scaling filter
-
-   ┌────────────────────────────┐
-   │ libavutil                  │
-   └────────────────────────────┘
-
-    • CAST128 symmetric block cipher, ECB mode
-
-   ┌────────────────────────────┐
-   │ libpostproc                │
-   └────────────────────────────┘
-
-    • visualization support

VERSION

@@ -0,0 +1 @@
+1.0.10

arch.mak

@@ -1,9 +1,9 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMV8)   += $(ARMV8-OBJS)   $(ARMV8-OBJS-yes)
-OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
+OBJS-$(HAVE_ARMVFP)  += $(ARMVFP-OBJS)  $(ARMVFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 
+OBJS-$(HAVE_MMI)     += $(MMI-OBJS)     $(MMI-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
@@ -11,5 +11,7 @@ OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 
+OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
+
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.h

@@ -24,13 +24,12 @@
 
 #include <stdint.h>
 
-#include "config.h"
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 
-#ifdef _WIN32
+#ifdef __MINGW32__
 #undef main /* We don't want SDL to override our main() */
 #endif
 
@@ -44,22 +43,16 @@ extern const char program_name[];
  */
 extern const int program_birth_year;
 
+/**
+ * this year, defined by the program for show_banner()
+ */
+extern const int this_year;
+
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern AVDictionary *swr_opts;
-extern AVDictionary *format_opts, *codec_opts, *resample_opts;
-extern int hide_banner;
-
-/**
- * Register a program-specific cleanup routine.
- */
-void register_exit(void (*cb)(int ret));
-
-/**
- * Wraps exit with a program-specific cleanup routine.
- */
-void exit_program(int ret) av_noreturn;
+extern struct SwrContext *swr_opts;
+extern AVDictionary *format_opts, *codec_opts;
 
 /**
  * Initialize the cmdutils option system, in particular
@@ -78,11 +71,6 @@ void uninit_opts(void);
  */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
 
-/**
- * Override the cpuflags.
- */
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
 /**
  * Fallback for options that are not explicitly handled, these will be
  * parsed through AVOptions.
@@ -98,14 +86,10 @@ int opt_report(const char *opt);
 
 int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 
+int opt_cpuflags(void *optctx, const char *opt, const char *arg);
+
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 
-#if CONFIG_OPENCL
-int opt_opencl(void *optctx, const char *opt, const char *arg);
-
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
-#endif
-
 /**
  * Limit the execution time.
  */
@@ -139,7 +123,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
  * not zero timestr is interpreted as a duration, otherwise as a
  * date
  *
- * @see av_parse_time()
+ * @see parse_date()
  */
 int64_t parse_time_or_die(const char *context, const char *timestr,
                           int is_duration);
@@ -155,7 +139,7 @@ typedef struct SpecifierOpt {
     } u;
 } SpecifierOpt;
 
-typedef struct OptionDef {
+typedef struct {
     const char *name;
     int flags;
 #define HAS_ARG    0x0001
@@ -178,8 +162,6 @@ typedef struct OptionDef {
                                    an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
-#define OPT_INPUT  0x40000
-#define OPT_OUTPUT 0x80000
      union {
         void *dst_ptr;
         int (*func_arg)(void *, const char *, const char *);
@@ -222,8 +204,6 @@ int show_help(void *optctx, const char *opt, const char *arg);
  * Parse the command line arguments.
  *
  * @param optctx an opaque options context
- * @param argc   number of command line arguments
- * @param argv   values of command line arguments
  * @param options Array with the definitions required to interpret every
  * option of the form: -option_name [argument]
  * @param parse_arg_function Name of the function called to process every
@@ -241,101 +221,6 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 int parse_option(void *optctx, const char *opt, const char *arg,
                  const OptionDef *options);
 
-/**
- * An option extracted from the commandline.
- * Cannot use AVDictionary because of options like -map which can be
- * used multiple times.
- */
-typedef struct Option {
-    const OptionDef  *opt;
-    const char       *key;
-    const char       *val;
-} Option;
-
-typedef struct OptionGroupDef {
-    /**< group name */
-    const char *name;
-    /**
-     * Option to be used as group separator. Can be NULL for groups which
-     * are terminated by a non-option argument (e.g. ffmpeg output files)
-     */
-    const char *sep;
-    /**
-     * Option flags that must be set on each option that is
-     * applied to this group
-     */
-    int flags;
-} OptionGroupDef;
-
-typedef struct OptionGroup {
-    const OptionGroupDef *group_def;
-    const char *arg;
-
-    Option *opts;
-    int  nb_opts;
-
-    AVDictionary *codec_opts;
-    AVDictionary *format_opts;
-    AVDictionary *resample_opts;
-    struct SwsContext *sws_opts;
-    AVDictionary *swr_opts;
-} OptionGroup;
-
-/**
- * A list of option groups that all have the same group type
- * (e.g. input files or output files)
- */
-typedef struct OptionGroupList {
-    const OptionGroupDef *group_def;
-
-    OptionGroup *groups;
-    int       nb_groups;
-} OptionGroupList;
-
-typedef struct OptionParseContext {
-    OptionGroup global_opts;
-
-    OptionGroupList *groups;
-    int           nb_groups;
-
-    /* parsing state */
-    OptionGroup cur_group;
-} OptionParseContext;
-
-/**
- * Parse an options group and write results into optctx.
- *
- * @param optctx an app-specific options context. NULL for global options group
- */
-int parse_optgroup(void *optctx, OptionGroup *g);
-
-/**
- * Split the commandline into an intermediate form convenient for further
- * processing.
- *
- * The commandline is assumed to be composed of options which either belong to a
- * group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
- * (everything else).
- *
- * A group (defined by an OptionGroupDef struct) is a sequence of options
- * terminated by either a group separator option (e.g. -i) or a parameter that
- * is not an option (doesn't start with -). A group without a separator option
- * must always be first in the supplied groups list.
- *
- * All options within the same group are stored in one OptionGroup struct in an
- * OptionGroupList, all groups with the same group definition are stored in one
- * OptionGroupList in OptionParseContext.groups. The order of group lists is the
- * same as the order of group definitions.
- */
-int split_commandline(OptionParseContext *octx, int argc, char *argv[],
-                      const OptionDef *options,
-                      const OptionGroupDef *groups, int nb_groups);
-
-/**
- * Free all allocated memory in an OptionParseContext.
- */
-void uninit_parse_context(OptionParseContext *octx);
-
 /**
  * Find the '-loglevel' option in the command line args and apply it.
  */
@@ -364,8 +249,6 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
  * Create a new options dictionary containing only the options from
  * opts which apply to the codec with ID codec_id.
  *
- * @param opts     dictionary to place options in
- * @param codec_id ID of the codec that should be filtered for
  * @param s Corresponding format context.
  * @param st A stream from s for which the options should be filtered.
  * @param codec The particular codec for which the options should be filtered.
@@ -415,13 +298,6 @@ void show_banner(int argc, char **argv, const OptionDef *options);
  */
 int show_version(void *optctx, const char *opt, const char *arg);
 
-/**
- * Print the build configuration of the program to stdout. The contents
- * depend on the definition of FFMPEG_CONFIGURATION.
- * This option processing function does not utilize the arguments.
- */
-int show_buildconf(void *optctx, const char *opt, const char *arg);
-
 /**
  * Print the license of the program to stdout. The license depends on
  * the license of the libraries compiled into the program.
@@ -431,31 +307,10 @@ int show_license(void *optctx, const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the formats supported by the
- * program (including devices).
- * This option processing function does not utilize the arguments.
- */
-int show_formats(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the devices supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_devices(void *optctx, const char *opt, const char *arg);
-
-#if CONFIG_AVDEVICE
-/**
- * Print a listing containing audodetected sinks of the output device.
- * Device name with options may be passed as an argument to limit results.
- */
-int show_sinks(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing audodetected sources of the input device.
- * Device name with options may be passed as an argument to limit results.
- */
-int show_sources(void *optctx, const char *opt, const char *arg);
-#endif
+int show_formats(void *optctx, const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the codecs supported by the
@@ -517,12 +372,6 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
  */
 int show_sample_fmts(void *optctx, const char *opt, const char *arg);
 
-/**
- * Print a listing containing all the color names and values recognized
- * by the program.
- */
-int show_colors(void *optctx, const char *opt, const char *arg);
-
 /**
  * Return a positive value if a line read from standard input
  * starts with [yY], otherwise return 0.
@@ -533,10 +382,9 @@ int read_yesno(void);
  * Read the file with name filename, and put its content in a newly
  * allocated 0-terminated buffer.
  *
- * @param filename file to read from
  * @param bufptr location where pointer to buffer is returned
  * @param size   location where size of buffer is returned
- * @return >= 0 in case of success, a negative value corresponding to an
+ * @return 0 in case of success, a negative value corresponding to an
  * AVERROR error code in case of failure.
  */
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
@@ -562,22 +410,64 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
 FILE *get_preset_file(char *filename, size_t filename_size,
                       const char *preset_name, int is_path, const char *codec_name);
 
+/**
+ * Do all the necessary cleanup and abort.
+ * This function is implemented in the avtools, not cmdutils.
+ */
+av_noreturn void exit_program(int ret);
+
 /**
  * Realloc array to hold new_size elements of elem_size.
- * Calls exit() on failure.
+ * Calls exit_program() on failure.
  *
- * @param array array to reallocate
  * @param elem_size size in bytes of each element
  * @param size new element count will be written here
- * @param new_size number of elements to place in reallocated array
  * @return reallocated array
  */
 void *grow_array(void *array, int elem_size, int *size, int new_size);
 
-#define media_type_string av_get_media_type_string
+typedef struct FrameBuffer {
+    uint8_t *base[4];
+    uint8_t *data[4];
+    int  linesize[4];
 
-#define GROW_ARRAY(array, nb_elems)\
-    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
+    int h, w;
+    enum PixelFormat pix_fmt;
+
+    int refcount;
+    struct FrameBuffer **pool;  ///< head of the buffer pool
+    struct FrameBuffer *next;
+} FrameBuffer;
+
+/**
+ * Get a frame from the pool. This is intended to be used as a callback for
+ * AVCodecContext.get_buffer.
+ *
+ * @param s codec context. s->opaque must be a pointer to the head of the
+ *          buffer pool.
+ * @param frame frame->opaque will be set to point to the FrameBuffer
+ *              containing the frame data.
+ */
+int codec_get_buffer(AVCodecContext *s, AVFrame *frame);
+
+/**
+ * A callback to be used for AVCodecContext.release_buffer along with
+ * codec_get_buffer().
+ */
+void codec_release_buffer(AVCodecContext *s, AVFrame *frame);
+
+/**
+ * A callback to be used for AVFilterBuffer.free.
+ * @param fb buffer to free. fb->priv must be a pointer to the FrameBuffer
+ *           containing the buffer data.
+ */
+void filter_release_buffer(AVFilterBuffer *fb);
+
+/**
+ * Free all the buffers in the pool. This must be called after all the
+ * buffers have been released.
+ */
+void free_buffer_pool(FrameBuffer **pool);
 
 #define GET_PIX_FMT_NAME(pix_fmt)\
     const char *name = av_get_pix_fmt_name(pix_fmt);

cmdutils_common_opts.h

@@ -4,9 +4,7 @@
     { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
-    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
     { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
-    { "devices"    , OPT_EXIT, {.func_arg = show_devices  },    "show available devices" },
     { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
     { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
     { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
@@ -16,20 +14,10 @@
     { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
     { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
     { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
-    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
-    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
+    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
+    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
+    { "debug"      , HAS_ARG,  {.func_arg = opt_codec_debug},   "set debug flags", "flags" },
+    { "fdebug"     , HAS_ARG,  {.func_arg = opt_codec_debug},   "set debug flags", "flags" },
     { "report"     , 0,        {(void*)opt_report}, "generate a report" },
     { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
-    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
-#if CONFIG_OPENCL
-    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
-    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
-#endif
-#if CONFIG_AVDEVICE
-    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
-      "list sources of the input device", "device" },
-    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
-      "list sinks of the output device", "device" },
-#endif
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },

cmdutils_opencl.c

@@ -1,276 +0,0 @@
-/*
- * Copyright (C) 2013 Lenny Wang
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/opt.h"
-#include "libavutil/time.h"
-#include "libavutil/log.h"
-#include "libavutil/opencl.h"
-#include "libavutil/avstring.h"
-#include "cmdutils.h"
-
-typedef struct {
-    int platform_idx;
-    int device_idx;
-    char device_name[64];
-    int64_t runtime;
-} OpenCLDeviceBenchmark;
-
-const char *ocl_bench_source = AV_OPENCL_KERNEL(
-inline unsigned char clip_uint8(int a)
-{
-    if (a & (~0xFF))
-        return (-a)>>31;
-    else
-        return a;
-}
-
-kernel void unsharp_bench(
-                    global unsigned char *src,
-                    global unsigned char *dst,
-                    global int *mask,
-                    int width,
-                    int height)
-{
-    int i, j, local_idx, lc_idx, sum = 0;
-    int2 thread_idx, block_idx, global_idx, lm_idx;
-    thread_idx.x = get_local_id(0);
-    thread_idx.y = get_local_id(1);
-    block_idx.x = get_group_id(0);
-    block_idx.y = get_group_id(1);
-    global_idx.x = get_global_id(0);
-    global_idx.y = get_global_id(1);
-    local uchar data[32][32];
-    local int lc[128];
-
-    for (i = 0; i <= 1; i++) {
-        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
-        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
-        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
-        for (j = 0; j <= 1; j++) {
-            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
-            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
-            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
-            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
-        }
-    }
-    local_idx = thread_idx.y*16 + thread_idx.x;
-    if (local_idx < 128)
-        lc[local_idx] = mask[local_idx];
-    barrier(CLK_LOCAL_MEM_FENCE);
-
-    \n#pragma unroll\n
-    for (i = -4; i <= 4; i++) {
-        lm_idx.y = 8 + i + thread_idx.y;
-        \n#pragma unroll\n
-        for (j = -4; j <= 4; j++) {
-            lm_idx.x = 8 + j + thread_idx.x;
-            lc_idx = (i + 4)*8 + j + 4;
-            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
-        }
-    }
-    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
-    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
-    if (global_idx.x < width && global_idx.y < height)
-        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
-}
-);
-
-#define OCLCHECK(method, ... )                                                 \
-do {                                                                           \
-    status = method(__VA_ARGS__);                                              \
-    if (status != CL_SUCCESS) {                                                \
-        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
-               av_opencl_errstr(status));                                      \
-        ret = AVERROR_EXTERNAL;                                                \
-        goto end;                                                              \
-    }                                                                          \
-} while (0)
-
-#define CREATEBUF(out, flags, size)                                            \
-do {                                                                           \
-    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
-    if (status != CL_SUCCESS) {                                                \
-        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
-        ret = AVERROR_EXTERNAL;                                                \
-        goto end;                                                              \
-    }                                                                          \
-} while (0)
-
-static void fill_rand_int(int *data, int n)
-{
-    int i;
-    srand(av_gettime());
-    for (i = 0; i < n; i++)
-        data[i] = rand();
-}
-
-#define OPENCL_NB_ITER 5
-static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
-{
-    int i, arg = 0, width = 1920, height = 1088;
-    int64_t start, ret = 0;
-    cl_int status;
-    size_t kernel_len;
-    char *inbuf;
-    int *mask;
-    int buf_size = width * height * sizeof(char);
-    int mask_size = sizeof(uint32_t) * 128;
-
-    cl_mem cl_mask, cl_inbuf, cl_outbuf;
-    cl_kernel kernel = NULL;
-    cl_program program = NULL;
-    size_t local_work_size_2d[2] = {16, 16};
-    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
-
-    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
-        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    fill_rand_int((int*)inbuf, buf_size/4);
-    fill_rand_int(mask, mask_size/4);
-
-    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
-    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
-    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
-
-    kernel_len = strlen(ocl_bench_source);
-    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
-                                        &kernel_len, &status);
-    if (status != CL_SUCCESS || !program) {
-        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
-        ret = AVERROR_EXTERNAL;
-        goto end;
-    }
-    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
-    if (status != CL_SUCCESS) {
-        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
-        ret = AVERROR_EXTERNAL;
-        goto end;
-    }
-    kernel = clCreateKernel(program, "unsharp_bench", &status);
-    if (status != CL_SUCCESS) {
-        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
-        ret = AVERROR_EXTERNAL;
-        goto end;
-    }
-
-    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
-             buf_size, inbuf, 0, NULL, NULL);
-    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
-             mask_size, mask, 0, NULL, NULL);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
-
-    start = av_gettime_relative();
-    for (i = 0; i < OPENCL_NB_ITER; i++)
-        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
-                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
-    clFinish(ext_opencl_env->command_queue);
-    ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
-end:
-    if (kernel)
-        clReleaseKernel(kernel);
-    if (program)
-        clReleaseProgram(program);
-    if (cl_inbuf)
-        clReleaseMemObject(cl_inbuf);
-    if (cl_outbuf)
-        clReleaseMemObject(cl_outbuf);
-    if (cl_mask)
-        clReleaseMemObject(cl_mask);
-    av_free(inbuf);
-    av_free(mask);
-    return ret;
-}
-
-static int compare_ocl_device_desc(const void *a, const void *b)
-{
-    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
-}
-
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
-{
-    int i, j, nb_devices = 0, count = 0;
-    int64_t score = 0;
-    AVOpenCLDeviceList *device_list;
-    AVOpenCLDeviceNode *device_node = NULL;
-    OpenCLDeviceBenchmark *devices = NULL;
-    cl_platform_id platform;
-
-    av_opencl_get_device_list(&device_list);
-    for (i = 0; i < device_list->platform_num; i++)
-        nb_devices += device_list->platform_node[i]->device_num;
-    if (!nb_devices) {
-        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
-        return AVERROR(EINVAL);
-    }
-    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
-        return AVERROR(ENOMEM);
-    }
-
-    for (i = 0; i < device_list->platform_num; i++) {
-        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
-            device_node = device_list->platform_node[i]->device_node[j];
-            platform = device_list->platform_node[i]->platform_id;
-            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
-            if (score > 0) {
-                devices[count].platform_idx = i;
-                devices[count].device_idx = j;
-                devices[count].runtime = score;
-                av_strlcpy(devices[count].device_name, device_node->device_name,
-                           sizeof(devices[count].device_name));
-                count++;
-            }
-        }
-    }
-    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
-    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
-    for (i = 0; i < count; i++)
-        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
-                devices[i].platform_idx, devices[i].device_idx,
-                devices[i].device_name, devices[i].runtime);
-
-    av_opencl_free_device_list(&device_list);
-    av_free(devices);
-    return 0;
-}
-
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}

common.mak

@@ -10,7 +10,7 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 
 MSG    = $@
@@ -30,9 +30,9 @@ CFLAGS     += $(ECFLAGS)
 CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
-YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
+YASMFLAGS  += $(IFLAGS:%=%/) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
 
-HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
+HOSTCCFLAGS = $(IFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 
 define COMPILE
@@ -43,7 +43,6 @@ endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
-COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -51,24 +50,12 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.cpp
 	$(COMPILE_CXX)
 
-%.o: %.m
-	$(COMPILE_C)
-
 %.s: %.c
 	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
 
 %.o: %.S
 	$(COMPILE_S)
 
-%_host.o: %.c
-	$(COMPILE_HOSTC)
-
-%.o: %.rc
-	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
-
-%.i: %.c
-	$(CC) $(CCFLAGS) $(CC_E) $<
-
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
 
@@ -92,15 +79,14 @@ endif
 include $(SRC_PATH)/arch.mak
 
 OBJS      += $(OBJS-yes)
-SLIBOBJS  += $(SLIBOBJS-yes)
-FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
+FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 
+EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
-SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -108,12 +94,9 @@ HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
 TOOLS     += $(TOOLS-yes)
 TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
-HEADERS   += $(HEADERS-yes)
 
-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
 
-SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
@@ -124,31 +107,24 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)
 
 $(HOSTOBJS): %.o: %.c
-	$(COMPILE_HOSTC)
+	$(call COMPILE,HOSTCC)
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
-$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
+$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HOBJS))
 
 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
-define RULES
 clean::
 	$(RM) $(OBJS) $(OBJS:.o=.d)
-	$(RM) $(HOSTPROGS)
-	$(RM) $(TOOLS)
-endef
 
-$(eval $(RULES))
-
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

compat/aix/math.h

@@ -1,31 +0,0 @@
-/*
- * Work around the class() function in AIX math.h clashing with
- * identifiers named "class".
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_COMPAT_AIX_MATH_H
-#define FFMPEG_COMPAT_AIX_MATH_H
-
-#define class class_in_math_h_causes_problems
-
-#include_next <math.h>
-
-#undef class
-
-#endif /* FFMPEG_COMPAT_AIX_MATH_H */

compat/avisynth/avisynth_c.h

@@ -1,880 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-// NOTE: this is a partial update of the Avisynth C interface to recognize
-// new color spaces added in Avisynth 2.60. By no means is this document
-// completely Avisynth 2.60 compliant.
-
-#ifndef __AVISYNTH_C__
-#define __AVISYNTH_C__
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-typedef unsigned char BYTE;
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 4 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_A=1<<4,
-      AVS_PLANAR_R=1<<5,
-      AVS_PLANAR_G=1<<6,
-      AVS_PLANAR_B=1<<7,
-      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31,
-
-      AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
-
-      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
-      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
-      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
-      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
-
-      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
-      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
-
-      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
-      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
-      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
-      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
-
-      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
-
-      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
-      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  //  AVS_CS_YV12  = 1<<3  Reserved
-  //  AVS_CS_I420  = 1<<4  Reserved
-  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
-
-  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
-  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
-  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
-  AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
-  AVS_CS_IYUV  = AVS_CS_I420,
-  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
-  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
-  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-  };
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  BYTE * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  volatile long sequence_number;
-
-  volatile long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  volatile long refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-  int row_sizeUV, heightUV;
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_sizeUV;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_sizeUV;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->heightUV;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on an AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                             INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                   int cachehints, int frame_range);
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                       AVS_FilterInfo * * fi,
-                                       AVS_Value child, int store_child);
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
-  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
-  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
-  AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
-};
-
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                const char * name, const char * params,
-                                AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-
-#ifdef AVSC_NO_DECLSPEC
-// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
-/*
-  The following functions needs to have been declared, probably from windows.h
-
-  void* malloc(size_t)
-  void free(void*);
-
-  HMODULE LoadLibrary(const char*);
-  void* GetProcAddress(HMODULE, const char*);
-  FreeLibrary(HMODULE);
-*/
-
-
-typedef struct AVS_Library AVS_Library;
-
-#define AVSC_DECLARE_FUNC(name) name##_func name
-
-struct AVS_Library {
-  HMODULE handle;
-
-  AVSC_DECLARE_FUNC(avs_add_function);
-  AVSC_DECLARE_FUNC(avs_at_exit);
-  AVSC_DECLARE_FUNC(avs_bit_blt);
-  AVSC_DECLARE_FUNC(avs_check_version);
-  AVSC_DECLARE_FUNC(avs_clip_get_error);
-  AVSC_DECLARE_FUNC(avs_copy_clip);
-  AVSC_DECLARE_FUNC(avs_copy_value);
-  AVSC_DECLARE_FUNC(avs_copy_video_frame);
-  AVSC_DECLARE_FUNC(avs_create_script_environment);
-  AVSC_DECLARE_FUNC(avs_delete_script_environment);
-  AVSC_DECLARE_FUNC(avs_function_exists);
-  AVSC_DECLARE_FUNC(avs_get_audio);
-  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
-  AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_get_frame);
-  AVSC_DECLARE_FUNC(avs_get_parity);
-  AVSC_DECLARE_FUNC(avs_get_var);
-  AVSC_DECLARE_FUNC(avs_get_version);
-  AVSC_DECLARE_FUNC(avs_get_video_info);
-  AVSC_DECLARE_FUNC(avs_invoke);
-  AVSC_DECLARE_FUNC(avs_make_writable);
-  AVSC_DECLARE_FUNC(avs_new_c_filter);
-  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
-  AVSC_DECLARE_FUNC(avs_release_clip);
-  AVSC_DECLARE_FUNC(avs_release_value);
-  AVSC_DECLARE_FUNC(avs_release_video_frame);
-  AVSC_DECLARE_FUNC(avs_save_string);
-  AVSC_DECLARE_FUNC(avs_set_cache_hints);
-  AVSC_DECLARE_FUNC(avs_set_global_var);
-  AVSC_DECLARE_FUNC(avs_set_memory_max);
-  AVSC_DECLARE_FUNC(avs_set_to_clip);
-  AVSC_DECLARE_FUNC(avs_set_var);
-  AVSC_DECLARE_FUNC(avs_set_working_dir);
-  AVSC_DECLARE_FUNC(avs_sprintf);
-  AVSC_DECLARE_FUNC(avs_subframe);
-  AVSC_DECLARE_FUNC(avs_subframe_planar);
-  AVSC_DECLARE_FUNC(avs_take_clip);
-  AVSC_DECLARE_FUNC(avs_vsprintf);
-};
-
-#undef AVSC_DECLARE_FUNC
-
-
-AVSC_INLINE AVS_Library * avs_load_library() {
-  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (!library)
-    return NULL;
-  library->handle = LoadLibrary("avisynth");
-  if (library->handle == NULL)
-    goto fail;
-
-#define __AVSC_STRINGIFY(x) #x
-#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
-#define AVSC_LOAD_FUNC(name) {\
-  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
-  if (library->name == NULL)\
-    goto fail;\
-}
-
-  AVSC_LOAD_FUNC(avs_add_function);
-  AVSC_LOAD_FUNC(avs_at_exit);
-  AVSC_LOAD_FUNC(avs_bit_blt);
-  AVSC_LOAD_FUNC(avs_check_version);
-  AVSC_LOAD_FUNC(avs_clip_get_error);
-  AVSC_LOAD_FUNC(avs_copy_clip);
-  AVSC_LOAD_FUNC(avs_copy_value);
-  AVSC_LOAD_FUNC(avs_copy_video_frame);
-  AVSC_LOAD_FUNC(avs_create_script_environment);
-  AVSC_LOAD_FUNC(avs_delete_script_environment);
-  AVSC_LOAD_FUNC(avs_function_exists);
-  AVSC_LOAD_FUNC(avs_get_audio);
-  AVSC_LOAD_FUNC(avs_get_cpu_flags);
-  AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_get_frame);
-  AVSC_LOAD_FUNC(avs_get_parity);
-  AVSC_LOAD_FUNC(avs_get_var);
-  AVSC_LOAD_FUNC(avs_get_version);
-  AVSC_LOAD_FUNC(avs_get_video_info);
-  AVSC_LOAD_FUNC(avs_invoke);
-  AVSC_LOAD_FUNC(avs_make_writable);
-  AVSC_LOAD_FUNC(avs_new_c_filter);
-  AVSC_LOAD_FUNC(avs_new_video_frame_a);
-  AVSC_LOAD_FUNC(avs_release_clip);
-  AVSC_LOAD_FUNC(avs_release_value);
-  AVSC_LOAD_FUNC(avs_release_video_frame);
-  AVSC_LOAD_FUNC(avs_save_string);
-  AVSC_LOAD_FUNC(avs_set_cache_hints);
-  AVSC_LOAD_FUNC(avs_set_global_var);
-  AVSC_LOAD_FUNC(avs_set_memory_max);
-  AVSC_LOAD_FUNC(avs_set_to_clip);
-  AVSC_LOAD_FUNC(avs_set_var);
-  AVSC_LOAD_FUNC(avs_set_working_dir);
-  AVSC_LOAD_FUNC(avs_sprintf);
-  AVSC_LOAD_FUNC(avs_subframe);
-  AVSC_LOAD_FUNC(avs_subframe_planar);
-  AVSC_LOAD_FUNC(avs_take_clip);
-  AVSC_LOAD_FUNC(avs_vsprintf);
-
-#undef __AVSC_STRINGIFY
-#undef AVSC_STRINGIFY
-#undef AVSC_LOAD_FUNC
-
-  return library;
-
-fail:
-  free(library);
-  return NULL;
-}
-
-AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (!library)
-    return;
-  FreeLibrary(library->handle);
-  free(library);
-}
-#endif
-
-#endif

compat/avisynth/avisynth_c_25.h

@@ -1,68 +0,0 @@
-//  Copyright (c) 2011 FFmpegSource Project
-//
-//  Permission is hereby granted, free of charge, to any person obtaining a copy
-//  of this software and associated documentation files (the "Software"), to deal
-//  in the Software without restriction, including without limitation the rights
-//  to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
-//  copies of the Software, and to permit persons to whom the Software is
-//  furnished to do so, subject to the following conditions:
-//
-//  The above copyright notice and this permission notice shall be included in
-//  all copies or substantial portions of the Software.
-//
-//  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-//  IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-//  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-//  AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-//  LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-//  OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
-//  THE SOFTWARE.
-
-/* these are defines/functions that are used and were changed in the switch to 2.6
- * and are needed to maintain full compatility with 2.5 */
-
-enum {
-  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-};
-
-AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
-    switch (plane)
-    {
-        case AVS_PLANAR_U: case AVS_PLANAR_V:
-            if (p->pitchUV)
-                return p->height>>1;
-            return 0;
-    }
-    return p->height;}
-
-AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
-    int r;
-    switch (plane)
-    {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-        if (p->pitchUV)
-            return p->row_size>>1;
-        else
-            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-        if (p->pitchUV)
-        {
-            r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-            if (r < p->pitchUV)
-                return r;
-            return p->row_size>>1;
-        }
-        else
-            return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-        r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-        if (r <= p->pitch)
-            return r;
-        return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
-    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }

compat/avisynth/avxsynth_c.h

@@ -1,728 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-#ifndef __AVXSYNTH_C__
-#define __AVXSYNTH_C__
-
-#include "windowsPorts/windows2linux.h"
-#include <stdarg.h>
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVXSYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 3 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-};
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  unsigned char * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  long sequence_number;
-
-  long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  int refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_size>>1;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_size>>1;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->height>>1;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    INT64 integer64; // match addition of __int64 to avxplugin.h
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on am AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                                  INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                        int cachehints, size_t frame_range);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                      AVS_FilterInfo * * fi,
-                                      AVS_Value child, int store_child);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-};
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                     const char * name, const char * params,
-                                     AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#endif //__AVXSYNTH_C__

compat/avisynth/windowsPorts/basicDataTypeConversions.h

@@ -1,85 +0,0 @@
-#ifndef __DATA_TYPE_CONVERSIONS_H__
-#define __DATA_TYPE_CONVERSIONS_H__
-
-#include <stdint.h>
-#include <wchar.h>
-
-#ifdef __cplusplus
-namespace avxsynth {
-#endif // __cplusplus
-
-typedef int64_t __int64;
-typedef int32_t __int32;
-#ifdef __cplusplus
-typedef bool BOOL;
-#else
-typedef uint32_t BOOL;
-#endif // __cplusplus
-typedef void* HMODULE;
-typedef void* LPVOID;
-typedef void* PVOID;
-typedef PVOID HANDLE;
-typedef HANDLE HWND;
-typedef HANDLE HINSTANCE;
-typedef void* HDC;
-typedef void* HBITMAP;
-typedef void* HICON;
-typedef void* HFONT;
-typedef void* HGDIOBJ;
-typedef void* HBRUSH;
-typedef void* HMMIO;
-typedef void* HACMSTREAM;
-typedef void* HACMDRIVER;
-typedef void* HIC;
-typedef void* HACMOBJ;
-typedef HACMSTREAM* LPHACMSTREAM;
-typedef void* HACMDRIVERID;
-typedef void* LPHACMDRIVER;
-typedef unsigned char BYTE;
-typedef BYTE* LPBYTE;
-typedef char TCHAR;
-typedef TCHAR* LPTSTR;
-typedef const TCHAR* LPCTSTR;
-typedef char* LPSTR;
-typedef LPSTR LPOLESTR;
-typedef const char* LPCSTR;
-typedef LPCSTR LPCOLESTR;
-typedef wchar_t WCHAR;
-typedef unsigned short WORD;
-typedef unsigned int UINT;
-typedef UINT MMRESULT;
-typedef uint32_t DWORD;
-typedef DWORD COLORREF;
-typedef DWORD FOURCC;
-typedef DWORD HRESULT;
-typedef DWORD* LPDWORD;
-typedef DWORD* DWORD_PTR;
-typedef int32_t LONG;
-typedef int32_t* LONG_PTR;
-typedef LONG_PTR LRESULT;
-typedef uint32_t ULONG;
-typedef uint32_t* ULONG_PTR;
-//typedef __int64_t intptr_t;
-typedef uint64_t _fsize_t;
-
-
-//
-// Structures
-//
-
-typedef struct _GUID {
-  DWORD Data1;
-  WORD  Data2;
-  WORD  Data3;
-  BYTE  Data4[8];
-} GUID;
-
-typedef GUID REFIID;
-typedef GUID CLSID;
-typedef CLSID* LPCLSID;
-typedef GUID IID;
-
-#ifdef __cplusplus
-}; // namespace avxsynth
-#endif // __cplusplus
-#endif //  __DATA_TYPE_CONVERSIONS_H__

compat/avisynth/windowsPorts/windows2linux.h

@@ -1,77 +0,0 @@
-#ifndef __WINDOWS2LINUX_H__
-#define __WINDOWS2LINUX_H__
-
-/*
- * LINUX SPECIFIC DEFINITIONS
-*/
-//
-// Data types conversions
-//
-#include <stdlib.h>
-#include <string.h>
-#include "basicDataTypeConversions.h"
-
-#ifdef __cplusplus
-namespace avxsynth {
-#endif // __cplusplus
-//
-// purposefully define the following MSFT definitions
-// to mean nothing (as they do not mean anything on Linux)
-//
-#define __stdcall
-#define __cdecl
-#define noreturn
-#define __declspec(x)
-#define STDAPI       extern "C" HRESULT
-#define STDMETHODIMP HRESULT __stdcall
-#define STDMETHODIMP_(x) x __stdcall
-
-#define STDMETHOD(x)    virtual HRESULT x
-#define STDMETHOD_(a, x) virtual a x
-
-#ifndef TRUE
-#define TRUE  true
-#endif
-
-#ifndef FALSE
-#define FALSE false
-#endif
-
-#define S_OK                (0x00000000)
-#define S_FALSE             (0x00000001)
-#define E_NOINTERFACE       (0X80004002)
-#define E_POINTER           (0x80004003)
-#define E_FAIL              (0x80004005)
-#define E_OUTOFMEMORY       (0x8007000E)
-
-#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
-#define FAILED(hr)              ((hr) & 0x80000000)
-#define SUCCEEDED(hr)           (!FAILED(hr))
-
-
-//
-// Functions
-//
-#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
-#define MAKEWORD(a,b) ((a << 8) | (b))
-
-#define lstrlen                             strlen
-#define lstrcpy                             strcpy
-#define lstrcmpi                            strcasecmp
-#define _stricmp                            strcasecmp
-#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
-#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
-// Windows uses (new, old) ordering but GCC has (old, new)
-#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
-
-#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
-#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
-#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
-
-#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
-
-#ifdef __cplusplus
-}; // namespace avxsynth
-#endif // __cplusplus
-
-#endif //  __WINDOWS2LINUX_H__

compat/float/float.h

@@ -1,35 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include_next <float.h>
-
-#ifdef FLT_MAX
-#undef  FLT_MAX
-#define FLT_MAX 3.40282346638528859812e+38F
-
-#undef  FLT_MIN
-#define FLT_MIN 1.17549435082228750797e-38F
-
-#undef  DBL_MAX
-#define DBL_MAX ((double)1.79769313486231570815e+308L)
-
-#undef  DBL_MIN
-#define DBL_MIN ((double)2.22507385850720138309e-308L)
-#endif

compat/float/limits.h

@@ -1,22 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include_next <limits.h>
-#include <float.h>

compat/getopt.c

@@ -38,13 +38,15 @@ static int optind = 1;
 static int optopt;
 static char *optarg;
 
+#undef fprintf
+
 static int getopt(int argc, char *argv[], char *opts)
 {
     static int sp = 1;
     int c;
     char *cp;
 
-    if (sp == 1) {
+    if (sp == 1)
         if (optind >= argc ||
             argv[optind][0] != '-' || argv[optind][1] == '\0')
             return EOF;
@@ -52,9 +54,8 @@ static int getopt(int argc, char *argv[], char *opts)
             optind++;
             return EOF;
         }
-    }
     optopt = c = argv[optind][sp];
-    if (c == ':' || !(cp = strchr(opts, c))) {
+    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
         fprintf(stderr, ": illegal option -- %c\n", c);
         if (argv[optind][++sp] == '\0') {
             optind++;

compat/plan9/head

@@ -1,10 +0,0 @@
-#!/bin/sh
-
-n=10
-
-case "$1" in
-    -n)  n=$2;      shift 2 ;;
-    -n*) n=${1#-n}; shift   ;;
-esac
-
-exec sed ${n}q "$@"

compat/plan9/main.c

@@ -1,34 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-int plan9_main(int argc, char **argv);
-
-#undef main
-int main(int argc, char **argv)
-{
-    /* The setfcr() function in lib9 is broken, must use asm. */
-#ifdef __i386
-    short fcr;
-    __asm__ volatile ("fstcw        %0 \n"
-                      "or      $63, %0 \n"
-                      "fldcw        %0 \n"
-                      : "=m"(fcr));
-#endif
-
-    return plan9_main(argc, argv);
-}

compat/plan9/printf

@@ -1,2 +0,0 @@
-#!/bin/sh
-exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"

compat/strtod.c

@@ -19,6 +19,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include <ctype.h>
 #include <limits.h>
 #include <stdlib.h>
 
@@ -48,7 +49,7 @@ double avpriv_strtod(const char *nptr, char **endptr)
     double res;
 
     /* Skip leading spaces */
-    while (av_isspace(*nptr))
+    while (isspace(*nptr))
         nptr++;
 
     if (!av_strncasecmp(nptr, "infinity", 8)) {

compat/tms470/math.h

@@ -1,30 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_COMPAT_TMS470_MATH_H
-#define FFMPEG_COMPAT_TMS470_MATH_H
-
-#include_next <math.h>
-
-#undef INFINITY
-#undef NAN
-
-#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
-#define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
-
-#endif /* FFMPEG_COMPAT_TMS470_MATH_H */

compat/va_copy.h

@@ -24,6 +24,3 @@
 #if !defined(va_copy) && defined(_MSC_VER)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
-#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
-#define va_copy(dst, src) __va_copy(dst, src)
-#endif

compat/w32pthreads.h

@@ -1,316 +0,0 @@
-/*
- * Copyright (C) 2010-2011 x264 project
- *
- * Authors: Steven Walters <kemuri9@gmail.com>
- *          Pegasys Inc. <http://www.pegasys-inc.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * w32threads to pthreads wrapper
- */
-
-#ifndef FFMPEG_COMPAT_W32PTHREADS_H
-#define FFMPEG_COMPAT_W32PTHREADS_H
-
-/* Build up a pthread-like API using underlying Windows API. Have only static
- * methods so as to not conflict with a potentially linked in pthread-win32
- * library.
- * As most functions here are used without checking return values,
- * only implement return values as necessary. */
-
-#define WIN32_LEAN_AND_MEAN
-#include <windows.h>
-#include <process.h>
-
-#include "libavutil/attributes.h"
-#include "libavutil/common.h"
-#include "libavutil/internal.h"
-#include "libavutil/mem.h"
-
-typedef struct pthread_t {
-    void *handle;
-    void *(*func)(void* arg);
-    void *arg;
-    void *ret;
-} pthread_t;
-
-/* the conditional variable api for windows 6.0+ uses critical sections and
- * not mutexes */
-typedef CRITICAL_SECTION pthread_mutex_t;
-
-/* This is the CONDITION_VARIABLE typedef for using Windows' native
- * conditional variables on kernels 6.0+. */
-#if HAVE_CONDITION_VARIABLE_PTR
-typedef CONDITION_VARIABLE pthread_cond_t;
-#else
-typedef struct pthread_cond_t {
-    void *Ptr;
-} pthread_cond_t;
-#endif
-
-#if _WIN32_WINNT >= 0x0600
-#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
-#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
-#endif
-
-static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
-{
-    pthread_t *h = arg;
-    h->ret = h->func(h->arg);
-    return 0;
-}
-
-static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
-                                    void *(*start_routine)(void*), void *arg)
-{
-    thread->func   = start_routine;
-    thread->arg    = arg;
-    thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
-                                           0, NULL);
-    return !thread->handle;
-}
-
-static av_unused void pthread_join(pthread_t thread, void **value_ptr)
-{
-    DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
-    if (ret != WAIT_OBJECT_0)
-        return;
-    if (value_ptr)
-        *value_ptr = thread.ret;
-    CloseHandle(thread.handle);
-}
-
-static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
-{
-    InitializeCriticalSection(m);
-    return 0;
-}
-static inline int pthread_mutex_destroy(pthread_mutex_t *m)
-{
-    DeleteCriticalSection(m);
-    return 0;
-}
-static inline int pthread_mutex_lock(pthread_mutex_t *m)
-{
-    EnterCriticalSection(m);
-    return 0;
-}
-static inline int pthread_mutex_unlock(pthread_mutex_t *m)
-{
-    LeaveCriticalSection(m);
-    return 0;
-}
-
-#if _WIN32_WINNT >= 0x0600
-static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
-{
-    InitializeConditionVariable(cond);
-    return 0;
-}
-
-/* native condition variables do not destroy */
-static inline void pthread_cond_destroy(pthread_cond_t *cond)
-{
-    return;
-}
-
-static inline void pthread_cond_broadcast(pthread_cond_t *cond)
-{
-    WakeAllConditionVariable(cond);
-}
-
-static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
-{
-    SleepConditionVariableCS(cond, mutex, INFINITE);
-    return 0;
-}
-
-static inline void pthread_cond_signal(pthread_cond_t *cond)
-{
-    WakeConditionVariable(cond);
-}
-
-#else // _WIN32_WINNT < 0x0600
-/* for pre-Windows 6.0 platforms we need to define and use our own condition
- * variable and api */
-typedef struct  win32_cond_t {
-    pthread_mutex_t mtx_broadcast;
-    pthread_mutex_t mtx_waiter_count;
-    volatile int waiter_count;
-    HANDLE semaphore;
-    HANDLE waiters_done;
-    volatile int is_broadcast;
-} win32_cond_t;
-
-/* function pointers to conditional variable API on windows 6.0+ kernels */
-static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
-static void (WINAPI *cond_init)(pthread_cond_t *cond);
-static void (WINAPI *cond_signal)(pthread_cond_t *cond);
-static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
-                                DWORD milliseconds);
-
-static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
-{
-    win32_cond_t *win32_cond = NULL;
-    if (cond_init) {
-        cond_init(cond);
-        return 0;
-    }
-
-    /* non native condition variables */
-    win32_cond = av_mallocz(sizeof(win32_cond_t));
-    if (!win32_cond)
-        return ENOMEM;
-    cond->Ptr = win32_cond;
-    win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
-    if (!win32_cond->semaphore)
-        return ENOMEM;
-    win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
-    if (!win32_cond->waiters_done)
-        return ENOMEM;
-
-    pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
-    pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
-    return 0;
-}
-
-static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
-{
-    win32_cond_t *win32_cond = cond->Ptr;
-    /* native condition variables do not destroy */
-    if (cond_init)
-        return;
-
-    /* non native condition variables */
-    CloseHandle(win32_cond->semaphore);
-    CloseHandle(win32_cond->waiters_done);
-    pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
-    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
-    av_freep(&win32_cond);
-    cond->Ptr = NULL;
-}
-
-static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
-{
-    win32_cond_t *win32_cond = cond->Ptr;
-    int have_waiter;
-
-    if (cond_broadcast) {
-        cond_broadcast(cond);
-        return;
-    }
-
-    /* non native condition variables */
-    pthread_mutex_lock(&win32_cond->mtx_broadcast);
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    have_waiter = 0;
-
-    if (win32_cond->waiter_count) {
-        win32_cond->is_broadcast = 1;
-        have_waiter = 1;
-    }
-
-    if (have_waiter) {
-        ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
-        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
-        ResetEvent(win32_cond->waiters_done);
-        win32_cond->is_broadcast = 0;
-    } else
-        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-}
-
-static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
-{
-    win32_cond_t *win32_cond = cond->Ptr;
-    int last_waiter;
-    if (cond_wait) {
-        cond_wait(cond, mutex, INFINITE);
-        return 0;
-    }
-
-    /* non native condition variables */
-    pthread_mutex_lock(&win32_cond->mtx_broadcast);
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    win32_cond->waiter_count++;
-    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-
-    // unlock the external mutex
-    pthread_mutex_unlock(mutex);
-    WaitForSingleObject(win32_cond->semaphore, INFINITE);
-
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    win32_cond->waiter_count--;
-    last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
-    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-
-    if (last_waiter)
-        SetEvent(win32_cond->waiters_done);
-
-    // lock the external mutex
-    return pthread_mutex_lock(mutex);
-}
-
-static av_unused void pthread_cond_signal(pthread_cond_t *cond)
-{
-    win32_cond_t *win32_cond = cond->Ptr;
-    int have_waiter;
-    if (cond_signal) {
-        cond_signal(cond);
-        return;
-    }
-
-    pthread_mutex_lock(&win32_cond->mtx_broadcast);
-
-    /* non-native condition variables */
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    have_waiter = win32_cond->waiter_count;
-    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-
-    if (have_waiter) {
-        ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
-        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
-        ResetEvent(win32_cond->waiters_done);
-    }
-
-    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-}
-#endif
-
-static av_unused void w32thread_init(void)
-{
-#if _WIN32_WINNT < 0x0600
-    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
-    /* if one is available, then they should all be available */
-    cond_init      =
-        (void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
-    cond_broadcast =
-        (void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
-    cond_signal    =
-        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
-    cond_wait      =
-        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
-#endif
-
-}
-
-#endif /* FFMPEG_COMPAT_W32PTHREADS_H */

compat/windows/makedef

@@ -1,132 +0,0 @@
-#!/bin/sh
-
-# Copyright (c) 2013, Derek Buitenhuis
-#
-# Permission to use, copy, modify, and/or distribute this software for any
-# purpose with or without fee is hereby granted, provided that the above
-# copyright notice and this permission notice appear in all copies.
-#
-# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
-# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
-# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
-# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
-# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
-# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
-# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
-
-# mktemp isn't POSIX, so supply an implementation
-mktemp() {
-    echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
-}
-
-if [ $# -lt 2 ]; then
-    echo "Usage: makedef <version_script> <objects>" >&2
-    exit 0
-fi
-
-vscript=$1
-shift
-
-if [ ! -f "$vscript" ]; then
-    echo "Version script does not exist" >&2
-    exit 1
-fi
-
-for object in "$@"; do
-    if [ ! -f "$object" ]; then
-        echo "Object does not exist: ${object}" >&2
-        exit 1
-    fi
-done
-
-# Create a lib temporarily to dump symbols from.
-# It's just much easier to do it this way
-libname=$(mktemp -u "library").lib
-
-trap 'rm -f -- $libname' EXIT
-
-lib -out:${libname} $@ >/dev/null
-if [ $? != 0 ]; then
-    echo "Could not create temporary library." >&2
-    exit 1
-fi
-
-IFS='
-'
-
-# Determine if we're building for x86 or x86_64 and
-# set the symbol prefix accordingly.
-prefix=""
-arch=$(dumpbin -headers ${libname} |
-       tr '\t' ' ' |
-       grep '^ \+.\+machine \+(.\+)' |
-       head -1 |
-       sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
-
-if [ "${arch}" = "x86" ]; then
-    prefix="_"
-else
-    if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
-        echo "Unknown machine type." >&2
-        exit 1
-    fi
-fi
-
-started=0
-regex="none"
-
-for line in $(cat ${vscript} | tr '\t' ' '); do
-    # We only care about global symbols
-    echo "${line}" | grep -q '^ \+global:'
-    if [ $? = 0 ]; then
-        started=1
-        line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
-    else
-        echo "${line}" | grep -q '^ \+local:'
-        if [ $? = 0 ]; then
-            started=0
-        fi
-    fi
-
-    if [ ${started} = 0 ]; then
-        continue
-    fi
-
-    # Handle multiple symbols on one line
-    IFS=';'
-
-    # Work around stupid expansion to filenames
-    line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
-    for exp in ${line}; do
-        # Remove leading and trailing whitespace
-        exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
-
-        if [ "${regex}" = "none" ]; then
-            regex="${exp}"
-        else
-            regex="${regex};${exp}"
-        fi
-    done
-
-    IFS='
-'
-done
-
-dump=$(dumpbin -linkermember:1 ${libname})
-
-rm ${libname}
-
-IFS=';'
-list=""
-for exp in ${regex}; do
-    list="${list}"'
-'$(echo "${dump}" |
-          sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
-          tail -n +2 |
-          cut -d' ' -f3 |
-          grep "^${exp}" |
-          sed -e 's/^/    /')
-done
-
-echo "EXPORTS"
-echo "${list}" | sort | uniq | tail -n +2

doc/Makefile

@@ -1,31 +1,13 @@
-LIBRARIES-$(CONFIG_AVUTIL)     += libavutil
-LIBRARIES-$(CONFIG_SWSCALE)    += libswscale
-LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
-LIBRARIES-$(CONFIG_AVCODEC)    += libavcodec
-LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
-LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
-LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter
-
-COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
-COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
-COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
-COMPONENTS-$(CONFIG_AVCODEC)    += ffmpeg-codecs ffmpeg-bitstream-filters
-COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
-COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
-COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters
-
-MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
-MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
-MANPAGES    = $(MANPAGES1) $(MANPAGES3)
-PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
+MANPAGES    = $(PROGS-yes:%=doc/%.1)
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
               doc/developer.html                                        \
               doc/faq.html                                              \
               doc/fate.html                                             \
               doc/general.html                                          \
               doc/git-howto.html                                        \
-              doc/nut.html                                              \
               doc/platform.html                                         \
+              doc/syntax.html                                           \
 
 TXTPAGES    = doc/fate.txt                                              \
 
@@ -36,38 +18,13 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
-DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
-ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
-
-DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
-PROGS              += $(DOC_EXAMPLES)
-
 all-$(CONFIG_DOC): doc
 
 doc: documentation
 
-apidoc: doc/doxy/html
 documentation: $(DOCS)
 
-examples: $(DOC_EXAMPLES)
-
-TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
+TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
 
 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -82,101 +39,39 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 	$(M)doc/print_options $* > $@
 
 doc/%.html: TAG = HTML
-doc/%-all.html: TAG = HTML
-
-ifdef HAVE_MAKEINFO_HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
-
-doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
-else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
-
-doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
-endif
+	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
+doc/%.pod: doc/%.texi $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
+	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
 
-doc/%-all.pod: TAG = POD
-doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
-
-doc/%.1 doc/%.3: TAG = MAN
+doc/%.1: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
-doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
 
-$(DOCS) doc/doxy/html: | doc/
-$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
-OBJDIRS += doc/examples
-
-DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
-
-doc/doxy/html: TAG = DOXY
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
-
-install-doc: install-html install-man
-
-install-html:
+$(DOCS): | doc/
 
 install-man:
 
-ifdef CONFIG_HTMLPAGES
-install-progs-$(CONFIG_DOC): install-html
-
-install-html: $(HTMLPAGES)
-	$(Q)mkdir -p "$(DOCDIR)"
-	$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
-endif
-
 ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man
 
 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
-	$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
-	$(Q)mkdir -p "$(MANDIR)/man3"
-	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
+	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
 endif
 
-uninstall: uninstall-doc
-
-uninstall-doc: uninstall-html uninstall-man
-
-uninstall-html:
-	$(RM) -r "$(DOCDIR)"
+uninstall: uninstall-man
 
 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
-	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 
-clean:: docclean
-
-distclean:: docclean
-	$(RM) doc/config.texi
-
-examplesclean:
-	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
-	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
-
-docclean: examplesclean
-	$(RM) $(CLEANSUFFIXES:%=doc/%)
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
-	$(RM) -r doc/doxy/html
+clean::
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 
 -include $(wildcard $(DOCS:%=%.d))
 
-.PHONY: apidoc doc documentation
+.PHONY: doc documentation

doc/RELEASE_NOTES

@@ -0,0 +1,16 @@
+Release Notes
+=============
+
+* 1.0 "Angel"  September, 2012
+
+
+General notes
+-------------
+See the Changelog file for a list of significant changes. Note, there
+are many more new features and bugfixes than whats listed there.
+
+Bugreports against FFmpeg git master or the most recent FFmpeg release are
+accepted. If you are experiencing issues with any formally released version of
+FFmpeg, please try git master to check if the issue still exists. If it does,
+make your report against the development code following the usual bug reporting
+guidelines.

doc/authors.texi

@@ -1,11 +0,0 @@
-@chapter Authors
-
-The FFmpeg developers.
-
-For details about the authorship, see the Git history of the project
-(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
-@command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{http://source.ffmpeg.org}.
-
-Maintainers for the specific components are listed in the file
-@file{MAINTAINERS} in the source code tree.

doc/avtools-common-opts.texi

@@ -0,0 +1,200 @@
+All the numerical options, if not specified otherwise, accept in input
+a string representing a number, which may contain one of the
+International System number postfixes, for example 'K', 'M', 'G'.
+If 'i' is appended after the postfix, powers of 2 are used instead of
+powers of 10. The 'B' postfix multiplies the value for 8, and can be
+appended after another postfix or used alone. This allows using for
+example 'KB', 'MiB', 'G' and 'B' as postfix.
+
+Options which do not take arguments are boolean options, and set the
+corresponding value to true. They can be set to false by prefixing
+with "no" the option name, for example using "-nofoo" in the
+command line will set to false the boolean option with name "foo".
+
+@anchor{Stream specifiers}
+@section Stream specifiers
+Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
+are used to precisely specify which stream(s) does a given option belong to.
+
+A stream specifier is a string generally appended to the option name and
+separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
+@code{a:1} stream specifier, which matches the second audio stream. Therefore it
+would select the ac3 codec for the second audio stream.
+
+A stream specifier can match several streams, the option is then applied to all
+of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
+streams.
+
+An empty stream specifier matches all streams, for example @code{-codec copy}
+or @code{-codec: copy} would copy all the streams without reencoding.
+
+Possible forms of stream specifiers are:
+@table @option
+@item @var{stream_index}
+Matches the stream with this index. E.g. @code{-threads:1 4} would set the
+thread count for the second stream to 4.
+@item @var{stream_type}[:@var{stream_index}]
+@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data and 't' for attachments. If @var{stream_index} is given, then
+matches stream number @var{stream_index} of this type. Otherwise matches all
+streams of this type.
+@item p:@var{program_id}[:@var{stream_index}]
+If @var{stream_index} is given, then matches stream number @var{stream_index} in
+program with id @var{program_id}. Otherwise matches all streams in this program.
+@item #@var{stream_id}
+Matches the stream by format-specific ID.
+@end table
+
+@section Generic options
+
+These options are shared amongst the av* tools.
+
+@table @option
+
+@item -L
+Show license.
+
+@item -h, -?, -help, --help [@var{arg}]
+Show help. An optional parameter may be specified to print help about a specific
+item.
+
+Possible values of @var{arg} are:
+@table @option
+@item decoder=@var{decoder_name}
+Print detailed information about the decoder named @var{decoder_name}. Use the
+@option{-decoders} option to get a list of all decoders.
+
+@item encoder=@var{encoder_name}
+Print detailed information about the encoder named @var{encoder_name}. Use the
+@option{-encoders} option to get a list of all encoders.
+
+@item demuxer=@var{demuxer_name}
+Print detailed information about the demuxer named @var{demuxer_name}. Use the
+@option{-formats} option to get a list of all demuxers and muxers.
+
+@item muxer=@var{muxer_name}
+Print detailed information about the muxer named @var{muxer_name}. Use the
+@option{-formats} option to get a list of all muxers and demuxers.
+
+@end table
+
+@item -version
+Show version.
+
+@item -formats
+Show available formats.
+
+The fields preceding the format names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@end table
+
+@item -codecs
+Show all codecs known to libavcodec.
+
+Note that the term 'codec' is used throughout this documentation as a shortcut
+for what is more correctly called a media bitstream format.
+
+@item -decoders
+Show available decoders.
+
+@item -encoders
+Show all available encoders.
+
+@item -bsfs
+Show available bitstream filters.
+
+@item -protocols
+Show available protocols.
+
+@item -filters
+Show available libavfilter filters.
+
+@item -pix_fmts
+Show available pixel formats.
+
+@item -sample_fmts
+Show available sample formats.
+
+@item -layouts
+Show channel names and standard channel layouts.
+
+@item -loglevel @var{loglevel} | -v @var{loglevel}
+Set the logging level used by the library.
+@var{loglevel} is a number or a string containing one of the following values:
+@table @samp
+@item quiet
+@item panic
+@item fatal
+@item error
+@item warning
+@item info
+@item verbose
+@item debug
+@end table
+
+By default the program logs to stderr, if coloring is supported by the
+terminal, colors are used to mark errors and warnings. Log coloring
+can be disabled setting the environment variable
+@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
+the environment variable @env{AV_LOG_FORCE_COLOR}.
+The use of the environment variable @env{NO_COLOR} is deprecated and
+will be dropped in a following FFmpeg version.
+
+@item -report
+Dump full command line and console output to a file named
+@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
+directory.
+This file can be useful for bug reports.
+It also implies @code{-loglevel verbose}.
+
+Note: setting the environment variable @code{FFREPORT} to any value has the
+same effect.
+
+@item -cpuflags flags (@emph{global})
+Allows setting and clearing cpu flags. This option is intended
+for testing. Do not use it unless you know what you're doing.
+@example
+ffmpeg -cpuflags -sse+mmx ...
+ffmpeg -cpuflags mmx ...
+ffmpeg -cpuflags 0 ...
+@end example
+
+@end table
+
+@section AVOptions
+
+These options are provided directly by the libavformat, libavdevice and
+libavcodec libraries. To see the list of available AVOptions, use the
+@option{-help} option. They are separated into two categories:
+@table @option
+@item generic
+These options can be set for any container, codec or device. Generic options
+are listed under AVFormatContext options for containers/devices and under
+AVCodecContext options for codecs.
+@item private
+These options are specific to the given container, device or codec. Private
+options are listed under their corresponding containers/devices/codecs.
+@end table
+
+For example to write an ID3v2.3 header instead of a default ID3v2.4 to
+an MP3 file, use the @option{id3v2_version} private option of the MP3
+muxer:
+@example
+ffmpeg -i input.flac -id3v2_version 3 out.mp3
+@end example
+
+All codec AVOptions are obviously per-stream, so the chapter on stream
+specifiers applies to them
+
+Note @option{-nooption} syntax cannot be used for boolean AVOptions,
+use @option{-option 0}/@option{-option 1}.
+
+Note2 old undocumented way of specifying per-stream AVOptions by prepending
+v/a/s to the options name is now obsolete and will be removed soon.
+
+@include avoptions_codec.texi
+@include avoptions_format.texi

doc/avutil.txt

@@ -0,0 +1,36 @@
+AVUtil
+======
+libavutil is a small lightweight library of generally useful functions.
+It is not a library for code needed by both libavcodec and libavformat.
+
+
+Overview:
+=========
+adler32.c               adler32 checksum
+aes.c                   AES encryption and decryption
+fifo.c                  resizeable first in first out buffer
+intfloat_readwrite.c    portable reading and writing of floating point values
+log.c                   "printf" with context and level
+md5.c                   MD5 Message-Digest Algorithm
+rational.c              code to perform exact calculations with rational numbers
+tree.c                  generic AVL tree
+crc.c                   generic CRC checksumming code
+integer.c               128bit integer math
+lls.c
+mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
+mem.c                   memory allocation routines with guaranteed alignment
+
+Headers:
+bswap.h                 big/little/native-endian conversion code
+x86_cpu.h               a few useful macros for unifying x86-64 and x86-32 code
+avutil.h
+common.h
+intreadwrite.h          reading and writing of unaligned big/little/native-endian integers
+
+
+Goals:
+======
+* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
+* Small (source and object)
+* Efficient (low CPU and memory usage)
+* Useful (avoid useless features almost no one needs)

doc/bitstream_filters.texi

@@ -13,59 +13,13 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
 The option @code{-bsfs} of the ff* tools will display the list of
 all the supported bitstream filters included in your build.
 
-The ff* tools have a -bsf option applied per stream, taking a
-comma-separated list of filters, whose parameters follow the filter
-name after a '='.
-
-@example
-ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
-@end example
-
-Below is a description of the currently available bitstream filters,
-with their parameters, if any.
+Below is a description of the currently available bitstream filters.
 
 @section aac_adtstoasc
 
-Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
-bitstream filter.
-
-This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
-ADTS header and removes the ADTS header.
-
-This is required for example when copying an AAC stream from a raw
-ADTS AAC container to a FLV or a MOV/MP4 file.
-
 @section chomp
 
-Remove zero padding at the end of a packet.
-
-@section dump_extra
-
-Add extradata to the beginning of the filtered packets.
-
-The additional argument specifies which packets should be filtered.
-It accepts the values:
-@table @samp
-@item a
-add extradata to all key packets, but only if @var{local_header} is
-set in the @option{flags2} codec context field
-
-@item k
-add extradata to all key packets
-
-@item e
-add extradata to all packets
-@end table
-
-If not specified it is assumed @samp{k}.
-
-For example the following @command{ffmpeg} command forces a global
-header (thus disabling individual packet headers) in the H.264 packets
-generated by the @code{libx264} encoder, but corrects them by adding
-the header stored in extradata to the key packets:
-@example
-ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
-@end example
+@section dump_extradata
 
 @section h264_mp4toannexb
 
@@ -83,18 +37,7 @@ format with @command{ffmpeg}, you can use the command:
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
 @end example
 
-@section imxdump
-
-Modifies the bitstream to fit in MOV and to be usable by the Final Cut
-Pro decoder. This filter only applies to the mpeg2video codec, and is
-likely not needed for Final Cut Pro 7 and newer with the appropriate
-@option{-tag:v}.
-
-For example, to remux 30 MB/sec NTSC IMX to MOV:
-
-@example
-ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
-@end example
+@section imx_dump_header
 
 @section mjpeg2jpeg
 
@@ -137,24 +80,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section movsub
 
+@section mp3_header_compress
+
 @section mp3_header_decompress
 
 @section noise
 
-Damages the contents of packets without damaging the container. Can be
-used for fuzzing or testing error resilience/concealment.
-
-Parameters:
-A numeral string, whose value is related to how often output bytes will
-be modified. Therefore, values below or equal to 0 are forbidden, and
-the lower the more frequent bytes will be modified, with 1 meaning
-every byte is modified.
-
-@example
-ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
-@end example
-applies the modification to every byte.
-
-@section remove_extra
+@section remove_extradata
 
 @c man end BITSTREAM FILTERS

doc/build_system.txt

@@ -25,9 +25,6 @@ fate-list
 install
     Install headers, libraries and programs.
 
-examples
-    Build all examples located in doc/examples.
-
 libavformat/output-example
     Build the libavformat basic example.
 
@@ -37,9 +34,6 @@ libavcodec/api-example
 libswscale/swscale-test
     Build the swscale self-test (useful also as example).
 
-config
-    Reconfigure the project with current configuration.
-
 
 Useful standard make commands:
 make -t <target>

doc/decoders.texi

@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
 @code{--disable-decoder=@var{DECODER}}.
 
-The option @code{-decoders} of the ff* tools will display the list of
+The option @code{-codecs} of the ff* tools will display the list of
 enabled decoders.
 
 @c man end DECODERS
@@ -52,37 +52,6 @@ top-field-first is assumed
 @chapter Audio Decoders
 @c man begin AUDIO DECODERS
 
-A description of some of the currently available audio decoders
-follows.
-
-@section ac3
-
-AC-3 audio decoder.
-
-This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
-the undocumented RealAudio 3 (a.k.a. dnet).
-
-@subsection AC-3 Decoder Options
-
-@table @option
-
-@item -drc_scale @var{value}
-Dynamic Range Scale Factor. The factor to apply to dynamic range values
-from the AC-3 stream. This factor is applied exponentially.
-There are 3 notable scale factor ranges:
-@table @option
-@item drc_scale == 0
-DRC disabled. Produces full range audio.
-@item 0 < drc_scale <= 1
-DRC enabled.  Applies a fraction of the stream DRC value.
-Audio reproduction is between full range and full compression.
-@item drc_scale > 1
-DRC enabled. Applies drc_scale asymmetrically.
-Loud sounds are fully compressed.  Soft sounds are enhanced.
-@end table
-
-@end table
-
 @section ffwavesynth
 
 Internal wave synthetizer.
@@ -93,7 +62,7 @@ documented.
 
 @section libcelt
 
-libcelt decoder wrapper.
+libcelt decoder wrapper
 
 libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
 Requires the presence of the libcelt headers and library during configuration.
@@ -101,7 +70,7 @@ You need to explicitly configure the build with @code{--enable-libcelt}.
 
 @section libgsm
 
-libgsm decoder wrapper.
+libgsm decoder wrapper
 
 libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
 the presence of the libgsm headers and library during configuration. You need
@@ -111,7 +80,7 @@ This decoder supports both the ordinary GSM and the Microsoft variant.
 
 @section libilbc
 
-libilbc decoder wrapper.
+libilbc decoder wrapper
 
 libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
 audio codec. Requires the presence of the libilbc headers and library during
@@ -132,7 +101,7 @@ value is 0 (disabled).
 
 @section libopencore-amrnb
 
-libopencore-amrnb decoder wrapper.
+libopencore-amrnb decoder wrapper
 
 libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
 Narrowband audio codec. Using it requires the presence of the
@@ -154,92 +123,4 @@ explicitly configure the build with @code{--enable-libopencore-amrwb}.
 An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
 without this library.
 
-@section libopus
-
-libopus decoder wrapper.
-
-libopus allows libavcodec to decode the Opus Interactive Audio Codec.
-Requires the presence of the libopus headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libopus}.
-
-An FFmpeg native decoder for Opus exists, so users can decode Opus
-without this library.
-
 @c man end AUDIO DECODERS
-
-@chapter Subtitles Decoders
-@c man begin SUBTILES DECODERS
-
-@section dvdsub
-
-This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
-also be found in VobSub file pairs and in some Matroska files.
-
-@subsection Options
-
-@table @option
-@item palette
-Specify the global palette used by the bitmaps. When stored in VobSub, the
-palette is normally specified in the index file; in Matroska, the palette is
-stored in the codec extra-data in the same format as in VobSub. In DVDs, the
-palette is stored in the IFO file, and therefore not available when reading
-from dumped VOB files.
-
-The format for this option is a string containing 16 24-bits hexadecimal
-numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
-ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
-7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
-
-@item ifo_palette
-Specify the IFO file from which the global palette is obtained.
-(experimental)
-
-@item forced_subs_only
-Only decode subtitle entries marked as forced. Some titles have forced
-and non-forced subtitles in the same track. Setting this flag to @code{1}
-will only keep the forced subtitles. Default value is @code{0}.
-@end table
-
-@section libzvbi-teletext
-
-Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
-subtitles. Requires the presence of the libzvbi headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libzvbi}.
-
-@subsection Options
-
-@table @option
-@item txt_page
-List of teletext page numbers to decode. You may use the special * string to
-match all pages. Pages that do not match the specified list are dropped.
-Default value is *.
-@item txt_chop_top
-Discards the top teletext line. Default value is 1.
-@item txt_format
-Specifies the format of the decoded subtitles. The teletext decoder is capable
-of decoding the teletext pages to bitmaps or to simple text, you should use
-"bitmap" for teletext pages, because certain graphics and colors cannot be
-expressed in simple text. You might use "text" for teletext based subtitles if
-your application can handle simple text based subtitles. Default value is
-bitmap.
-@item txt_left
-X offset of generated bitmaps, default is 0.
-@item txt_top
-Y offset of generated bitmaps, default is 0.
-@item txt_chop_spaces
-Chops leading and trailing spaces and removes empty lines from the generated
-text. This option is useful for teletext based subtitles where empty spaces may
-be present at the start or at the end of the lines or empty lines may be
-present between the subtitle lines because of double-sized teletext charactes.
-Default value is 1.
-@item txt_duration
-Sets the display duration of the decoded teletext pages or subtitles in
-miliseconds. Default value is 30000 which is 30 seconds.
-@item txt_transparent
-Force transparent background of the generated teletext bitmaps. Default value
-is 0 which means an opaque (black) background.
-@end table
-
-@c man end SUBTILES DECODERS

doc/default.css

@@ -1,165 +0,0 @@
-a.summary-letter {
-    text-decoration: none;
-}
-
-a {
-    color: #2D6198;
-}
-
-a:visited {
-    color: #884488;
-}
-
-#banner {
-    background-color: white;
-    position: relative;
-    text-align: center;
-}
-
-#banner img {
-    margin-bottom: 1px;
-    margin-top: 5px;
-}
-
-#body {
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-body {
-    background-color: #313131;
-    margin: 0;
-    text-align: justify;
-}
-
-.center {
-    margin-left: auto;
-    margin-right: auto;
-    text-align: center;
-}
-
-#container {
-    background-color: white;
-    color: #202020;
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-#footer {
-    text-align: center;
-}
-
-h1 a, h2 a, h3 a, h4 a {
-    text-decoration: inherit;
-    color: inherit;
-}
-
-h1, h2, h3, h4 {
-    padding-left: 0.4em;
-    border-radius: 4px;
-    padding-bottom: 0.25em;
-    padding-top: 0.25em;
-    border: 1px solid #6A996A;
-}
-
-h1 {
-    background-color: #7BB37B;
-    color: #151515;
-    font-size: 1.2em;
-    padding-bottom: 0.3em;
-    padding-top: 0.3em;
-}
-
-h2 {
-    color: #313131;
-    font-size: 1.0em;
-    background-color: #ABE3AB;
-}
-
-h3 {
-    color: #313131;
-    font-size: 0.9em;
-    margin-bottom: -6px;
-    background-color: #BBF3BB;
-}
-
-h4 {
-    color: #313131;
-    font-size: 0.8em;
-    margin-bottom: -8px;
-    background-color: #D1FDD1;
-}
-
-img {
-    border: 0;
-}
-
-#navbar {
-    background-color: #738073;
-    border-bottom: 1px solid #5C665C;
-    border-top: 1px solid #5C665C;
-    margin-top: 12px;
-    padding: 0.3em;
-    position: relative;
-    text-align: center;
-}
-
-#navbar a, #navbar_secondary a {
-    color: white;
-    padding: 0.3em;
-    text-decoration: none;
-}
-
-#navbar a:hover, #navbar_secondary a:hover {
-    background-color: #313131;
-    color: white;
-    text-decoration: none;
-}
-
-#navbar_secondary {
-    background-color: #738073;
-    border-bottom: 1px solid #5C665C;
-    border-left: 1px solid #5C665C;
-    border-right: 1px solid #5C665C;
-    padding: 0.3em;
-    position: relative;
-    text-align: center;
-}
-
-p {
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-pre {
-    margin-left: 3em;
-    margin-right: 3em;
-    padding: 0.3em;
-    border: 1px solid #bbb;
-    background-color: #f7f7f7;
-}
-
-dl dt {
-    font-weight: bold;
-}
-
-#proj_desc {
-    font-size: 1.2em;
-}
-
-#repos {
-    margin-left: 1em;
-    margin-right: 1em;
-    border-collapse: collapse;
-    border: solid 1px #6A996A;
-}
-
-#repos th {
-    background-color: #7BB37B;
-    border: solid 1px #6A996A;
-}
-
-#repos td {
-    padding: 0.2em;
-    border: solid 1px #6A996A;
-}

doc/demuxers.texi

@@ -1,235 +1,23 @@
 @chapter Demuxers
 @c man begin DEMUXERS
 
-Demuxers are configured elements in FFmpeg that can read the
+Demuxers are configured elements in FFmpeg which allow to read the
 multimedia streams from a particular type of file.
 
 When you configure your FFmpeg build, all the supported demuxers
 are enabled by default. You can list all available ones using the
-configure option @code{--list-demuxers}.
+configure option "--list-demuxers".
 
 You can disable all the demuxers using the configure option
-@code{--disable-demuxers}, and selectively enable a single demuxer with
-the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
-with the option @code{--disable-demuxer=@var{DEMUXER}}.
+"--disable-demuxers", and selectively enable a single demuxer with
+the option "--enable-demuxer=@var{DEMUXER}", or disable it
+with the option "--disable-demuxer=@var{DEMUXER}".
 
-The option @code{-formats} of the ff* tools will display the list of
+The option "-formats" of the ff* tools will display the list of
 enabled demuxers.
 
 The description of some of the currently available demuxers follows.
 
-@section applehttp
-
-Apple HTTP Live Streaming demuxer.
-
-This demuxer presents all AVStreams from all variant streams.
-The id field is set to the bitrate variant index number. By setting
-the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-the caller can decide which variant streams to actually receive.
-The total bitrate of the variant that the stream belongs to is
-available in a metadata key named "variant_bitrate".
-
-@section apng
-
-Animated Portable Network Graphics demuxer.
-
-This demuxer is used to demux APNG files.
-All headers, but the PNG signature, up to (but not including) the first
-fcTL chunk are transmitted as extradata.
-Frames are then split as being all the chunks between two fcTL ones, or
-between the last fcTL and IEND chunks.
-
-@table @option
-@item -ignore_loop @var{bool}
-Ignore the loop variable in the file if set.
-@item -max_fps @var{int}
-Maximum framerate in frames per second (0 for no limit).
-@item -default_fps @var{int}
-Default framerate in frames per second when none is specified in the file
-(0 meaning as fast as possible).
-@end table
-
-@section asf
-
-Advanced Systems Format demuxer.
-
-This demuxer is used to demux ASF files and MMS network streams.
-
-@table @option
-@item -no_resync_search @var{bool}
-Do not try to resynchronize by looking for a certain optional start code.
-@end table
-
-@anchor{concat}
-@section concat
-
-Virtual concatenation script demuxer.
-
-This demuxer reads a list of files and other directives from a text file and
-demuxes them one after the other, as if all their packet had been muxed
-together.
-
-The timestamps in the files are adjusted so that the first file starts at 0
-and each next file starts where the previous one finishes. Note that it is
-done globally and may cause gaps if all streams do not have exactly the same
-length.
-
-All files must have the same streams (same codecs, same time base, etc.).
-
-The duration of each file is used to adjust the timestamps of the next file:
-if the duration is incorrect (because it was computed using the bit-rate or
-because the file is truncated, for example), it can cause artifacts. The
-@code{duration} directive can be used to override the duration stored in
-each file.
-
-@subsection Syntax
-
-The script is a text file in extended-ASCII, with one directive per line.
-Empty lines, leading spaces and lines starting with '#' are ignored. The
-following directive is recognized:
-
-@table @option
-
-@item @code{file @var{path}}
-Path to a file to read; special characters and spaces must be escaped with
-backslash or single quotes.
-
-All subsequent file-related directives apply to that file.
-
-@item @code{ffconcat version 1.0}
-Identify the script type and version. It also sets the @option{safe} option
-to 1 if it was -1.
-
-To make FFmpeg recognize the format automatically, this directive must
-appears exactly as is (no extra space or byte-order-mark) on the very first
-line of the script.
-
-@item @code{duration @var{dur}}
-Duration of the file. This information can be specified from the file;
-specifying it here may be more efficient or help if the information from the
-file is not available or accurate.
-
-If the duration is set for all files, then it is possible to seek in the
-whole concatenated video.
-
-@item @code{stream}
-Introduce a stream in the virtual file.
-All subsequent stream-related directives apply to the last introduced
-stream.
-Some streams properties must be set in order to allow identifying the
-matching streams in the subfiles.
-If no streams are defined in the script, the streams from the first file are
-copied.
-
-@item @code{exact_stream_id @var{id}}
-Set the id of the stream.
-If this directive is given, the string with the corresponding id in the
-subfiles will be used.
-This is especially useful for MPEG-PS (VOB) files, where the order of the
-streams is not reliable.
-
-@end table
-
-@subsection Options
-
-This demuxer accepts the following option:
-
-@table @option
-
-@item safe
-If set to 1, reject unsafe file paths. A file path is considered safe if it
-does not contain a protocol specification and is relative and all components
-only contain characters from the portable character set (letters, digits,
-period, underscore and hyphen) and have no period at the beginning of a
-component.
-
-If set to 0, any file name is accepted.
-
-The default is 1.
-
--1 is equivalent to 1 if the format was automatically
-probed and 0 otherwise.
-
-@item auto_convert
-If set to 1, try to perform automatic conversions on packet data to make the
-streams concatenable.
-
-Currently, the only conversion is adding the h264_mp4toannexb bitstream
-filter to H.264 streams in MP4 format. This is necessary in particular if
-there are resolution changes.
-
-@end table
-
-@section flv
-
-Adobe Flash Video Format demuxer.
-
-This demuxer is used to demux FLV files and RTMP network streams.
-
-@table @option
-@item -flv_metadata @var{bool}
-Allocate the streams according to the onMetaData array content.
-@end table
-
-@section libgme
-
-The Game Music Emu library is a collection of video game music file emulators.
-
-See @url{http://code.google.com/p/game-music-emu/} for more information.
-
-Some files have multiple tracks. The demuxer will pick the first track by
-default. The @option{track_index} option can be used to select a different
-track. Track indexes start at 0. The demuxer exports the number of tracks as
-@var{tracks} meta data entry.
-
-For very large files, the @option{max_size} option may have to be adjusted.
-
-@section libquvi
-
-Play media from Internet services using the quvi project.
-
-The demuxer accepts a @option{format} option to request a specific quality. It
-is by default set to @var{best}.
-
-See @url{http://quvi.sourceforge.net/} for more information.
-
-FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
-enabled.
-
-@section gif
-
-Animated GIF demuxer.
-
-It accepts the following options:
-
-@table @option
-@item min_delay
-Set the minimum valid delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 2.
-
-@item default_delay
-Set the default delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 10.
-
-@item ignore_loop
-GIF files can contain information to loop a certain number of times (or
-infinitely). If @option{ignore_loop} is set to 1, then the loop setting
-from the input will be ignored and looping will not occur. If set to 0,
-then looping will occur and will cycle the number of times according to
-the GIF. Default value is 1.
-@end table
-
-For example, with the overlay filter, place an infinitely looping GIF
-over another video:
-@example
-ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
-@end example
-
-Note that in the above example the shortest option for overlay filter is
-used to end the output video at the length of the shortest input file,
-which in this case is @file{input.mp4} as the GIF in this example loops
-infinitely.
-
 @section image2
 
 Image file demuxer.
@@ -247,7 +35,7 @@ same for all the files in the sequence.
 This demuxer accepts the following options:
 @table @option
 @item framerate
-Set the frame rate for the video stream. It defaults to 25.
+Set the framerate for the video stream. It defaults to 25.
 @item loop
 If set to 1, loop over the input. Default value is 0.
 @item pattern_type
@@ -325,12 +113,6 @@ to read from. Default value is 0.
 Set the index interval range to check when looking for the first image
 file in the sequence, starting from @var{start_number}. Default value
 is 5.
-@item ts_from_file
-If set to 1, will set frame timestamp to modification time of image file. Note
-that monotonity of timestamps is not provided: images go in the same order as
-without this option. Default value is 0.
-If set to 2, will set frame timestamp to the modification time of the image file in
-nanosecond precision.
 @item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
@@ -344,81 +126,33 @@ Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
 @example
-ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
 @end example
 
 @item
 As above, but start by reading from a file with index 100 in the sequence:
 @example
-ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
 @end example
 
 @item
 Read images matching the "*.png" glob pattern , that is all the files
 terminating with the ".png" suffix:
 @example
-ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
+ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
 @end example
 @end itemize
 
-@section mov/mp4/3gp/Quicktme
+@section applehttp
 
-Quicktime / MP4 demuxer.
+Apple HTTP Live Streaming demuxer.
 
-This demuxer accepts the following options:
-@table @option
-@item enable_drefs
-Enable loading of external tracks, disabled by default.
-Enabling this can theoretically leak information in some use cases.
-
-@item use_absolute_path
-Allows loading of external tracks via absolute paths, disabled by default.
-Enabling this poses a security risk. It should only be enabled if the source
-is known to be non malicious.
-
-@end table
-
-@section mpegts
-
-MPEG-2 transport stream demuxer.
-
-@table @option
-
-@item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
-from the PCR of the first program which the teletext stream is part of and is
-not discarded. Default value is 1, set this option to 0 if you want your
-teletext packet PTS and DTS values untouched.
-@end table
-
-@section rawvideo
-
-Raw video demuxer.
-
-This demuxer allows one to read raw video data. Since there is no header
-specifying the assumed video parameters, the user must specify them
-in order to be able to decode the data correctly.
-
-This demuxer accepts the following options:
-@table @option
-
-@item framerate
-Set input video frame rate. Default value is 25.
-
-@item pixel_format
-Set the input video pixel format. Default value is @code{yuv420p}.
-
-@item video_size
-Set the input video size. This value must be specified explicitly.
-@end table
-
-For example to read a rawvideo file @file{input.raw} with
-@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
-size of @code{320x240}, and a frame rate of 10 images per second, use
-the command:
-@example
-ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
-@end example
+This demuxer presents all AVStreams from all variant streams.
+The id field is set to the bitrate variant index number. By setting
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
+the caller can decide which variant streams to actually receive.
+The total bitrate of the variant that the stream belongs to is
+available in a metadata key named "variant_bitrate".
 
 @section sbg
 
@@ -450,25 +184,4 @@ the script is directly played, the actual times will match the absolute
 timestamps up to the sound controller's clock accuracy, but if the user
 somehow pauses the playback or seeks, all times will be shifted accordingly.
 
-@section tedcaptions
-
-JSON captions used for @url{http://www.ted.com/, TED Talks}.
-
-TED does not provide links to the captions, but they can be guessed from the
-page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
-contains a bookmarklet to expose them.
-
-This demuxer accepts the following option:
-@table @option
-@item start_time
-Set the start time of the TED talk, in milliseconds. The default is 15000
-(15s). It is used to sync the captions with the downloadable videos, because
-they include a 15s intro.
-@end table
-
-Example: convert the captions to a format most players understand:
-@example
-ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
-@end example
-
-@c man end DEMUXERS
+@c man end INPUT DEVICES

doc/developer.texi

@@ -11,23 +11,29 @@
 
 @chapter Developers Guide
 
-@section Notes for external developers
+@section API
+@itemize @bullet
+@item libavcodec is the library containing the codecs (both encoding and
+decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
+it.
 
-This document is mostly useful for internal FFmpeg developers.
-External developers who need to use the API in their application should
-refer to the API doxygen documentation in the public headers, and
-check the examples in @file{doc/examples} and in the source code to
-see how the public API is employed.
+@item libavformat is the library containing the file format handling (mux and
+demux code for several formats). Look at @file{ffplay.c} to use it in a
+player. See @file{doc/examples/muxing.c} to use it to generate audio or video
+streams.
 
-You can use the FFmpeg libraries in your commercial program, but you
-are encouraged to @emph{publish any patch you make}. In this case the
-best way to proceed is to send your patches to the ffmpeg-devel
-mailing list following the guidelines illustrated in the remainder of
-this document.
+@end itemize
 
-For more detailed legal information about the use of FFmpeg in
-external programs read the @file{LICENSE} file in the source tree and
-consult @url{http://ffmpeg.org/legal.html}.
+@section Integrating libavcodec or libavformat in your program
+
+You can integrate all the source code of the libraries to link them
+statically to avoid any version problem. All you need is to provide a
+'config.mak' and a 'config.h' in the parent directory. See the defines
+generated by ./configure to understand what is needed.
+
+You can use libavcodec or libavformat in your commercial program, but
+@emph{any patch you make must be published}. The best way to proceed is
+to send your patches to the FFmpeg mailing list.
 
 @section Contributing
 
@@ -51,16 +57,13 @@ and should try to fix issues their commit causes.
 @subsection Code formatting conventions
 
 There are the following guidelines regarding the indentation in files:
-
 @itemize @bullet
 @item
 Indent size is 4.
-
 @item
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 rejected by the git repository.
-
 @item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@@ -92,7 +95,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
  * more text ...
  * ...
  */
-typedef struct Foobar @{
+typedef struct Foobar@{
     int var1; /**< var1 description */
     int var2; ///< var2 description
     /** var3 description */
@@ -114,17 +117,13 @@ int myfunc(int my_parameter)
 
 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
-
 @itemize @bullet
 @item
 the @samp{inline} keyword;
-
 @item
 @samp{//} comments;
-
 @item
 designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
-
 @item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};})
 @end itemize
@@ -136,72 +135,46 @@ clarity and performance.
 All code must compile with recent versions of GCC and a number of other
 currently supported compilers. To ensure compatibility, please do not use
 additional C99 features or GCC extensions. Especially watch out for:
-
 @itemize @bullet
 @item
 mixing statements and declarations;
-
 @item
 @samp{long long} (use @samp{int64_t} instead);
-
 @item
 @samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
-
 @item
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 @end itemize
 
 @subsection Naming conventions
-All names should be composed with underscores (_), not CamelCase. For example,
-@samp{avfilter_get_video_buffer} is an acceptable function name and
-@samp{AVFilterGetVideo} is not. The exception from this are type names, like
+All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
+a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
 for example structs and enums; they should always be in the CamelCase
 
-There are the following conventions for naming variables and functions:
 
+There are following conventions for naming variables and functions:
 @itemize @bullet
 @item
 For local variables no prefix is required.
-
 @item
-For file-scope variables and functions declared as @code{static}, no prefix
-is required.
-
+For variables and functions declared as @code{static} no prefixes are required.
 @item
-For variables and functions visible outside of file scope, but only used
-internally by a library, an @code{ff_} prefix should be used,
-e.g. @samp{ff_w64_demuxer}.
-
+For variables and functions used internally by the library, @code{ff_} prefix
+should be used.
+For example, @samp{ff_w64_demuxer}.
 @item
-For variables and functions visible outside of file scope, used internally
-across multiple libraries, use @code{avpriv_} as prefix, for example,
-@samp{avpriv_aac_parse_header}.
-
+For variables and functions used internally across multiple libraries, use
+@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
 @item
-Each library has its own prefix for public symbols, in addition to the
-commonly used @code{av_} (@code{avformat_} for libavformat,
-@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
-Check the existing code and choose names accordingly.
-Note that some symbols without these prefixes are also exported for
-retro-compatibility reasons. These exceptions are declared in the
-@code{lib<name>/lib<name>.v} files.
+For exported names, each library has its own prefixes. Just check the existing
+code and name accordingly.
 @end itemize
 
-Furthermore, name space reserved for the system should not be invaded.
-Identifiers ending in @code{_t} are reserved by
-@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
-Also avoid names starting with @code{__} or @code{_} followed by an uppercase
-letter as they are reserved by the C standard. Names starting with @code{_}
-are reserved at the file level and may not be used for externally visible
-symbols. If in doubt, just avoid names starting with @code{_} altogether.
-
-@subsection Miscellaneous conventions
-
+@subsection Miscellanous conventions
 @itemize @bullet
 @item
 fprintf and printf are forbidden in libavformat and libavcodec,
 please use av_log() instead.
-
 @item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@@ -217,8 +190,8 @@ set shiftwidth=4
 set softtabstop=4
 set cindent
 set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" allow tabs in Makefiles
+autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
 " Trailing whitespace and tabs are forbidden, so highlight them.
 highlight ForbiddenWhitespace ctermbg=red guibg=red
 match ForbiddenWhitespace /\s\+$\|\t/
@@ -244,161 +217,124 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 
 @enumerate
 @item
-Contributions should be licensed under the
-@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
-including an "or any later version" clause, or, if you prefer
-a gift-style license, the
-@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
-@uref{http://mit-license.org/, MIT} license.
-@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
-an "or any later version" clause is also acceptable, but LGPL is
-preferred.
-If you add a new file, give it a proper license header. Do not copy and
-paste it from a random place, use an existing file as template.
-
+   Contributions should be licensed under the LGPL 2.1, including an
+   "or any later version" clause, or the MIT license.  GPL 2 including
+   an "or any later version" clause is also acceptable, but LGPL is
+   preferred.
 @item
-You must not commit code which breaks FFmpeg! (Meaning unfinished but
-enabled code which breaks compilation or compiles but does not work or
-breaks the regression tests)
-You can commit unfinished stuff (for testing etc), but it must be disabled
-(#ifdef etc) by default so it does not interfere with other developers'
-work.
-
+   You must not commit code which breaks FFmpeg! (Meaning unfinished but
+   enabled code which breaks compilation or compiles but does not work or
+   breaks the regression tests)
+   You can commit unfinished stuff (for testing etc), but it must be disabled
+   (#ifdef etc) by default so it does not interfere with other developers'
+   work.
 @item
-The commit message should have a short first line in the form of
-a @samp{topic: short description} as a header, separated by a newline
-from the body consisting of an explanation of why the change is necessary.
-If the commit fixes a known bug on the bug tracker, the commit message
-should include its bug ID. Referring to the issue on the bug tracker does
-not exempt you from writing an excerpt of the bug in the commit message.
-
+   You do not have to over-test things. If it works for you, and you think it
+   should work for others, then commit. If your code has problems
+   (portability, triggers compiler bugs, unusual environment etc) they will be
+   reported and eventually fixed.
 @item
-You do not have to over-test things. If it works for you, and you think it
-should work for others, then commit. If your code has problems
-(portability, triggers compiler bugs, unusual environment etc) they will be
-reported and eventually fixed.
-
+   Do not commit unrelated changes together, split them into self-contained
+   pieces. Also do not forget that if part B depends on part A, but A does not
+   depend on B, then A can and should be committed first and separate from B.
+   Keeping changes well split into self-contained parts makes reviewing and
+   understanding them on the commit log mailing list easier. This also helps
+   in case of debugging later on.
+   Also if you have doubts about splitting or not splitting, do not hesitate to
+   ask/discuss it on the developer mailing list.
 @item
-Do not commit unrelated changes together, split them into self-contained
-pieces. Also do not forget that if part B depends on part A, but A does not
-depend on B, then A can and should be committed first and separate from B.
-Keeping changes well split into self-contained parts makes reviewing and
-understanding them on the commit log mailing list easier. This also helps
-in case of debugging later on.
-Also if you have doubts about splitting or not splitting, do not hesitate to
-ask/discuss it on the developer mailing list.
+   Do not change behavior of the programs (renaming options etc) or public
+   API or ABI without first discussing it on the ffmpeg-devel mailing list.
+   Do not remove functionality from the code. Just improve!
 
+   Note: Redundant code can be removed.
 @item
-Do not change behavior of the programs (renaming options etc) or public
-API or ABI without first discussing it on the ffmpeg-devel mailing list.
-Do not remove functionality from the code. Just improve!
-
-Note: Redundant code can be removed.
-
+   Do not commit changes to the build system (Makefiles, configure script)
+   which change behavior, defaults etc, without asking first. The same
+   applies to compiler warning fixes, trivial looking fixes and to code
+   maintained by other developers. We usually have a reason for doing things
+   the way we do. Send your changes as patches to the ffmpeg-devel mailing
+   list, and if the code maintainers say OK, you may commit. This does not
+   apply to files you wrote and/or maintain.
 @item
-Do not commit changes to the build system (Makefiles, configure script)
-which change behavior, defaults etc, without asking first. The same
-applies to compiler warning fixes, trivial looking fixes and to code
-maintained by other developers. We usually have a reason for doing things
-the way we do. Send your changes as patches to the ffmpeg-devel mailing
-list, and if the code maintainers say OK, you may commit. This does not
-apply to files you wrote and/or maintain.
+   We refuse source indentation and other cosmetic changes if they are mixed
+   with functional changes, such commits will be rejected and removed. Every
+   developer has his own indentation style, you should not change it. Of course
+   if you (re)write something, you can use your own style, even though we would
+   prefer if the indentation throughout FFmpeg was consistent (Many projects
+   force a given indentation style - we do not.). If you really need to make
+   indentation changes (try to avoid this), separate them strictly from real
+   changes.
 
+   NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
+   then either do NOT change the indentation of the inner part within (do not
+   move it to the right)! or do so in a separate commit
 @item
-We refuse source indentation and other cosmetic changes if they are mixed
-with functional changes, such commits will be rejected and removed. Every
-developer has his own indentation style, you should not change it. Of course
-if you (re)write something, you can use your own style, even though we would
-prefer if the indentation throughout FFmpeg was consistent (Many projects
-force a given indentation style - we do not.). If you really need to make
-indentation changes (try to avoid this), separate them strictly from real
-changes.
-
-NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
-then either do NOT change the indentation of the inner part within (do not
-move it to the right)! or do so in a separate commit
+   Always fill out the commit log message. Describe in a few lines what you
+   changed and why. You can refer to mailing list postings if you fix a
+   particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+   Recommended format:
+   area changed: Short 1 line description
 
+   details describing what and why and giving references.
 @item
-Always fill out the commit log message. Describe in a few lines what you
-changed and why. You can refer to mailing list postings if you fix a
-particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
-Recommended format:
-
-@example
-area changed: Short 1 line description
-
-details describing what and why and giving references.
-@end example
-
+   Make sure the author of the commit is set correctly. (see git commit --author)
+   If you apply a patch, send an
+   answer to ffmpeg-devel (or wherever you got the patch from) saying that
+   you applied the patch.
 @item
-Make sure the author of the commit is set correctly. (see git commit --author)
-If you apply a patch, send an
-answer to ffmpeg-devel (or wherever you got the patch from) saying that
-you applied the patch.
-
+   When applying patches that have been discussed (at length) on the mailing
+   list, reference the thread in the log message.
 @item
-When applying patches that have been discussed (at length) on the mailing
-list, reference the thread in the log message.
-
+    Do NOT commit to code actively maintained by others without permission.
+    Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+    timeframe (12h for build failures and security fixes, 3 days small changes,
+    1 week for big patches) then commit your patch if you think it is OK.
+    Also note, the maintainer can simply ask for more time to review!
 @item
-Do NOT commit to code actively maintained by others without permission.
-Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-timeframe (12h for build failures and security fixes, 3 days small changes,
-1 week for big patches) then commit your patch if you think it is OK.
-Also note, the maintainer can simply ask for more time to review!
-
+    Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+    are sent there and reviewed by all the other developers. Bugs and possible
+    improvements or general questions regarding commits are discussed there. We
+    expect you to react if problems with your code are uncovered.
 @item
-Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-are sent there and reviewed by all the other developers. Bugs and possible
-improvements or general questions regarding commits are discussed there. We
-expect you to react if problems with your code are uncovered.
-
+    Update the documentation if you change behavior or add features. If you are
+    unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+    maintainer(s) will review and commit your stuff.
 @item
-Update the documentation if you change behavior or add features. If you are
-unsure how best to do this, send a patch to ffmpeg-devel, the documentation
-maintainer(s) will review and commit your stuff.
-
+    Try to keep important discussions and requests (also) on the public
+    developer mailing list, so that all developers can benefit from them.
 @item
-Try to keep important discussions and requests (also) on the public
-developer mailing list, so that all developers can benefit from them.
-
+    Never write to unallocated memory, never write over the end of arrays,
+    always check values read from some untrusted source before using them
+    as array index or other risky things.
 @item
-Never write to unallocated memory, never write over the end of arrays,
-always check values read from some untrusted source before using them
-as array index or other risky things.
-
+    Remember to check if you need to bump versions for the specific libav*
+    parts (libavutil, libavcodec, libavformat) you are changing. You need
+    to change the version integer.
+    Incrementing the first component means no backward compatibility to
+    previous versions (e.g. removal of a function from the public API).
+    Incrementing the second component means backward compatible change
+    (e.g. addition of a function to the public API or extension of an
+    existing data structure).
+    Incrementing the third component means a noteworthy binary compatible
+    change (e.g. encoder bug fix that matters for the decoder).
 @item
-Remember to check if you need to bump versions for the specific libav*
-parts (libavutil, libavcodec, libavformat) you are changing. You need
-to change the version integer.
-Incrementing the first component means no backward compatibility to
-previous versions (e.g. removal of a function from the public API).
-Incrementing the second component means backward compatible change
-(e.g. addition of a function to the public API or extension of an
-existing data structure).
-Incrementing the third component means a noteworthy binary compatible
-change (e.g. encoder bug fix that matters for the decoder). The third
-component always starts at 100 to distinguish FFmpeg from Libav.
-
+    Compiler warnings indicate potential bugs or code with bad style. If a type of
+    warning always points to correct and clean code, that warning should
+    be disabled, not the code changed.
+    Thus the remaining warnings can either be bugs or correct code.
+    If it is a bug, the bug has to be fixed. If it is not, the code should
+    be changed to not generate a warning unless that causes a slowdown
+    or obfuscates the code.
 @item
-Compiler warnings indicate potential bugs or code with bad style. If a type of
-warning always points to correct and clean code, that warning should
-be disabled, not the code changed.
-Thus the remaining warnings can either be bugs or correct code.
-If it is a bug, the bug has to be fixed. If it is not, the code should
-be changed to not generate a warning unless that causes a slowdown
-or obfuscates the code.
-
-@item
-Make sure that no parts of the codebase that you maintain are missing from the
-@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
-your name after it.
-If at some point you no longer want to maintain some code, then please help
-finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
+    If you add a new file, give it a proper license header. Do not copy and
+    paste it from a random place, use an existing file as template.
 @end enumerate
 
 We think our rules are not too hard. If you have comments, contact us.
 
+Note, these rules are mostly borrowed from the MPlayer project.
+
 @anchor{Submitting patches}
 @section Submitting patches
 
@@ -421,6 +357,11 @@ The tool is located in the tools directory.
 Run the @ref{Regression tests} before submitting a patch in order to verify
 it does not cause unexpected problems.
 
+Patches should be posted as base64 encoded attachments (or any other
+encoding which ensures that the patch will not be trashed during
+transmission) to the ffmpeg-devel mailing list, see
+@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
+
 It also helps quite a bit if you tell us what the patch does (for example
 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
 and has no lrint()')
@@ -428,13 +369,6 @@ and has no lrint()')
 Also please if you send several patches, send each patch as a separate mail,
 do not attach several unrelated patches to the same mail.
 
-Patches should be posted to the
-@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
-mailing list. Use @code{git send-email} when possible since it will properly
-send patches without requiring extra care. If you cannot, then send patches
-as base64-encoded attachments, so your patch is not trashed during
-transmission.
-
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -449,51 +383,40 @@ send a reminder by email. Your patch should eventually be dealt with.
 
 @enumerate
 @item
-Did you use av_cold for codec initialization and close functions?
-
+    Did you use av_cold for codec initialization and close functions?
 @item
-Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
-AVInputFormat/AVOutputFormat struct?
-
+    Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+    AVInputFormat/AVOutputFormat struct?
 @item
-Did you bump the minor version number (and reset the micro version
-number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
-
+    Did you bump the minor version number (and reset the micro version
+    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
 @item
-Did you register it in @file{allcodecs.c} or @file{allformats.c}?
-
+    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
 @item
-Did you add the AVCodecID to @file{avcodec.h}?
-When adding new codec IDs, also add an entry to the codec descriptor
-list in @file{libavcodec/codec_desc.c}.
-
+    Did you add the AVCodecID to @file{avcodec.h}?
+    When adding new codec IDs, also add an entry to the codec descriptor
+    list in @file{libavcodec/codec_desc.c}.
 @item
-If it has a FourCC, did you add it to @file{libavformat/riff.c},
-even if it is only a decoder?
-
+    If it has a fourCC, did you add it to @file{libavformat/riff.c},
+    even if it is only a decoder?
 @item
-Did you add a rule to compile the appropriate files in the Makefile?
-Remember to do this even if you're just adding a format to a file that is
-already being compiled by some other rule, like a raw demuxer.
-
+    Did you add a rule to compile the appropriate files in the Makefile?
+    Remember to do this even if you're just adding a format to a file that is
+    already being compiled by some other rule, like a raw demuxer.
 @item
-Did you add an entry to the table of supported formats or codecs in
-@file{doc/general.texi}?
-
+    Did you add an entry to the table of supported formats or codecs in
+    @file{doc/general.texi}?
 @item
-Did you add an entry in the Changelog?
-
+    Did you add an entry in the Changelog?
 @item
-If it depends on a parser or a library, did you add that dependency in
-configure?
-
+    If it depends on a parser or a library, did you add that dependency in
+    configure?
 @item
-Did you @code{git add} the appropriate files before committing?
-
+    Did you @code{git add} the appropriate files before committing?
 @item
-Did you make sure it compiles standalone, i.e. with
-@code{configure --disable-everything --enable-decoder=foo}
-(or @code{--enable-demuxer} or whatever your component is)?
+    Did you make sure it compiles standalone, i.e. with
+    @code{configure --disable-everything --enable-decoder=foo}
+    (or @code{--enable-demuxer} or whatever your component is)?
 @end enumerate
 
 
@@ -501,109 +424,77 @@ Did you make sure it compiles standalone, i.e. with
 
 @enumerate
 @item
-Does @code{make fate} pass with the patch applied?
-
+    Does @code{make fate} pass with the patch applied?
 @item
-Was the patch generated with git format-patch or send-email?
-
+    Was the patch generated with git format-patch or send-email?
 @item
-Did you sign off your patch? (git commit -s)
-See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-of sign off.
-
+    Did you sign off your patch? (git commit -s)
+    See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
+    of sign off.
 @item
-Did you provide a clear git commit log message?
-
+    Did you provide a clear git commit log message?
 @item
-Is the patch against latest FFmpeg git master branch?
-
+    Is the patch against latest FFmpeg git master branch?
 @item
-Are you subscribed to ffmpeg-devel?
-(the list is subscribers only due to spam)
-
+    Are you subscribed to ffmpeg-devel?
+    (the list is subscribers only due to spam)
 @item
-Have you checked that the changes are minimal, so that the same cannot be
-achieved with a smaller patch and/or simpler final code?
-
+    Have you checked that the changes are minimal, so that the same cannot be
+    achieved with a smaller patch and/or simpler final code?
 @item
-If the change is to speed critical code, did you benchmark it?
-
+    If the change is to speed critical code, did you benchmark it?
 @item
-If you did any benchmarks, did you provide them in the mail?
-
+    If you did any benchmarks, did you provide them in the mail?
 @item
-Have you checked that the patch does not introduce buffer overflows or
-other security issues?
-
+    Have you checked that the patch does not introduce buffer overflows or
+    other security issues?
 @item
-Did you test your decoder or demuxer against damaged data? If no, see
-tools/trasher, the noise bitstream filter, and
-@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
-should not crash, end in a (near) infinite loop, or allocate ridiculous
-amounts of memory when fed damaged data.
-
+    Did you test your decoder or demuxer against damaged data? If no, see
+    tools/trasher and the noise bitstream filter. Your decoder or demuxer
+    should not crash or end in a (near) infinite loop when fed damaged data.
 @item
-Does the patch not mix functional and cosmetic changes?
-
+    Does the patch not mix functional and cosmetic changes?
 @item
-Did you add tabs or trailing whitespace to the code? Both are forbidden.
-
+    Did you add tabs or trailing whitespace to the code? Both are forbidden.
 @item
-Is the patch attached to the email you send?
-
+    Is the patch attached to the email you send?
 @item
-Is the mime type of the patch correct? It should be text/x-diff or
-text/x-patch or at least text/plain and not application/octet-stream.
-
+    Is the mime type of the patch correct? It should be text/x-diff or
+    text/x-patch or at least text/plain and not application/octet-stream.
 @item
-If the patch fixes a bug, did you provide a verbose analysis of the bug?
-
+    If the patch fixes a bug, did you provide a verbose analysis of the bug?
 @item
-If the patch fixes a bug, did you provide enough information, including
-a sample, so the bug can be reproduced and the fix can be verified?
-Note please do not attach samples >100k to mails but rather provide a
-URL, you can upload to ftp://upload.ffmpeg.org
-
+    If the patch fixes a bug, did you provide enough information, including
+    a sample, so the bug can be reproduced and the fix can be verified?
+    Note please do not attach samples >100k to mails but rather provide a
+    URL, you can upload to ftp://upload.ffmpeg.org
 @item
-Did you provide a verbose summary about what the patch does change?
-
+    Did you provide a verbose summary about what the patch does change?
 @item
-Did you provide a verbose explanation why it changes things like it does?
-
+    Did you provide a verbose explanation why it changes things like it does?
 @item
-Did you provide a verbose summary of the user visible advantages and
-disadvantages if the patch is applied?
-
+    Did you provide a verbose summary of the user visible advantages and
+    disadvantages if the patch is applied?
 @item
-Did you provide an example so we can verify the new feature added by the
-patch easily?
-
+    Did you provide an example so we can verify the new feature added by the
+    patch easily?
 @item
-If you added a new file, did you insert a license header? It should be
-taken from FFmpeg, not randomly copied and pasted from somewhere else.
-
+    If you added a new file, did you insert a license header? It should be
+    taken from FFmpeg, not randomly copied and pasted from somewhere else.
 @item
-You should maintain alphabetical order in alphabetically ordered lists as
-long as doing so does not break API/ABI compatibility.
-
+    You should maintain alphabetical order in alphabetically ordered lists as
+    long as doing so does not break API/ABI compatibility.
 @item
-Lines with similar content should be aligned vertically when doing so
-improves readability.
-
+    Lines with similar content should be aligned vertically when doing so
+    improves readability.
 @item
-Consider to add a regression test for your code.
-
+    Consider to add a regression test for your code.
 @item
-If you added YASM code please check that things still work with --disable-yasm
-
+    If you added YASM code please check that things still work with --disable-yasm
 @item
-Make sure you check the return values of function and return appropriate
-error codes. Especially memory allocation functions like @code{av_malloc()}
-are notoriously left unchecked, which is a serious problem.
-
-@item
-Test your code with valgrind and or Address Sanitizer to ensure it's free
-of leaks, out of array accesses, etc.
+    Make sure you check the return values of function and return appropriate
+    error codes. Especially memory allocation functions like @code{av_malloc()}
+    are notoriously left unchecked, which is a serious problem.
 @end enumerate
 
 @section Patch review process
@@ -656,145 +547,4 @@ message or introductionary message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.
 
 
-@subsection Visualizing Test Coverage
-
-The FFmpeg build system allows visualizing the test coverage in an easy
-manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
-the following steps:
-
-@enumerate
-@item
-    Configure to compile with instrumentation enabled:
-    @code{configure --toolchain=gcov}.
-
-@item
-    Run your test case, either manually or via FATE. This can be either
-    the full FATE regression suite, or any arbitrary invocation of any
-    front-end tool provided by FFmpeg, in any combination.
-
-@item
-    Run @code{make lcov} to generate coverage data in HTML format.
-
-@item
-    View @code{lcov/index.html} in your preferred HTML viewer.
-@end enumerate
-
-You can use the command @code{make lcov-reset} to reset the coverage
-measurements. You will need to rerun @code{make lcov} after running a
-new test.
-
-@subsection Using Valgrind
-
-The configure script provides a shortcut for using valgrind to spot bugs
-related to memory handling. Just add the option
-@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
-to your configure line, and reasonable defaults will be set for running
-FATE under the supervision of either the @strong{memcheck} or the
-@strong{massif} tool of the valgrind suite.
-
-In case you need finer control over how valgrind is invoked, use the
-@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
-your configure line instead.
-
-@anchor{Release process}
-@section Release process
-
-FFmpeg maintains a set of @strong{release branches}, which are the
-recommended deliverable for system integrators and distributors (such as
-Linux distributions, etc.). At regular times, a @strong{release
-manager} prepares, tests and publishes tarballs on the
-@url{http://ffmpeg.org} website.
-
-There are two kinds of releases:
-
-@enumerate
-@item
-@strong{Major releases} always include the latest and greatest
-features and functionality.
-
-@item
-@strong{Point releases} are cut from @strong{release} branches,
-which are named @code{release/X}, with @code{X} being the release
-version number.
-@end enumerate
-
-Note that we promise to our users that shared libraries from any FFmpeg
-release never break programs that have been @strong{compiled} against
-previous versions of @strong{the same release series} in any case!
-
-However, from time to time, we do make API changes that require adaptations
-in applications. Such changes are only allowed in (new) major releases and
-require further steps such as bumping library version numbers and/or
-adjustments to the symbol versioning file. Please discuss such changes
-on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
-
-@anchor{Criteria for Point Releases}
-@subsection Criteria for Point Releases
-
-Changes that match the following criteria are valid candidates for
-inclusion into a point release:
-
-@enumerate
-@item
-Fixes a security issue, preferably identified by a @strong{CVE
-number} issued by @url{http://cve.mitre.org/}.
-
-@item
-Fixes a documented bug in @url{https://trac.ffmpeg.org}.
-
-@item
-Improves the included documentation.
-
-@item
-Retains both source code and binary compatibility with previous
-point releases of the same release branch.
-@end enumerate
-
-The order for checking the rules is (1 OR 2 OR 3) AND 4.
-
-
-@subsection Release Checklist
-
-The release process involves the following steps:
-
-@enumerate
-@item
-Ensure that the @file{RELEASE} file contains the version number for
-the upcoming release.
-
-@item
-Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
-
-@item
-Announce the intent to do a release to the mailing list.
-
-@item
-Make sure all relevant security fixes have been backported. See
-@url{https://ffmpeg.org/security.html}.
-
-@item
-Ensure that the FATE regression suite still passes in the release
-branch on at least @strong{i386} and @strong{amd64}
-(cf. @ref{Regression tests}).
-
-@item
-Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
-supplementing files that contain @code{gpg} signatures
-
-@item
-Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
-push an annotated tag in the form @code{nX}, with @code{X}
-containing the version number.
-
-@item
-Propose and send a patch to the @strong{ffmpeg-devel} mailing list
-with a news entry for the website.
-
-@item
-Publish the news entry.
-
-@item
-Send announcement to the mailing list.
-@end enumerate
-
 @bye

doc/devices.texi

@@ -1,25 +0,0 @@
-@chapter Device Options
-@c man begin DEVICE OPTIONS
-
-The libavdevice library provides the same interface as
-libavformat. Namely, an input device is considered like a demuxer, and
-an output device like a muxer, and the interface and generic device
-options are the same provided by libavformat (see the ffmpeg-formats
-manual).
-
-In addition each input or output device may support so-called private
-options, which are specific for that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the device
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-@c man end DEVICE OPTIONS
-
-@ifclear config-writeonly
-@include indevs.texi
-@end ifclear
-@ifclear config-readonly
-@include outdevs.texi
-@end ifclear

doc/doxy-wrapper.sh

@@ -1,13 +0,0 @@
-#!/bin/sh
-
-SRC_PATH="${1}"
-DOXYFILE="${2}"
-DOXYGEN="${3}"
-
-shift 3
-
-$DOXYGEN - <<EOF
-@INCLUDE        = ${DOXYFILE}
-INPUT           = $@
-EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
-EOF

doc/doxy/footer.html

@@ -0,0 +1,10 @@
+ </div>
+
+ <div id="footer">
+   Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+ </div>
+
+</div>
+
+</body>
+</html>

doc/doxy/header.html

@@ -0,0 +1,14 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head>
+<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
+<meta http-equiv="X-UA-Compatible" content="IE=9"/>
+<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
+<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
+<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
+</head>
+
+<div id="container">
+
+<div id="body">
+ <div>

doc/eval.texi

@@ -0,0 +1,252 @@
+@chapter Expression Evaluation
+@c man begin EXPRESSION EVALUATION
+
+When evaluating an arithmetic expression, FFmpeg uses an internal
+formula evaluator, implemented through the @file{libavutil/eval.h}
+interface.
+
+An expression may contain unary, binary operators, constants, and
+functions.
+
+Two expressions @var{expr1} and @var{expr2} can be combined to form
+another expression "@var{expr1};@var{expr2}".
+@var{expr1} and @var{expr2} are evaluated in turn, and the new
+expression evaluates to the value of @var{expr2}.
+
+The following binary operators are available: @code{+}, @code{-},
+@code{*}, @code{/}, @code{^}.
+
+The following unary operators are available: @code{+}, @code{-}.
+
+The following functions are available:
+@table @option
+@item sinh(x)
+Compute hyperbolic sine of @var{x}.
+
+@item cosh(x)
+Compute hyperbolic cosine of @var{x}.
+
+@item tanh(x)
+Compute hyperbolic tangent of @var{x}.
+
+@item sin(x)
+Compute sine of @var{x}.
+
+@item cos(x)
+Compute cosine of @var{x}.
+
+@item tan(x)
+Compute tangent of @var{x}.
+
+@item atan(x)
+Compute arctangent of @var{x}.
+
+@item asin(x)
+Compute arcsine of @var{x}.
+
+@item acos(x)
+Compute arccosine of @var{x}.
+
+@item exp(x)
+Compute exponential of @var{x} (with base @code{e}, the Euler's number).
+
+@item log(x)
+Compute natural logarithm of @var{x}.
+
+@item abs(x)
+Compute absolute value of @var{x}.
+
+@item squish(x)
+Compute expression @code{1/(1 + exp(4*x))}.
+
+@item gauss(x)
+Compute Gauss function of @var{x}, corresponding to
+@code{exp(-x*x/2) / sqrt(2*PI)}.
+
+@item isinf(x)
+Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
+
+@item isnan(x)
+Return 1.0 if @var{x} is NAN, 0.0 otherwise.
+
+@item mod(x, y)
+Compute the remainder of division of @var{x} by @var{y}.
+
+@item max(x, y)
+Return the maximum between @var{x} and @var{y}.
+
+@item min(x, y)
+Return the maximum between @var{x} and @var{y}.
+
+@item eq(x, y)
+Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
+
+@item gte(x, y)
+Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
+
+@item gt(x, y)
+Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
+
+@item lte(x, y)
+Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
+
+@item lt(x, y)
+Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
+
+@item st(var, expr)
+Allow to store the value of the expression @var{expr} in an internal
+variable. @var{var} specifies the number of the variable where to
+store the value, and it is a value ranging from 0 to 9. The function
+returns the value stored in the internal variable.
+Note, Variables are currently not shared between expressions.
+
+@item ld(var)
+Allow to load the value of the internal variable with number
+@var{var}, which was previously stored with st(@var{var}, @var{expr}).
+The function returns the loaded value.
+
+@item while(cond, expr)
+Evaluate expression @var{expr} while the expression @var{cond} is
+non-zero, and returns the value of the last @var{expr} evaluation, or
+NAN if @var{cond} was always false.
+
+@item ceil(expr)
+Round the value of expression @var{expr} upwards to the nearest
+integer. For example, "ceil(1.5)" is "2.0".
+
+@item floor(expr)
+Round the value of expression @var{expr} downwards to the nearest
+integer. For example, "floor(-1.5)" is "-2.0".
+
+@item trunc(expr)
+Round the value of expression @var{expr} towards zero to the nearest
+integer. For example, "trunc(-1.5)" is "-1.0".
+
+@item sqrt(expr)
+Compute the square root of @var{expr}. This is equivalent to
+"(@var{expr})^.5".
+
+@item not(expr)
+Return 1.0 if @var{expr} is zero, 0.0 otherwise.
+
+@item pow(x, y)
+Compute the power of @var{x} elevated @var{y}, it is equivalent to
+"(@var{x})^(@var{y})".
+
+@item random(x)
+Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
+internal variable which will be used to save the seed/state.
+
+@item hypot(x, y)
+This function is similar to the C function with the same name; it returns
+"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
+right triangle with sides of length @var{x} and @var{y}, or the distance of the
+point (@var{x}, @var{y}) from the origin.
+
+@item gcd(x, y)
+Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
+@var{y} are 0 or either or both are less than zero then behavior is undefined.
+
+@item if(x, y)
+Evaluate @var{x}, and if the result is non-zero return the result of
+the evaluation of @var{y}, return 0 otherwise.
+
+@item ifnot(x, y)
+Evaluate @var{x}, and if the result is zero return the result of the
+evaluation of @var{y}, return 0 otherwise.
+
+@item taylor(expr, x) taylor(expr, x, id)
+Evaluate a taylor series at x.
+expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
+then 0 is assumed.
+note, when you have the derivatives at y instead of 0
+taylor(expr, x-y) can be used
+When the series does not converge the results are undefined.
+
+@item root(expr, max)
+Finds x where f(x)=0 in the interval 0..max.
+f() must be continuous or the result is undefined.
+@end table
+
+The following constants are available:
+@table @option
+@item PI
+area of the unit disc, approximately 3.14
+@item E
+exp(1) (Euler's number), approximately 2.718
+@item PHI
+golden ratio (1+sqrt(5))/2, approximately 1.618
+@end table
+
+Assuming that an expression is considered "true" if it has a non-zero
+value, note that:
+
+@code{*} works like AND
+
+@code{+} works like OR
+
+and the construct:
+@example
+if A then B else C
+@end example
+is equivalent to
+@example
+if(A,B) + ifnot(A,C)
+@end example
+
+In your C code, you can extend the list of unary and binary functions,
+and define recognized constants, so that they are available for your
+expressions.
+
+The evaluator also recognizes the International System number
+postfixes. If 'i' is appended after the postfix, powers of 2 are used
+instead of powers of 10. The 'B' postfix multiplies the value for 8,
+and can be appended after another postfix or used alone. This allows
+using for example 'KB', 'MiB', 'G' and 'B' as postfix.
+
+Follows the list of available International System postfixes, with
+indication of the corresponding powers of 10 and of 2.
+@table @option
+@item y
+-24 / -80
+@item z
+-21 / -70
+@item a
+-18 / -60
+@item f
+-15 / -50
+@item p
+-12 / -40
+@item n
+-9 / -30
+@item u
+-6 / -20
+@item m
+-3 / -10
+@item c
+-2
+@item d
+-1
+@item h
+2
+@item k
+3 / 10
+@item K
+3 / 10
+@item M
+6 / 20
+@item G
+9 / 30
+@item T
+12 / 40
+@item P
+15 / 40
+@item E
+18 / 50
+@item Z
+21 / 60
+@item Y
+24 / 70
+@end table
+
+@c man end

doc/examples/Makefile

@@ -7,31 +7,23 @@ FFMPEG_LIBS=    libavdevice                        \
                 libswscale                         \
                 libavutil                          \
 
-CFLAGS += -Wall -g
+CFLAGS += -Wall -O2 -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_reading                       \
-                decoding_encoding                  \
-                demuxing_decoding                  \
-                extract_mvs                        \
+EXAMPLES=       decoding_encoding                  \
+                demuxing                           \
                 filtering_video                    \
                 filtering_audio                    \
                 metadata                           \
                 muxing                             \
-                remuxing                           \
-                resampling_audio                   \
                 scaling_video                      \
-                transcode_aac                      \
-                transcoding                        \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
-avcodec:           LDLIBS += -lm
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
-resampling_audio:  LDLIBS += -lm
 
 .phony: all clean-test clean
 

doc/examples/README

@@ -1,23 +0,0 @@
-FFmpeg examples README
-----------------------
-
-Both following use cases rely on pkg-config and make, thus make sure
-that you have them installed and working on your system.
-
-
-Method 1: build the installed examples in a generic read/write user directory
-
-Copy to a read/write user directory and just use "make", it will link
-to the libraries on your system, assuming the PKG_CONFIG_PATH is
-correctly configured.
-
-Method 2: build the examples in-tree
-
-Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then just run "make examples".
-This will build the examples using the FFmpeg build system. You can clean those
-examples using "make examplesclean"
-
-If you want to try the dedicated Makefile examples (to emulate the first
-method), go into doc/examples and run a command such as
-PKG_CONFIG_PATH=pc-uninstalled make.

doc/examples/avio_reading.c

@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2014 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat AVIOContext API example.
- *
- * Make libavformat demuxer access media content through a custom
- * AVIOContext read callback.
- * @example avio_reading.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavformat/avio.h>
-#include <libavutil/file.h>
-
-struct buffer_data {
-    uint8_t *ptr;
-    size_t size; ///< size left in the buffer
-};
-
-static int read_packet(void *opaque, uint8_t *buf, int buf_size)
-{
-    struct buffer_data *bd = (struct buffer_data *)opaque;
-    buf_size = FFMIN(buf_size, bd->size);
-
-    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
-
-    /* copy internal buffer data to buf */
-    memcpy(buf, bd->ptr, buf_size);
-    bd->ptr  += buf_size;
-    bd->size -= buf_size;
-
-    return buf_size;
-}
-
-int main(int argc, char *argv[])
-{
-    AVFormatContext *fmt_ctx = NULL;
-    AVIOContext *avio_ctx = NULL;
-    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
-    size_t buffer_size, avio_ctx_buffer_size = 4096;
-    char *input_filename = NULL;
-    int ret = 0;
-    struct buffer_data bd = { 0 };
-
-    if (argc != 2) {
-        fprintf(stderr, "usage: %s input_file\n"
-                "API example program to show how to read from a custom buffer "
-                "accessed through AVIOContext.\n", argv[0]);
-        return 1;
-    }
-    input_filename = argv[1];
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-
-    /* slurp file content into buffer */
-    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
-    if (ret < 0)
-        goto end;
-
-    /* fill opaque structure used by the AVIOContext read callback */
-    bd.ptr  = buffer;
-    bd.size = buffer_size;
-
-    if (!(fmt_ctx = avformat_alloc_context())) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
-    if (!avio_ctx_buffer) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
-                                  0, &bd, &read_packet, NULL, NULL);
-    if (!avio_ctx) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    fmt_ctx->pb = avio_ctx;
-
-    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open input\n");
-        goto end;
-    }
-
-    ret = avformat_find_stream_info(fmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        goto end;
-    }
-
-    av_dump_format(fmt_ctx, 0, input_filename, 0);
-
-end:
-    avformat_close_input(&fmt_ctx);
-    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
-    if (avio_ctx) {
-        av_freep(&avio_ctx->buffer);
-        av_freep(&avio_ctx);
-    }
-    av_file_unmap(buffer, buffer_size);
-
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}

doc/examples/decoding_encoding.c

@@ -24,7 +24,6 @@
  * @file
  * libavcodec API use example.
  *
- * @example decoding_encoding.c
  * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
  * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
  * format handling
@@ -34,7 +33,7 @@
 
 #include <libavutil/opt.h>
 #include <libavcodec/avcodec.h>
-#include <libavutil/channel_layout.h>
+#include <libavutil/audioconvert.h>
 #include <libavutil/common.h>
 #include <libavutil/imgutils.h>
 #include <libavutil/mathematics.h>
@@ -79,7 +78,7 @@ static int select_channel_layout(AVCodec *codec)
 {
     const uint64_t *p;
     uint64_t best_ch_layout = 0;
-    int best_nb_channels   = 0;
+    int best_nb_channells   = 0;
 
     if (!codec->channel_layouts)
         return AV_CH_LAYOUT_STEREO;
@@ -88,9 +87,9 @@ static int select_channel_layout(AVCodec *codec)
     while (*p) {
         int nb_channels = av_get_channel_layout_nb_channels(*p);
 
-        if (nb_channels > best_nb_channels) {
+        if (nb_channels > best_nb_channells) {
             best_ch_layout    = *p;
-            best_nb_channels = nb_channels;
+            best_nb_channells = nb_channels;
         }
         p++;
     }
@@ -122,10 +121,6 @@ static void audio_encode_example(const char *filename)
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
 
     /* put sample parameters */
     c->bit_rate = 64000;
@@ -156,7 +151,7 @@ static void audio_encode_example(const char *filename)
     }
 
     /* frame containing input raw audio */
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate audio frame\n");
         exit(1);
@@ -170,10 +165,6 @@ static void audio_encode_example(const char *filename)
      * we calculate the size of the samples buffer in bytes */
     buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                              c->sample_fmt, 0);
-    if (buffer_size < 0) {
-        fprintf(stderr, "Could not get sample buffer size\n");
-        exit(1);
-    }
     samples = av_malloc(buffer_size);
     if (!samples) {
         fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
@@ -191,7 +182,7 @@ static void audio_encode_example(const char *filename)
     /* encode a single tone sound */
     t = 0;
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for (i = 0; i < 200; i++) {
+    for(i=0;i<200;i++) {
         av_init_packet(&pkt);
         pkt.data = NULL; // packet data will be allocated by the encoder
         pkt.size = 0;
@@ -231,7 +222,7 @@ static void audio_encode_example(const char *filename)
     fclose(f);
 
     av_freep(&samples);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -261,10 +252,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
 
     /* open it */
     if (avcodec_open2(c, codec, NULL) < 0) {
@@ -288,15 +275,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
 
     while (avpkt.size > 0) {
-        int i, ch;
         int got_frame = 0;
 
         if (!decoded_frame) {
-            if (!(decoded_frame = av_frame_alloc())) {
+            if (!(decoded_frame = avcodec_alloc_frame())) {
                 fprintf(stderr, "Could not allocate audio frame\n");
                 exit(1);
             }
-        }
+        } else
+            avcodec_get_frame_defaults(decoded_frame);
 
         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
         if (len < 0) {
@@ -305,15 +292,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         }
         if (got_frame) {
             /* if a frame has been decoded, output it */
-            int data_size = av_get_bytes_per_sample(c->sample_fmt);
-            if (data_size < 0) {
-                /* This should not occur, checking just for paranoia */
-                fprintf(stderr, "Failed to calculate data size\n");
-                exit(1);
-            }
-            for (i=0; i<decoded_frame->nb_samples; i++)
-                for (ch=0; ch<c->channels; ch++)
-                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
+            int data_size = av_samples_get_buffer_size(NULL, c->channels,
+                                                       decoded_frame->nb_samples,
+                                                       c->sample_fmt, 1);
+            fwrite(decoded_frame->data[0], 1, data_size, outfile);
         }
         avpkt.size -= len;
         avpkt.data += len;
@@ -338,7 +320,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    av_frame_free(&decoded_frame);
+    avcodec_free_frame(&decoded_frame);
 }
 
 /*
@@ -364,10 +346,6 @@ static void video_encode_example(const char *filename, int codec_id)
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
 
     /* put sample parameters */
     c->bit_rate = 400000;
@@ -375,18 +353,12 @@ static void video_encode_example(const char *filename, int codec_id)
     c->width = 352;
     c->height = 288;
     /* frames per second */
-    c->time_base = (AVRational){1,25};
-    /* emit one intra frame every ten frames
-     * check frame pict_type before passing frame
-     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
-     * then gop_size is ignored and the output of encoder
-     * will always be I frame irrespective to gop_size
-     */
-    c->gop_size = 10;
-    c->max_b_frames = 1;
-    c->pix_fmt = AV_PIX_FMT_YUV420P;
+    c->time_base= (AVRational){1,25};
+    c->gop_size = 10; /* emit one intra frame every ten frames */
+    c->max_b_frames=1;
+    c->pix_fmt = PIX_FMT_YUV420P;
 
-    if (codec_id == AV_CODEC_ID_H264)
+    if(codec_id == AV_CODEC_ID_H264)
         av_opt_set(c->priv_data, "preset", "slow", 0);
 
     /* open it */
@@ -401,7 +373,7 @@ static void video_encode_example(const char *filename, int codec_id)
         exit(1);
     }
 
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -420,7 +392,7 @@ static void video_encode_example(const char *filename, int codec_id)
     }
 
     /* encode 1 second of video */
-    for (i = 0; i < 25; i++) {
+    for(i=0;i<25;i++) {
         av_init_packet(&pkt);
         pkt.data = NULL;    // packet data will be allocated by the encoder
         pkt.size = 0;
@@ -428,15 +400,15 @@ static void video_encode_example(const char *filename, int codec_id)
         fflush(stdout);
         /* prepare a dummy image */
         /* Y */
-        for (y = 0; y < c->height; y++) {
-            for (x = 0; x < c->width; x++) {
+        for(y=0;y<c->height;y++) {
+            for(x=0;x<c->width;x++) {
                 frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
             }
         }
 
         /* Cb and Cr */
-        for (y = 0; y < c->height/2; y++) {
-            for (x = 0; x < c->width/2; x++) {
+        for(y=0;y<c->height/2;y++) {
+            for(x=0;x<c->width/2;x++) {
                 frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                 frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
             }
@@ -482,7 +454,7 @@ static void video_encode_example(const char *filename, int codec_id)
     avcodec_close(c);
     av_free(c);
     av_freep(&frame->data[0]);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     printf("\n");
 }
 
@@ -496,49 +468,22 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     FILE *f;
     int i;
 
-    f = fopen(filename,"w");
-    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
-    for (i = 0; i < ysize; i++)
-        fwrite(buf + i * wrap, 1, xsize, f);
+    f=fopen(filename,"w");
+    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
+    for(i=0;i<ysize;i++)
+        fwrite(buf + i * wrap,1,xsize,f);
     fclose(f);
 }
 
-static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
-                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
-{
-    int len, got_frame;
-    char buf[1024];
-
-    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
-    if (len < 0) {
-        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
-        return len;
-    }
-    if (got_frame) {
-        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
-        fflush(stdout);
-
-        /* the picture is allocated by the decoder, no need to free it */
-        snprintf(buf, sizeof(buf), outfilename, *frame_count);
-        pgm_save(frame->data[0], frame->linesize[0],
-                 avctx->width, avctx->height, buf);
-        (*frame_count)++;
-    }
-    if (pkt->data) {
-        pkt->size -= len;
-        pkt->data += len;
-    }
-    return 0;
-}
-
 static void video_decode_example(const char *outfilename, const char *filename)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int frame_count;
+    int frame, got_picture, len;
     FILE *f;
-    AVFrame *frame;
+    AVFrame *picture;
     uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    char buf[1024];
     AVPacket avpkt;
 
     av_init_packet(&avpkt);
@@ -556,11 +501,6 @@ static void video_decode_example(const char *outfilename, const char *filename)
     }
 
     c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
-
     if(codec->capabilities&CODEC_CAP_TRUNCATED)
         c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
 
@@ -574,20 +514,22 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
+    /* the codec gives us the frame size, in samples */
+
     f = fopen(filename, "rb");
     if (!f) {
         fprintf(stderr, "Could not open %s\n", filename);
         exit(1);
     }
 
-    frame = av_frame_alloc();
-    if (!frame) {
+    picture = avcodec_alloc_frame();
+    if (!picture) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
 
-    frame_count = 0;
-    for (;;) {
+    frame = 0;
+    for(;;) {
         avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
         if (avpkt.size == 0)
             break;
@@ -608,9 +550,26 @@ static void video_decode_example(const char *outfilename, const char *filename)
         /* here, we use a stream based decoder (mpeg1video), so we
            feed decoder and see if it could decode a frame */
         avpkt.data = inbuf;
-        while (avpkt.size > 0)
-            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
+        while (avpkt.size > 0) {
+            len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+            if (len < 0) {
+                fprintf(stderr, "Error while decoding frame %d\n", frame);
                 exit(1);
+            }
+            if (got_picture) {
+                printf("Saving frame %3d\n", frame);
+                fflush(stdout);
+
+                /* the picture is allocated by the decoder. no need to
+                   free it */
+                snprintf(buf, sizeof(buf), outfilename, frame);
+                pgm_save(picture->data[0], picture->linesize[0],
+                         c->width, c->height, buf);
+                frame++;
+            }
+            avpkt.size -= len;
+            avpkt.data += len;
+        }
     }
 
     /* some codecs, such as MPEG, transmit the I and P frame with a
@@ -618,13 +577,24 @@ static void video_decode_example(const char *outfilename, const char *filename)
        chance to get the last frame of the video */
     avpkt.data = NULL;
     avpkt.size = 0;
-    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
+    len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+    if (got_picture) {
+        printf("Saving last frame %3d\n", frame);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder. no need to
+           free it */
+        snprintf(buf, sizeof(buf), outfilename, frame);
+        pgm_save(picture->data[0], picture->linesize[0],
+                 c->width, c->height, buf);
+        frame++;
+    }
 
     fclose(f);
 
     avcodec_close(c);
     av_free(c);
-    av_frame_free(&frame);
+    avcodec_free_frame(&picture);
     printf("\n");
 }
 
@@ -640,8 +610,8 @@ int main(int argc, char **argv)
                "API example program to decode/encode a media stream with libavcodec.\n"
                "This program generates a synthetic stream and encodes it to a file\n"
                "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
-               "The encoded stream is then decoded and written to a raw data output.\n"
-               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
+               "The encoded stream is then decoded and written to a raw data output\n."
+               "output_type must be choosen between 'h264', 'mp2', 'mpg'\n",
                argv[0]);
         return 1;
     }
@@ -651,7 +621,7 @@ int main(int argc, char **argv)
         video_encode_example("test.h264", AV_CODEC_ID_H264);
     } else if (!strcmp(output_type, "mp2")) {
         audio_encode_example("test.mp2");
-        audio_decode_example("test.pcm", "test.mp2");
+        audio_decode_example("test.sw", "test.mp2");
     } else if (!strcmp(output_type, "mpg")) {
         video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
         video_decode_example("test%02d.pgm", "test.mpg");

doc/examples/demuxing.c

@@ -0,0 +1,339 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat demuxing API use example.
+ *
+ * Show how to use the libavformat and libavcodec API to demux and
+ * decode audio and video data.
+ */
+
+#include <libavutil/imgutils.h>
+#include <libavutil/samplefmt.h>
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *fmt_ctx = NULL;
+static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
+static AVStream *video_stream = NULL, *audio_stream = NULL;
+static const char *src_filename = NULL;
+static const char *video_dst_filename = NULL;
+static const char *audio_dst_filename = NULL;
+static FILE *video_dst_file = NULL;
+static FILE *audio_dst_file = NULL;
+
+static uint8_t *video_dst_data[4] = {NULL};
+static int      video_dst_linesize[4];
+static int video_dst_bufsize;
+
+static uint8_t **audio_dst_data = NULL;
+static int       audio_dst_linesize;
+static int audio_dst_bufsize;
+
+static int video_stream_idx = -1, audio_stream_idx = -1;
+static AVFrame *frame = NULL;
+static AVPacket pkt;
+static int video_frame_count = 0;
+static int audio_frame_count = 0;
+
+static int decode_packet(int *got_frame, int cached)
+{
+    int ret = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        /* decode video frame */
+        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame\n");
+            return ret;
+        }
+
+        if (*got_frame) {
+            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   video_frame_count++, frame->coded_picture_number,
+                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
+
+            /* copy decoded frame to destination buffer:
+             * this is required since rawvideo expects non aligned data */
+            av_image_copy(video_dst_data, video_dst_linesize,
+                          (const uint8_t **)(frame->data), frame->linesize,
+                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
+
+            /* write to rawvideo file */
+            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
+        }
+    } else if (pkt.stream_index == audio_stream_idx) {
+        /* decode audio frame */
+        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding audio frame\n");
+            return ret;
+        }
+
+        if (*got_frame) {
+            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   audio_frame_count++, frame->nb_samples,
+                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
+
+            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, frame->channels,
+                                   frame->nb_samples, frame->format, 1);
+            if (ret < 0) {
+                fprintf(stderr, "Could not allocate audio buffer\n");
+                return AVERROR(ENOMEM);
+            }
+
+            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
+            audio_dst_bufsize =
+                av_samples_get_buffer_size(NULL, frame->channels,
+                                           frame->nb_samples, frame->format, 1);
+
+            /* copy audio data to destination buffer:
+             * this is required since rawaudio expects non aligned data */
+            av_samples_copy(audio_dst_data, frame->data, 0, 0,
+                            frame->nb_samples, frame->channels, frame->format);
+
+            /* write to rawaudio file */
+            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
+            av_freep(&audio_dst_data[0]);
+        }
+    }
+
+    return ret;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret;
+    AVStream *st;
+    AVCodecContext *dec_ctx = NULL;
+    AVCodec *dec = NULL;
+
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
+                av_get_media_type_string(type), src_filename);
+        return ret;
+    } else {
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];
+
+        /* find decoder for the stream */
+        dec_ctx = st->codec;
+        dec = avcodec_find_decoder(dec_ctx->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+
+        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+            fprintf(stderr, "Failed to open %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+    }
+
+    return 0;
+}
+
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "sample format %s is not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return -1;
+}
+
+int main (int argc, char **argv)
+{
+    int ret = 0, got_frame;
+
+    if (argc != 4) {
+        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
+                "API example program to show how to read frames from an input file.\n"
+                "This program reads frames from a file, decodes them, and writes decoded\n"
+                "video frames to a rawvideo file named video_output_file, and decoded\n"
+                "audio frames to a rawaudio file named audio_output_file.\n"
+                "\n", argv[0]);
+        exit(1);
+    }
+    src_filename = argv[1];
+    video_dst_filename = argv[2];
+    audio_dst_filename = argv[3];
+
+    /* register all formats and codecs */
+    av_register_all();
+
+    /* open input file, and allocated format context */
+    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
+        fprintf(stderr, "Could not open source file %s\n", src_filename);
+        exit(1);
+    }
+
+    /* retrieve stream information */
+    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        exit(1);
+    }
+
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = video_stream->codec;
+
+        video_dst_file = fopen(video_dst_filename, "wb");
+        if (!video_dst_file) {
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
+            ret = 1;
+            goto end;
+        }
+
+        /* allocate image where the decoded image will be put */
+        ret = av_image_alloc(video_dst_data, video_dst_linesize,
+                             video_dec_ctx->width, video_dec_ctx->height,
+                             video_dec_ctx->pix_fmt, 1);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate raw video buffer\n");
+            goto end;
+        }
+        video_dst_bufsize = ret;
+    }
+
+    /* dump input information to stderr */
+    av_dump_format(fmt_ctx, 0, src_filename, 0);
+
+    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        int nb_planes;
+
+        audio_stream = fmt_ctx->streams[audio_stream_idx];
+        audio_dec_ctx = audio_stream->codec;
+        audio_dst_file = fopen(audio_dst_filename, "wb");
+        if (!audio_dst_file) {
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
+            ret = 1;
+            goto end;
+        }
+
+        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
+            audio_dec_ctx->channels : 1;
+        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
+        if (!audio_dst_data) {
+            fprintf(stderr, "Could not allocate audio data buffers\n");
+            ret = AVERROR(ENOMEM);
+            goto end;
+        }
+    }
+
+    if (!audio_stream && !video_stream) {
+        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
+        ret = 1;
+        goto end;
+    }
+
+    frame = avcodec_alloc_frame();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    if (video_stream)
+        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
+    if (audio_stream)
+        printf("Demuxing video from file '%s' into '%s'\n", src_filename, audio_dst_filename);
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0)
+        decode_packet(&got_frame, 0);
+
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
+
+    printf("Demuxing succeeded.\n");
+
+    if (video_stream) {
+        printf("Play the output video file with the command:\n"
+               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
+               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
+               video_dst_filename);
+    }
+
+    if (audio_stream) {
+        const char *fmt;
+
+        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt) < 0))
+            goto end;
+        printf("Play the output audio file with the command:\n"
+               "ffplay -f %s -ac %d -ar %d %s\n",
+               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
+               audio_dst_filename);
+    }
+
+end:
+    if (video_dec_ctx)
+        avcodec_close(video_dec_ctx);
+    if (audio_dec_ctx)
+        avcodec_close(audio_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    if (video_dst_file)
+        fclose(video_dst_file);
+    if (audio_dst_file)
+        fclose(audio_dst_file);
+    av_free(frame);
+    av_free(video_dst_data[0]);
+    av_free(audio_dst_data);
+
+    return ret < 0;
+}

doc/examples/demuxing_decoding.c

@@ -1,386 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * Demuxing and decoding example.
- *
- * Show how to use the libavformat and libavcodec API to demux and
- * decode audio and video data.
- * @example demuxing_decoding.c
- */
-
-#include <libavutil/imgutils.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *fmt_ctx = NULL;
-static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
-static AVStream *video_stream = NULL, *audio_stream = NULL;
-static const char *src_filename = NULL;
-static const char *video_dst_filename = NULL;
-static const char *audio_dst_filename = NULL;
-static FILE *video_dst_file = NULL;
-static FILE *audio_dst_file = NULL;
-
-static uint8_t *video_dst_data[4] = {NULL};
-static int      video_dst_linesize[4];
-static int video_dst_bufsize;
-
-static int video_stream_idx = -1, audio_stream_idx = -1;
-static AVFrame *frame = NULL;
-static AVPacket pkt;
-static int video_frame_count = 0;
-static int audio_frame_count = 0;
-
-/* The different ways of decoding and managing data memory. You are not
- * supposed to support all the modes in your application but pick the one most
- * appropriate to your needs. Look for the use of api_mode in this example to
- * see what are the differences of API usage between them */
-enum {
-    API_MODE_OLD                  = 0, /* old method, deprecated */
-    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
-    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
-};
-
-static int api_mode = API_MODE_OLD;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int ret = 0;
-    int decoded = pkt.size;
-
-    *got_frame = 0;
-
-    if (pkt.stream_index == video_stream_idx) {
-        /* decode video frame */
-        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-
-        if (*got_frame) {
-            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   video_frame_count++, frame->coded_picture_number,
-                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
-
-            /* copy decoded frame to destination buffer:
-             * this is required since rawvideo expects non aligned data */
-            av_image_copy(video_dst_data, video_dst_linesize,
-                          (const uint8_t **)(frame->data), frame->linesize,
-                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
-
-            /* write to rawvideo file */
-            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
-        }
-    } else if (pkt.stream_index == audio_stream_idx) {
-        /* decode audio frame */
-        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-        /* Some audio decoders decode only part of the packet, and have to be
-         * called again with the remainder of the packet data.
-         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
-         * Also, some decoders might over-read the packet. */
-        decoded = FFMIN(ret, pkt.size);
-
-        if (*got_frame) {
-            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
-            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   audio_frame_count++, frame->nb_samples,
-                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
-
-            /* Write the raw audio data samples of the first plane. This works
-             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
-             * most audio decoders output planar audio, which uses a separate
-             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
-             * In other words, this code will write only the first audio channel
-             * in these cases.
-             * You should use libswresample or libavfilter to convert the frame
-             * to packed data. */
-            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
-        }
-    }
-
-    /* If we use the new API with reference counting, we own the data and need
-     * to de-reference it when we don't use it anymore */
-    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
-        av_frame_unref(frame);
-
-    return decoded;
-}
-
-static int open_codec_context(int *stream_idx,
-                              AVFormatContext *fmt_ctx, enum AVMediaType type)
-{
-    int ret;
-    AVStream *st;
-    AVCodecContext *dec_ctx = NULL;
-    AVCodec *dec = NULL;
-    AVDictionary *opts = NULL;
-
-    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
-                av_get_media_type_string(type), src_filename);
-        return ret;
-    } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
-
-        /* find decoder for the stream */
-        dec_ctx = st->codec;
-        dec = avcodec_find_decoder(dec_ctx->codec_id);
-        if (!dec) {
-            fprintf(stderr, "Failed to find %s codec\n",
-                    av_get_media_type_string(type));
-            return AVERROR(EINVAL);
-        }
-
-        /* Init the decoders, with or without reference counting */
-        if (api_mode == API_MODE_NEW_API_REF_COUNT)
-            av_dict_set(&opts, "refcounted_frames", "1", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "sample format %s is not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return -1;
-}
-
-int main (int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
-                "input_file video_output_file audio_output_file\n"
-                "API example program to show how to read frames from an input file.\n"
-                "This program reads frames from a file, decodes them, and writes decoded\n"
-                "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n\n"
-                "If the -refcount option is specified, the program use the\n"
-                "reference counting frame system which allows keeping a copy of\n"
-                "the data for longer than one decode call. If unset, it's using\n"
-                "the classic old method.\n"
-                "\n", argv[0]);
-        exit(1);
-    }
-    if (argc == 5) {
-        const char *mode = argv[1] + strlen("-refcount=");
-        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
-        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
-        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
-        else {
-            fprintf(stderr, "unknow mode '%s'\n", mode);
-            exit(1);
-        }
-        argv++;
-    }
-    src_filename = argv[1];
-    video_dst_filename = argv[2];
-    audio_dst_filename = argv[3];
-
-    /* register all formats and codecs */
-    av_register_all();
-
-    /* open input file, and allocate format context */
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    /* retrieve stream information */
-    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        exit(1);
-    }
-
-    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = video_stream->codec;
-
-        video_dst_file = fopen(video_dst_filename, "wb");
-        if (!video_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
-            ret = 1;
-            goto end;
-        }
-
-        /* allocate image where the decoded image will be put */
-        ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             video_dec_ctx->width, video_dec_ctx->height,
-                             video_dec_ctx->pix_fmt, 1);
-        if (ret < 0) {
-            fprintf(stderr, "Could not allocate raw video buffer\n");
-            goto end;
-        }
-        video_dst_bufsize = ret;
-    }
-
-    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        audio_stream = fmt_ctx->streams[audio_stream_idx];
-        audio_dec_ctx = audio_stream->codec;
-        audio_dst_file = fopen(audio_dst_filename, "wb");
-        if (!audio_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
-            ret = 1;
-            goto end;
-        }
-    }
-
-    /* dump input information to stderr */
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
-    if (!audio_stream && !video_stream) {
-        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
-        ret = 1;
-        goto end;
-    }
-
-    /* When using the new API, you need to use the libavutil/frame.h API, while
-     * the classic frame management is available in libavcodec */
-    if (api_mode == API_MODE_OLD)
-        frame = avcodec_alloc_frame();
-    else
-        frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate frame\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    if (video_stream)
-        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
-    if (audio_stream)
-        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_free_packet(&orig_pkt);
-    }
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-    printf("Demuxing succeeded.\n");
-
-    if (video_stream) {
-        printf("Play the output video file with the command:\n"
-               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
-               video_dst_filename);
-    }
-
-    if (audio_stream) {
-        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        int n_channels = audio_dec_ctx->channels;
-        const char *fmt;
-
-        if (av_sample_fmt_is_planar(sfmt)) {
-            const char *packed = av_get_sample_fmt_name(sfmt);
-            printf("Warning: the sample format the decoder produced is planar "
-                   "(%s). This example will output the first channel only.\n",
-                   packed ? packed : "?");
-            sfmt = av_get_packed_sample_fmt(sfmt);
-            n_channels = 1;
-        }
-
-        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
-            goto end;
-
-        printf("Play the output audio file with the command:\n"
-               "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, n_channels, audio_dec_ctx->sample_rate,
-               audio_dst_filename);
-    }
-
-end:
-    avcodec_close(video_dec_ctx);
-    avcodec_close(audio_dec_ctx);
-    avformat_close_input(&fmt_ctx);
-    if (video_dst_file)
-        fclose(video_dst_file);
-    if (audio_dst_file)
-        fclose(audio_dst_file);
-    if (api_mode == API_MODE_OLD)
-        avcodec_free_frame(&frame);
-    else
-        av_frame_free(&frame);
-    av_free(video_dst_data[0]);
-
-    return ret < 0;
-}

doc/examples/extract_mvs.c

@@ -1,185 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- * Copyright (c) 2014 Clément Bœsch
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-#include <libavutil/motion_vector.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *fmt_ctx = NULL;
-static AVCodecContext *video_dec_ctx = NULL;
-static AVStream *video_stream = NULL;
-static const char *src_filename = NULL;
-
-static int video_stream_idx = -1;
-static AVFrame *frame = NULL;
-static AVPacket pkt;
-static int video_frame_count = 0;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int decoded = pkt.size;
-
-    *got_frame = 0;
-
-    if (pkt.stream_index == video_stream_idx) {
-        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-
-        if (*got_frame) {
-            int i;
-            AVFrameSideData *sd;
-
-            video_frame_count++;
-            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
-            if (sd) {
-                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
-                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
-                    const AVMotionVector *mv = &mvs[i];
-                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
-                           video_frame_count, mv->source,
-                           mv->w, mv->h, mv->src_x, mv->src_y,
-                           mv->dst_x, mv->dst_y, mv->flags);
-                }
-            }
-        }
-    }
-
-    return decoded;
-}
-
-static int open_codec_context(int *stream_idx,
-                              AVFormatContext *fmt_ctx, enum AVMediaType type)
-{
-    int ret;
-    AVStream *st;
-    AVCodecContext *dec_ctx = NULL;
-    AVCodec *dec = NULL;
-    AVDictionary *opts = NULL;
-
-    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
-                av_get_media_type_string(type), src_filename);
-        return ret;
-    } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
-
-        /* find decoder for the stream */
-        dec_ctx = st->codec;
-        dec = avcodec_find_decoder(dec_ctx->codec_id);
-        if (!dec) {
-            fprintf(stderr, "Failed to find %s codec\n",
-                    av_get_media_type_string(type));
-            return AVERROR(EINVAL);
-        }
-
-        /* Init the video decoder */
-        av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
-        exit(1);
-    }
-    src_filename = argv[1];
-
-    av_register_all();
-
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        exit(1);
-    }
-
-    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = video_stream->codec;
-    }
-
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
-    if (!video_stream) {
-        fprintf(stderr, "Could not find video stream in the input, aborting\n");
-        ret = 1;
-        goto end;
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate frame\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_free_packet(&orig_pkt);
-    }
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-end:
-    avcodec_close(video_dec_ctx);
-    avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    return ret < 0;
-}

doc/examples/filter_audio.c

@@ -1,365 +0,0 @@
-/*
- * copyright (c) 2013 Andrew Kelley
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * libavfilter API usage example.
- *
- * @example filter_audio.c
- * This example will generate a sine wave audio,
- * pass it through a simple filter chain, and then compute the MD5 checksum of
- * the output data.
- *
- * The filter chain it uses is:
- * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
- *
- * abuffer: This provides the endpoint where you can feed the decoded samples.
- * volume: In this example we hardcode it to 0.90.
- * aformat: This converts the samples to the samplefreq, channel layout,
- *          and sample format required by the audio device.
- * abuffersink: This provides the endpoint where you can read the samples after
- *              they have passed through the filter chain.
- */
-
-#include <inttypes.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/md5.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/buffersink.h"
-#include "libavfilter/buffersrc.h"
-
-#define INPUT_SAMPLERATE     48000
-#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
-
-#define VOLUME_VAL 0.90
-
-static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
-                             AVFilterContext **sink)
-{
-    AVFilterGraph *filter_graph;
-    AVFilterContext *abuffer_ctx;
-    AVFilter        *abuffer;
-    AVFilterContext *volume_ctx;
-    AVFilter        *volume;
-    AVFilterContext *aformat_ctx;
-    AVFilter        *aformat;
-    AVFilterContext *abuffersink_ctx;
-    AVFilter        *abuffersink;
-
-    AVDictionary *options_dict = NULL;
-    uint8_t options_str[1024];
-    uint8_t ch_layout[64];
-
-    int err;
-
-    /* Create a new filtergraph, which will contain all the filters. */
-    filter_graph = avfilter_graph_alloc();
-    if (!filter_graph) {
-        fprintf(stderr, "Unable to create filter graph.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Create the abuffer filter;
-     * it will be used for feeding the data into the graph. */
-    abuffer = avfilter_get_by_name("abuffer");
-    if (!abuffer) {
-        fprintf(stderr, "Could not find the abuffer filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
-    if (!abuffer_ctx) {
-        fprintf(stderr, "Could not allocate the abuffer instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Set the filter options through the AVOptions API. */
-    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
-    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
-    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
-    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
-    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
-
-    /* Now initialize the filter; we pass NULL options, since we have already
-     * set all the options above. */
-    err = avfilter_init_str(abuffer_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffer filter.\n");
-        return err;
-    }
-
-    /* Create volume filter. */
-    volume = avfilter_get_by_name("volume");
-    if (!volume) {
-        fprintf(stderr, "Could not find the volume filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
-    if (!volume_ctx) {
-        fprintf(stderr, "Could not allocate the volume instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A different way of passing the options is as key/value pairs in a
-     * dictionary. */
-    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
-    err = avfilter_init_dict(volume_ctx, &options_dict);
-    av_dict_free(&options_dict);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the volume filter.\n");
-        return err;
-    }
-
-    /* Create the aformat filter;
-     * it ensures that the output is of the format we want. */
-    aformat = avfilter_get_by_name("aformat");
-    if (!aformat) {
-        fprintf(stderr, "Could not find the aformat filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
-    if (!aformat_ctx) {
-        fprintf(stderr, "Could not allocate the aformat instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A third way of passing the options is in a string of the form
-     * key1=value1:key2=value2.... */
-    snprintf(options_str, sizeof(options_str),
-             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
-             (uint64_t)AV_CH_LAYOUT_STEREO);
-    err = avfilter_init_str(aformat_ctx, options_str);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
-        return err;
-    }
-
-    /* Finally create the abuffersink filter;
-     * it will be used to get the filtered data out of the graph. */
-    abuffersink = avfilter_get_by_name("abuffersink");
-    if (!abuffersink) {
-        fprintf(stderr, "Could not find the abuffersink filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
-    if (!abuffersink_ctx) {
-        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* This filter takes no options. */
-    err = avfilter_init_str(abuffersink_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
-        return err;
-    }
-
-    /* Connect the filters;
-     * in this simple case the filters just form a linear chain. */
-    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
-    if (err < 0) {
-        fprintf(stderr, "Error connecting filters\n");
-        return err;
-    }
-
-    /* Configure the graph. */
-    err = avfilter_graph_config(filter_graph, NULL);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
-        return err;
-    }
-
-    *graph = filter_graph;
-    *src   = abuffer_ctx;
-    *sink  = abuffersink_ctx;
-
-    return 0;
-}
-
-/* Do something useful with the filtered data: this simple
- * example just prints the MD5 checksum of each plane to stdout. */
-static int process_output(struct AVMD5 *md5, AVFrame *frame)
-{
-    int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
-    int planes     = planar ? channels : 1;
-    int bps        = av_get_bytes_per_sample(frame->format);
-    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
-    int i, j;
-
-    for (i = 0; i < planes; i++) {
-        uint8_t checksum[16];
-
-        av_md5_init(md5);
-        av_md5_sum(checksum, frame->extended_data[i], plane_size);
-
-        fprintf(stdout, "plane %d: 0x", i);
-        for (j = 0; j < sizeof(checksum); j++)
-            fprintf(stdout, "%02X", checksum[j]);
-        fprintf(stdout, "\n");
-    }
-    fprintf(stdout, "\n");
-
-    return 0;
-}
-
-/* Construct a frame of audio data to be filtered;
- * this simple example just synthesizes a sine wave. */
-static int get_input(AVFrame *frame, int frame_num)
-{
-    int err, i, j;
-
-#define FRAME_SIZE 1024
-
-    /* Set up the frame properties and allocate the buffer for the data. */
-    frame->sample_rate    = INPUT_SAMPLERATE;
-    frame->format         = INPUT_FORMAT;
-    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
-    frame->nb_samples     = FRAME_SIZE;
-    frame->pts            = frame_num * FRAME_SIZE;
-
-    err = av_frame_get_buffer(frame, 0);
-    if (err < 0)
-        return err;
-
-    /* Fill the data for each channel. */
-    for (i = 0; i < 5; i++) {
-        float *data = (float*)frame->extended_data[i];
-
-        for (j = 0; j < frame->nb_samples; j++)
-            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
-    }
-
-    return 0;
-}
-
-int main(int argc, char *argv[])
-{
-    struct AVMD5 *md5;
-    AVFilterGraph *graph;
-    AVFilterContext *src, *sink;
-    AVFrame *frame;
-    uint8_t errstr[1024];
-    float duration;
-    int err, nb_frames, i;
-
-    if (argc < 2) {
-        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
-        return 1;
-    }
-
-    duration  = atof(argv[1]);
-    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
-    if (nb_frames <= 0) {
-        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
-        return 1;
-    }
-
-    avfilter_register_all();
-
-    /* Allocate the frame we will be using to store the data. */
-    frame  = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Error allocating the frame\n");
-        return 1;
-    }
-
-    md5 = av_md5_alloc();
-    if (!md5) {
-        fprintf(stderr, "Error allocating the MD5 context\n");
-        return 1;
-    }
-
-    /* Set up the filtergraph. */
-    err = init_filter_graph(&graph, &src, &sink);
-    if (err < 0) {
-        fprintf(stderr, "Unable to init filter graph:");
-        goto fail;
-    }
-
-    /* the main filtering loop */
-    for (i = 0; i < nb_frames; i++) {
-        /* get an input frame to be filtered */
-        err = get_input(frame, i);
-        if (err < 0) {
-            fprintf(stderr, "Error generating input frame:");
-            goto fail;
-        }
-
-        /* Send the frame to the input of the filtergraph. */
-        err = av_buffersrc_add_frame(src, frame);
-        if (err < 0) {
-            av_frame_unref(frame);
-            fprintf(stderr, "Error submitting the frame to the filtergraph:");
-            goto fail;
-        }
-
-        /* Get all the filtered output that is available. */
-        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
-            /* now do something with our filtered frame */
-            err = process_output(md5, frame);
-            if (err < 0) {
-                fprintf(stderr, "Error processing the filtered frame:");
-                goto fail;
-            }
-            av_frame_unref(frame);
-        }
-
-        if (err == AVERROR(EAGAIN)) {
-            /* Need to feed more frames in. */
-            continue;
-        } else if (err == AVERROR_EOF) {
-            /* Nothing more to do, finish. */
-            break;
-        } else if (err < 0) {
-            /* An error occurred. */
-            fprintf(stderr, "Error filtering the data:");
-            goto fail;
-        }
-    }
-
-    avfilter_graph_free(&graph);
-    av_frame_free(&frame);
-    av_freep(&md5);
-
-    return 0;
-
-fail:
-    av_strerror(err, errstr, sizeof(errstr));
-    fprintf(stderr, "%s\n", errstr);
-    return 1;
-}

doc/examples/filtering_audio.c

@@ -25,7 +25,6 @@
 /**
  * @file
  * API example for audio decoding and filtering
- * @example filtering_audio.c
  */
 
 #include <unistd.h>
@@ -36,10 +35,9 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
 
-static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
-static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+const char *filter_descr = "aresample=8000,aconvert=s16:mono";
+const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
 
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -71,7 +69,6 @@ static int open_input_file(const char *filename)
     }
     audio_stream_index = ret;
     dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the audio decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -85,22 +82,17 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret = 0;
+    int ret;
     AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
-    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
-    static const int out_sample_rates[] = { 8000, -1 };
+    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    AVABufferSinkParams *abuffersink_params;
     const AVFilterLink *outlink;
     AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
 
     filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
 
     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
     if (!dec_ctx->channel_layout)
@@ -113,36 +105,18 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        goto end;
+        return ret;
     }
 
     /* buffer audio sink: to terminate the filter chain. */
+    abuffersink_params = av_abuffersink_params_alloc();
+    abuffersink_params->sample_fmts     = sample_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, abuffersink_params, filter_graph);
+    av_free(abuffersink_params);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        goto end;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
-                              AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        goto end;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
-                              AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-        goto end;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
-                              AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        goto end;
+        return ret;
     }
 
     /* Endpoints for the filter graph. */
@@ -156,12 +130,12 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                        &inputs, &outputs, NULL)) < 0)
-        goto end;
+    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+                                    &inputs, &outputs, NULL)) < 0)
+        return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
+        return ret;
 
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
@@ -172,17 +146,14 @@ static int init_filters(const char *filters_descr)
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
            args);
 
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+    return 0;
 }
 
-static void print_frame(const AVFrame *frame)
+static void print_samplesref(AVFilterBufferRef *samplesref)
 {
-    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
-    const uint16_t *p     = (uint16_t*)frame->data[0];
+    const AVFilterBufferRefAudioProps *props = samplesref->audio;
+    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
+    const uint16_t *p     = (uint16_t*)samplesref->data[0];
     const uint16_t *p_end = p + n;
 
     while (p < p_end) {
@@ -196,20 +167,16 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket packet0, packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVPacket packet;
+    AVFrame frame;
     int got_frame;
 
-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
         exit(1);
     }
 
+    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -219,60 +186,53 @@ int main(int argc, char **argv)
         goto end;
 
     /* read all packets */
-    packet0.data = NULL;
-    packet.data = NULL;
     while (1) {
-        if (!packet0.data) {
-            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-                break;
-            packet0 = packet;
-        }
+        AVFilterBufferRef *samplesref;
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+            break;
 
         if (packet.stream_index == audio_stream_index) {
+            avcodec_get_frame_defaults(&frame);
             got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
+            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                 continue;
             }
-            packet.size -= ret;
-            packet.data += ret;
 
             if (got_frame) {
                 /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
                     av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                     break;
                 }
 
                 /* pull filtered audio from the filtergraph */
                 while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
+                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
-                    if (ret < 0)
+                    if(ret < 0)
                         goto end;
-                    print_frame(filt_frame);
-                    av_frame_unref(filt_frame);
+                    if (samplesref) {
+                        print_samplesref(samplesref);
+                        avfilter_unref_bufferp(&samplesref);
+                    }
                 }
             }
-
-            if (packet.size <= 0)
-                av_free_packet(&packet0);
-        } else {
-            /* discard non-wanted packets */
-            av_free_packet(&packet0);
         }
+        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    avcodec_close(dec_ctx);
+    if (dec_ctx)
+        avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
         exit(1);
     }
 

doc/examples/filtering_video.c

@@ -24,7 +24,6 @@
 /**
  * @file
  * API example for decoding and filtering
- * @example filtering_video.c
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -36,7 +35,6 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -71,7 +69,6 @@ static int open_input_file(const char *filename)
     }
     video_stream_index = ret;
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -85,47 +82,39 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret = 0;
+    int ret;
     AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
-    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
+    enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
+    AVBufferSinkParams *buffersink_params;
 
     filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
 
     /* buffer video source: the decoded frames from the decoder will be inserted here. */
     snprintf(args, sizeof(args),
             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
             dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-            time_base.num, time_base.den,
+            dec_ctx->time_base.num, dec_ctx->time_base.den,
             dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
 
     ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        goto end;
+        return ret;
     }
 
     /* buffer video sink: to terminate the filter chain. */
+    buffersink_params = av_buffersink_params_alloc();
+    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, buffersink_params, filter_graph);
+    av_free(buffersink_params);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        goto end;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
-                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-        goto end;
+        return ret;
     }
 
     /* Endpoints for the filter graph. */
@@ -139,47 +128,42 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
                                     &inputs, &outputs, NULL)) < 0)
-        goto end;
+        return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+        return ret;
+    return 0;
 }
 
-static void display_frame(const AVFrame *frame, AVRational time_base)
+static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
 {
     int x, y;
     uint8_t *p0, *p;
     int64_t delay;
 
-    if (frame->pts != AV_NOPTS_VALUE) {
+    if (picref->pts != AV_NOPTS_VALUE) {
         if (last_pts != AV_NOPTS_VALUE) {
             /* sleep roughly the right amount of time;
              * usleep is in microseconds, just like AV_TIME_BASE. */
-            delay = av_rescale_q(frame->pts - last_pts,
+            delay = av_rescale_q(picref->pts - last_pts,
                                  time_base, AV_TIME_BASE_Q);
             if (delay > 0 && delay < 1000000)
                 usleep(delay);
         }
-        last_pts = frame->pts;
+        last_pts = picref->pts;
     }
 
     /* Trivial ASCII grayscale display. */
-    p0 = frame->data[0];
+    p0 = picref->data[0];
     puts("\033c");
-    for (y = 0; y < frame->height; y++) {
+    for (y = 0; y < picref->video->h; y++) {
         p = p0;
-        for (x = 0; x < frame->width; x++)
+        for (x = 0; x < picref->video->w; x++)
             putchar(" .-+#"[*(p++) / 52]);
         putchar('\n');
-        p0 += frame->linesize[0];
+        p0 += picref->linesize[0];
     }
     fflush(stdout);
 }
@@ -188,19 +172,15 @@ int main(int argc, char **argv)
 {
     int ret;
     AVPacket packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVFrame frame;
     int got_frame;
 
-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file\n", argv[0]);
         exit(1);
     }
 
+    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -211,50 +191,55 @@ int main(int argc, char **argv)
 
     /* read all packets */
     while (1) {
+        AVFilterBufferRef *picref;
         if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
             break;
 
         if (packet.stream_index == video_stream_index) {
+            avcodec_get_frame_defaults(&frame);
             got_frame = 0;
-            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
+            ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                 break;
             }
 
             if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
+                frame.pts = av_frame_get_best_effort_timestamp(&frame);
 
                 /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
                     av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                     break;
                 }
 
-                /* pull filtered frames from the filtergraph */
+                /* pull filtered pictures from the filtergraph */
                 while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
                     if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
                     if (ret < 0)
                         goto end;
-                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
-                    av_frame_unref(filt_frame);
+
+                    if (picref) {
+                        display_picref(picref, buffersink_ctx->inputs[0]->time_base);
+                        avfilter_unref_bufferp(&picref);
+                    }
                 }
-                av_frame_unref(frame);
             }
         }
         av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    avcodec_close(dec_ctx);
+    if (dec_ctx)
+        avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
         exit(1);
     }
 

doc/examples/metadata.c

@@ -23,7 +23,6 @@
 /**
  * @file
  * Shows how the metadata API can be used in application programs.
- * @example metadata.c
  */
 
 #include <stdio.h>

doc/examples/muxing.c

@@ -24,9 +24,8 @@
  * @file
  * libavformat API example.
  *
- * Output a media file in any supported libavformat format. The default
- * codecs are used.
- * @example muxing.c
+ * Output a media file in any supported libavformat format.
+ * The default codecs are used.
  */
 
 #include <stdlib.h>
@@ -34,401 +33,267 @@
 #include <string.h>
 #include <math.h>
 
-#include <libavutil/avassert.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
-#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
-#include <libswresample/swresample.h>
 
-#define STREAM_DURATION   10.0
+/* 5 seconds stream duration */
+#define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
+#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
+#define STREAM_PIX_FMT    PIX_FMT_YUV420P /* default pix_fmt */
 
-#define SCALE_FLAGS SWS_BICUBIC
-
-// a wrapper around a single output AVStream
-typedef struct OutputStream {
-    AVStream *st;
-
-    /* pts of the next frame that will be generated */
-    int64_t next_pts;
-    int samples_count;
-
-    AVFrame *frame;
-    AVFrame *tmp_frame;
-
-    float t, tincr, tincr2;
-
-    struct SwsContext *sws_ctx;
-    struct SwrContext *swr_ctx;
-} OutputStream;
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
-{
-    /* rescale output packet timestamp values from codec to stream timebase */
-    av_packet_rescale_ts(pkt, *time_base, st->time_base);
-    pkt->stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    log_packet(fmt_ctx, pkt);
-    return av_interleaved_write_frame(fmt_ctx, pkt);
-}
-
-/* Add an output stream. */
-static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       AVCodec **codec,
-                       enum AVCodecID codec_id)
-{
-    AVCodecContext *c;
-    int i;
-
-    /* find the encoder */
-    *codec = avcodec_find_encoder(codec_id);
-    if (!(*codec)) {
-        fprintf(stderr, "Could not find encoder for '%s'\n",
-                avcodec_get_name(codec_id));
-        exit(1);
-    }
-
-    ost->st = avformat_new_stream(oc, *codec);
-    if (!ost->st) {
-        fprintf(stderr, "Could not allocate stream\n");
-        exit(1);
-    }
-    ost->st->id = oc->nb_streams-1;
-    c = ost->st->codec;
-
-    switch ((*codec)->type) {
-    case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = (*codec)->sample_fmts ?
-            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
-        c->bit_rate    = 64000;
-        c->sample_rate = 44100;
-        if ((*codec)->supported_samplerates) {
-            c->sample_rate = (*codec)->supported_samplerates[0];
-            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
-                if ((*codec)->supported_samplerates[i] == 44100)
-                    c->sample_rate = 44100;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        c->channel_layout = AV_CH_LAYOUT_STEREO;
-        if ((*codec)->channel_layouts) {
-            c->channel_layout = (*codec)->channel_layouts[0];
-            for (i = 0; (*codec)->channel_layouts[i]; i++) {
-                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
-                    c->channel_layout = AV_CH_LAYOUT_STEREO;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        ost->st->time_base = (AVRational){ 1, c->sample_rate };
-        break;
-
-    case AVMEDIA_TYPE_VIDEO:
-        c->codec_id = codec_id;
-
-        c->bit_rate = 400000;
-        /* Resolution must be a multiple of two. */
-        c->width    = 352;
-        c->height   = 288;
-        /* timebase: This is the fundamental unit of time (in seconds) in terms
-         * of which frame timestamps are represented. For fixed-fps content,
-         * timebase should be 1/framerate and timestamp increments should be
-         * identical to 1. */
-        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
-        c->time_base       = ost->st->time_base;
-
-        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
-        c->pix_fmt       = STREAM_PIX_FMT;
-        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
-            /* just for testing, we also add B frames */
-            c->max_b_frames = 2;
-        }
-        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
-            /* Needed to avoid using macroblocks in which some coeffs overflow.
-             * This does not happen with normal video, it just happens here as
-             * the motion of the chroma plane does not match the luma plane. */
-            c->mb_decision = 2;
-        }
-    break;
-
-    default:
-        break;
-    }
-
-    /* Some formats want stream headers to be separate. */
-    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
-}
+static int sws_flags = SWS_BICUBIC;
 
 /**************************************************************/
 /* audio output */
 
-static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  uint64_t channel_layout,
-                                  int sample_rate, int nb_samples)
-{
-    AVFrame *frame = av_frame_alloc();
-    int ret;
+static float t, tincr, tincr2;
+static int16_t *samples;
+static int audio_input_frame_size;
 
-    if (!frame) {
-        fprintf(stderr, "Error allocating an audio frame\n");
+/*
+ * add an audio output stream
+ */
+static AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
+                                  enum AVCodecID codec_id)
+{
+    AVCodecContext *c;
+    AVStream *st;
+
+    /* find the audio encoder */
+    *codec = avcodec_find_encoder(codec_id);
+    if (!(*codec)) {
+        fprintf(stderr, "Could not find codec\n");
         exit(1);
     }
 
-    frame->format = sample_fmt;
-    frame->channel_layout = channel_layout;
-    frame->sample_rate = sample_rate;
-    frame->nb_samples = nb_samples;
-
-    if (nb_samples) {
-        ret = av_frame_get_buffer(frame, 0);
-        if (ret < 0) {
-            fprintf(stderr, "Error allocating an audio buffer\n");
-            exit(1);
-        }
+    st = avformat_new_stream(oc, *codec);
+    if (!st) {
+        fprintf(stderr, "Could not allocate stream\n");
+        exit(1);
     }
+    st->id = 1;
 
-    return frame;
+    c = st->codec;
+
+    /* put sample parameters */
+    c->sample_fmt  = AV_SAMPLE_FMT_S16;
+    c->bit_rate    = 64000;
+    c->sample_rate = 44100;
+    c->channels    = 2;
+
+    // some formats want stream headers to be separate
+    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    return st;
 }
 
-static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
     AVCodecContext *c;
-    int nb_samples;
-    int ret;
-    AVDictionary *opt = NULL;
 
-    c = ost->st->codec;
+    c = st->codec;
 
     /* open it */
-    av_dict_copy(&opt, opt_arg, 0);
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     /* init signal generator */
-    ost->t     = 0;
-    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
+    t     = 0;
+    tincr = 2 * M_PI * 110.0 / c->sample_rate;
     /* increment frequency by 110 Hz per second */
-    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
     if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        nb_samples = 10000;
+        audio_input_frame_size = 10000;
     else
-        nb_samples = c->frame_size;
-
-    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
-                                       c->sample_rate, nb_samples);
-    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
-                                       c->sample_rate, nb_samples);
-
-    /* create resampler context */
-        ost->swr_ctx = swr_alloc();
-        if (!ost->swr_ctx) {
-            fprintf(stderr, "Could not allocate resampler context\n");
-            exit(1);
-        }
-
-        /* set options */
-        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
-
-        /* initialize the resampling context */
-        if ((ret = swr_init(ost->swr_ctx)) < 0) {
-            fprintf(stderr, "Failed to initialize the resampling context\n");
-            exit(1);
-        }
+        audio_input_frame_size = c->frame_size;
+    samples = av_malloc(audio_input_frame_size *
+                        av_get_bytes_per_sample(c->sample_fmt) *
+                        c->channels);
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
  * 'nb_channels' channels. */
-static AVFrame *get_audio_frame(OutputStream *ost)
+static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
-    AVFrame *frame = ost->tmp_frame;
     int j, i, v;
-    int16_t *q = (int16_t*)frame->data[0];
+    int16_t *q;
 
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    for (j = 0; j <frame->nb_samples; j++) {
-        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->st->codec->channels; i++)
+    q = samples;
+    for (j = 0; j < frame_size; j++) {
+        v = (int)(sin(t) * 10000);
+        for (i = 0; i < nb_channels; i++)
             *q++ = v;
-        ost->t     += ost->tincr;
-        ost->tincr += ost->tincr2;
+        t     += tincr;
+        tincr += tincr2;
     }
-
-    frame->pts = ost->next_pts;
-    ost->next_pts  += frame->nb_samples;
-
-    return frame;
 }
 
-/*
- * encode one audio frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame;
-    int ret;
+    AVFrame *frame = avcodec_alloc_frame();
     int got_packet;
-    int dst_nb_samples;
 
     av_init_packet(&pkt);
-    c = ost->st->codec;
+    c = st->codec;
 
-    frame = get_audio_frame(ost);
+    get_audio_frame(samples, audio_input_frame_size, c->channels);
+    frame->nb_samples = audio_input_frame_size;
+    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                             (uint8_t *)samples,
+                             audio_input_frame_size *
+                             av_get_bytes_per_sample(c->sample_fmt) *
+                             c->channels, 1);
 
-    if (frame) {
-        /* convert samples from native format to destination codec format, using the resampler */
-            /* compute destination number of samples */
-            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
-                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
-            av_assert0(dst_nb_samples == frame->nb_samples);
+    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (!got_packet)
+        return;
 
-        /* when we pass a frame to the encoder, it may keep a reference to it
-         * internally;
-         * make sure we do not overwrite it here
-         */
-        ret = av_frame_make_writable(ost->frame);
-        if (ret < 0)
-            exit(1);
+    pkt.stream_index = st->index;
 
-            /* convert to destination format */
-            ret = swr_convert(ost->swr_ctx,
-                              ost->frame->data, dst_nb_samples,
-                              (const uint8_t **)frame->data, frame->nb_samples);
-            if (ret < 0) {
-                fprintf(stderr, "Error while converting\n");
-                exit(1);
-            }
-            frame = ost->frame;
-
-        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        ost->samples_count += dst_nb_samples;
-    }
-
-    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
-    if (ret < 0) {
-        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
+    /* Write the compressed frame to the media file. */
+    if (av_interleaved_write_frame(oc, &pkt) != 0) {
+        fprintf(stderr, "Error while writing audio frame\n");
         exit(1);
     }
+    avcodec_free_frame(&frame);
+}
 
-    if (got_packet) {
-        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error while writing audio frame: %s\n",
-                    av_err2str(ret));
-            exit(1);
-        }
-    }
+static void close_audio(AVFormatContext *oc, AVStream *st)
+{
+    avcodec_close(st->codec);
 
-    return (frame || got_packet) ? 0 : 1;
+    av_free(samples);
 }
 
 /**************************************************************/
 /* video output */
 
-static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
+static AVFrame *frame;
+static AVPicture src_picture, dst_picture;
+static uint8_t *video_outbuf;
+static int frame_count, video_outbuf_size;
+
+/* Add a video output stream. */
+static AVStream *add_video_stream(AVFormatContext *oc, AVCodec **codec,
+                                  enum AVCodecID codec_id)
 {
-    AVFrame *picture;
-    int ret;
+    AVCodecContext *c;
+    AVStream *st;
 
-    picture = av_frame_alloc();
-    if (!picture)
-        return NULL;
-
-    picture->format = pix_fmt;
-    picture->width  = width;
-    picture->height = height;
-
-    /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate frame data.\n");
+    /* find the video encoder */
+    *codec = avcodec_find_encoder(codec_id);
+    if (!(*codec)) {
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
-    return picture;
+    st = avformat_new_stream(oc, *codec);
+    if (!st) {
+        fprintf(stderr, "Could not alloc stream\n");
+        exit(1);
+    }
+
+    c = st->codec;
+
+    avcodec_get_context_defaults3(c, *codec);
+
+    c->codec_id = codec_id;
+
+    /* Put sample parameters. */
+    c->bit_rate = 400000;
+    /* Resolution must be a multiple of two. */
+    c->width    = 352;
+    c->height   = 288;
+    /* timebase: This is the fundamental unit of time (in seconds) in terms
+     * of which frame timestamps are represented. For fixed-fps content,
+     * timebase should be 1/framerate and timestamp increments should be
+     * identical to 1. */
+    c->time_base.den = STREAM_FRAME_RATE;
+    c->time_base.num = 1;
+    c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
+    c->pix_fmt       = STREAM_PIX_FMT;
+    if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
+        /* just for testing, we also add B frames */
+        c->max_b_frames = 2;
+    }
+    if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
+        /* Needed to avoid using macroblocks in which some coeffs overflow.
+         * This does not happen with normal video, it just happens here as
+         * the motion of the chroma plane does not match the luma plane. */
+        c->mb_decision = 2;
+    }
+    /* Some formats want stream headers to be separate. */
+    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    return st;
 }
 
-static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
     int ret;
-    AVCodecContext *c = ost->st->codec;
-    AVDictionary *opt = NULL;
-
-    av_dict_copy(&opt, opt_arg, 0);
+    AVCodecContext *c = st->codec;
 
     /* open the codec */
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
         exit(1);
     }
 
+    video_outbuf = NULL;
+    if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
+        /* Allocate output buffer. */
+        /* XXX: API change will be done. */
+        /* Buffers passed into lav* can be allocated any way you prefer,
+         * as long as they're aligned enough for the architecture, and
+         * they're freed appropriately (such as using av_free for buffers
+         * allocated with av_malloc). */
+        video_outbuf_size = 200000;
+        video_outbuf      = av_malloc(video_outbuf_size);
+    }
+
     /* allocate and init a re-usable frame */
-    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
-    if (!ost->frame) {
+    frame = avcodec_alloc_frame();
+    if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
 
+    /* Allocate the encoded raw picture. */
+    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate picture\n");
+        exit(1);
+    }
+
     /* If the output format is not YUV420P, then a temporary YUV420P
      * picture is needed too. It is then converted to the required
      * output format. */
-    ost->tmp_frame = NULL;
-    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
-        if (!ost->tmp_frame) {
+    if (c->pix_fmt != PIX_FMT_YUV420P) {
+        ret = avpicture_alloc(&src_picture, PIX_FMT_YUV420P, c->width, c->height);
+        if (ret < 0) {
             fprintf(stderr, "Could not allocate temporary picture\n");
             exit(1);
         }
     }
+
+    /* copy data and linesize picture pointers to frame */
+    *((AVPicture *)frame) = dst_picture;
 }
 
 /* Prepare a dummy image. */
-static void fill_yuv_image(AVFrame *pict, int frame_index,
+static void fill_yuv_image(AVPicture *pict, int frame_index,
                            int width, int height)
 {
-    int x, y, i, ret;
-
-    /* when we pass a frame to the encoder, it may keep a reference to it
-     * internally;
-     * make sure we do not overwrite it here
-     */
-    ret = av_frame_make_writable(pict);
-    if (ret < 0)
-        exit(1);
+    int x, y, i;
 
     i = frame_index;
 
@@ -446,108 +311,96 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
     }
 }
 
-static AVFrame *get_video_frame(OutputStream *ost)
-{
-    AVCodecContext *c = ost->st->codec;
-
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        /* as we only generate a YUV420P picture, we must convert it
-         * to the codec pixel format if needed */
-        if (!ost->sws_ctx) {
-            ost->sws_ctx = sws_getContext(c->width, c->height,
-                                          AV_PIX_FMT_YUV420P,
-                                          c->width, c->height,
-                                          c->pix_fmt,
-                                          SCALE_FLAGS, NULL, NULL, NULL);
-            if (!ost->sws_ctx) {
-                fprintf(stderr,
-                        "Could not initialize the conversion context\n");
-                exit(1);
-            }
-        }
-        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
-        sws_scale(ost->sws_ctx,
-                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
-                  0, c->height, ost->frame->data, ost->frame->linesize);
-    } else {
-        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
-    }
-
-    ost->frame->pts = ost->next_pts++;
-
-    return ost->frame;
-}
-
-/*
- * encode one video frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
     int ret;
-    AVCodecContext *c;
-    AVFrame *frame;
-    int got_packet = 0;
+    static struct SwsContext *sws_ctx;
+    AVCodecContext *c = st->codec;
 
-    c = ost->st->codec;
-
-    frame = get_video_frame(ost);
+    if (frame_count >= STREAM_NB_FRAMES) {
+        /* No more frames to compress. The codec has a latency of a few
+         * frames if using B-frames, so we get the last frames by
+         * passing the same picture again. */
+    } else {
+        if (c->pix_fmt != PIX_FMT_YUV420P) {
+            /* as we only generate a YUV420P picture, we must convert it
+             * to the codec pixel format if needed */
+            if (!sws_ctx) {
+                sws_ctx = sws_getContext(c->width, c->height, PIX_FMT_YUV420P,
+                                         c->width, c->height, c->pix_fmt,
+                                         sws_flags, NULL, NULL, NULL);
+                if (!sws_ctx) {
+                    fprintf(stderr,
+                            "Could not initialize the conversion context\n");
+                    exit(1);
+                }
+            }
+            fill_yuv_image(&src_picture, frame_count, c->width, c->height);
+            sws_scale(sws_ctx,
+                      (const uint8_t * const *)src_picture.data, src_picture.linesize,
+                      0, c->height, dst_picture.data, dst_picture.linesize);
+        } else {
+            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
+        }
+    }
 
     if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* a hack to avoid data copy with some raw video muxers */
+        /* Raw video case - the API will change slightly in the near
+         * future for that. */
         AVPacket pkt;
         av_init_packet(&pkt);
 
-        if (!frame)
-            return 1;
-
         pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = ost->st->index;
-        pkt.data          = (uint8_t *)frame;
+        pkt.stream_index  = st->index;
+        pkt.data          = dst_picture.data[0];
         pkt.size          = sizeof(AVPicture);
 
-        pkt.pts = pkt.dts = frame->pts;
-        av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
-
         ret = av_interleaved_write_frame(oc, &pkt);
     } else {
-        AVPacket pkt = { 0 };
-        av_init_packet(&pkt);
-
         /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+        AVPacket pkt;
+        int got_output;
+
+        av_init_packet(&pkt);
+        pkt.data = NULL;    // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
         if (ret < 0) {
-            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
+            fprintf(stderr, "Error encoding video frame\n");
             exit(1);
         }
 
-        if (got_packet) {
-            ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        /* If size is zero, it means the image was buffered. */
+        if (got_output) {
+            if (c->coded_frame->pts != AV_NOPTS_VALUE)
+                pkt.pts = av_rescale_q(c->coded_frame->pts,
+                                       c->time_base, st->time_base);
+            if (c->coded_frame->key_frame)
+                pkt.flags |= AV_PKT_FLAG_KEY;
+
+            pkt.stream_index = st->index;
+
+            /* Write the compressed frame to the media file. */
+            ret = av_interleaved_write_frame(oc, &pkt);
         } else {
             ret = 0;
         }
     }
-
-    if (ret < 0) {
-        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+    if (ret != 0) {
+        fprintf(stderr, "Error while writing video frame\n");
         exit(1);
     }
-
-    return (frame || got_packet) ? 0 : 1;
+    frame_count++;
 }
 
-static void close_stream(AVFormatContext *oc, OutputStream *ost)
+static void close_video(AVFormatContext *oc, AVStream *st)
 {
-    avcodec_close(ost->st->codec);
-    av_frame_free(&ost->frame);
-    av_frame_free(&ost->tmp_frame);
-    sws_freeContext(ost->sws_ctx);
-    swr_free(&ost->swr_ctx);
+    avcodec_close(st->codec);
+    av_free(src_picture.data[0]);
+    av_free(dst_picture.data[0]);
+    av_free(frame);
+    av_free(video_outbuf);
 }
 
 /**************************************************************/
@@ -555,34 +408,27 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 
 int main(int argc, char **argv)
 {
-    OutputStream video_st = { 0 }, audio_st = { 0 };
     const char *filename;
     AVOutputFormat *fmt;
     AVFormatContext *oc;
+    AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
-    int ret;
-    int have_video = 0, have_audio = 0;
-    int encode_video = 0, encode_audio = 0;
-    AVDictionary *opt = NULL;
+    double audio_pts, video_pts;
+    int i;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
 
-    if (argc < 2) {
+    if (argc != 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"
-               "This program generates a synthetic audio and video stream, encodes and\n"
-               "muxes them into a file named output_file.\n"
                "The output format is automatically guessed according to the file extension.\n"
-               "Raw images can also be output by using '%%d' in the filename.\n"
+               "Raw images can also be output by using '%%d' in the filename\n"
                "\n", argv[0]);
         return 1;
     }
 
     filename = argv[1];
-    if (argc > 3 && !strcmp(argv[2], "-flags")) {
-        av_dict_set(&opt, argv[2]+1, argv[3], 0);
-    }
 
     /* allocate the output media context */
     avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -590,60 +436,69 @@ int main(int argc, char **argv)
         printf("Could not deduce output format from file extension: using MPEG.\n");
         avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
     }
-    if (!oc)
+    if (!oc) {
         return 1;
-
+    }
     fmt = oc->oformat;
 
     /* Add the audio and video streams using the default format codecs
      * and initialize the codecs. */
+    video_st = NULL;
+    audio_st = NULL;
     if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
-        have_video = 1;
-        encode_video = 1;
+        video_st = add_video_stream(oc, &video_codec, fmt->video_codec);
     }
     if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
-        have_audio = 1;
-        encode_audio = 1;
+        audio_st = add_audio_stream(oc, &audio_codec, fmt->audio_codec);
     }
 
     /* Now that all the parameters are set, we can open the audio and
      * video codecs and allocate the necessary encode buffers. */
-    if (have_video)
-        open_video(oc, video_codec, &video_st, opt);
-
-    if (have_audio)
-        open_audio(oc, audio_codec, &audio_st, opt);
+    if (video_st)
+        open_video(oc, video_codec, video_st);
+    if (audio_st)
+        open_audio(oc, audio_codec, audio_st);
 
     av_dump_format(oc, 0, filename, 1);
 
     /* open the output file, if needed */
     if (!(fmt->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            fprintf(stderr, "Could not open '%s': %s\n", filename,
-                    av_err2str(ret));
+        if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
+            fprintf(stderr, "Could not open '%s'\n", filename);
             return 1;
         }
     }
 
     /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, &opt);
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred when opening output file: %s\n",
-                av_err2str(ret));
+    if (avformat_write_header(oc, NULL) < 0) {
+        fprintf(stderr, "Error occurred when opening output file\n");
         return 1;
     }
 
-    while (encode_video || encode_audio) {
-        /* select the stream to encode */
-        if (encode_video &&
-            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
-                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
-            encode_video = !write_video_frame(oc, &video_st);
+    frame->pts = 0;
+    for (;;) {
+        /* Compute current audio and video time. */
+        if (audio_st)
+            audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
+        else
+            audio_pts = 0.0;
+
+        if (video_st)
+            video_pts = (double)video_st->pts.val * video_st->time_base.num /
+                        video_st->time_base.den;
+        else
+            video_pts = 0.0;
+
+        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
+            (!video_st || video_pts >= STREAM_DURATION))
+            break;
+
+        /* write interleaved audio and video frames */
+        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+            write_audio_frame(oc, audio_st);
         } else {
-            encode_audio = !write_audio_frame(oc, &audio_st);
+            write_video_frame(oc, video_st);
+            frame->pts++;
         }
     }
 
@@ -654,17 +509,23 @@ int main(int argc, char **argv)
     av_write_trailer(oc);
 
     /* Close each codec. */
-    if (have_video)
-        close_stream(oc, &video_st);
-    if (have_audio)
-        close_stream(oc, &audio_st);
+    if (video_st)
+        close_video(oc, video_st);
+    if (audio_st)
+        close_audio(oc, audio_st);
+
+    /* Free the streams. */
+    for (i = 0; i < oc->nb_streams; i++) {
+        av_freep(&oc->streams[i]->codec);
+        av_freep(&oc->streams[i]);
+    }
 
     if (!(fmt->flags & AVFMT_NOFILE))
         /* Close the output file. */
         avio_close(oc->pb);
 
     /* free the stream */
-    avformat_free_context(oc);
+    av_free(oc);
 
     return 0;
 }

doc/examples/remuxing.c

@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2013 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat/libavcodec demuxing and muxing API example.
- *
- * Remux streams from one container format to another.
- * @example remuxing.c
- */
-
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           tag,
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-int main(int argc, char **argv)
-{
-    AVOutputFormat *ofmt = NULL;
-    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket pkt;
-    const char *in_filename, *out_filename;
-    int ret, i;
-
-    if (argc < 3) {
-        printf("usage: %s input output\n"
-               "API example program to remux a media file with libavformat and libavcodec.\n"
-               "The output format is guessed according to the file extension.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    in_filename  = argv[1];
-    out_filename = argv[2];
-
-    av_register_all();
-
-    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
-        fprintf(stderr, "Could not open input file '%s'", in_filename);
-        goto end;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
-        fprintf(stderr, "Failed to retrieve input stream information");
-        goto end;
-    }
-
-    av_dump_format(ifmt_ctx, 0, in_filename, 0);
-
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
-    if (!ofmt_ctx) {
-        fprintf(stderr, "Could not create output context\n");
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    ofmt = ofmt_ctx->oformat;
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *in_stream = ifmt_ctx->streams[i];
-        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
-        if (!out_stream) {
-            fprintf(stderr, "Failed allocating output stream\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
-            goto end;
-        }
-        out_stream->codec->codec_tag = 0;
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
-    }
-    av_dump_format(ofmt_ctx, 0, out_filename, 1);
-
-    if (!(ofmt->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            fprintf(stderr, "Could not open output file '%s'", out_filename);
-            goto end;
-        }
-    }
-
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred when opening output file\n");
-        goto end;
-    }
-
-    while (1) {
-        AVStream *in_stream, *out_stream;
-
-        ret = av_read_frame(ifmt_ctx, &pkt);
-        if (ret < 0)
-            break;
-
-        in_stream  = ifmt_ctx->streams[pkt.stream_index];
-        out_stream = ofmt_ctx->streams[pkt.stream_index];
-
-        log_packet(ifmt_ctx, &pkt, "in");
-
-        /* copy packet */
-        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
-        pkt.pos = -1;
-        log_packet(ofmt_ctx, &pkt, "out");
-
-        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error muxing packet\n");
-            break;
-        }
-        av_free_packet(&pkt);
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-
-    avformat_close_input(&ifmt_ctx);
-
-    /* close output */
-    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_close(ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}

doc/examples/resampling_audio.c

@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @example resampling_audio.c
- * libswresample API use example.
- */
-
-#include <libavutil/opt.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/samplefmt.h>
-#include <libswresample/swresample.h>
-
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "Sample format %s not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return AVERROR(EINVAL);
-}
-
-/**
- * Fill dst buffer with nb_samples, generated starting from t.
- */
-static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
-{
-    int i, j;
-    double tincr = 1.0 / sample_rate, *dstp = dst;
-    const double c = 2 * M_PI * 440.0;
-
-    /* generate sin tone with 440Hz frequency and duplicated channels */
-    for (i = 0; i < nb_samples; i++) {
-        *dstp = sin(c * *t);
-        for (j = 1; j < nb_channels; j++)
-            dstp[j] = dstp[0];
-        dstp += nb_channels;
-        *t += tincr;
-    }
-}
-
-int main(int argc, char **argv)
-{
-    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
-    int src_rate = 48000, dst_rate = 44100;
-    uint8_t **src_data = NULL, **dst_data = NULL;
-    int src_nb_channels = 0, dst_nb_channels = 0;
-    int src_linesize, dst_linesize;
-    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
-    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
-    const char *dst_filename = NULL;
-    FILE *dst_file;
-    int dst_bufsize;
-    const char *fmt;
-    struct SwrContext *swr_ctx;
-    double t;
-    int ret;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s output_file\n"
-                "API example program to show how to resample an audio stream with libswresample.\n"
-                "This program generates a series of audio frames, resamples them to a specified "
-                "output format and rate and saves them to an output file named output_file.\n",
-            argv[0]);
-        exit(1);
-    }
-    dst_filename = argv[1];
-
-    dst_file = fopen(dst_filename, "wb");
-    if (!dst_file) {
-        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
-        exit(1);
-    }
-
-    /* create resampler context */
-    swr_ctx = swr_alloc();
-    if (!swr_ctx) {
-        fprintf(stderr, "Could not allocate resampler context\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    /* set options */
-    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
-    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
-    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
-
-    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
-    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
-    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
-
-    /* initialize the resampling context */
-    if ((ret = swr_init(swr_ctx)) < 0) {
-        fprintf(stderr, "Failed to initialize the resampling context\n");
-        goto end;
-    }
-
-    /* allocate source and destination samples buffers */
-
-    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
-                                             src_nb_samples, src_sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate source samples\n");
-        goto end;
-    }
-
-    /* compute the number of converted samples: buffering is avoided
-     * ensuring that the output buffer will contain at least all the
-     * converted input samples */
-    max_dst_nb_samples = dst_nb_samples =
-        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
-
-    /* buffer is going to be directly written to a rawaudio file, no alignment */
-    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
-    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
-                                             dst_nb_samples, dst_sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate destination samples\n");
-        goto end;
-    }
-
-    t = 0;
-    do {
-        /* generate synthetic audio */
-        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
-
-        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
-                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
-        if (dst_nb_samples > max_dst_nb_samples) {
-            av_freep(&dst_data[0]);
-            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
-                                   dst_nb_samples, dst_sample_fmt, 1);
-            if (ret < 0)
-                break;
-            max_dst_nb_samples = dst_nb_samples;
-        }
-
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            goto end;
-        }
-        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
-                                                 ret, dst_sample_fmt, 1);
-        if (dst_bufsize < 0) {
-            fprintf(stderr, "Could not get sample buffer size\n");
-            goto end;
-        }
-        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
-        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
-    } while (t < 10);
-
-    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
-        goto end;
-    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
-            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
-
-end:
-    fclose(dst_file);
-
-    if (src_data)
-        av_freep(&src_data[0]);
-    av_freep(&src_data);
-
-    if (dst_data)
-        av_freep(&dst_data[0]);
-    av_freep(&dst_data);
-
-    swr_free(&swr_ctx);
-    return ret < 0;
-}

doc/examples/scaling_video.c

@@ -23,7 +23,6 @@
 /**
  * @file
  * libswscale API use example.
- * @example scaling_video.c
  */
 
 #include <libavutil/imgutils.h>
@@ -33,18 +32,20 @@
 static void fill_yuv_image(uint8_t *data[4], int linesize[4],
                            int width, int height, int frame_index)
 {
-    int x, y;
+    int x, y, i;
+
+    i = frame_index;
 
     /* Y */
     for (y = 0; y < height; y++)
         for (x = 0; x < width; x++)
-            data[0][y * linesize[0] + x] = x + y + frame_index * 3;
+            data[0][y * linesize[0] + x] = x + y + i * 3;
 
     /* Cb and Cr */
     for (y = 0; y < height / 2; y++) {
         for (x = 0; x < width / 2; x++) {
-            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
-            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
+            data[1][y * linesize[1] + x] = 128 + y + i * 2;
+            data[2][y * linesize[2] + x] = 64 + x + i * 5;
         }
     }
 }
@@ -54,7 +55,7 @@ int main(int argc, char **argv)
     uint8_t *src_data[4], *dst_data[4];
     int src_linesize[4], dst_linesize[4];
     int src_w = 320, src_h = 240, dst_w, dst_h;
-    enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
+    enum PixelFormat src_pix_fmt = PIX_FMT_YUV420P, dst_pix_fmt = PIX_FMT_RGB24;
     const char *dst_size = NULL;
     const char *dst_filename = NULL;
     FILE *dst_file;
@@ -107,7 +108,7 @@ int main(int argc, char **argv)
         goto end;
     }
 
-    /* buffer is going to be written to rawvideo file, no alignment */
+    /* buffer is going to be written to rawvideo file, no alignmnet */
     if ((ret = av_image_alloc(dst_data, dst_linesize,
                               dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
         fprintf(stderr, "Could not allocate destination image\n");
@@ -132,7 +133,8 @@ int main(int argc, char **argv)
            av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
 
 end:
-    fclose(dst_file);
+    if (dst_file)
+        fclose(dst_file);
     av_freep(&src_data[0]);
     av_freep(&dst_data[0]);
     sws_freeContext(sws_ctx);

doc/examples/transcode_aac.c

@@ -1,755 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * simple audio converter
- *
- * @example transcode_aac.c
- * Convert an input audio file to AAC in an MP4 container using FFmpeg.
- * @author Andreas Unterweger (dustsigns@gmail.com)
- */
-
-#include <stdio.h>
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-
-#include "libavutil/audio_fifo.h"
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/frame.h"
-#include "libavutil/opt.h"
-
-#include "libswresample/swresample.h"
-
-/** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 48000
-/** The number of output channels */
-#define OUTPUT_CHANNELS 2
-/** The audio sample output format */
-#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
-
-/**
- * Convert an error code into a text message.
- * @param error Error code to be converted
- * @return Corresponding error text (not thread-safe)
- */
-static const char *get_error_text(const int error)
-{
-    static char error_buffer[255];
-    av_strerror(error, error_buffer, sizeof(error_buffer));
-    return error_buffer;
-}
-
-/** Open an input file and the required decoder. */
-static int open_input_file(const char *filename,
-                           AVFormatContext **input_format_context,
-                           AVCodecContext **input_codec_context)
-{
-    AVCodec *input_codec;
-    int error;
-
-    /** Open the input file to read from it. */
-    if ((error = avformat_open_input(input_format_context, filename, NULL,
-                                     NULL)) < 0) {
-        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        *input_format_context = NULL;
-        return error;
-    }
-
-    /** Get information on the input file (number of streams etc.). */
-    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not open find stream info (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Make sure that there is only one stream in the input file. */
-    if ((*input_format_context)->nb_streams != 1) {
-        fprintf(stderr, "Expected one audio input stream, but found %d\n",
-                (*input_format_context)->nb_streams);
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
-        fprintf(stderr, "Could not find input codec\n");
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Open the decoder for the audio stream to use it later. */
-    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
-                               input_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open input codec (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Save the decoder context for easier access later. */
-    *input_codec_context = (*input_format_context)->streams[0]->codec;
-
-    return 0;
-}
-
-/**
- * Open an output file and the required encoder.
- * Also set some basic encoder parameters.
- * Some of these parameters are based on the input file's parameters.
- */
-static int open_output_file(const char *filename,
-                            AVCodecContext *input_codec_context,
-                            AVFormatContext **output_format_context,
-                            AVCodecContext **output_codec_context)
-{
-    AVIOContext *output_io_context = NULL;
-    AVStream *stream               = NULL;
-    AVCodec *output_codec          = NULL;
-    int error;
-
-    /** Open the output file to write to it. */
-    if ((error = avio_open(&output_io_context, filename,
-                           AVIO_FLAG_WRITE)) < 0) {
-        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        return error;
-    }
-
-    /** Create a new format context for the output container format. */
-    if (!(*output_format_context = avformat_alloc_context())) {
-        fprintf(stderr, "Could not allocate output format context\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /** Associate the output file (pointer) with the container format context. */
-    (*output_format_context)->pb = output_io_context;
-
-    /** Guess the desired container format based on the file extension. */
-    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
-                                                              NULL))) {
-        fprintf(stderr, "Could not find output file format\n");
-        goto cleanup;
-    }
-
-    av_strlcpy((*output_format_context)->filename, filename,
-               sizeof((*output_format_context)->filename));
-
-    /** Find the encoder to be used by its name. */
-    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
-        fprintf(stderr, "Could not find an AAC encoder.\n");
-        goto cleanup;
-    }
-
-    /** Create a new audio stream in the output file container. */
-    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
-        fprintf(stderr, "Could not create new stream\n");
-        error = AVERROR(ENOMEM);
-        goto cleanup;
-    }
-
-    /** Save the encoder context for easiert access later. */
-    *output_codec_context = stream->codec;
-
-    /**
-     * Set the basic encoder parameters.
-     * The input file's sample rate is used to avoid a sample rate conversion.
-     */
-    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
-    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
-    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
-    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
-    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
-
-    /**
-     * Some container formats (like MP4) require global headers to be present
-     * Mark the encoder so that it behaves accordingly.
-     */
-    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
-    /** Open the encoder for the audio stream to use it later. */
-    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open output codec (error '%s')\n",
-                get_error_text(error));
-        goto cleanup;
-    }
-
-    return 0;
-
-cleanup:
-    avio_close((*output_format_context)->pb);
-    avformat_free_context(*output_format_context);
-    *output_format_context = NULL;
-    return error < 0 ? error : AVERROR_EXIT;
-}
-
-/** Initialize one data packet for reading or writing. */
-static void init_packet(AVPacket *packet)
-{
-    av_init_packet(packet);
-    /** Set the packet data and size so that it is recognized as being empty. */
-    packet->data = NULL;
-    packet->size = 0;
-}
-
-/** Initialize one audio frame for reading from the input file */
-static int init_input_frame(AVFrame **frame)
-{
-    if (!(*frame = av_frame_alloc())) {
-        fprintf(stderr, "Could not allocate input frame\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-/**
- * Initialize the audio resampler based on the input and output codec settings.
- * If the input and output sample formats differ, a conversion is required
- * libswresample takes care of this, but requires initialization.
- */
-static int init_resampler(AVCodecContext *input_codec_context,
-                          AVCodecContext *output_codec_context,
-                          SwrContext **resample_context)
-{
-        int error;
-
-        /**
-         * Create a resampler context for the conversion.
-         * Set the conversion parameters.
-         * Default channel layouts based on the number of channels
-         * are assumed for simplicity (they are sometimes not detected
-         * properly by the demuxer and/or decoder).
-         */
-        *resample_context = swr_alloc_set_opts(NULL,
-                                              av_get_default_channel_layout(output_codec_context->channels),
-                                              output_codec_context->sample_fmt,
-                                              output_codec_context->sample_rate,
-                                              av_get_default_channel_layout(input_codec_context->channels),
-                                              input_codec_context->sample_fmt,
-                                              input_codec_context->sample_rate,
-                                              0, NULL);
-        if (!*resample_context) {
-            fprintf(stderr, "Could not allocate resample context\n");
-            return AVERROR(ENOMEM);
-        }
-        /**
-        * Perform a sanity check so that the number of converted samples is
-        * not greater than the number of samples to be converted.
-        * If the sample rates differ, this case has to be handled differently
-        */
-        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
-
-        /** Open the resampler with the specified parameters. */
-        if ((error = swr_init(*resample_context)) < 0) {
-            fprintf(stderr, "Could not open resample context\n");
-            swr_free(resample_context);
-            return error;
-        }
-    return 0;
-}
-
-/** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int init_fifo(AVAudioFifo **fifo)
-{
-    /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
-        fprintf(stderr, "Could not allocate FIFO\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-/** Write the header of the output file container. */
-static int write_output_file_header(AVFormatContext *output_format_context)
-{
-    int error;
-    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not write output file header (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-    return 0;
-}
-
-/** Decode one audio frame from the input file. */
-static int decode_audio_frame(AVFrame *frame,
-                              AVFormatContext *input_format_context,
-                              AVCodecContext *input_codec_context,
-                              int *data_present, int *finished)
-{
-    /** Packet used for temporary storage. */
-    AVPacket input_packet;
-    int error;
-    init_packet(&input_packet);
-
-    /** Read one audio frame from the input file into a temporary packet. */
-    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /** If we are at the end of the file, flush the decoder below. */
-        if (error == AVERROR_EOF)
-            *finished = 1;
-        else {
-            fprintf(stderr, "Could not read frame (error '%s')\n",
-                    get_error_text(error));
-            return error;
-        }
-    }
-
-    /**
-     * Decode the audio frame stored in the temporary packet.
-     * The input audio stream decoder is used to do this.
-     * If we are at the end of the file, pass an empty packet to the decoder
-     * to flush it.
-     */
-    if ((error = avcodec_decode_audio4(input_codec_context, frame,
-                                       data_present, &input_packet)) < 0) {
-        fprintf(stderr, "Could not decode frame (error '%s')\n",
-                get_error_text(error));
-        av_free_packet(&input_packet);
-        return error;
-    }
-
-    /**
-     * If the decoder has not been flushed completely, we are not finished,
-     * so that this function has to be called again.
-     */
-    if (*finished && *data_present)
-        *finished = 0;
-    av_free_packet(&input_packet);
-    return 0;
-}
-
-/**
- * Initialize a temporary storage for the specified number of audio samples.
- * The conversion requires temporary storage due to the different format.
- * The number of audio samples to be allocated is specified in frame_size.
- */
-static int init_converted_samples(uint8_t ***converted_input_samples,
-                                  AVCodecContext *output_codec_context,
-                                  int frame_size)
-{
-    int error;
-
-    /**
-     * Allocate as many pointers as there are audio channels.
-     * Each pointer will later point to the audio samples of the corresponding
-     * channels (although it may be NULL for interleaved formats).
-     */
-    if (!(*converted_input_samples = calloc(output_codec_context->channels,
-                                            sizeof(**converted_input_samples)))) {
-        fprintf(stderr, "Could not allocate converted input sample pointers\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /**
-     * Allocate memory for the samples of all channels in one consecutive
-     * block for convenience.
-     */
-    if ((error = av_samples_alloc(*converted_input_samples, NULL,
-                                  output_codec_context->channels,
-                                  frame_size,
-                                  output_codec_context->sample_fmt, 0)) < 0) {
-        fprintf(stderr,
-                "Could not allocate converted input samples (error '%s')\n",
-                get_error_text(error));
-        av_freep(&(*converted_input_samples)[0]);
-        free(*converted_input_samples);
-        return error;
-    }
-    return 0;
-}
-
-/**
- * Convert the input audio samples into the output sample format.
- * The conversion happens on a per-frame basis, the size of which is specified
- * by frame_size.
- */
-static int convert_samples(const uint8_t **input_data,
-                           uint8_t **converted_data, const int frame_size,
-                           SwrContext *resample_context)
-{
-    int error;
-
-    /** Convert the samples using the resampler. */
-    if ((error = swr_convert(resample_context,
-                             converted_data, frame_size,
-                             input_data    , frame_size)) < 0) {
-        fprintf(stderr, "Could not convert input samples (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-
-    return 0;
-}
-
-/** Add converted input audio samples to the FIFO buffer for later processing. */
-static int add_samples_to_fifo(AVAudioFifo *fifo,
-                               uint8_t **converted_input_samples,
-                               const int frame_size)
-{
-    int error;
-
-    /**
-     * Make the FIFO as large as it needs to be to hold both,
-     * the old and the new samples.
-     */
-    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
-        fprintf(stderr, "Could not reallocate FIFO\n");
-        return error;
-    }
-
-    /** Store the new samples in the FIFO buffer. */
-    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
-                            frame_size) < frame_size) {
-        fprintf(stderr, "Could not write data to FIFO\n");
-        return AVERROR_EXIT;
-    }
-    return 0;
-}
-
-/**
- * Read one audio frame from the input file, decodes, converts and stores
- * it in the FIFO buffer.
- */
-static int read_decode_convert_and_store(AVAudioFifo *fifo,
-                                         AVFormatContext *input_format_context,
-                                         AVCodecContext *input_codec_context,
-                                         AVCodecContext *output_codec_context,
-                                         SwrContext *resampler_context,
-                                         int *finished)
-{
-    /** Temporary storage of the input samples of the frame read from the file. */
-    AVFrame *input_frame = NULL;
-    /** Temporary storage for the converted input samples. */
-    uint8_t **converted_input_samples = NULL;
-    int data_present;
-    int ret = AVERROR_EXIT;
-
-    /** Initialize temporary storage for one input frame. */
-    if (init_input_frame(&input_frame))
-        goto cleanup;
-    /** Decode one frame worth of audio samples. */
-    if (decode_audio_frame(input_frame, input_format_context,
-                           input_codec_context, &data_present, finished))
-        goto cleanup;
-    /**
-     * If we are at the end of the file and there are no more samples
-     * in the decoder which are delayed, we are actually finished.
-     * This must not be treated as an error.
-     */
-    if (*finished && !data_present) {
-        ret = 0;
-        goto cleanup;
-    }
-    /** If there is decoded data, convert and store it */
-    if (data_present) {
-        /** Initialize the temporary storage for the converted input samples. */
-        if (init_converted_samples(&converted_input_samples, output_codec_context,
-                                   input_frame->nb_samples))
-            goto cleanup;
-
-        /**
-         * Convert the input samples to the desired output sample format.
-         * This requires a temporary storage provided by converted_input_samples.
-         */
-        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
-                            input_frame->nb_samples, resampler_context))
-            goto cleanup;
-
-        /** Add the converted input samples to the FIFO buffer for later processing. */
-        if (add_samples_to_fifo(fifo, converted_input_samples,
-                                input_frame->nb_samples))
-            goto cleanup;
-        ret = 0;
-    }
-    ret = 0;
-
-cleanup:
-    if (converted_input_samples) {
-        av_freep(&converted_input_samples[0]);
-        free(converted_input_samples);
-    }
-    av_frame_free(&input_frame);
-
-    return ret;
-}
-
-/**
- * Initialize one input frame for writing to the output file.
- * The frame will be exactly frame_size samples large.
- */
-static int init_output_frame(AVFrame **frame,
-                             AVCodecContext *output_codec_context,
-                             int frame_size)
-{
-    int error;
-
-    /** Create a new frame to store the audio samples. */
-    if (!(*frame = av_frame_alloc())) {
-        fprintf(stderr, "Could not allocate output frame\n");
-        return AVERROR_EXIT;
-    }
-
-    /**
-     * Set the frame's parameters, especially its size and format.
-     * av_frame_get_buffer needs this to allocate memory for the
-     * audio samples of the frame.
-     * Default channel layouts based on the number of channels
-     * are assumed for simplicity.
-     */
-    (*frame)->nb_samples     = frame_size;
-    (*frame)->channel_layout = output_codec_context->channel_layout;
-    (*frame)->format         = output_codec_context->sample_fmt;
-    (*frame)->sample_rate    = output_codec_context->sample_rate;
-
-    /**
-     * Allocate the samples of the created frame. This call will make
-     * sure that the audio frame can hold as many samples as specified.
-     */
-    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
-        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
-                get_error_text(error));
-        av_frame_free(frame);
-        return error;
-    }
-
-    return 0;
-}
-
-/** Encode one frame worth of audio to the output file. */
-static int encode_audio_frame(AVFrame *frame,
-                              AVFormatContext *output_format_context,
-                              AVCodecContext *output_codec_context,
-                              int *data_present)
-{
-    /** Packet used for temporary storage. */
-    AVPacket output_packet;
-    int error;
-    init_packet(&output_packet);
-
-    /**
-     * Encode the audio frame and store it in the temporary packet.
-     * The output audio stream encoder is used to do this.
-     */
-    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
-                                       frame, data_present)) < 0) {
-        fprintf(stderr, "Could not encode frame (error '%s')\n",
-                get_error_text(error));
-        av_free_packet(&output_packet);
-        return error;
-    }
-
-    /** Write one audio frame from the temporary packet to the output file. */
-    if (*data_present) {
-        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
-            fprintf(stderr, "Could not write frame (error '%s')\n",
-                    get_error_text(error));
-            av_free_packet(&output_packet);
-            return error;
-        }
-
-        av_free_packet(&output_packet);
-    }
-
-    return 0;
-}
-
-/**
- * Load one audio frame from the FIFO buffer, encode and write it to the
- * output file.
- */
-static int load_encode_and_write(AVAudioFifo *fifo,
-                                 AVFormatContext *output_format_context,
-                                 AVCodecContext *output_codec_context)
-{
-    /** Temporary storage of the output samples of the frame written to the file. */
-    AVFrame *output_frame;
-    /**
-     * Use the maximum number of possible samples per frame.
-     * If there is less than the maximum possible frame size in the FIFO
-     * buffer use this number. Otherwise, use the maximum possible frame size
-     */
-    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
-                                 output_codec_context->frame_size);
-    int data_written;
-
-    /** Initialize temporary storage for one output frame. */
-    if (init_output_frame(&output_frame, output_codec_context, frame_size))
-        return AVERROR_EXIT;
-
-    /**
-     * Read as many samples from the FIFO buffer as required to fill the frame.
-     * The samples are stored in the frame temporarily.
-     */
-    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
-        fprintf(stderr, "Could not read data from FIFO\n");
-        av_frame_free(&output_frame);
-        return AVERROR_EXIT;
-    }
-
-    /** Encode one frame worth of audio samples. */
-    if (encode_audio_frame(output_frame, output_format_context,
-                           output_codec_context, &data_written)) {
-        av_frame_free(&output_frame);
-        return AVERROR_EXIT;
-    }
-    av_frame_free(&output_frame);
-    return 0;
-}
-
-/** Write the trailer of the output file container. */
-static int write_output_file_trailer(AVFormatContext *output_format_context)
-{
-    int error;
-    if ((error = av_write_trailer(output_format_context)) < 0) {
-        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-    return 0;
-}
-
-/** Convert an audio file to an AAC file in an MP4 container. */
-int main(int argc, char **argv)
-{
-    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
-    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
-    SwrContext *resample_context = NULL;
-    AVAudioFifo *fifo = NULL;
-    int ret = AVERROR_EXIT;
-
-    if (argc < 3) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-        exit(1);
-    }
-
-    /** Register all codecs and formats so that they can be used. */
-    av_register_all();
-    /** Open the input file for reading. */
-    if (open_input_file(argv[1], &input_format_context,
-                        &input_codec_context))
-        goto cleanup;
-    /** Open the output file for writing. */
-    if (open_output_file(argv[2], input_codec_context,
-                         &output_format_context, &output_codec_context))
-        goto cleanup;
-    /** Initialize the resampler to be able to convert audio sample formats. */
-    if (init_resampler(input_codec_context, output_codec_context,
-                       &resample_context))
-        goto cleanup;
-    /** Initialize the FIFO buffer to store audio samples to be encoded. */
-    if (init_fifo(&fifo))
-        goto cleanup;
-    /** Write the header of the output file container. */
-    if (write_output_file_header(output_format_context))
-        goto cleanup;
-
-    /**
-     * Loop as long as we have input samples to read or output samples
-     * to write; abort as soon as we have neither.
-     */
-    while (1) {
-        /** Use the encoder's desired frame size for processing. */
-        const int output_frame_size = output_codec_context->frame_size;
-        int finished                = 0;
-
-        /**
-         * Make sure that there is one frame worth of samples in the FIFO
-         * buffer so that the encoder can do its work.
-         * Since the decoder's and the encoder's frame size may differ, we
-         * need to FIFO buffer to store as many frames worth of input samples
-         * that they make up at least one frame worth of output samples.
-         */
-        while (av_audio_fifo_size(fifo) < output_frame_size) {
-            /**
-             * Decode one frame worth of audio samples, convert it to the
-             * output sample format and put it into the FIFO buffer.
-             */
-            if (read_decode_convert_and_store(fifo, input_format_context,
-                                              input_codec_context,
-                                              output_codec_context,
-                                              resample_context, &finished))
-                goto cleanup;
-
-            /**
-             * If we are at the end of the input file, we continue
-             * encoding the remaining audio samples to the output file.
-             */
-            if (finished)
-                break;
-        }
-
-        /**
-         * If we have enough samples for the encoder, we encode them.
-         * At the end of the file, we pass the remaining samples to
-         * the encoder.
-         */
-        while (av_audio_fifo_size(fifo) >= output_frame_size ||
-               (finished && av_audio_fifo_size(fifo) > 0))
-            /**
-             * Take one frame worth of audio samples from the FIFO buffer,
-             * encode it and write it to the output file.
-             */
-            if (load_encode_and_write(fifo, output_format_context,
-                                      output_codec_context))
-                goto cleanup;
-
-        /**
-         * If we are at the end of the input file and have encoded
-         * all remaining samples, we can exit this loop and finish.
-         */
-        if (finished) {
-            int data_written;
-            /** Flush the encoder as it may have delayed frames. */
-            do {
-                if (encode_audio_frame(NULL, output_format_context,
-                                       output_codec_context, &data_written))
-                    goto cleanup;
-            } while (data_written);
-            break;
-        }
-    }
-
-    /** Write the trailer of the output file container. */
-    if (write_output_file_trailer(output_format_context))
-        goto cleanup;
-    ret = 0;
-
-cleanup:
-    if (fifo)
-        av_audio_fifo_free(fifo);
-    swr_free(&resample_context);
-    if (output_codec_context)
-        avcodec_close(output_codec_context);
-    if (output_format_context) {
-        avio_close(output_format_context->pb);
-        avformat_free_context(output_format_context);
-    }
-    if (input_codec_context)
-        avcodec_close(input_codec_context);
-    if (input_format_context)
-        avformat_close_input(&input_format_context);
-
-    return ret;
-}

doc/examples/transcoding.c

@@ -1,583 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2014 Andrey Utkin
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * API example for demuxing, decoding, filtering, encoding and muxing
- * @example transcoding.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
-#include <libavutil/pixdesc.h>
-
-static AVFormatContext *ifmt_ctx;
-static AVFormatContext *ofmt_ctx;
-typedef struct FilteringContext {
-    AVFilterContext *buffersink_ctx;
-    AVFilterContext *buffersrc_ctx;
-    AVFilterGraph *filter_graph;
-} FilteringContext;
-static FilteringContext *filter_ctx;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    unsigned int i;
-
-    ifmt_ctx = NULL;
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *stream;
-        AVCodecContext *codec_ctx;
-        stream = ifmt_ctx->streams[i];
-        codec_ctx = stream->codec;
-        /* Reencode video & audio and remux subtitles etc. */
-        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* Open decoder */
-            ret = avcodec_open2(codec_ctx,
-                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
-                return ret;
-            }
-        }
-    }
-
-    av_dump_format(ifmt_ctx, 0, filename, 0);
-    return 0;
-}
-
-static int open_output_file(const char *filename)
-{
-    AVStream *out_stream;
-    AVStream *in_stream;
-    AVCodecContext *dec_ctx, *enc_ctx;
-    AVCodec *encoder;
-    int ret;
-    unsigned int i;
-
-    ofmt_ctx = NULL;
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
-    if (!ofmt_ctx) {
-        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
-        return AVERROR_UNKNOWN;
-    }
-
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        out_stream = avformat_new_stream(ofmt_ctx, NULL);
-        if (!out_stream) {
-            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
-            return AVERROR_UNKNOWN;
-        }
-
-        in_stream = ifmt_ctx->streams[i];
-        dec_ctx = in_stream->codec;
-        enc_ctx = out_stream->codec;
-
-        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* in this example, we choose transcoding to same codec */
-            encoder = avcodec_find_encoder(dec_ctx->codec_id);
-            if (!encoder) {
-                av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
-                return AVERROR_INVALIDDATA;
-            }
-
-            /* In this example, we transcode to same properties (picture size,
-             * sample rate etc.). These properties can be changed for output
-             * streams easily using filters */
-            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-                enc_ctx->height = dec_ctx->height;
-                enc_ctx->width = dec_ctx->width;
-                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
-                /* take first format from list of supported formats */
-                enc_ctx->pix_fmt = encoder->pix_fmts[0];
-                /* video time_base can be set to whatever is handy and supported by encoder */
-                enc_ctx->time_base = dec_ctx->time_base;
-            } else {
-                enc_ctx->sample_rate = dec_ctx->sample_rate;
-                enc_ctx->channel_layout = dec_ctx->channel_layout;
-                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
-                /* take first format from list of supported formats */
-                enc_ctx->sample_fmt = encoder->sample_fmts[0];
-                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
-            }
-
-            /* Third parameter can be used to pass settings to encoder */
-            ret = avcodec_open2(enc_ctx, encoder, NULL);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
-                return ret;
-            }
-        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
-            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
-            return AVERROR_INVALIDDATA;
-        } else {
-            /* if this stream must be remuxed */
-            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
-                    ifmt_ctx->streams[i]->codec);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
-                return ret;
-            }
-        }
-
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
-    }
-    av_dump_format(ofmt_ctx, 0, filename, 1);
-
-    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
-            return ret;
-        }
-    }
-
-    /* init muxer, write output file header */
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
-        AVCodecContext *enc_ctx, const char *filter_spec)
-{
-    char args[512];
-    int ret = 0;
-    AVFilter *buffersrc = NULL;
-    AVFilter *buffersink = NULL;
-    AVFilterContext *buffersrc_ctx = NULL;
-    AVFilterContext *buffersink_ctx = NULL;
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVFilterGraph *filter_graph = avfilter_graph_alloc();
-
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-        buffersrc = avfilter_get_by_name("buffer");
-        buffersink = avfilter_get_by_name("buffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        snprintf(args, sizeof(args),
-                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
-                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-                dec_ctx->time_base.num, dec_ctx->time_base.den,
-                dec_ctx->sample_aspect_ratio.num,
-                dec_ctx->sample_aspect_ratio.den);
-
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
-                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-            goto end;
-        }
-    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        buffersrc = avfilter_get_by_name("abuffer");
-        buffersink = avfilter_get_by_name("abuffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        if (!dec_ctx->channel_layout)
-            dec_ctx->channel_layout =
-                av_get_default_channel_layout(dec_ctx->channels);
-        snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
-                av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                dec_ctx->channel_layout);
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
-                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
-                (uint8_t*)&enc_ctx->channel_layout,
-                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
-                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-            goto end;
-        }
-    } else {
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if (!outputs->name || !inputs->name) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
-                    &inputs, &outputs, NULL)) < 0)
-        goto end;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-    /* Fill FilteringContext */
-    fctx->buffersrc_ctx = buffersrc_ctx;
-    fctx->buffersink_ctx = buffersink_ctx;
-    fctx->filter_graph = filter_graph;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
-}
-
-static int init_filters(void)
-{
-    const char *filter_spec;
-    unsigned int i;
-    int ret;
-    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
-    if (!filter_ctx)
-        return AVERROR(ENOMEM);
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        filter_ctx[i].buffersrc_ctx  = NULL;
-        filter_ctx[i].buffersink_ctx = NULL;
-        filter_ctx[i].filter_graph   = NULL;
-        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
-                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
-            continue;
-
-
-        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
-            filter_spec = "null"; /* passthrough (dummy) filter for video */
-        else
-            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
-        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
-                ofmt_ctx->streams[i]->codec, filter_spec);
-        if (ret)
-            return ret;
-    }
-    return 0;
-}
-
-static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
-    int ret;
-    int got_frame_local;
-    AVPacket enc_pkt;
-    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
-        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
-         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
-
-    if (!got_frame)
-        got_frame = &got_frame_local;
-
-    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
-    /* encode filtered frame */
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-    av_init_packet(&enc_pkt);
-    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
-            filt_frame, got_frame);
-    av_frame_free(&filt_frame);
-    if (ret < 0)
-        return ret;
-    if (!(*got_frame))
-        return 0;
-
-    /* prepare packet for muxing */
-    enc_pkt.stream_index = stream_index;
-    av_packet_rescale_ts(&enc_pkt,
-                         ofmt_ctx->streams[stream_index]->codec->time_base,
-                         ofmt_ctx->streams[stream_index]->time_base);
-
-    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
-    /* mux encoded frame */
-    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
-    return ret;
-}
-
-static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
-{
-    int ret;
-    AVFrame *filt_frame;
-
-    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
-    /* push the decoded frame into the filtergraph */
-    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
-            frame, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-        return ret;
-    }
-
-    /* pull filtered frames from the filtergraph */
-    while (1) {
-        filt_frame = av_frame_alloc();
-        if (!filt_frame) {
-            ret = AVERROR(ENOMEM);
-            break;
-        }
-        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
-        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
-                filt_frame);
-        if (ret < 0) {
-            /* if no more frames for output - returns AVERROR(EAGAIN)
-             * if flushed and no more frames for output - returns AVERROR_EOF
-             * rewrite retcode to 0 to show it as normal procedure completion
-             */
-            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                ret = 0;
-            av_frame_free(&filt_frame);
-            break;
-        }
-
-        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(filt_frame, stream_index, NULL);
-        if (ret < 0)
-            break;
-    }
-
-    return ret;
-}
-
-static int flush_encoder(unsigned int stream_index)
-{
-    int ret;
-    int got_frame;
-
-    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                CODEC_CAP_DELAY))
-        return 0;
-
-    while (1) {
-        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
-        ret = encode_write_frame(NULL, stream_index, &got_frame);
-        if (ret < 0)
-            break;
-        if (!got_frame)
-            return 0;
-    }
-    return ret;
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet = { .data = NULL, .size = 0 };
-    AVFrame *frame = NULL;
-    enum AVMediaType type;
-    unsigned int stream_index;
-    unsigned int i;
-    int got_frame;
-    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
-
-    if (argc != 3) {
-        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
-        return 1;
-    }
-
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = open_output_file(argv[2])) < 0)
-        goto end;
-    if ((ret = init_filters()) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
-            break;
-        stream_index = packet.stream_index;
-        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
-        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
-                stream_index);
-
-        if (filter_ctx[stream_index].filter_graph) {
-            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-            frame = av_frame_alloc();
-            if (!frame) {
-                ret = AVERROR(ENOMEM);
-                break;
-            }
-            av_packet_rescale_ts(&packet,
-                                 ifmt_ctx->streams[stream_index]->time_base,
-                                 ifmt_ctx->streams[stream_index]->codec->time_base);
-            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
-                avcodec_decode_audio4;
-            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
-                    &got_frame, &packet);
-            if (ret < 0) {
-                av_frame_free(&frame);
-                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
-                break;
-            }
-
-            if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
-                ret = filter_encode_write_frame(frame, stream_index);
-                av_frame_free(&frame);
-                if (ret < 0)
-                    goto end;
-            } else {
-                av_frame_free(&frame);
-            }
-        } else {
-            /* remux this frame without reencoding */
-            av_packet_rescale_ts(&packet,
-                                 ifmt_ctx->streams[stream_index]->time_base,
-                                 ofmt_ctx->streams[stream_index]->time_base);
-
-            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
-            if (ret < 0)
-                goto end;
-        }
-        av_free_packet(&packet);
-    }
-
-    /* flush filters and encoders */
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        /* flush filter */
-        if (!filter_ctx[i].filter_graph)
-            continue;
-        ret = filter_encode_write_frame(NULL, i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
-            goto end;
-        }
-
-        /* flush encoder */
-        ret = flush_encoder(i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
-            goto end;
-        }
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-    av_free_packet(&packet);
-    av_frame_free(&frame);
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        avcodec_close(ifmt_ctx->streams[i]->codec);
-        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
-            avcodec_close(ofmt_ctx->streams[i]->codec);
-        if (filter_ctx && filter_ctx[i].filter_graph)
-            avfilter_graph_free(&filter_ctx[i].filter_graph);
-    }
-    av_free(filter_ctx);
-    avformat_close_input(&ifmt_ctx);
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_close(ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
-
-    return ret ? 1 : 0;
-}

doc/faq.texi

@@ -79,17 +79,6 @@ not a bug they should fix:
 Then again, some of them do not know the difference between an undecidable
 problem and an NP-hard problem...
 
-@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
-
-Distributions usually split libraries in several packages. The main package
-contains the files necessary to run programs using the library. The
-development package contains the files necessary to build programs using the
-library. Sometimes, docs and/or data are in a separate package too.
-
-To build FFmpeg, you need to install the development package. It is usually
-called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
-build is finished, but be sure to keep the main package.
-
 @chapter Usage
 
 @section ffmpeg does not work; what is wrong?
@@ -105,21 +94,12 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
 Then you may run:
 
 @example
-ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
 @end example
 
 Notice that @samp{%d} is replaced by the image number.
 
-@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
-
-Use the @option{-start_number} option to declare a starting number for
-the sequence. This is useful if your sequence does not start with
-@file{img001.jpg} but is still in a numerical order. The following
-example will start with @file{img100.jpg}:
-
-@example
-ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
-@end example
+@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
 
 If you have large number of pictures to rename, you can use the
 following command to ease the burden. The command, using the bourne
@@ -128,7 +108,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
 @file{img001.jpg}, @file{img002.jpg} and so on.
 
 @example
-x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
 @end example
 
 If you want to sequence them by oldest modified first, substitute
@@ -137,23 +117,17 @@ If you want to sequence them by oldest modified first, substitute
 Then run:
 
 @example
-ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
 @end example
 
 The same logic is used for any image format that ffmpeg reads.
 
-You can also use @command{cat} to pipe images to ffmpeg:
-
-@example
-cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
-@end example
-
 @section How do I encode movie to single pictures?
 
 Use:
 
 @example
-ffmpeg -i movie.mpg movie%d.jpg
+  ffmpeg -i movie.mpg movie%d.jpg
 @end example
 
 The @file{movie.mpg} used as input will be converted to
@@ -169,7 +143,7 @@ to force the encoding.
 
 Applying that to the previous example:
 @example
-ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+  ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
 @end example
 
 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,15 +201,15 @@ then you may use any file that DirectShow can read as input.
 
 Just create an "input.avs" text file with this single line ...
 @example
-DirectShowSource("C:\path to your file\yourfile.asf")
+  DirectShowSource("C:\path to your file\yourfile.asf")
 @end example
 ... and then feed that text file to ffmpeg:
 @example
-ffmpeg -i input.avs
+  ffmpeg -i input.avs
 @end example
 
-For ANY other help on AviSynth, please visit the
-@uref{http://www.avisynth.org/, AviSynth homepage}.
+For ANY other help on Avisynth, please visit the
+@uref{http://www.avisynth.org/, Avisynth homepage}.
 
 @section How can I join video files?
 
@@ -260,18 +234,18 @@ invoking ffmpeg with several @option{-i} options.
 For audio, to put all channels together in a single stream (example: two
 mono streams into one stereo stream): this is sometimes called to
 @emph{merge} them, and can be done using the
-@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
+@url{http://ffmpeg.org/ffmpeg.html#amerge, @code{amerge}} filter.
 
 @item
 For audio, to play one on top of the other: this is called to @emph{mix}
 them, and can be done by first merging them into a single stream and then
-using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
+using the @url{http://ffmpeg.org/ffmpeg.html#pan, @code{pan}} filter to mix
 the channels at will.
 
 @item
 For video, to display both together, side by side or one on top of a part of
 the other; it can be done using the
-@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
+@url{http://ffmpeg.org/ffmpeg.html#overlay, @code{overlay}} video filter.
 
 @end itemize
 
@@ -280,26 +254,15 @@ the other; it can be done using the
 
 There are several solutions, depending on the exact circumstances.
 
-@subsection Concatenating using the concat @emph{filter}
+@subsection Concatenating using filters
 
-FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
-@code{concat}} filter designed specifically for that, with examples in the
-documentation. This operation is recommended if you need to re-encode.
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg.html#concat-1, @code{concat}}
+filter designed specifically for that, with examples in the documentation.
 
-@subsection Concatenating using the concat @emph{demuxer}
+@subsection Concatenating at the file level
 
-FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
-@code{concat}} demuxer which you can use when you want to avoid a re-encode and
-your format doesn't support file level concatenation.
-
-@subsection Concatenating using the concat @emph{protocol} (file level)
-
-FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
-@code{concat}} protocol designed specifically for that, with examples in the
-documentation.
-
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
-video by merely concatenating the files containing them.
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
+video by merely concatenating the files them.
 
 Hence you may concatenate your multimedia files by first transcoding them to
 these privileged formats, then using the humble @code{cat} command (or the
@@ -334,7 +297,7 @@ mkfifo intermediate2.mpg
 ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
 ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
+ffmpeg -f mpeg -i - -qscale:v 2 -c:v mpeg4 -acodec libmp3lame -q:a 4 output.avi
 @end example
 
 @subsection Concatenating using raw audio and video
@@ -364,26 +327,46 @@ cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
 ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
        -f yuv4mpegpipe -i all.v \
-       -y output.flv
+       -qscale:v 2 -y output.flv
 rm temp[12].[av] all.[av]
 @end example
 
+@section -profile option fails when encoding H.264 video with AAC audio
+
+@command{ffmpeg} prints an error like
+
+@example
+Undefined constant or missing '(' in 'baseline'
+Unable to parse option value "baseline"
+Error setting option profile to value baseline.
+@end example
+
+Short answer: write @option{-profile:v} instead of @option{-profile}.
+
+Long answer: this happens because the @option{-profile} option can apply to both
+video and audio.  Specifically the AAC encoder also defines some profiles, none
+of which are named @var{baseline}.
+
+The solution is to apply the @option{-profile} option to the video stream only
+by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
+Appending @code{:v} to it will do exactly that.
+
 @section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
 
 Use @option{-dumpgraph -} to find out exactly where the channel layout is
 lost.
 
-Most likely, it is through @code{auto-inserted aresample}. Try to understand
+Most likely, it is through @code{auto-inserted aconvert}. Try to understand
 why the converting filter was needed at that place.
 
 Just before the output is a likely place, as @option{-f lavfi} currently
 only support packed S16.
 
-Then insert the correct @code{aformat} explicitly in the filtergraph,
+Then insert the correct @code{aconvert} explicitly in the filter graph,
 specifying the exact format.
 
 @example
-aformat=sample_fmts=s16:channel_layouts=stereo
+aconvert=s16:stereo:packed
 @end example
 
 @section Why does FFmpeg not see the subtitles in my VOB file?
@@ -392,7 +375,7 @@ VOB and a few other formats do not have a global header that describes
 everything present in the file. Instead, applications are supposed to scan
 the file to see what it contains. Since VOB files are frequently large, only
 the beginning is scanned. If the subtitles happen only later in the file,
-they will not be initially detected.
+they will not be initally detected.
 
 Some applications, including the @code{ffmpeg} command-line tool, can only
 work with streams that were detected during the initial scan; streams that
@@ -402,20 +385,6 @@ The size of the initial scan is controlled by two options: @code{probesize}
 (default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
 the subtitle stream to be detected, both values must be large enough.
 
-@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
-
-The @option{-sameq} option meant "same quantizer", and made sense only in a
-very limited set of cases. Unfortunately, a lot of people mistook it for
-"same quality" and used it in places where it did not make sense: it had
-roughly the expected visible effect, but achieved it in a very inefficient
-way.
-
-Each encoder has its own set of options to set the quality-vs-size balance,
-use the options for the encoder you are using to set the quality level to a
-point acceptable for your tastes. The most common options to do that are
-@option{-qscale} and @option{-qmax}, but you should peruse the documentation
-of the encoder you chose.
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -439,8 +408,31 @@ with @code{#ifdef}s related to the compiler.
 
 @section Is Microsoft Visual C++ supported?
 
-Yes. Please see the @uref{platform.html, Microsoft Visual C++}
-section in the FFmpeg documentation.
+No. Microsoft Visual C++ is not compliant to the C99 standard and does
+not - among other things - support the inline assembly used in FFmpeg.
+If you wish to use MSVC++ for your
+project then you can link the MSVC++ code with libav* as long as
+you compile the latter with a working C compiler. For more information, see
+the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
+documentation.
+
+There have been efforts to make FFmpeg compatible with MSVC++ in the
+past. However, they have all been rejected as too intrusive, especially
+since MinGW does the job adequately. None of the core developers
+work with MSVC++ and thus this item is low priority. Should you find
+the silver bullet that solves this problem, feel free to shoot it at us.
+
+We strongly recommend you to move over from MSVC++ to MinGW tools.
+
+@section Can I use FFmpeg or libavcodec under Windows?
+
+Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
+Read the @emph{Windows} section in the FFmpeg documentation to find more
+information.
+
+To get help and instructions for building FFmpeg under Windows, check out
+the FFmpeg Windows Help Forum at
+@url{http://ffmpeg.arrozcru.org/}.
 
 @section Can you add automake, libtool or autoconf support?
 
@@ -455,10 +447,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
 
 @section Why are the ffmpeg programs devoid of debugging symbols?
 
-The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
-contain full debug information. Those binaries are stripped to create
-@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
-the *_g versions.
+The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
+information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
+you need the debug information, use the *_g versions.
 
 @section I do not like the LGPL, can I contribute code under the GPL instead?
 
@@ -478,7 +469,7 @@ An easy way to get the full list of required libraries in dependency order
 is to use @code{pkg-config}.
 
 @example
-c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
 @end example
 
 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@@ -499,8 +490,12 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 
 @section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
 
-You have to create a custom AVIOContext using @code{avio_alloc_context},
-see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
+You have to implement a URLProtocol, see @file{libavformat/file.c} in
+FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
+
+@section Where can I find libav* headers for Pascal/Delphi?
+
+see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
 
 @section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
 
@@ -514,12 +509,11 @@ In this specific case please look at RFC 4629 to see how it should be done.
 
 @section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
 
-@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average frame rate, it is the smallest frame rate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
-will be 150 (it is the least common multiple).
-If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
+For example, if you have mixed 25 and 30 fps content, then r_frame_rate
+will be 150.
 
 @section Why is @code{make fate} not running all tests?
 

doc/fate.texi

@@ -27,7 +27,7 @@ by visiting this website:
 
   This is especially recommended for all people contributing source
 code to FFmpeg, as it can be seen if some test on some platform broke
-with their recent contribution. This usually happens on the platforms
+with there recent contribution. This usually happens on the platforms
 the developers could not test on.
 
   The second part of this document describes how you can run FATE to
@@ -93,11 +93,11 @@ tests/fate.sh /path/to/fate_config
 @end example
 
   A configuration file template with comments describing the individual
-configuration variables can be found at @file{doc/fate_config.sh.template}.
+configuration variables can be found at @file{tests/fate_config.sh.template}.
 
 @ifhtml
   The mentioned configuration template is also available here:
-@verbatiminclude fate_config.sh.template
+@verbatiminclude ../tests/fate_config.sh.template
 @end ifhtml
 
   Create a configuration that suits your needs, based on the configuration
@@ -131,12 +131,7 @@ of the server and to accept its host key. This can usually be achieved by
 running your SSH client manually and killing it after you accepted the key.
 The FATE server's fingerprint is:
 
-@table @option
-@item RSA
-   d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
-@item ECDSA
-   76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
-@end table
+  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
 
   If you have problems connecting to the FATE server, it may help to try out
 the @command{ssh} command with one or more @option{-v} options. You should
@@ -153,20 +148,20 @@ the synchronisation of the samples directory.
 
 @table @option
 @item fate-rsync
-Download/synchronize sample files to the configured samples directory.
+    Download/synchronize sample files to the configured samples directory.
 
 @item fate-list
-Will list all fate/regression test targets.
+    Will list all fate/regression test targets.
 
 @item fate
-Run the FATE test suite (requires the fate-suite dataset).
+    Run the FATE test suite (requires the fate-suite dataset).
 @end table
 
 @section Makefile variables
 
 @table @option
 @item V
-Verbosity level, can be set to 0, 1 or 2.
+    Verbosity level, can be set to 0, 1 or 2.
     @itemize
         @item 0: show just the test arguments
         @item 1: show just the command used in the test
@@ -174,28 +169,22 @@ Verbosity level, can be set to 0, 1 or 2.
     @end itemize
 
 @item SAMPLES
-Specify or override the path to the FATE samples at make time, it has a
-meaning only while running the regression tests.
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.
 
 @item THREADS
-Specify how many threads to use while running regression tests, it is
-quite useful to detect thread-related regressions.
-
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
 @item THREAD_TYPE
-Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @var{slice+frame}
-
+    Specify which threading strategy test, either @var{slice} or @var{frame},
+    by default @var{slice+frame}
 @item CPUFLAGS
-Specify CPU flags.
-
+    Specify CPU flags.
 @item TARGET_EXEC
-Specify or override the wrapper used to run the tests.
-The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-through @command{ssh}.
-
-@item GEN
-Set to @var{1} to generate the missing or mismatched references.
+    Specify or override the wrapper used to run the tests.
+    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
+    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
+    through @command{ssh}.
 @end table
 
 @section Examples

doc/ffmpeg-bitstream-filters.texi

@@ -1,45 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Bitstream Filters Documentation
-@titlepage
-@center @titlefont{FFmpeg Bitstream Filters Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the bitstream filters provided by the
-libavcodec library.
-
-A bitstream filter operates on the encoded stream data, and performs
-bitstream level modifications without performing decoding.
-
-@c man end DESCRIPTION
-
-@include bitstream_filters.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-bitstream-filters
-@settitle FFmpeg bitstream filters
-
-@end ignore
-
-@bye

doc/ffmpeg-codecs.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Codecs Documentation
-@titlepage
-@center @titlefont{FFmpeg Codecs Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the codecs (decoders and encoders) provided by
-the libavcodec library.
-
-@c man end DESCRIPTION
-
-@include codecs.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-codecs
-@settitle FFmpeg codecs
-
-@end ignore
-
-@bye

doc/ffmpeg-devices.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Devices Documentation
-@titlepage
-@center @titlefont{FFmpeg Devices Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the input and output devices provided by the
-libavdevice library.
-
-@c man end DESCRIPTION
-
-@include devices.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavdevice.html,libavdevice}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-devices
-@settitle FFmpeg devices
-
-@end ignore
-
-@bye

doc/ffmpeg-filters.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Filters Documentation
-@titlepage
-@center @titlefont{FFmpeg Filters Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes filters, sources, and sinks provided by the
-libavfilter library.
-
-@c man end DESCRIPTION
-
-@include filters.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavfilter.html,libavfilter}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-filters
-@settitle FFmpeg filters
-
-@end ignore
-
-@bye

doc/ffmpeg-formats.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Formats Documentation
-@titlepage
-@center @titlefont{FFmpeg Formats Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the supported formats (muxers and demuxers)
-provided by the libavformat library.
-
-@c man end DESCRIPTION
-
-@include formats.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-formats
-@settitle FFmpeg formats
-
-@end ignore
-
-@bye

doc/ffmpeg-protocols.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Protocols Documentation
-@titlepage
-@center @titlefont{FFmpeg Protocols Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the input and output protocols provided by the
-libavformat library.
-
-@c man end DESCRIPTION
-
-@include protocols.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-protocols
-@settitle FFmpeg protocols
-
-@end ignore
-
-@bye

doc/ffmpeg-resampler.texi

@@ -1,44 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Resampler Documentation
-@titlepage
-@center @titlefont{FFmpeg Resampler Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The FFmpeg resampler provides a high-level interface to the
-libswresample library audio resampling utilities. In particular it
-allows one to perform audio resampling, audio channel layout rematrixing,
-and convert audio format and packing layout.
-
-@c man end DESCRIPTION
-
-@include resampler.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libswresample.html,libswresample}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-resampler
-@settitle FFmpeg Resampler
-
-@end ignore
-
-@bye

doc/ffmpeg-scaler.texi

@@ -1,43 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Scaler Documentation
-@titlepage
-@center @titlefont{FFmpeg Scaler Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The FFmpeg rescaler provides a high-level interface to the libswscale
-library image conversion utilities. In particular it allows one to perform
-image rescaling and pixel format conversion.
-
-@c man end DESCRIPTION
-
-@include scaler.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libswscale.html,libswscale}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-scaler
-@settitle FFmpeg video scaling and pixel format converter
-
-@end ignore
-
-@bye

doc/ffmpeg-utils.texi

@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Utilities Documentation
-@titlepage
-@center @titlefont{FFmpeg Utilities Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes some generic features and utilities provided
-by the libavutil library.
-
-@c man end DESCRIPTION
-
-@include utils.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-utils
-@settitle FFmpeg utilities
-
-@end ignore
-
-@bye

doc/ffmpeg.texi

@@ -11,31 +11,37 @@
 
 @chapter Synopsis
 
-ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffmpeg [global options] [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
+@c man end
+@end example
 
 @chapter Description
 @c man begin DESCRIPTION
 
-@command{ffmpeg} is a very fast video and audio converter that can also grab from
+ffmpeg is a very fast video and audio converter that can also grab from
 a live audio/video source. It can also convert between arbitrary sample
 rates and resize video on the fly with a high quality polyphase filter.
 
-@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
+ffmpeg reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
 @code{-i} option, and writes to an arbitrary number of output "files", which are
 specified by a plain output filename. Anything found on the command line which
 cannot be interpreted as an option is considered to be an output filename.
 
-Each input or output file can, in principle, contain any number of streams of
-different types (video/audio/subtitle/attachment/data). The allowed number and/or
-types of streams may be limited by the container format. Selecting which
-streams from which inputs will go into which output is either done automatically
-or with the @code{-map} option (see the Stream selection chapter).
+Each input or output file can in principle contain any number of streams of
+different types (video/audio/subtitle/attachment/data). Allowed number and/or
+types of streams can be limited by the container format. Selecting, which
+streams from which inputs go into output, is done either automatically or with
+the @code{-map} option (see the Stream selection chapter).
 
 To refer to input files in options, you must use their indices (0-based). E.g.
-the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams
+the first input file is @code{0}, the second is @code{1} etc. Similarly, streams
 within a file are referred to by their indices. E.g. @code{2:3} refers to the
-fourth stream in the third input file. Also see the Stream specifiers chapter.
+fourth stream in the third input file. See also the Stream specifiers chapter.
 
 As a general rule, options are applied to the next specified
 file. Therefore, order is important, and you can have the same
@@ -50,9 +56,9 @@ options apply ONLY to the next input or output file and are reset between files.
 
 @itemize
 @item
-To set the video bitrate of the output file to 64 kbit/s:
+To set the video bitrate of the output file to 64kbit/s:
 @example
-ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
+ffmpeg -i input.avi -b:v 64k output.avi
 @end example
 
 @item
@@ -80,23 +86,11 @@ The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:
 
 @example
- _______              ______________
-|       |            |              |
-| input |  demuxer   | encoded data |   decoder
-| file  | ---------> | packets      | -----+
-|_______|            |______________|      |
-                                           v
-                                       _________
-                                      |         |
-                                      | decoded |
-                                      | frames  |
-                                      |_________|
- ________             ______________       |
-|        |           |              |      |
-| output | <-------- | encoded data | <----+
-| file   |   muxer   | packets      |   encoder
-|________|           |______________|
-
+ _______              ______________               _________              ______________            ________
+|       |            |              |             |         |            |              |          |        |
+| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
+| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
+|_______|            |______________|             |_________|            |______________|          |________|
 
 @end example
 
@@ -108,14 +102,14 @@ tracking lowest timestamp on any active input stream.
 Encoded packets are then passed to the decoder (unless streamcopy is selected
 for the stream, see further for a description). The decoder produces
 uncompressed frames (raw video/PCM audio/...) which can be processed further by
-filtering (see next section). After filtering, the frames are passed to the
-encoder, which encodes them and outputs encoded packets. Finally those are
+filtering (see next section). After filtering the frames are passed to the
+encoder, which encodes them and outputs encoded packets again. Finally those are
 passed to the muxer, which writes the encoded packets to the output file.
 
 @section Filtering
 Before encoding, @command{ffmpeg} can process raw audio and video frames using
 filters from the libavfilter library. Several chained filters form a filter
-graph. @command{ffmpeg} distinguishes between two types of filtergraphs:
+graph.  @command{ffmpeg} distinguishes between two types of filtergraphs -
 simple and complex.
 
 @subsection Simple filtergraphs
@@ -124,16 +118,11 @@ the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:
 
 @example
- _________                        ______________
-|         |                      |              |
-| decoded |                      | encoded data |
-| frames  |\                   _ | packets      |
-|_________| \                  /||______________|
-             \   __________   /
-  simple     _\||          | /  encoder
-  filtergraph   | filtered |/
-                | frames   |
-                |__________|
+ _________                        __________              ______________
+|         |                      |          |            |              |
+| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
+| frames  | -------------------> | frames   | ---------> | packets      |
+|_________|                      |__________|            |______________|
 
 @end example
 
@@ -142,10 +131,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
 A simple filtergraph for video can look for example like this:
 
 @example
- _______        _____________        _______        ________
-|       |      |             |      |       |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | output |
-|_______|      |_____________|      |_______|      |________|
+ _______        _____________        _______        _____        ________
+|       |      |             |      |       |      |     |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
+|_______|      |_____________|      |_______|      |_____|      |________|
 
 @end example
 
@@ -156,7 +145,7 @@ only sets timestamps and otherwise passes the frames unchanged.
 
 @subsection Complex filtergraphs
 Complex filtergraphs are those which cannot be described as simply a linear
-processing chain applied to one stream. This is the case, for example, when the graph has
+processing chain applied to one stream. This is the case e.g. when the graph has
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:
 
@@ -181,11 +170,9 @@ input. They can be represented with the following diagram:
 @end example
 
 Complex filtergraphs are configured with the @option{-filter_complex} option.
-Note that this option is global, since a complex filtergraph, by its nature,
+Note that this option is global, since a complex filtergraph by its nature
 cannot be unambiguously associated with a single stream or file.
 
-The @option{-lavfi} option is equivalent to @option{-filter_complex}.
-
 A trivial example of a complex filtergraph is the @code{overlay} filter, which
 has two video inputs and one video output, containing one video overlaid on top
 of the other. Its audio counterpart is the @code{amix} filter.
@@ -195,7 +182,7 @@ Stream copy is a mode selected by supplying the @code{copy} parameter to the
 @option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding
 step for the specified stream, so it does only demuxing and muxing. It is useful
 for changing the container format or modifying container-level metadata. The
-diagram above will, in this case, simplify to this:
+diagram above will in this case simplify to this:
 
 @example
  _______              ______________            ________
@@ -207,7 +194,7 @@ diagram above will, in this case, simplify to this:
 @end example
 
 Since there is no decoding or encoding, it is very fast and there is no quality
-loss. However, it might not work in some cases because of many factors. Applying
+loss. However it might not work in some cases because of many factors. Applying
 filters is obviously also impossible, since filters work on uncompressed data.
 
 @c man end DETAILED DESCRIPTION
@@ -215,14 +202,14 @@ filters is obviously also impossible, since filters work on uncompressed data.
 @chapter Stream selection
 @c man begin STREAM SELECTION
 
-By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
+By default ffmpeg includes only one stream of each type (video, audio, subtitle)
 present in the input files and adds them to each output file.  It picks the
-"best" of each based upon the following criteria: for video, it is the stream
-with the highest resolution, for audio, it is the stream with the most channels, for
-subtitles, it is the first subtitle stream. In the case where several streams of
-the same type rate equally, the stream with the lowest index is chosen.
+"best" of each based upon the following criteria; for video it is the stream
+with the highest resolution, for audio the stream with the most channels, for
+subtitle it's the first subtitle stream. In the case where several streams of
+the same type rate equally, the lowest numbered stream is chosen.
 
-You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
+You can disable some of those defaults by using @code{-vn/-an/-sn} options. For
 full manual control, use the @code{-map} option, which disables the defaults just
 described.
 
@@ -231,7 +218,7 @@ described.
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
@@ -239,7 +226,7 @@ described.
 
 @item -f @var{fmt} (@emph{input/output})
 Force input or output file format. The format is normally auto detected for input
-files and guessed from the file extension for output files, so this option is not
+files and guessed from file extension for output files, so this option is not
 needed in most cases.
 
 @item -i @var{filename} (@emph{input})
@@ -249,8 +236,7 @@ input file name
 Overwrite output files without asking.
 
 @item -n (@emph{global})
-Do not overwrite output files, and exit immediately if a specified
-output file already exists.
+Do not overwrite output files but exit if file exists.
 
 @item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 @itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@@ -272,55 +258,39 @@ ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT
 will copy all the streams except the second video, which will be encoded with
 libx264, and the 138th audio, which will be encoded with libvorbis.
 
-@item -t @var{duration} (@emph{input/output})
-When used as an input option (before @code{-i}), limit the @var{duration} of
-data read from the input file.
-
-When used as an output option (before an output filename), stop writing the
-output after its duration reaches @var{duration}.
-
+@item -t @var{duration} (@emph{output})
+Stop writing the output after its duration reaches @var{duration}.
 @var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 
--to and -t are mutually exclusive and -t has priority.
-
-@item -to @var{position} (@emph{output})
-Stop writing the output at @var{position}.
-@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
-
--to and -t are mutually exclusive and -t has priority.
-
 @item -fs @var{limit_size} (@emph{output})
 Set the file size limit, expressed in bytes.
 
 @item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. Note the in most formats it is not possible to seek exactly, so
-@command{ffmpeg} will seek to the closest seek point before @var{position}.
-When transcoding and @option{-accurate_seek} is enabled (the default), this
-extra segment between the seek point and @var{position} will be decoded and
-discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
-will be preserved.
-
-When used as an output option (before an output filename), decodes but discards
-input until the timestamps reach @var{position}.
+@var{position}. When used as an output option (before an output filename),
+decodes but discards input until the timestamps reach @var{position}. This is
+slower, but more accurate.
 
 @var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
 
 @item -itsoffset @var{offset} (@emph{input})
-Set the input time offset.
+Set the input time offset in seconds.
+@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
+The offset is added to the timestamps of the input files.
+Specifying a positive offset means that the corresponding
+streams are delayed by @var{offset} seconds.
 
-@var{offset} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-The offset is added to the timestamps of the input files. Specifying
-a positive offset means that the corresponding streams are delayed by
-the time duration specified in @var{offset}.
-
-@item -timestamp @var{date} (@emph{output})
+@item -timestamp @var{time} (@emph{output})
 Set the recording timestamp in the container.
-
-@var{date} must be a time duration specification,
-see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+The syntax for @var{time} is:
+@example
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
+@end example
+If the value is "now" it takes the current time.
+Time is local time unless 'Z' or 'z' is appended, in which case it is
+interpreted as UTC.
+If the year-month-day part is not specified it takes the current
+year-month-day.
 
 @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.
@@ -339,7 +309,7 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
 
 To set the language of the first audio stream:
 @example
-ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
 @end example
 
 @item -target @var{type} (@emph{output})
@@ -360,47 +330,28 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
 @item -dframes @var{number} (@emph{output})
-Set the number of data frames to output. This is an alias for @code{-frames:d}.
+Set the number of data frames to record. This is an alias for @code{-frames:d}.
 
 @item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
 Stop writing to the stream after @var{framecount} frames.
 
 @item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
 @itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
+Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.
-If @var{qscale} is used without a @var{stream_specifier} then it applies only
-to the video stream, this is to maintain compatibility with previous behavior
-and as specifying the same codec specific value to 2 different codecs that is
-audio and video generally is not what is intended when no stream_specifier is
-used.
 
-@anchor{filter_option}
-@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the stream.
-
-@var{filtergraph} is a description of the filtergraph to apply to
-the stream, and must have a single input and a single output of the
-same type of the stream. In the filtergraph, the input is associated
-to the label @code{in}, and the output to the label @code{out}. See
-the ffmpeg-filters manual for more information about the filtergraph
-syntax.
-
-See the @ref{filter_complex_option,,-filter_complex option} if you
-want to create filtergraphs with multiple inputs and/or outputs.
-
-@item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream})
-This option is similar to @option{-filter}, the only difference is that its
-argument is the name of the file from which a filtergraph description is to be
-read.
+@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream})
+@var{filter_graph} is a description of the filter graph to apply to
+the stream. Use @code{-filters} to show all the available filters
+(including also sources and sinks).
 
+See also the @option{-filter_complex} option if you want to create filter graphs
+with multiple inputs and/or outputs.
 @item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).
 
 @item -stats (@emph{global})
-Print encoding progress/statistics. It is on by default, to explicitly
-disable it you need to specify @code{-nostats}.
+Print encoding progress/statistics. On by default.
 
 @item -progress @var{url} (@emph{global})
 Send program-friendly progress information to @var{url}.
@@ -450,11 +401,11 @@ will be used.
 
 E.g. to extract the first attachment to a file named 'out.ttf':
 @example
-ffmpeg -dump_attachment:t:0 out.ttf -i INPUT
+ffmpeg -dump_attachment:t:0 out.ttf INPUT
 @end example
 To extract all attachments to files determined by the @code{filename} tag:
 @example
-ffmpeg -dump_attachment:t "" -i INPUT
+ffmpeg -dump_attachment:t "" INPUT
 @end example
 
 Technical note -- attachments are implemented as codec extradata, so this
@@ -467,18 +418,16 @@ attachments.
 
 @table @option
 @item -vframes @var{number} (@emph{output})
-Set the number of video frames to output. This is an alias for @code{-frames:v}.
+Set the number of video frames to record. This is an alias for @code{-frames:v}.
 @item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
 Set frame rate (Hz value, fraction or abbreviation).
 
 As an input option, ignore any timestamps stored in the file and instead
 generate timestamps assuming constant frame rate @var{fps}.
-This is not the same as the @option{-framerate} option used for some input formats
-like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
-If in doubt use @option{-framerate} instead of the input option @option{-r}.
 
 As an output option, duplicate or drop input frames to achieve constant output
-frame rate @var{fps}.
+frame rate @var{fps} (note that this actually causes the @code{fps} filter to be
+inserted to the end of the corresponding filtergraph).
 
 @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
 Set frame size.
@@ -501,15 +450,31 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the
 numerator and denominator of the aspect ratio. For example "4:3",
 "16:9", "1.3333", and "1.7777" are valid argument values.
 
-If used together with @option{-vcodec copy}, it will affect the aspect ratio
-stored at container level, but not the aspect ratio stored in encoded
-frames, if it exists.
+@item -croptop @var{size}
+@item -cropbottom @var{size}
+@item -cropleft @var{size}
+@item -cropright @var{size}
+All the crop options have been removed. Use -vf
+crop=width:height:x:y instead.
+
+@item -padtop @var{size}
+@item -padbottom @var{size}
+@item -padleft @var{size}
+@item -padright @var{size}
+@item -padcolor @var{hex_color}
+All the pad options have been removed. Use -vf
+pad=width:height:x:y:color instead.
 
 @item -vn (@emph{output})
 Disable video recording.
 
 @item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
+@item -same_quant
+Use same quantizer as source (implies VBR).
+
+Note that this is NOT SAME QUALITY. Do not use this option unless you know you
+need it.
 
 @item -pass[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
 Select the pass number (1 or 2). It is used to do two-pass
@@ -530,14 +495,18 @@ prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
-@item -vf @var{filtergraph} (@emph{output})
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the stream.
+@item -vlang @var{code}
+Set the ISO 639 language code (3 letters) of the current video stream.
+
+@item -vf @var{filter_graph} (@emph{output})
+@var{filter_graph} is a description of the filter graph to apply to
+the input video.
+Use the option "-filters" to show all the available filters (including
+also sources and sinks).  This is an alias for @code{-filter:v}.
 
-This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
 @end table
 
-@section Advanced Video options
+@section Advanced Video Options
 
 @table @option
 @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
@@ -547,7 +516,7 @@ If the selected pixel format can not be selected, ffmpeg will print a
 warning and select the best pixel format supported by the encoder.
 If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error
 if the requested pixel format can not be selected, and automatic conversions
-inside filtergraphs are disabled.
+inside filter graphs are disabled.
 If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
 as the input (or graph output) and automatic conversions are disabled.
 
@@ -562,6 +531,10 @@ list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.
 
+@item -deinterlace
+Deinterlace pictures.
+This option is deprecated since the deinterlacing is very low quality.
+Use the yadif filter with @code{-filter:v yadif}.
 @item -ilme
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
@@ -584,115 +557,23 @@ Force video tag/fourcc. This is an alias for @code{-tag:v}.
 Show QP histogram
 @item -vbsf @var{bitstream_filter}
 Deprecated see -bsf
-
 @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
-@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
 Force key frames at the specified timestamps, more precisely at the first
 frames after each specified time.
-
-If the argument is prefixed with @code{expr:}, the string @var{expr}
-is interpreted like an expression and is evaluated for each frame. A
-key frame is forced in case the evaluation is non-zero.
-
-If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
-the time of the beginning of all chapters in the file, shifted by
-@var{delta}, expressed as a time in seconds.
 This option can be useful to ensure that a seek point is present at a
 chapter mark or any other designated place in the output file.
-
-For example, to insert a key frame at 5 minutes, plus key frames 0.1 second
-before the beginning of every chapter:
-@example
--force_key_frames 0:05:00,chapters-0.1
-@end example
-
-The expression in @var{expr} can contain the following constants:
-@table @option
-@item n
-the number of current processed frame, starting from 0
-@item n_forced
-the number of forced frames
-@item prev_forced_n
-the number of the previous forced frame, it is @code{NAN} when no
-keyframe was forced yet
-@item prev_forced_t
-the time of the previous forced frame, it is @code{NAN} when no
-keyframe was forced yet
-@item t
-the time of the current processed frame
-@end table
-
-For example to force a key frame every 5 seconds, you can specify:
-@example
--force_key_frames expr:gte(t,n_forced*5)
-@end example
-
-To force a key frame 5 seconds after the time of the last forced one,
-starting from second 13:
-@example
--force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
-@end example
-
-Note that forcing too many keyframes is very harmful for the lookahead
-algorithms of certain encoders: using fixed-GOP options or similar
-would be more efficient.
+The timestamps must be specified in ascending order.
 
 @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
 beginning.
-
-@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
-Use hardware acceleration to decode the matching stream(s). The allowed values
-of @var{hwaccel} are:
-@table @option
-@item none
-Do not use any hardware acceleration (the default).
-
-@item auto
-Automatically select the hardware acceleration method.
-
-@item vda
-Use Apple VDA hardware acceleration.
-
-@item vdpau
-Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
-
-@item dxva2
-Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
-@end table
-
-This option has no effect if the selected hwaccel is not available or not
-supported by the chosen decoder.
-
-Note that most acceleration methods are intended for playback and will not be
-faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
-will usually need to copy the decoded frames from the GPU memory into the system
-memory, resulting in further performance loss. This option is thus mainly
-useful for testing.
-
-@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
-Select a device to use for hardware acceleration.
-
-This option only makes sense when the @option{-hwaccel} option is also
-specified. Its exact meaning depends on the specific hardware acceleration
-method chosen.
-
-@table @option
-@item vdpau
-For VDPAU, this option specifies the X11 display/screen to use. If this option
-is not specified, the value of the @var{DISPLAY} environment variable is used
-
-@item dxva2
-For DXVA2, this option should contain the number of the display adapter to use.
-If this option is not specified, the default adapter is used.
-@end table
 @end table
 
 @section Audio Options
 
 @table @option
 @item -aframes @var{number} (@emph{output})
-Set the number of audio frames to output. This is an alias for @code{-frames:a}.
+Set the number of audio frames to record. This is an alias for @code{-frames:a}.
 @item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
 Set the audio sampling frequency. For output streams it is set by
 default to the frequency of the corresponding input stream. For input
@@ -712,32 +593,27 @@ Set the audio codec. This is an alias for @code{-codec:a}.
 @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.
-
-@item -af @var{filtergraph} (@emph{output})
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the stream.
-
-This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
+@item -af @var{filter_graph} (@emph{output})
+@var{filter_graph} is a description of the filter graph to apply to
+the input audio.
+Use the option "-filters" to show all the available filters (including
+also sources and sinks).  This is an alias for @code{-filter:a}.
 @end table
 
-@section Advanced Audio options
+@section Advanced Audio options:
 
 @table @option
 @item -atag @var{fourcc/tag} (@emph{output})
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
 @item -absf @var{bitstream_filter}
 Deprecated, see -bsf
-@item -guess_layout_max @var{channels} (@emph{input,per-stream})
-If some input channel layout is not known, try to guess only if it
-corresponds to at most the specified number of channels. For example, 2
-tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as
-stereo but not 6 channels as 5.1. The default is to always try to guess. Use
-0 to disable all guessing.
 @end table
 
-@section Subtitle options
+@section Subtitle options:
 
 @table @option
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
 @item -sn (@emph{output})
@@ -746,7 +622,7 @@ Disable subtitle recording.
 Deprecated, see -bsf
 @end table
 
-@section Advanced Subtitle options
+@section Advanced Subtitle options:
 
 @table @option
 
@@ -763,9 +639,6 @@ Note that this option will delay the output of all data until the next
 subtitle packet is decoded: it may increase memory consumption and latency a
 lot.
 
-@item -canvas_size @var{size}
-Set the size of the canvas used to render subtitles.
-
 @end table
 
 @section Advanced options
@@ -824,11 +697,6 @@ To map all the streams except the second audio, use negative mappings
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
 @end example
 
-To pick the English audio stream:
-@example
-ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
-@end example
-
 Note that using this option disables the default mappings for this output file.
 
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
@@ -933,7 +801,39 @@ Copy chapters from input file with index @var{input_file_index} to the next
 output file. If no chapter mapping is specified, then chapters are copied from
 the first input file with at least one chapter. Use a negative file index to
 disable any chapter copying.
-
+@item -debug @var{category}
+Print specific debug info.
+@var{category} is a number or a string containing one of the following values:
+@table @samp
+@item bitstream
+@item buffers
+picture buffer allocations
+@item bugs
+@item dct_coeff
+@item er
+error recognition
+@item mb_type
+macroblock (MB) type
+@item mmco
+memory management control operations (H.264)
+@item mv
+motion vector
+@item pict
+picture info
+@item pts
+@item qp
+per-block quantization parameter (QP)
+@item rc
+rate control
+@item skip
+@item startcode
+@item thread_ops
+threading operations
+@item vis_mb_type
+visualize block types
+@item vis_qp
+visualize quantization parameter (QP), lower QP are tinted greener
+@end table
 @item -benchmark (@emph{global})
 Show benchmarking information at the end of an encode.
 Shows CPU time used and maximum memory consumption.
@@ -950,12 +850,13 @@ Dump each input packet to stderr.
 When dumping packets, also dump the payload.
 @item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
-or live input stream (e.g. when reading from a file). Should not be used
-with actual grab devices or live input streams (where it can cause packet
-loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming).
+of the input(s). It is useful for real-time output (e.g. live streaming). If
+your input(s) is coming from some other live streaming source (through HTTP or
+UDP for example) the server might already be in real-time, thus the option will
+likely not be required. On the other hand, this is meaningful if your input(s)
+is a file you are trying to push in real-time.
 @item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -974,7 +875,7 @@ Newly added values will have to be specified as strings always.
 Each frame is passed with its timestamp from the demuxer to the muxer.
 @item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
-constant frame rate.
+constant framerate.
 @item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
@@ -986,10 +887,6 @@ Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
 @end table
 
-Note that the timestamps may be further modified by the muxer, after this.
-For example, in the case that the format option @option{avoid_negative_ts}
-is enabled.
-
 With -map you can select from which stream the timestamps should be
 taken. You can leave either video or audio unchanged and sync the
 remaining stream(s) to the unchanged one.
@@ -999,29 +896,9 @@ Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps
 the parameter is the maximum samples per second by which the audio is changed.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
-
-Note that the timestamps may be further modified by the muxer, after this.
-For example, in the case that the format option @option{avoid_negative_ts}
-is enabled.
-
-This option has been deprecated. Use the @code{aresample} audio filter instead.
-
+This option has been deprecated. Use the @code{asyncts} audio filter instead.
 @item -copyts
-Do not process input timestamps, but keep their values without trying
-to sanitize them. In particular, do not remove the initial start time
-offset value.
-
-Note that, depending on the @option{vsync} option or on specific muxer
-processing (e.g. in case the format option @option{avoid_negative_ts}
-is enabled) the output timestamps may mismatch with the input
-timestamps even when this option is selected.
-
-@item -start_at_zero
-When used with @option{copyts}, shift input timestamps so they start at zero.
-
-This means that using e.g. @code{-ss 50} will make output timestamps start at
-50 seconds, regardless of what timestamp the input file started at.
-
+Copy timestamps from input to output.
 @item -copytb @var{mode}
 Specify how to set the encoder timebase when stream copying.  @var{mode} is an
 integer numeric value, and can assume one of the following values:
@@ -1067,7 +944,7 @@ ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
 @end example
 
 @item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
-Set bitstream filters for matching streams. @var{bitstream_filters} is
+Set bitstream filters for matching streams. @var{bistream_filters} is
 a comma-separated list of bitstream filters. Use the @code{-bsfs} option
 to get the list of bitstream filters.
 @example
@@ -1077,7 +954,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
 @end example
 
-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
 Force a tag/fourcc for matching streams.
 
 @item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -1087,13 +964,11 @@ Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
 ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
 @end example
 
-@anchor{filter_complex_option}
 @item -filter_complex @var{filtergraph} (@emph{global})
-Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
+Define a complex filter graph, i.e. one with arbitrary number of inputs and/or
 outputs. For simple graphs -- those with one input and one output of the same
 type -- see the @option{-filter} options. @var{filtergraph} is a description of
-the filtergraph, as described in the ``Filtergraph syntax'' section of the
-ffmpeg-filters manual.
+the filter graph, as described in @ref{Filtergraph syntax}.
 
 Input link labels must refer to input streams using the
 @code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map}
@@ -1132,58 +1007,8 @@ ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
 
 To generate 5 seconds of pure red video using lavfi @code{color} source:
 @example
-ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
+ffmpeg -filter_complex 'color=red' -t 5 out.mkv
 @end example
-
-@item -lavfi @var{filtergraph} (@emph{global})
-Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
-outputs. Equivalent to @option{-filter_complex}.
-
-@item -filter_complex_script @var{filename} (@emph{global})
-This option is similar to @option{-filter_complex}, the only difference is that
-its argument is the name of the file from which a complex filtergraph
-description is to be read.
-
-@item -accurate_seek (@emph{input})
-This option enables or disables accurate seeking in input files with the
-@option{-ss} option. It is enabled by default, so seeking is accurate when
-transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
-e.g. when copying some streams and transcoding the others.
-
-@item -override_ffserver (@emph{global})
-Overrides the input specifications from @command{ffserver}. Using this
-option you can map any input stream to @command{ffserver} and control
-many aspects of the encoding from @command{ffmpeg}. Without this
-option @command{ffmpeg} will transmit to @command{ffserver} what is
-requested by @command{ffserver}.
-
-The option is intended for cases where features are needed that cannot be
-specified to @command{ffserver} but can be to @command{ffmpeg}.
-
-@item -discard (@emph{input})
-Allows discarding specific streams or frames of streams at the demuxer.
-Not all demuxers support this.
-
-@table @option
-@item none
-Discard no frame.
-
-@item default
-Default, which discards no frames.
-
-@item noref
-Discard all non-reference frames.
-
-@item bidir
-Discard all bidirectional frames.
-
-@item nokey
-Discard all frames excepts keyframes.
-
-@item all
-Discard all frames.
-@end table
-
 @end table
 
 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1241,7 +1066,7 @@ then it will search for the file @file{libvpx-1080p.ffpreset}.
 
 @itemize
 @item
-For streaming at very low bitrates, use a low frame rate
+For streaming at very low bitrate application, use a low frame rate
 and a small GOP size. This is especially true for RealVideo where
 the Linux player does not seem to be very fast, so it can miss
 frames. An example is:
@@ -1320,14 +1145,14 @@ standard mixer.
 Grab the X11 display with ffmpeg via
 
 @example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
 @end example
 
 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.
 
 @example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
 @end example
 
 0.0 is display.screen number of your X11 server, same as the DISPLAY environment
@@ -1464,11 +1289,11 @@ ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
 You can put many streams of the same type in the output:
 
 @example
-ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy test12.nut
 @end example
 
-The resulting output file @file{test12.nut} will contain the first four streams
-from the input files in reverse order.
+The resulting output file @file{test12.avi} will contain first four streams from
+the input file in reverse order.
 
 @item
 To force CBR video output:
@@ -1486,74 +1311,32 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
 @end itemize
 @c man end EXAMPLES
 
-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
+@include syntax.texi
+@include eval.texi
+@include decoders.texi
+@include encoders.texi
+@include demuxers.texi
+@include muxers.texi
+@include indevs.texi
+@include outdevs.texi
 @include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
+@include bitstream_filters.texi
 @include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffmpeg.html,ffmpeg}
-@end ifset
-@ifset config-not-all
-@url{ffmpeg-all.html,ffmpeg-all},
-@end ifset
-@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffmpeg(1),
-@end ifset
-@ifset config-not-all
-ffmpeg-all(1),
-@end ifset
-ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
+@include metadata.texi
 
 @ignore
 
 @setfilename ffmpeg
 @settitle ffmpeg video converter
 
+@c man begin SEEALSO
+ffplay(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+See git history
+@c man end
+
 @end ignore
 
 @bye

doc/ffplay.texi

@@ -11,7 +11,11 @@
 
 @chapter Synopsis
 
-ffplay [@var{options}] [@file{input_file}]
+@example
+@c man begin SYNOPSIS
+ffplay [options] [@file{input_file}]
+@c man end
+@end example
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -24,7 +28,7 @@ various FFmpeg APIs.
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
@@ -37,14 +41,10 @@ Force displayed height.
 Set frame size (WxH or abbreviation), needed for videos which do
 not contain a header with the frame size like raw YUV.  This option
 has been deprecated in favor of private options, try -video_size.
-@item -fs
-Start in fullscreen mode.
 @item -an
 Disable audio.
 @item -vn
 Disable video.
-@item -sn
-Disable subtitles.
 @item -ss @var{pos}
 Seek to a given position in seconds.
 @item -t @var{duration}
@@ -77,25 +77,11 @@ Default value is "video", if video is not present or cannot be played
 You can interactively cycle through the available show modes by
 pressing the key @key{w}.
 
-@item -vf @var{filtergraph}
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the video stream.
-
-@var{filtergraph} is a description of the filtergraph to apply to
-the stream, and must have a single video input and a single video
-output. In the filtergraph, the input is associated to the label
-@code{in}, and the output to the label @code{out}. See the
-ffmpeg-filters manual for more information about the filtergraph
-syntax.
-
-You can specify this parameter multiple times and cycle through the specified
-filtergraphs along with the show modes by pressing the key @key{w}.
-
-@item -af @var{filtergraph}
-@var{filtergraph} is a description of the filtergraph to apply to
-the input audio.
+@item -vf @var{filter_graph}
+@var{filter_graph} is a description of the filter graph to apply to
+the input video.
 Use the option "-filters" to show all the available filters (including
-sources and sinks).
+also sources and sinks).
 
 @item -i @var{input_file}
 Read @var{input_file}.
@@ -106,17 +92,18 @@ Read @var{input_file}.
 @item -pix_fmt @var{format}
 Set pixel format.
 This option has been deprecated in favor of private options, try -pixel_format.
-
 @item -stats
-Print several playback statistics, in particular show the stream
-duration, the codec parameters, the current position in the stream and
-the audio/video synchronisation drift. It is on by default, to
-explicitly disable it you need to specify @code{-nostats}.
-
+Show the stream duration, the codec parameters, the current position in
+the stream and the audio/video synchronisation drift.
+@item -bug
+Work around bugs.
 @item -fast
 Non-spec-compliant optimizations.
 @item -genpts
 Generate pts.
+@item -rtp_tcp
+Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
+if you are streaming with the RTSP protocol.
 @item -sync @var{type}
 Set the master clock to audio (@code{type=audio}), video
 (@code{type=video}) or external (@code{type=ext}). Default is audio. The
@@ -125,8 +112,7 @@ players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
 @item -threads @var{count}
-Set the thread count. By default, @command{ffplay} automatically detects a
-suitable number of threads to use.
+Set the thread count.
 @item -ast @var{audio_stream_number}
 Select the desired audio stream number, counting from 0. The number
 refers to the list of all the input audio streams. If it is greater
@@ -148,36 +134,8 @@ Exit when video is done playing.
 Exit if any key is pressed.
 @item -exitonmousedown
 Exit if any mouse button is pressed.
-
-@item -codec:@var{media_specifier} @var{codec_name}
-Force a specific decoder implementation for the stream identified by
-@var{media_specifier}, which can assume the values @code{a} (audio),
-@code{v} (video), and @code{s} subtitle.
-
-@item -acodec @var{codec_name}
-Force a specific audio decoder.
-
-@item -vcodec @var{codec_name}
-Force a specific video decoder.
-
-@item -scodec @var{codec_name}
-Force a specific subtitle decoder.
-
-@item -autorotate
-Automatically rotate the video according to presentation metadata. Enabled by
-default, use @option{-noautorotate} to disable it.
-
-@item -framedrop
-Drop video frames if video is out of sync. Enabled by default if the master
-clock is not set to video. Use this option to enable frame dropping for all
-master clock sources, use @option{-noframedrop} to disable it.
-
-@item -infbuf
-Do not limit the input buffer size, read as much data as possible from the
-input as soon as possible. Enabled by default for realtime streams, where data
-may be dropped if not read in time. Use this option to enable infinite buffers
-for all inputs, use @option{-noinfbuf} to disable it.
-
+@item -codec:@var{stream_type}
+Force a specific decoder implementation
 @end table
 
 @section While playing
@@ -193,25 +151,16 @@ Toggle full screen.
 Pause.
 
 @item a
-Cycle audio channel in the current program.
+Cycle audio channel.
 
 @item v
 Cycle video channel.
 
 @item t
-Cycle subtitle channel in the current program.
-
-@item c
-Cycle program.
+Cycle subtitle channel.
 
 @item w
-Cycle video filters or show modes.
-
-@item s
-Step to the next frame.
-
-Pause if the stream is not already paused, step to the next video
-frame, and pause.
+Show audio waves.
 
 @item left/right
 Seek backward/forward 10 seconds.
@@ -220,8 +169,6 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.
 
 @item page down/page up
-Seek to the previous/next chapter.
-or if there are no chapters
 Seek backward/forward 10 minutes.
 
 @item mouse click
@@ -231,75 +178,29 @@ Seek to percentage in file corresponding to fraction of width.
 
 @c man end
 
-@include config.texi
-@ifset config-all
-@set config-readonly
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
+@include syntax.texi
+@include eval.texi
+@include decoders.texi
+@include demuxers.texi
+@include muxers.texi
+@include indevs.texi
+@include outdevs.texi
 @include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
 @include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffplay.html,ffplay},
-@end ifset
-@ifset config-not-all
-@url{ffplay-all.html,ffmpeg-all},
-@end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffplay(1),
-@end ifset
-@ifset config-not-all
-ffplay-all(1),
-@end ifset
-ffmpeg(1), ffprobe(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
 
 @ignore
 
 @setfilename ffplay
 @settitle FFplay media player
 
+@c man begin SEEALSO
+ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
 @end ignore
 
 @bye

doc/ffprobe.texi

@@ -11,7 +11,13 @@
 
 @chapter Synopsis
 
-ffprobe [@var{options}] [@file{input_file}]
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffprobe [options] [@file{input_file}]
+@c man end
+@end example
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -39,20 +45,15 @@ ffprobe output is designed to be easily parsable by a textual filter,
 and consists of one or more sections of a form defined by the selected
 writer, which is specified by the @option{print_format} option.
 
-Sections may contain other nested sections, and are identified by a
-name (which may be shared by other sections), and an unique
-name. See the output of @option{sections}.
-
 Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
-section.
+and printed in the corresponding "FORMAT" or "STREAM" section.
 
 @c man end
 
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
@@ -93,36 +94,13 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.
 
-@item -sections
-Print sections structure and section information, and exit. The output
-is not meant to be parsed by a machine.
-
-@item -select_streams @var{stream_specifier}
-Select only the streams specified by @var{stream_specifier}. This
-option affects only the options related to streams
-(e.g. @code{show_streams}, @code{show_packets}, etc.).
-
-For example to show only audio streams, you can use the command:
-@example
-ffprobe -show_streams -select_streams a INPUT
-@end example
-
-To show only video packets belonging to the video stream with index 1:
-@example
-ffprobe -show_packets -select_streams v:1 INPUT
-@end example
-
 @item -show_data
-Show payload data, as a hexadecimal and ASCII dump. Coupled with
+Show payload data, as an hexadecimal and ASCII dump. Coupled with
 @option{-show_packets}, it will dump the packets' data. Coupled with
 @option{-show_streams}, it will dump the codec extradata.
 
 The dump is printed as the "data" field. It may contain newlines.
 
-@item -show_data_hash @var{algorithm}
-Show a hash of payload data, for packets with @option{-show_packets} and for
-codec extradata with @option{-show_streams}.
-
 @item -show_error
 Show information about the error found when trying to probe the input.
 
@@ -140,59 +118,6 @@ Like @option{-show_format}, but only prints the specified entry of the
 container format information, rather than all. This option may be given more
 than once, then all specified entries will be shown.
 
-This option is deprecated, use @code{show_entries} instead.
-
-@item -show_entries @var{section_entries}
-Set list of entries to show.
-
-Entries are specified according to the following
-syntax. @var{section_entries} contains a list of section entries
-separated by @code{:}. Each section entry is composed by a section
-name (or unique name), optionally followed by a list of entries local
-to that section, separated by @code{,}.
-
-If section name is specified but is followed by no @code{=}, all
-entries are printed to output, together with all the contained
-sections. Otherwise only the entries specified in the local section
-entries list are printed. In particular, if @code{=} is specified but
-the list of local entries is empty, then no entries will be shown for
-that section.
-
-Note that the order of specification of the local section entries is
-not honored in the output, and the usual display order will be
-retained.
-
-The formal syntax is given by:
-@example
-@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
-@var{SECTION_ENTRY}         ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
-@var{SECTION_ENTRIES}       ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
-@end example
-
-For example, to show only the index and type of each stream, and the PTS
-time, duration time, and stream index of the packets, you can specify
-the argument:
-@example
-packet=pts_time,duration_time,stream_index : stream=index,codec_type
-@end example
-
-To show all the entries in the section "format", but only the codec
-type in the section "stream", specify the argument:
-@example
-format : stream=codec_type
-@end example
-
-To show all the tags in the stream and format sections:
-@example
-stream_tags : format_tags
-@end example
-
-To show only the @code{title} tag (if available) in the stream
-sections:
-@example
-stream_tags=title
-@end example
-
 @item -show_packets
 Show information about each packet contained in the input multimedia
 stream.
@@ -201,11 +126,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".
 
 @item -show_frames
-Show information about each frame and subtitle contained in the input
-multimedia stream.
+Show information about each frame contained in the input multimedia
+stream.
 
 The information for each single frame is printed within a dedicated
-section with name "FRAME" or "SUBTITLE".
+section with name "FRAME".
 
 @item -show_streams
 Show information about each media stream contained in the input
@@ -214,18 +139,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".
 
-@item -show_programs
-Show information about programs and their streams contained in the input
-multimedia stream.
-
-Each media stream information is printed within a dedicated section
-with name "PROGRAM_STREAM".
-
-@item -show_chapters
-Show information about chapters stored in the format.
-
-Each chapter is printed within a dedicated section with name "CHAPTER".
-
 @item -count_frames
 Count the number of frames per stream and report it in the
 corresponding stream section.
@@ -234,70 +147,6 @@ corresponding stream section.
 Count the number of packets per stream and report it in the
 corresponding stream section.
 
-@item -read_intervals @var{read_intervals}
-
-Read only the specified intervals. @var{read_intervals} must be a
-sequence of interval specifications separated by ",".
-@command{ffprobe} will seek to the interval starting point, and will
-continue reading from that.
-
-Each interval is specified by two optional parts, separated by "%".
-
-The first part specifies the interval start position. It is
-interpreted as an abolute position, or as a relative offset from the
-current position if it is preceded by the "+" character. If this first
-part is not specified, no seeking will be performed when reading this
-interval.
-
-The second part specifies the interval end position. It is interpreted
-as an absolute position, or as a relative offset from the current
-position if it is preceded by the "+" character. If the offset
-specification starts with "#", it is interpreted as the number of
-packets to read (not including the flushing packets) from the interval
-start. If no second part is specified, the program will read until the
-end of the input.
-
-Note that seeking is not accurate, thus the actual interval start
-point may be different from the specified position. Also, when an
-interval duration is specified, the absolute end time will be computed
-by adding the duration to the interval start point found by seeking
-the file, rather than to the specified start value.
-
-The formal syntax is given by:
-@example
-@var{INTERVAL}  ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
-@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
-@end example
-
-A few examples follow.
-@itemize
-@item
-Seek to time 10, read packets until 20 seconds after the found seek
-point, then seek to position @code{01:30} (1 minute and thirty
-seconds) and read packets until position @code{01:45}.
-@example
-10%+20,01:30%01:45
-@end example
-
-@item
-Read only 42 packets after seeking to position @code{01:23}:
-@example
-01:23%+#42
-@end example
-
-@item
-Read only the first 20 seconds from the start:
-@example
-%+20
-@end example
-
-@item
-Read from the start until position @code{02:30}:
-@example
-%02:30
-@end example
-@end itemize
-
 @item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -321,12 +170,6 @@ Show information related to program and library versions. This is the
 equivalent of setting both @option{-show_program_version} and
 @option{-show_library_versions} options.
 
-@item -show_pixel_formats
-Show information about all pixel formats supported by FFmpeg.
-
-Pixel format information for each format is printed within a section
-with name "PIXEL_FORMAT".
-
 @item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -343,42 +186,8 @@ Read @var{input_file}.
 A writer defines the output format adopted by @command{ffprobe}, and will be
 used for printing all the parts of the output.
 
-A writer may accept one or more arguments, which specify the options
-to adopt. The options are specified as a list of @var{key}=@var{value}
-pairs, separated by ":".
-
-All writers support the following options:
-
-@table @option
-@item string_validation, sv
-Set string validation mode.
-
-The following values are accepted.
-@table @samp
-@item fail
-The writer will fail immediately in case an invalid string (UTF-8)
-sequence or code point is found in the input. This is especially
-useful to validate input metadata.
-
-@item ignore
-Any validation error will be ignored. This will result in possibly
-broken output, especially with the json or xml writer.
-
-@item replace
-The writer will substitute invalid UTF-8 sequences or code points with
-the string specified with the @option{string_validation_replacement}.
-@end table
-
-Default value is @samp{replace}.
-
-@item string_validation_replacement, svr
-Set replacement string to use in case @option{string_validation} is
-set to @samp{replace}.
-
-In case the option is not specified, the writer will assume the empty
-string, that is it will remove the invalid sequences from the input
-strings.
-@end table
+A writer may accept one or more arguments, which specify the options to
+adopt.
 
 A description of the currently available writers follows.
 
@@ -394,8 +203,11 @@ keyN=valN
 [/SECTION]
 @end example
 
-Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
-PROGRAM_STREAM section, and are prefixed by the string "TAG:".
+Metadata tags are printed as a line in the corresponding FORMAT or
+STREAM section, and are prefixed by the string "TAG:".
+
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
 
 A description of the accepted options follows.
 
@@ -426,6 +238,9 @@ Metadata tags are printed in the corresponding "format" or "stream"
 section. A metadata tag key, if printed, is prefixed by the string
 "tag:".
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
@@ -477,6 +292,9 @@ A free-form output where each line contains an explicit key=value, such as
 directly embedded in sh scripts as long as the separator character is an
 alphanumeric character or an underscore (see @var{sep_char} option).
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
@@ -539,6 +357,9 @@ JSON based format.
 
 Each section is printed using JSON notation.
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
@@ -565,6 +386,9 @@ Note that the output issued will be compliant to the
 (@option{unit}, @option{prefix}, @option{byte_binary_prefix},
 @option{sexagesimal} etc.) are specified.
 
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.
 
 @table @option
@@ -607,75 +431,25 @@ DV, GXF and AVI timecodes are available in format metadata
 @end itemize
 @c man end TIMECODE
 
-@include config.texi
-@ifset config-all
-@set config-readonly
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
+@include syntax.texi
+@include decoders.texi
+@include demuxers.texi
 @include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffprobe.html,ffprobe},
-@end ifset
-@ifset config-not-all
-@url{ffprobe-all.html,ffprobe-all},
-@end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffprobe(1),
-@end ifset
-@ifset config-not-all
-ffprobe-all(1),
-@end ifset
-ffmpeg(1), ffplay(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
+@include indevs.texi
 
 @ignore
 
 @setfilename ffprobe
 @settitle ffprobe media prober
 
+@c man begin SEEALSO
+ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
 @end ignore
 
 @bye

doc/ffprobe.xsd

@@ -8,17 +8,13 @@
 
     <xsd:complexType name="ffprobeType">
         <xsd:sequence>
-            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="pixel_formats"    type="ffprobe:pixelFormatsType"    minOccurs="0" maxOccurs="1" />
             <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
             <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
+            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
+            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
         </xsd:sequence>
     </xsd:complexType>
 
@@ -30,20 +26,7 @@
 
     <xsd:complexType name="framesType">
         <xsd:sequence>
-            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
-            </xsd:choice>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetsAndFramesType">
-        <xsd:sequence>
-            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-                <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
-            </xsd:choice>
+            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
         </xsd:sequence>
     </xsd:complexType>
 
@@ -62,15 +45,9 @@
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
       <xsd:attribute name="data"          type="xsd:string" />
-      <xsd:attribute name="data_hash"     type="xsd:string" />
     </xsd:complexType>
 
     <xsd:complexType name="frameType">
-      <xsd:sequence>
-            <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-            <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
-      </xsd:sequence>
-
       <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
@@ -79,12 +56,9 @@
       <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
       <xsd:attribute name="pkt_duration"  type="xsd:long" />
       <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
-      <xsd:attribute name="pkt_size"      type="xsd:int" />
 
       <!-- audio attributes -->
       <xsd:attribute name="sample_fmt"             type="xsd:string"/>
@@ -103,26 +77,7 @@
       <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
       <xsd:attribute name="top_field_first"        type="xsd:int"   />
       <xsd:attribute name="repeat_pict"            type="xsd:int"   />
-    </xsd:complexType>
-
-    <xsd:complexType name="frameSideDataListType">
-        <xsd:sequence>
-            <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-    <xsd:complexType name="frameSideDataType">
-        <xsd:attribute name="side_data_type"              type="xsd:string"/>
-        <xsd:attribute name="side_data_size"              type="xsd:int"   />
-    </xsd:complexType>
-
-    <xsd:complexType name="subtitleType">
-      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
-      <xsd:attribute name="pts"                type="xsd:long" />
-      <xsd:attribute name="pts_time"           type="xsd:float"/>
-      <xsd:attribute name="format"             type="xsd:int"  />
-      <xsd:attribute name="start_display_time" type="xsd:int"  />
-      <xsd:attribute name="end_display_time"   type="xsd:int"  />
-      <xsd:attribute name="num_rects"          type="xsd:int"  />
+      <xsd:attribute name="reference"              type="xsd:int"   />
     </xsd:complexType>
 
     <xsd:complexType name="streamsType">
@@ -131,29 +86,8 @@
         </xsd:sequence>
     </xsd:complexType>
 
-    <xsd:complexType name="programsType">
-        <xsd:sequence>
-            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="streamDispositionType">
-      <xsd:attribute name="default"          type="xsd:int" use="required" />
-      <xsd:attribute name="dub"              type="xsd:int" use="required" />
-      <xsd:attribute name="original"         type="xsd:int" use="required" />
-      <xsd:attribute name="comment"          type="xsd:int" use="required" />
-      <xsd:attribute name="lyrics"           type="xsd:int" use="required" />
-      <xsd:attribute name="karaoke"          type="xsd:int" use="required" />
-      <xsd:attribute name="forced"           type="xsd:int" use="required" />
-      <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
-      <xsd:attribute name="visual_impaired"  type="xsd:int" use="required" />
-      <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
-      <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
-    </xsd:complexType>
-
     <xsd:complexType name="streamType">
       <xsd:sequence>
-        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
         <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
       </xsd:sequence>
 
@@ -166,7 +100,6 @@
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
@@ -176,18 +109,12 @@
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
       <xsd:attribute name="pix_fmt"              type="xsd:string"/>
       <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="color_range"          type="xsd:string"/>
-      <xsd:attribute name="color_space"          type="xsd:string"/>
-      <xsd:attribute name="color_transfer"       type="xsd:string"/>
-      <xsd:attribute name="color_primaries"      type="xsd:string"/>
-      <xsd:attribute name="chroma_location"      type="xsd:string"/>
       <xsd:attribute name="timecode"             type="xsd:string"/>
 
       <!-- audio attributes -->
       <xsd:attribute name="sample_fmt"       type="xsd:string"/>
       <xsd:attribute name="sample_rate"      type="xsd:int"/>
       <xsd:attribute name="channels"         type="xsd:int"/>
-      <xsd:attribute name="channel_layout"   type="xsd:string"/>
       <xsd:attribute name="bits_per_sample"  type="xsd:int"/>
 
       <xsd:attribute name="id"               type="xsd:string"/>
@@ -199,30 +126,11 @@
       <xsd:attribute name="duration_ts"      type="xsd:long"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="bit_rate"         type="xsd:int"/>
-      <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
-      <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
       <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
       <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
     </xsd:complexType>
 
-    <xsd:complexType name="programType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="start_time"           type="xsd:float"/>
-      <xsd:attribute name="start_pts"            type="xsd:long"/>
-      <xsd:attribute name="end_time"             type="xsd:float"/>
-      <xsd:attribute name="end_pts"              type="xsd:long"/>
-      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
-      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
     <xsd:complexType name="formatType">
       <xsd:sequence>
         <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -230,14 +138,12 @@
 
       <xsd:attribute name="filename"         type="xsd:string" use="required"/>
       <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
       <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
       <xsd:attribute name="format_long_name" type="xsd:string"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="size"             type="xsd:long"/>
       <xsd:attribute name="bit_rate"         type="xsd:long"/>
-      <xsd:attribute name="probe_score"      type="xsd:int"/>
     </xsd:complexType>
 
     <xsd:complexType name="tagType">
@@ -255,36 +161,17 @@
       <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
       <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
       <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
+      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
+      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
       <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
     </xsd:complexType>
 
-    <xsd:complexType name="chaptersType">
-      <xsd:sequence>
-        <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="chapterType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="id"         type="xsd:int" use="required"/>
-      <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
-      <xsd:attribute name="start"      type="xsd:int" use="required"/>
-      <xsd:attribute name="start_time" type="xsd:float"/>
-      <xsd:attribute name="end"        type="xsd:int" use="required"/>
-      <xsd:attribute name="end_time"   type="xsd:float" use="required"/>
-    </xsd:complexType>
-
     <xsd:complexType name="libraryVersionType">
       <xsd:attribute name="name"        type="xsd:string" use="required"/>
       <xsd:attribute name="major"       type="xsd:int"    use="required"/>
       <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
       <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
       <xsd:attribute name="version"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="ident"       type="xsd:string" use="required"/>
     </xsd:complexType>
 
     <xsd:complexType name="libraryVersionsType">
@@ -292,45 +179,4 @@
           <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
         </xsd:sequence>
     </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatFlagsType">
-      <xsd:attribute name="big_endian" type="xsd:int" use="required"/>
-      <xsd:attribute name="palette"    type="xsd:int" use="required"/>
-      <xsd:attribute name="bitstream"  type="xsd:int" use="required"/>
-      <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
-      <xsd:attribute name="planar"     type="xsd:int" use="required"/>
-      <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
-      <xsd:attribute name="pseudopal"  type="xsd:int" use="required"/>
-      <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatComponentType">
-      <xsd:attribute name="index"      type="xsd:int" use="required"/>
-      <xsd:attribute name="bit_depth"  type="xsd:int" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatComponentsType">
-      <xsd:sequence>
-        <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatType">
-      <xsd:sequence>
-        <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
-        <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="name"                type="xsd:string" use="required"/>
-      <xsd:attribute name="nb_components"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="log2_chroma_w"       type="xsd:int"/>
-      <xsd:attribute name="log2_chroma_h"       type="xsd:int"/>
-      <xsd:attribute name="bits_per_pixel"      type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatsType">
-      <xsd:sequence>
-        <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
 </xsd:schema>

doc/ffserver.conf

@@ -1,11 +1,11 @@
 # Port on which the server is listening. You must select a different
 # port from your standard HTTP web server if it is running on the same
 # computer.
-HTTPPort 8090
+Port 8090
 
 # Address on which the server is bound. Only useful if you have
 # several network interfaces.
-HTTPBindAddress 0.0.0.0
+BindAddress 0.0.0.0
 
 # Number of simultaneous HTTP connections that can be handled. It has
 # to be defined *before* the MaxClients parameter, since it defines the
@@ -25,6 +25,10 @@ MaxBandwidth 1000
 # '-' is the standard output.
 CustomLog -
 
+# Suppress that if you want to launch ffserver as a daemon.
+NoDaemon
+
+
 ##################################################################
 # Definition of the live feeds. Each live feed contains one video
 # and/or audio sequence coming from an ffmpeg encoder or another
@@ -235,7 +239,7 @@ StartSendOnKey
 
 #<Stream test.ogg>
 #Feed feed1.ffm
-#Metadata title "Stream title"
+#Title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +284,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Metadata author "Me"
-#Metadata copyright "Super MegaCorp"
-#Metadata title "Test stream from disk"
-#Metadata comment "Test comment"
+#Author "Me"
+#Copyright "Super MegaCorp"
+#Title "Test stream from disk"
+#Comment "Test comment"
 #</Stream>
 
 

doc/ffserver.texi

@@ -9,159 +9,57 @@
 
 @contents
 
-@chapter Synopsis
+@chapter Synopsys
 
-ffserver [@var{options}]
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffserver [options]
+@c man end
+@end example
 
 @chapter Description
 @c man begin DESCRIPTION
 
-@command{ffserver} is a streaming server for both audio and video.
-It supports several live feeds, streaming from files and time shifting
-on live feeds. You can seek to positions in the past on each live
-feed, provided you specify a big enough feed storage.
+ffserver is a streaming server for both audio and video. It supports
 
-@command{ffserver} is configured through a configuration file, which
-is read at startup. If not explicitly specified, it will read from
-@file{/etc/ffserver.conf}.
+several live feeds, streaming from files and time shifting on live feeds
+(you can seek to positions in the past on each live feed, provided you
+specify a big enough feed storage in ffserver.conf).
 
-@command{ffserver} receives prerecorded files or FFM streams from some
-@command{ffmpeg} instance as input, then streams them over
-RTP/RTSP/HTTP.
+ffserver runs in daemon mode by default; that is, it puts itself in
+the background and detaches from its TTY, unless it is launched in
+debug mode or a NoDaemon option is specified in the configuration
+file.
 
-An @command{ffserver} instance will listen on some port as specified
-in the configuration file. You can launch one or more instances of
-@command{ffmpeg} and send one or more FFM streams to the port where
-ffserver is expecting to receive them. Alternately, you can make
-@command{ffserver} launch such @command{ffmpeg} instances at startup.
+This documentation covers only the streaming aspects of ffserver /
+ffmpeg. All questions about parameters for ffmpeg, codec questions,
+etc. are not covered here. Read @file{ffmpeg.html} for more
+information.
 
-Input streams are called feeds, and each one is specified by a
-@code{<Feed>} section in the configuration file.
+@section How does it work?
+
+ffserver receives prerecorded files or FFM streams from some ffmpeg
+instance as input, then streams them over RTP/RTSP/HTTP.
+
+An ffserver instance will listen on some port as specified in the
+configuration file. You can launch one or more instances of ffmpeg and
+send one or more FFM streams to the port where ffserver is expecting
+to receive them. Alternately, you can make ffserver launch such ffmpeg
+instances at startup.
+
+Input streams are called feeds, and each one is specified by a <Feed>
+section in the configuration file.
 
 For each feed you can have different output streams in various
-formats, each one specified by a @code{<Stream>} section in the
-configuration file.
-
-@chapter Detailed description
-
-@command{ffserver} works by forwarding streams encoded by
-@command{ffmpeg}, or pre-recorded streams which are read from disk.
-
-Precisely, @command{ffserver} acts as an HTTP server, accepting POST
-requests from @command{ffmpeg} to acquire the stream to publish, and
-serving RTSP clients or HTTP clients GET requests with the stream
-media content.
-
-A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
-a port where @command{ffserver} is listening.
-
-Each feed is identified by a unique name, corresponding to the name
-of the resource published on @command{ffserver}, and is configured by
-a dedicated @code{Feed} section in the configuration file.
-
-The feed publish URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
-@end example
-
-where @var{ffserver_ip_address} is the IP address of the machine where
-@command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{HTTPPort} option), and
-@var{feed_name} is the name of the corresponding feed defined in the
-configuration file.
-
-Each feed is associated to a file which is stored on disk. This stored
-file is used to send pre-recorded data to a player as fast as
-possible when new content is added in real-time to the stream.
-
-A "live-stream" or "stream" is a resource published by
-@command{ffserver}, and made accessible through the HTTP protocol to
-clients.
-
-A stream can be connected to a feed, or to a file. In the first case,
-the published stream is forwarded from the corresponding feed
-generated by a running instance of @command{ffmpeg}, in the second
-case the stream is read from a pre-recorded file.
-
-Each stream is identified by a unique name, corresponding to the name
-of the resource served by @command{ffserver}, and is configured by
-a dedicated @code{Stream} section in the configuration file.
-
-The stream access HTTP URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
-@end example
-
-The stream access RTSP URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
-@end example
-
-@var{stream_name} is the name of the corresponding stream defined in
-the configuration file. @var{options} is a list of options specified
-after the URL which affects how the stream is served by
-@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
-and RTSP ports configured with the options @var{HTTPPort} and
-@var{RTSPPort} respectively.
-
-In case the stream is associated to a feed, the encoding parameters
-must be configured in the stream configuration. They are sent to
-@command{ffmpeg} when setting up the encoding. This allows
-@command{ffserver} to define the encoding parameters used by
-the @command{ffmpeg} encoders.
-
-The @command{ffmpeg} @option{override_ffserver} commandline option
-allows one to override the encoding parameters set by the server.
-
-Multiple streams can be connected to the same feed.
-
-For example, you can have a situation described by the following
-graph:
-@example
-               _________       __________
-              |         |     |          |
-ffmpeg 1 -----| feed 1  |-----| stream 1 |
-    \         |_________|\    |__________|
-     \                    \
-      \                    \   __________
-       \                    \ |          |
-        \                    \| stream 2 |
-         \                    |__________|
-          \
-           \   _________       __________
-            \ |         |     |          |
-             \| feed 2  |-----| stream 3 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-ffmpeg 2 -----| feed 3  |-----| stream 4 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-              | file 1  |-----| stream 5 |
-              |_________|     |__________|
-@end example
-
-@anchor{FFM}
-@section FFM, FFM2 formats
-
-FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
-video and audio streams and encoding options, and can store a moving time segment
-of an infinite movie or a whole movie.
-
-FFM is version specific, and there is limited compatibility of FFM files
-generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but it is not guaranteed to work.
-
-FFM2 is extensible while maintaining compatibility and should work between
-differing versions of tools. FFM2 is the default.
+formats, each one specified by a <Stream> section in the configuration
+file.
 
 @section Status stream
 
-@command{ffserver} supports an HTTP interface which exposes the
-current status of the server.
+ffserver supports an HTTP interface which exposes the current status
+of the server.
 
 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -180,8 +78,35 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.
 
+@section What can this do?
+
+When properly configured and running, you can capture video and audio in real
+time from a suitable capture card, and stream it out over the Internet to
+either Windows Media Player or RealAudio player (with some restrictions).
+
+It can also stream from files, though that is currently broken. Very often, a
+web server can be used to serve up the files just as well.
+
+It can stream prerecorded video from .ffm files, though it is somewhat tricky
+to make it work correctly.
+
+@section What do I need?
+
+I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm
+using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
+I needed some special drivers for my motherboard-based sound card.]
+
+I understand that FreeBSD systems work just fine as well.
+
 @section How do I make it work?
 
+First, build the kit. It *really* helps to have installed LAME first. Then when
+you run the ffserver ./configure, make sure that you have the
+@code{--enable-libmp3lame} flag turned on.
+
+LAME is important as it allows for streaming audio to Windows Media Player.
+Don't ask why the other audio types do not work.
+
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:
 
@@ -203,9 +128,40 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
 transfer the entire file before starting to play.
 The same is true of AVI files.
 
-You should edit the @file{ffserver.conf} file to suit your needs (in
-terms of frame rates etc). Then install @command{ffserver} and
-@command{ffmpeg}, write a script to start them up, and off you go.
+@section What happens next?
+
+You should edit the ffserver.conf file to suit your needs (in terms of
+frame rates etc). Then install ffserver and ffmpeg, write a script to start
+them up, and off you go.
+
+@section Troubleshooting
+
+@subsection I don't hear any audio, but video is fine.
+
+Maybe you didn't install LAME, or got your ./configure statement wrong. Check
+the ffmpeg output to see if a line referring to MP3 is present. If not, then
+your configuration was incorrect. If it is, then maybe your wiring is not
+set up correctly. Maybe the sound card is not getting data from the right
+input source. Maybe you have a really awful audio interface (like I do)
+that only captures in stereo and also requires that one channel be flipped.
+If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
+starting ffmpeg.
+
+@subsection The audio and video lose sync after a while.
+
+Yes, they do.
+
+@subsection After a long while, the video update rate goes way down in WMP.
+
+Yes, it does. Who knows why?
+
+@subsection WMP 6.4 behaves differently to WMP 7.
+
+Yes, it does. Any thoughts on this would be gratefully received. These
+differences extend to embedding WMP into a web page. [There are two
+object IDs that you can use: The old one, which does not play well, and
+the new one, which does (both tested on the same system). However,
+I suspect that the new one is not available unless you have installed WMP 7].
 
 @section What else can it do?
 
@@ -246,6 +202,9 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.
 
+* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
+the amount of bandwidth consumed by live streams.
+
 @section Why does the ?buffer / Preroll stop working after a time?
 
 It turns out that (on my machine at least) the number of frames successfully
@@ -282,639 +241,39 @@ For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
 @table @option
 @item -f @var{configfile}
-Read configuration file @file{configfile}. If not specified it will
-read by default from @file{/etc/ffserver.conf}.
-
+Use @file{configfile} instead of @file{/etc/ffserver.conf}.
 @item -n
-Enable no-launch mode. This option disables all the @code{Launch}
-directives within the various @code{<Feed>} sections. Since
-@command{ffserver} will not launch any @command{ffmpeg} instances, you
-will have to launch them manually.
-
+Enable no-launch mode. This option disables all the Launch directives
+within the various <Stream> sections. Since ffserver will not launch
+any ffmpeg instances, you will have to launch them manually.
 @item -d
-Enable debug mode. This option increases log verbosity, and directs
-log messages to stdout. When specified, the @option{CustomLog} option
-is ignored.
+Enable debug mode. This option increases log verbosity, directs log
+messages to stdout and causes ffserver to run in the foreground
+rather than as a daemon.
 @end table
-
-@chapter Configuration file syntax
-
-@command{ffserver} reads a configuration file containing global
-options and settings for each stream and feed.
-
-The configuration file consists of global options and dedicated
-sections, which must be introduced by "<@var{SECTION_NAME}
-@var{ARGS}>" on a separate line and must be terminated by a line in
-the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
-
-Currently the following sections are recognized: @samp{Feed},
-@samp{Stream}, @samp{Redirect}.
-
-A line starting with @code{#} is ignored and treated as a comment.
-
-Name of options and sections are case-insensitive.
-
-@section ACL syntax
-An ACL (Access Control List) specifies the address which are allowed
-to access a given stream, or to write a given feed.
-
-It accepts the folling forms
-@itemize
-@item
-Allow/deny access to @var{address}.
-@example
-ACL ALLOW <address>
-ACL DENY <address>
-@end example
-
-@item
-Allow/deny access to ranges of addresses from @var{first_address} to
-@var{last_address}.
-@example
-ACL ALLOW <first_address> <last_address>
-ACL DENY <first_address> <last_address>
-@end example
-@end itemize
-
-You can repeat the ACL allow/deny as often as you like. It is on a per
-stream basis. The first match defines the action. If there are no matches,
-then the default is the inverse of the last ACL statement.
-
-Thus 'ACL allow localhost' only allows access from localhost.
-'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
-allow everybody else.
-
-@section Global options
-@table @option
-@item HTTPPort @var{port_number}
-@item Port @var{port_number}
-@item RTSPPort @var{port_number}
-
-@var{HTTPPort} sets the HTTP server listening TCP port number,
-@var{RTSPPort} sets the RTSP server listening TCP port number.
-
-@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
-
-You must select a different port from your standard HTTP web server if
-it is running on the same computer.
-
-If not specified, no corresponding server will be created.
-
-@item HTTPBindAddress @var{ip_address}
-@item BindAddress @var{ip_address}
-@item RTSPBindAddress @var{ip_address}
-Set address on which the HTTP/RTSP server is bound. Only useful if you
-have several network interfaces.
-
-@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
-deprecated.
-
-@item MaxHTTPConnections @var{n}
-Set number of simultaneous HTTP connections that can be handled. It
-has to be defined @emph{before} the @option{MaxClients} parameter,
-since it defines the @option{MaxClients} maximum limit.
-
-Default value is 2000.
-
-@item MaxClients @var{n}
-Set number of simultaneous requests that can be handled. Since
-@command{ffserver} is very fast, it is more likely that you will want
-to leave this high and use @option{MaxBandwidth}.
-
-Default value is 5.
-
-@item MaxBandwidth @var{kbps}
-Set the maximum amount of kbit/sec that you are prepared to consume
-when streaming to clients.
-
-Default value is 1000.
-
-@item CustomLog @var{filename}
-Set access log file (uses standard Apache log file format). '-' is the
-standard output.
-
-If not specified @command{ffserver} will produce no log.
-
-In case the commandline option @option{-d} is specified this option is
-ignored, and the log is written to standard output.
-
-@item NoDaemon
-Set no-daemon mode. This option is currently ignored since now
-@command{ffserver} will always work in no-daemon mode, and is
-deprecated.
-
-@item UseDefaults
-@item NoDefaults
-Control whether default codec options are used for the all streams or not.
-Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
-The lastest occurrence overrides previous if multiple definitions.
-@end table
-
-@section Feed section
-
-A Feed section defines a feed provided to @command{ffserver}.
-
-Each live feed contains one video and/or audio sequence coming from an
-@command{ffmpeg} encoder or another @command{ffserver}. This sequence
-may be encoded simultaneously with several codecs at several
-resolutions.
-
-A feed instance specification is introduced by a line in the form:
-@example
-<Feed FEED_FILENAME>
-@end example
-
-where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
-
-The following options are recognized within a Feed section.
-
-@table @option
-@item File @var{filename}
-@item ReadOnlyFile @var{filename}
-Set the path where the feed file is stored on disk.
-
-If not specified, the @file{/tmp/FEED.ffm} is assumed, where
-@var{FEED} is the feed name.
-
-If @option{ReadOnlyFile} is used the file is marked as read-only and
-it will not be deleted or updated.
-
-@item Truncate
-Truncate the feed file, rather than appending to it. By default
-@command{ffserver} will append data to the file, until the maximum
-file size value is reached (see @option{FileMaxSize} option).
-
-@item FileMaxSize @var{size}
-Set maximum size of the feed file in bytes. 0 means unlimited. The
-postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
-recognized.
-
-Default value is 5M.
-
-@item Launch @var{args}
-Launch an @command{ffmpeg} command when creating @command{ffserver}.
-
-@var{args} must be a sequence of arguments to be provided to an
-@command{ffmpeg} instance. The first provided argument is ignored, and
-it is replaced by a path with the same dirname of the @command{ffserver}
-instance, followed by the remaining argument and terminated with a
-path corresponding to the feed.
-
-When the launched process exits, @command{ffserver} will launch
-another program instance.
-
-In case you need a more complex @command{ffmpeg} configuration,
-e.g. if you need to generate multiple FFM feeds with a single
-@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
-
-This option is ignored in case the commandline option @option{-n} is
-specified.
-
-@item ACL @var{spec}
-Specify the list of IP address which are allowed or denied to write
-the feed. Multiple ACL options can be specified.
-@end table
-
-@section Stream section
-
-A Stream section defines a stream provided by @command{ffserver}, and
-identified by a single name.
-
-The stream is sent when answering a request containing the stream
-name.
-
-A stream section must be introduced by the line:
-@example
-<Stream STREAM_NAME>
-@end example
-
-where @var{STREAM_NAME} specifies the unique name of the stream.
-
-The following options are recognized within a Stream section.
-
-Encoding options are marked with the @emph{encoding} tag, and they are
-used to set the encoding parameters, and are mapped to libavcodec
-encoding options. Not all encoding options are supported, in
-particular it is not possible to set encoder private options. In order
-to override the encoding options specified by @command{ffserver}, you
-can use the @command{ffmpeg} @option{override_ffserver} commandline
-option.
-
-Only one of the @option{Feed} and @option{File} options should be set.
-
-@table @option
-@item Feed @var{feed_name}
-Set the input feed. @var{feed_name} must correspond to an existing
-feed defined in a @code{Feed} section.
-
-When this option is set, encoding options are used to setup the
-encoding operated by the remote @command{ffmpeg} process.
-
-@item File @var{filename}
-Set the filename of the pre-recorded input file to stream.
-
-When this option is set, encoding options are ignored and the input
-file content is re-streamed as is.
-
-@item Format @var{format_name}
-Set the format of the output stream.
-
-Must be the name of a format recognized by FFmpeg. If set to
-@samp{status}, it is treated as a status stream.
-
-@item InputFormat @var{format_name}
-Set input format. If not specified, it is automatically guessed.
-
-@item Preroll @var{n}
-Set this to the number of seconds backwards in time to start. Note that
-most players will buffer 5-10 seconds of video, and also you need to allow
-for a keyframe to appear in the data stream.
-
-Default value is 0.
-
-@item StartSendOnKey
-Do not send stream until it gets the first key frame. By default
-@command{ffserver} will send data immediately.
-
-@item MaxTime @var{n}
-Set the number of seconds to run. This value set the maximum duration
-of the stream a client will be able to receive.
-
-A value of 0 means that no limit is set on the stream duration.
-
-@item ACL @var{spec}
-Set ACL for the stream.
-
-@item DynamicACL @var{spec}
-
-@item RTSPOption @var{option}
-
-@item MulticastAddress @var{address}
-
-@item MulticastPort @var{port}
-
-@item MulticastTTL @var{integer}
-
-@item NoLoop
-
-@item FaviconURL @var{url}
-Set favicon (favourite icon) for the server status page. It is ignored
-for regular streams.
-
-@item Author @var{value}
-@item Comment @var{value}
-@item Copyright @var{value}
-@item Title @var{value}
-Set metadata corresponding to the option. All these options are
-deprecated in favor of @option{Metadata}.
-
-@item Metadata @var{key} @var{value}
-Set metadata value on the output stream.
-
-@item UseDefaults
-@item NoDefaults
-Control whether default codec options are used for the stream or not.
-Default is @var{UseDefaults} unless disabled globally.
-
-@item NoAudio
-@item NoVideo
-Suppress audio/video.
-
-@item AudioCodec @var{codec_name} (@emph{encoding,audio})
-Set audio codec.
-
-@item AudioBitRate @var{rate} (@emph{encoding,audio})
-Set bitrate for the audio stream in kbits per second.
-
-@item AudioChannels @var{n} (@emph{encoding,audio})
-Set number of audio channels.
-
-@item AudioSampleRate @var{n} (@emph{encoding,audio})
-Set sampling frequency for audio. When using low bitrates, you should
-lower this frequency to 22050 or 11025. The supported frequencies
-depend on the selected audio codec.
-
-@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
-Set generic or private option for audio stream.
-Private option must be prefixed with codec name or codec must be defined before.
-
-@item AVPresetAudio @var{preset} (@emph{encoding,audio})
-Set preset for audio stream.
-
-@item VideoCodec @var{codec_name} (@emph{encoding,video})
-Set video codec.
-
-@item VideoBitRate @var{n} (@emph{encoding,video})
-Set bitrate for the video stream in kbits per second.
-
-@item VideoBitRateRange @var{range} (@emph{encoding,video})
-Set video bitrate range.
-
-A range must be specified in the form @var{minrate}-@var{maxrate}, and
-specifies the @option{minrate} and @option{maxrate} encoding options
-expressed in kbits per second.
-
-@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
-Set video bitrate tolerance in kbits per second.
-
-@item PixelFormat @var{pixel_format} (@emph{encoding,video})
-Set video pixel format.
-
-@item Debug @var{integer} (@emph{encoding,video})
-Set video @option{debug} encoding option.
-
-@item Strict @var{integer} (@emph{encoding,video})
-Set video @option{strict} encoding option.
-
-@item VideoBufferSize @var{n} (@emph{encoding,video})
-Set ratecontrol buffer size, expressed in KB.
-
-@item VideoFrameRate @var{n} (@emph{encoding,video})
-Set number of video frames per second.
-
-@item VideoSize (@emph{encoding,video})
-Set size of the video frame, must be an abbreviation or in the form
-@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
-in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-Default value is @code{160x128}.
-
-@item VideoIntraOnly (@emph{encoding,video})
-Transmit only intra frames (useful for low bitrates, but kills frame rate).
-
-@item VideoGopSize @var{n} (@emph{encoding,video})
-If non-intra only, an intra frame is transmitted every VideoGopSize
-frames. Video synchronization can only begin at an intra frame.
-
-@item VideoTag @var{tag} (@emph{encoding,video})
-Set video tag.
-
-@item VideoHighQuality (@emph{encoding,video})
-@item Video4MotionVector (@emph{encoding,video})
-
-@item BitExact (@emph{encoding,video})
-Set bitexact encoding flag.
-
-@item IdctSimple (@emph{encoding,video})
-Set simple IDCT algorithm.
-
-@item Qscale @var{n} (@emph{encoding,video})
-Enable constant quality encoding, and set video qscale (quantization
-scale) value, expressed in @var{n} QP units.
-
-@item VideoQMin @var{n} (@emph{encoding,video})
-@item VideoQMax @var{n} (@emph{encoding,video})
-Set video qmin/qmax.
-
-@item VideoQDiff @var{integer} (@emph{encoding,video})
-Set video @option{qdiff} encoding option.
-
-@item LumiMask @var{float} (@emph{encoding,video})
-@item DarkMask @var{float} (@emph{encoding,video})
-Set @option{lumi_mask}/@option{dark_mask} encoding options.
-
-@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
-Set generic or private option for video stream.
-Private option must be prefixed with codec name or codec must be defined before.
-
-@item AVPresetVideo @var{preset} (@emph{encoding,video})
-Set preset for video stream.
-
-@var{preset} must be the path of a preset file.
-@end table
-
-@subsection Server status stream
-
-A server status stream is a special stream which is used to show
-statistics about the @command{ffserver} operations.
-
-It must be specified setting the option @option{Format} to
-@samp{status}.
-
-@section Redirect section
-
-A redirect section specifies where to redirect the requested URL to
-another page.
-
-A redirect section must be introduced by the line:
-@example
-<Redirect NAME>
-@end example
-
-where @var{NAME} is the name of the page which should be redirected.
-
-It only accepts the option @option{URL}, which specify the redirection
-URL.
-
-@chapter Stream examples
-
-@itemize
-@item
-Multipart JPEG
-@example
-<Stream test.mjpg>
-Feed feed1.ffm
-Format mpjpeg
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Single JPEG
-@example
-<Stream test.jpg>
-Feed feed1.ffm
-Format jpeg
-VideoFrameRate 2
-VideoIntraOnly
-VideoSize 352x240
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Flash
-@example
-<Stream test.swf>
-Feed feed1.ffm
-Format swf
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-</Stream>
-@end example
-
-@item
-ASF compatible
-@example
-<Stream test.asf>
-Feed feed1.ffm
-Format asf
-VideoFrameRate 15
-VideoSize 352x240
-VideoBitRate 256
-VideoBufferSize 40
-VideoGopSize 30
-AudioBitRate 64
-StartSendOnKey
-</Stream>
-@end example
-
-@item
-MP3 audio
-@example
-<Stream test.mp3>
-Feed feed1.ffm
-Format mp2
-AudioCodec mp3
-AudioBitRate 64
-AudioChannels 1
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Ogg Vorbis audio
-@example
-<Stream test.ogg>
-Feed feed1.ffm
-Metadata title "Stream title"
-AudioBitRate 64
-AudioChannels 2
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio only at 32 kbits
-@example
-<Stream test.ra>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio and video at 64 kbits
-@example
-<Stream test.rm>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-VideoBitRate 128
-VideoFrameRate 25
-VideoGopSize 25
-</Stream>
-@end example
-
-@item
-For stream coming from a file: you only need to set the input filename
-and optionally a new format.
-
-@example
-<Stream file.rm>
-File "/usr/local/httpd/htdocs/tlive.rm"
-NoAudio
-</Stream>
-@end example
-
-@example
-<Stream file.asf>
-File "/usr/local/httpd/htdocs/test.asf"
-NoAudio
-Metadata author "Me"
-Metadata copyright "Super MegaCorp"
-Metadata title "Test stream from disk"
-Metadata comment "Test comment"
-</Stream>
-@end example
-@end itemize
-
 @c man end
 
-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
-@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffserver.html,ffserver},
-@end ifset
-@ifset config-not-all
-@url{ffserver-all.html,ffserver-all},
-@end ifset
-the @file{doc/ffserver.conf} example,
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffserver(1),
-@end ifset
-@ifset config-not-all
-ffserver-all(1),
-@end ifset
-the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
-
 @ignore
 
 @setfilename ffserver
 @settitle ffserver video server
 
+@c man begin SEEALSO
+
+ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
+example and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
 @end ignore
 
 @bye

doc/fftools-common-opts.texi

@@ -1,342 +0,0 @@
-All the numerical options, if not specified otherwise, accept a string
-representing a number as input, which may be followed by one of the SI
-unit prefixes, for example: 'K', 'M', or 'G'.
-
-If 'i' is appended to the SI unit prefix, the complete prefix will be
-interpreted as a unit prefix for binary multiples, which are based on
-powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
-prefix multiplies the value by 8. This allows using, for example:
-'KB', 'MiB', 'G' and 'B' as number suffixes.
-
-Options which do not take arguments are boolean options, and set the
-corresponding value to true. They can be set to false by prefixing
-the option name with "no". For example using "-nofoo"
-will set the boolean option with name "foo" to false.
-
-@anchor{Stream specifiers}
-@section Stream specifiers
-Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
-are used to precisely specify which stream(s) a given option belongs to.
-
-A stream specifier is a string generally appended to the option name and
-separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
-@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
-would select the ac3 codec for the second audio stream.
-
-A stream specifier can match several streams, so that the option is applied to all
-of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
-streams.
-
-An empty stream specifier matches all streams. For example, @code{-codec copy}
-or @code{-codec: copy} would copy all the streams without reencoding.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index. E.g. @code{-threads:1 4} would set the
-thread count for the second stream to 4.
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
-stream number @var{stream_index} of this type. Otherwise, it matches all
-streams of this type.
-@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
-in the program with the id @var{program_id}. Otherwise, it matches all streams in the
-program.
-@item #@var{stream_id} or i:@var{stream_id}
-Match the stream by stream id (e.g. PID in MPEG-TS container).
-@item m:@var{key}[:@var{value}]
-Matches streams with the metadata tag @var{key} having the specified value. If
-@var{value} is not given, matches streams that contain the given tag with any
-value.
-
-Note that in @command{ffmpeg}, matching by metadata will only work properly for
-input files.
-@end table
-
-@section Generic options
-
-These options are shared amongst the ff* tools.
-
-@table @option
-
-@item -L
-Show license.
-
-@item -h, -?, -help, --help [@var{arg}]
-Show help. An optional parameter may be specified to print help about a specific
-item. If no argument is specified, only basic (non advanced) tool
-options are shown.
-
-Possible values of @var{arg} are:
-@table @option
-@item long
-Print advanced tool options in addition to the basic tool options.
-
-@item full
-Print complete list of options, including shared and private options
-for encoders, decoders, demuxers, muxers, filters, etc.
-
-@item decoder=@var{decoder_name}
-Print detailed information about the decoder named @var{decoder_name}. Use the
-@option{-decoders} option to get a list of all decoders.
-
-@item encoder=@var{encoder_name}
-Print detailed information about the encoder named @var{encoder_name}. Use the
-@option{-encoders} option to get a list of all encoders.
-
-@item demuxer=@var{demuxer_name}
-Print detailed information about the demuxer named @var{demuxer_name}. Use the
-@option{-formats} option to get a list of all demuxers and muxers.
-
-@item muxer=@var{muxer_name}
-Print detailed information about the muxer named @var{muxer_name}. Use the
-@option{-formats} option to get a list of all muxers and demuxers.
-
-@item filter=@var{filter_name}
-Print detailed information about the filter name @var{filter_name}. Use the
-@option{-filters} option to get a list of all filters.
-@end table
-
-@item -version
-Show version.
-
-@item -formats
-Show available formats (including devices).
-
-@item -devices
-Show available devices.
-
-@item -codecs
-Show all codecs known to libavcodec.
-
-Note that the term 'codec' is used throughout this documentation as a shortcut
-for what is more correctly called a media bitstream format.
-
-@item -decoders
-Show available decoders.
-
-@item -encoders
-Show all available encoders.
-
-@item -bsfs
-Show available bitstream filters.
-
-@item -protocols
-Show available protocols.
-
-@item -filters
-Show available libavfilter filters.
-
-@item -pix_fmts
-Show available pixel formats.
-
-@item -sample_fmts
-Show available sample formats.
-
-@item -layouts
-Show channel names and standard channel layouts.
-
-@item -colors
-Show recognized color names.
-
-@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sources of the intput device.
-Some devices may provide system-dependent source names that cannot be autodetected.
-The returned list cannot be assumed to be always complete.
-@example
-ffmpeg -sources pulse,server=192.168.0.4
-@end example
-
-@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sinks of the output device.
-Some devices may provide system-dependent sink names that cannot be autodetected.
-The returned list cannot be assumed to be always complete.
-@example
-ffmpeg -sinks pulse,server=192.168.0.4
-@end example
-
-@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
-Set the logging level used by the library.
-Adding "repeat+" indicates that repeated log output should not be compressed
-to the first line and the "Last message repeated n times" line will be
-omitted. "repeat" can also be used alone.
-If "repeat" is used alone, and with no prior loglevel set, the default
-loglevel will be used. If multiple loglevel parameters are given, using
-'repeat' will not change the loglevel.
-@var{loglevel} is a number or a string containing one of the following values:
-@table @samp
-@item quiet
-Show nothing at all; be silent.
-@item panic
-Only show fatal errors which could lead the process to crash, such as
-and assert failure. This is not currently used for anything.
-@item fatal
-Only show fatal errors. These are errors after which the process absolutely
-cannot continue after.
-@item error
-Show all errors, including ones which can be recovered from.
-@item warning
-Show all warnings and errors. Any message related to possibly
-incorrect or unexpected events will be shown.
-@item info
-Show informative messages during processing. This is in addition to
-warnings and errors. This is the default value.
-@item verbose
-Same as @code{info}, except more verbose.
-@item debug
-Show everything, including debugging information.
-@end table
-
-By default the program logs to stderr, if coloring is supported by the
-terminal, colors are used to mark errors and warnings. Log coloring
-can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
-the environment variable @env{AV_LOG_FORCE_COLOR}.
-The use of the environment variable @env{NO_COLOR} is deprecated and
-will be dropped in a following FFmpeg version.
-
-@item -report
-Dump full command line and console output to a file named
-@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
-directory.
-This file can be useful for bug reports.
-It also implies @code{-loglevel verbose}.
-
-Setting the environment variable @code{FFREPORT} to any value has the
-same effect. If the value is a ':'-separated key=value sequence, these
-options will affect the report; options values must be escaped if they
-contain special characters or the options delimiter ':' (see the
-``Quoting and escaping'' section in the ffmpeg-utils manual). The
-following option is recognized:
-@table @option
-@item file
-set the file name to use for the report; @code{%p} is expanded to the name
-of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
-to a plain @code{%}
-@item level
-set the log level
-@end table
-
-Errors in parsing the environment variable are not fatal, and will not
-appear in the report.
-
-@item -hide_banner
-Suppress printing banner.
-
-All FFmpeg tools will normally show a copyright notice, build options
-and library versions. This option can be used to suppress printing
-this information.
-
-@item -cpuflags flags (@emph{global})
-Allows setting and clearing cpu flags. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpuflags -sse+mmx ...
-ffmpeg -cpuflags mmx ...
-ffmpeg -cpuflags 0 ...
-@end example
-Possible flags for this option are:
-@table @samp
-@item x86
-@table @samp
-@item mmx
-@item mmxext
-@item sse
-@item sse2
-@item sse2slow
-@item sse3
-@item sse3slow
-@item ssse3
-@item atom
-@item sse4.1
-@item sse4.2
-@item avx
-@item avx2
-@item xop
-@item fma3
-@item fma4
-@item 3dnow
-@item 3dnowext
-@item bmi1
-@item bmi2
-@item cmov
-@end table
-@item ARM
-@table @samp
-@item armv5te
-@item armv6
-@item armv6t2
-@item vfp
-@item vfpv3
-@item neon
-@item setend
-@end table
-@item AArch64
-@table @samp
-@item armv8
-@item vfp
-@item neon
-@end table
-@item PowerPC
-@table @samp
-@item altivec
-@end table
-@item Specific Processors
-@table @samp
-@item pentium2
-@item pentium3
-@item pentium4
-@item k6
-@item k62
-@item athlon
-@item athlonxp
-@item k8
-@end table
-@end table
-
-@item -opencl_bench
-Benchmark all available OpenCL devices and show the results. This option
-is only available when FFmpeg has been compiled with @code{--enable-opencl}.
-
-@item -opencl_options options (@emph{global})
-Set OpenCL environment options. This option is only available when
-FFmpeg has been compiled with @code{--enable-opencl}.
-
-@var{options} must be a list of @var{key}=@var{value} option pairs
-separated by ':'. See the ``OpenCL Options'' section in the
-ffmpeg-utils manual for the list of supported options.
-@end table
-
-@section AVOptions
-
-These options are provided directly by the libavformat, libavdevice and
-libavcodec libraries. To see the list of available AVOptions, use the
-@option{-help} option. They are separated into two categories:
-@table @option
-@item generic
-These options can be set for any container, codec or device. Generic options
-are listed under AVFormatContext options for containers/devices and under
-AVCodecContext options for codecs.
-@item private
-These options are specific to the given container, device or codec. Private
-options are listed under their corresponding containers/devices/codecs.
-@end table
-
-For example to write an ID3v2.3 header instead of a default ID3v2.4 to
-an MP3 file, use the @option{id3v2_version} private option of the MP3
-muxer:
-@example
-ffmpeg -i input.flac -id3v2_version 3 out.mp3
-@end example
-
-All codec AVOptions are per-stream, and thus a stream specifier
-should be attached to them.
-
-Note: the @option{-nooption} syntax cannot be used for boolean
-AVOptions, use @option{-option 0}/@option{-option 1}.
-
-Note: the old undocumented way of specifying per-stream AVOptions by
-prepending v/a/s to the options name is now obsolete and will be
-removed soon.

doc/filter_design.txt

@@ -15,13 +15,13 @@ Format negotiation
   the list of supported formats.
 
   For video links, that means pixel format. For audio links, that means
-  channel layout, sample format (the sample packing is implied by the sample
-  format) and sample rate.
+  channel layout, and sample format (the sample packing is implied by the
+  sample format).
 
   The lists are not just lists, they are references to shared objects. When
   the negotiation mechanism computes the intersection of the formats
-  supported at each end of a link, all references to both lists are replaced
-  with a reference to the intersection. And when a single format is
+  supported at each ends of a link, all references to both lists are
+  replaced with a reference to the intersection. And when a single format is
   eventually chosen for a link amongst the remaining list, again, all
   references to the list are updated.
 
@@ -29,11 +29,6 @@ Format negotiation
   same format amongst a supported list, all it has to do is use a reference
   to the same list of formats.
 
-  query_formats can leave some formats unset and return AVERROR(EAGAIN) to
-  cause the negotiation mechanism to try again later. That can be used by
-  filters with complex requirements to use the format negotiated on one link
-  to set the formats supported on another.
-
 
 Buffer references ownership and permissions
 ===========================================
@@ -57,7 +52,7 @@ Buffer references ownership and permissions
     point to only a part of a video buffer.
 
     A reference is usually obtained as input to the start_frame or
-    filter_frame method or requested using the ff_get_video_buffer or
+    filter_samples method or requested using the ff_get_video_buffer or
     ff_get_audio_buffer functions. A new reference on an existing buffer can
     be created with the avfilter_ref_buffer. A reference is destroyed using
     the avfilter_unref_bufferp function.
@@ -73,15 +68,15 @@ Buffer references ownership and permissions
 
     Here are the (fairly obvious) rules for reference ownership:
 
-    * A reference received by the filter_frame method (or its start_frame
-      deprecated version) belongs to the corresponding filter.
+    * A reference received by the start_frame or filter_samples method
+      belong to the corresponding filter.
 
       Special exception: for video references: the reference may be used
       internally for automatic copying and must not be destroyed before
       end_frame; it can be given away to ff_start_frame.
 
-    * A reference passed to ff_filter_frame (or the deprecated
-      ff_start_frame) is given away and must no longer be used.
+    * A reference passed to ff_start_frame or ff_filter_samples is given
+      away and must no longer be used.
 
     * A reference created with avfilter_ref_buffer belongs to the code that
       created it.
@@ -95,11 +90,27 @@ Buffer references ownership and permissions
   Link reference fields
   ---------------------
 
-    The AVFilterLink structure has a few AVFilterBufferRef fields. The
-    cur_buf and out_buf were used with the deprecated
-    start_frame/draw_slice/end_frame API and should no longer be used.
-    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
-    and must not be accessed by filters.
+    The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
+    the rules to handle them:
+
+    * cur_buf is set before the start_frame and filter_samples methods to
+      the same reference given as argument to the methods and belongs to the
+      destination filter of the link. If it has not been cleared after
+      end_frame or filter_samples, libavfilter will automatically destroy
+      the reference; therefore, any filter that needs to keep the reference
+      for longer must set cur_buf to NULL.
+
+    * out_buf belongs to the source filter of the link and can be used to
+      store a reference to the buffer that has been sent to the destination.
+      If it is not NULL after end_frame or filter_samples, libavfilter will
+      automatically destroy the reference.
+
+      If a video input pad does not have a start_frame method, the default
+      method will request a buffer on the first output of the filter, store
+      the reference in out_buf and push a second reference to the output.
+
+    * src_buf, cur_buf_copy and partial_buf are used by libavfilter
+      internally and must not be accessed by filters.
 
   Reference permissions
   ---------------------
@@ -108,10 +119,8 @@ Buffer references ownership and permissions
     the code that owns the reference is allowed to do to the buffer data.
     Different references for the same buffer can have different permissions.
 
-    For video filters that implement the deprecated
-    start_frame/draw_slice/end_frame API, the permissions only apply to the
-    parts of the buffer that have already been covered by the draw_slice
-    method.
+    For video filters, the permissions only apply to the parts of the buffer
+    that have already been covered by the draw_slice method.
 
     The value is a binary OR of the following constants:
 
@@ -166,13 +175,13 @@ Buffer references ownership and permissions
       WRITE permission.
 
     * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and must request a buffer
+      scale) need the READ permission on input and and must request a buffer
       with the WRITE permission.
 
     * Filters that intend to keep a reference after the filtering process
-      is finished (after filter_frame returns) must have the PRESERVE
-      permission on it and remove the WRITE permission if they create a new
-      reference to give it away.
+      is finished (after end_frame or filter_samples returns) must have the
+      PRESERVE permission on it and remove the WRITE permission if they
+      create a new reference to give it away.
 
     * Filters that intend to modify a reference they have kept after the end
       of the filtering process need the REUSE2 permission and must remove
@@ -189,11 +198,11 @@ Frame scheduling
   Simple filters that output one frame for each input frame should not have
   to worry about it.
 
-  filter_frame
-  ------------
+  start_frame / filter_samples
+  ----------------------------
 
-    This method is called when a frame is pushed to the filter's input. It
-    can be called at any time except in a reentrant way.
+    These methods are called when a frame is pushed to the filter's input.
+    They can be called at any time except in a reentrant way.
 
     If the input frame is enough to produce output, then the filter should
     push the output frames on the output link immediately.
@@ -204,7 +213,7 @@ Frame scheduling
     filter; these buffered frames must be flushed immediately if a new input
     produces new output.
 
-    (Example: frame rate-doubling filter: filter_frame must (1) flush the
+    (Example: framerate-doubling filter: start_frame must (1) flush the
     second copy of the previous frame, if it is still there, (2) push the
     first copy of the incoming frame, (3) keep the second copy for later.)
 
@@ -224,8 +233,8 @@ Frame scheduling
 
     This method is called when a frame is wanted on an output.
 
-    For an input, it should directly call filter_frame on the corresponding
-    output.
+    For an input, it should directly call start_frame or filter_samples on
+    the corresponding output.
 
     For a filter, if there are queued frames already ready, one of these
     frames should be pushed. If not, the filter should request a frame on
@@ -246,7 +255,7 @@ Frame scheduling
         }
         while (!frame_pushed) {
             input = input_where_a_frame_is_most_needed();
-            ret = ff_request_frame(input);
+            ret = avfilter_request_frame(input);
             if (ret == AVERROR_EOF) {
                 process_eof_on_input();
             } else if (ret < 0) {
@@ -257,14 +266,4 @@ Frame scheduling
 
     Note that, except for filters that can have queued frames, request_frame
     does not push frames: it requests them to its input, and as a reaction,
-    the filter_frame method will be called and do the work.
-
-Legacy API
-==========
-
-  Until libavfilter 3.23, the filter_frame method was split:
-
-  - for video filters, it was made of start_frame, draw_slice (that could be
-    called several times on distinct parts of the frame) and end_frame;
-
-  - for audio filters, it was called filter_samples.
+    the start_frame / filter_samples method will be called and do the work.

doc/formats.texi

@@ -1,228 +0,0 @@
-@chapter Format Options
-@c man begin FORMAT OPTIONS
-
-The libavformat library provides some generic global options, which
-can be set on all the muxers and demuxers. In addition each muxer or
-demuxer may support so-called private options, which are specific for
-that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-The list of supported options follows:
-
-@table @option
-@item avioflags @var{flags} (@emph{input/output})
-Possible values:
-@table @samp
-@item direct
-Reduce buffering.
-@end table
-
-@item probesize @var{integer} (@emph{input})
-Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will enable detecting more
-information in case it is dispersed into the stream, but will increase
-latency. Must be an integer not lesser than 32. It is 5000000 by default.
-
-@item packetsize @var{integer} (@emph{output})
-Set packet size.
-
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
-
-Possible values:
-@table @samp
-@item ignidx
-Ignore index.
-@item genpts
-Generate PTS.
-@item nofillin
-Do not fill in missing values that can be exactly calculated.
-@item noparse
-Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
-@item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
-@item bitexact
-Only write platform-, build- and time-independent data.
-This ensures that file and data checksums are reproducible and match between
-platforms. Its primary use is for regression testing.
-@end table
-
-@item seek2any @var{integer} (@emph{input})
-Allow seeking to non-keyframes on demuxer level when supported if set to 1.
-Default is 0.
-
-@item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to probe the input. A
-higher value will enable detecting more accurate information, but will
-increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
-
-@item cryptokey @var{hexadecimal string} (@emph{input})
-Set decryption key.
-
-@item indexmem @var{integer} (@emph{input})
-Set max memory used for timestamp index (per stream).
-
-@item rtbufsize @var{integer} (@emph{input})
-Set max memory used for buffering real-time frames.
-
-@item fdebug @var{flags} (@emph{input/output})
-Print specific debug info.
-
-Possible values:
-@table @samp
-@item ts
-@end table
-
-@item max_delay @var{integer} (@emph{input/output})
-Set maximum muxing or demuxing delay in microseconds.
-
-@item fpsprobesize @var{integer} (@emph{input})
-Set number of frames used to probe fps.
-
-@item audio_preload @var{integer} (@emph{output})
-Set microseconds by which audio packets should be interleaved earlier.
-
-@item chunk_duration @var{integer} (@emph{output})
-Set microseconds for each chunk.
-
-@item chunk_size @var{integer} (@emph{output})
-Set size in bytes for each chunk.
-
-@item err_detect, f_err_detect @var{flags} (@emph{input})
-Set error detection flags. @code{f_err_detect} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item crccheck
-Verify embedded CRCs.
-@item bitstream
-Detect bitstream specification deviations.
-@item buffer
-Detect improper bitstream length.
-@item explode
-Abort decoding on minor error detection.
-@item careful
-Consider things that violate the spec and have not been seen in the
-wild as errors.
-@item compliant
-Consider all spec non compliancies as errors.
-@item aggressive
-Consider things that a sane encoder should not do as an error.
-@end table
-
-@item use_wallclock_as_timestamps @var{integer} (@emph{input})
-Use wallclock as timestamps.
-
-@item avoid_negative_ts @var{integer} (@emph{output})
-
-Possible values:
-@table @samp
-@item make_non_negative
-Shift timestamps to make them non-negative.
-Also note that this affects only leading negative timestamps, and not
-non-monotonic negative timestamps.
-@item make_zero
-Shift timestamps so that the first timestamp is 0.
-@item auto (default)
-Enables shifting when required by the target format.
-@item disabled
-Disables shifting of timestamp.
-@end table
-
-When shifting is enabled, all output timestamps are shifted by the
-same amount. Audio, video, and subtitles desynching and relative
-timestamp differences are preserved compared to how they would have
-been without shifting.
-
-@item skip_initial_bytes @var{integer} (@emph{input})
-Set number of bytes to skip before reading header and frames if set to 1.
-Default is 0.
-
-@item correct_ts_overflow @var{integer} (@emph{input})
-Correct single timestamp overflows if set to 1. Default is 1.
-
-@item flush_packets @var{integer} (@emph{output})
-Flush the underlying I/O stream after each packet. Default 1 enables it, and
-has the effect of reducing the latency; 0 disables it and may slightly
-increase performance in some cases.
-
-@item output_ts_offset @var{offset} (@emph{output})
-Set the output time offset.
-
-@var{offset} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-The offset is added by the muxer to the output timestamps.
-
-Specifying a positive offset means that the corresponding streams are
-delayed bt the time duration specified in @var{offset}. Default value
-is @code{0} (meaning that no offset is applied).
-
-@item format_whitelist @var{list} (@emph{input})
-"," separated List of allowed demuxers. By default all are allowed.
-
-@item dump_separator @var{string} (@emph{input})
-Separator used to separate the fields printed on the command line about the
-Stream parameters.
-For example to separate the fields with newlines and indention:
-@example
-ffprobe -dump_separator "
-                          "  -i ~/videos/matrixbench_mpeg2.mpg
-@end example
-@end table
-
-@c man end FORMAT OPTIONS
-
-@anchor{Format stream specifiers}
-@section Format stream specifiers
-
-Format stream specifiers allow selection of one or more streams that
-match specific properties.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index.
-
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio,
-'s' for subtitle, 'd' for data, and 't' for attachments. If
-@var{stream_index} is given, then it matches the stream number
-@var{stream_index} of this type. Otherwise, it matches all streams of
-this type.
-
-@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number
-@var{stream_index} in the program with the id
-@var{program_id}. Otherwise, it matches all streams in the program.
-
-@item #@var{stream_id}
-Matches the stream by a format-specific ID.
-@end table
-
-The exact semantics of stream specifiers is defined by the
-@code{avformat_match_stream_specifier()} function declared in the
-@file{libavformat/avformat.h} header.
-
-@ifclear config-writeonly
-@include demuxers.texi
-@end ifclear
-@ifclear config-readonly
-@include muxers.texi
-@end ifclear
-@include metadata.texi

doc/general.texi

@@ -94,20 +94,12 @@ Then pass @code{--enable-libtwolame} to configure to enable it.
 
 @section libvpx
 
-FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
+FFmpeg can make use of the libvpx library for VP8 encoding.
 
 Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
 enable it.
 
-@section libwavpack
-
-FFmpeg can make use of the libwavpack library for WavPack encoding.
-
-Go to @url{http://www.wavpack.com/} and follow the instructions for
-installing the library. Then pass @code{--enable-libwavpack} to configure to
-enable it.
-
 @section x264
 
 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -122,20 +114,6 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section x265
-
-FFmpeg can make use of the x265 library for HEVC encoding.
-
-Go to @url{http://x265.org/developers.html} and follow the instructions
-for installing the library. Then pass @code{--enable-libx265} to configure
-to enable it.
-
-@float NOTE
-x265 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
 @section libilbc
 
 iLBC is a narrowband speech codec that has been made freely available
@@ -147,41 +125,6 @@ Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.
 
-@section libzvbi
-
-libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
-teletext pages and DVB teletext subtitles.
-
-Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
-installing the library. Then pass @code{--enable-libzvbi} to configure to
-enable it.
-
-@float NOTE
-libzvbi is licensed under the GNU General Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
-you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
-@section AviSynth
-
-FFmpeg can read AviSynth scripts as input. To enable support, pass
-@code{--enable-avisynth} to configure.  The correct headers are
-included in compat/avisynth/, which allows the user to enable support
-without needing to search for these headers themselves.
-
-For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
-
-For Linux and OS X, the supported AviSynth variant is
-@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
-
-@float NOTE
-AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
-with @code{--enable-avisynth}, and the binaries will work regardless of the
-end user having AviSynth or AvxSynth installed - they'll only need to be
-installed to use AviSynth scripts (obviously).
-@end float
 
 
 @chapter Supported File Formats, Codecs or Features
@@ -205,21 +148,13 @@ library:
 @item American Laser Games MM   @tab   @tab X
     @tab Multimedia format used in games like Mad Dog McCree.
 @item 3GPP AMR                  @tab X @tab X
-@item Amazing Studio Packed Animation File  @tab   @tab X
+@item Amazing Studio Packed Animation File        @tab   @tab X
     @tab Multimedia format used in game Heart Of Darkness.
 @item Apple HTTP Live Streaming @tab   @tab X
 @item Artworx Data Format       @tab   @tab X
-@item ADP                       @tab   @tab X
-    @tab Audio format used on the Nintendo Gamecube.
-@item AFC                       @tab   @tab X
-    @tab Audio format used on the Nintendo Gamecube.
 @item ASF                       @tab X @tab X
-@item AST                       @tab X @tab X
-    @tab Audio format used on the Nintendo Wii.
 @item AVI                       @tab X @tab X
-@item AviSynth                  @tab   @tab X
-@item AVR                       @tab   @tab X
-    @tab Audio format used on Mac.
+@item AVISynth                  @tab   @tab X
 @item AVS                       @tab   @tab X
     @tab Multimedia format used by the Creature Shock game.
 @item Beam Software SIFF        @tab   @tab X
@@ -233,8 +168,6 @@ library:
     @tab Used in Z and Z95 games.
 @item Brute Force & Ignorance   @tab   @tab X
     @tab Used in the game Flash Traffic: City of Angels.
-@item BRSTM                     @tab   @tab X
-    @tab Audio format used on the Nintendo Wii.
 @item BWF                       @tab X @tab X
 @item CRI ADX                   @tab X @tab X
     @tab Audio-only format used in console video games.
@@ -245,7 +178,6 @@ library:
     @tab Multimedia format used by Delphine Software games.
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
-@item Phantom Cine              @tab   @tab X
 @item Commodore CDXL            @tab   @tab X
     @tab Amiga CD video format
 @item Core Audio Format         @tab X @tab X
@@ -259,7 +191,6 @@ library:
 @item Deluxe Paint Animation    @tab   @tab X
 @item DFA                       @tab   @tab X
     @tab This format is used in Chronomaster game
-@item DSD Stream File (DSF)     @tab   @tab X
 @item DV video                  @tab X @tab X
 @item DXA                       @tab   @tab X
     @tab This format is used in the non-Windows version of the Feeble Files
@@ -267,7 +198,6 @@ library:
 @item Electronic Arts cdata  @tab    @tab X
 @item Electronic Arts Multimedia  @tab    @tab X
     @tab Used in various EA games; files have extensions like WVE and UV2.
-@item Ensoniq Paris Audio File  @tab   @tab X
 @item FFM (FFserver live feed)  @tab X @tab X
 @item Flash (SWF)               @tab X @tab X
 @item Flash 9 (AVM2)            @tab X @tab X
@@ -282,12 +212,10 @@ library:
 @item G.723.1                   @tab X @tab X
 @item G.729 BIT                 @tab X @tab X
 @item G.729 raw                 @tab   @tab X
-@item GIF Animation             @tab X @tab X
+@item GIF Animation             @tab X @tab
 @item GXF                       @tab X @tab X
     @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
          playout servers.
-@item HNM @tab   @tab X
-    @tab Only version 4 supported, used in some games from Cryo Interactive
 @item iCEDraw File              @tab   @tab X
 @item ICO                       @tab X @tab X
     @tab Microsoft Windows ICO
@@ -304,17 +232,13 @@ library:
     @tab A format generated by IndigoVision 8000 video server.
 @item IVF (On2)                 @tab X @tab X
     @tab A format used by libvpx
-@item IRCAM                     @tab X @tab X
 @item LATM                      @tab X @tab X
 @item LMLM4                     @tab   @tab X
     @tab Used by Linux Media Labs MPEG-4 PCI boards
 @item LOAS                      @tab   @tab X
     @tab contains LATM multiplexed AAC audio
-@item LRC                       @tab X @tab X
-@item LVF                       @tab   @tab X
 @item LXF                       @tab   @tab X
     @tab VR native stream format, used by Leitch/Harris' video servers.
-@item Magic Lantern Video (MLV) @tab   @tab X
 @item Matroska                  @tab X @tab X
 @item Matroska audio            @tab X @tab
 @item FFmpeg metadata           @tab X @tab X
@@ -323,8 +247,6 @@ library:
     @tab Used in Sim City 3000; file extension .xa.
 @item MD Studio                 @tab   @tab X
 @item Metal Gear Solid: The Twin Snakes @tab @tab X
-@item Megalux Frame             @tab   @tab X
-    @tab Used by Megalux Ultimate Paint
 @item Mobotix .mxg              @tab   @tab X
 @item Monkey's Audio            @tab   @tab X
 @item Motion Pixels MVI         @tab   @tab X
@@ -340,8 +262,6 @@ library:
     @tab also known as DVB Transport Stream
 @item MPEG-4                    @tab X @tab X
     @tab MPEG-4 is a variant of QuickTime.
-@item Mirillis FIC video        @tab   @tab X
-    @tab No cursor rendering.
 @item MIME multipart JPEG       @tab X @tab
 @item MSN TCP webcam            @tab   @tab X
     @tab Used by MSN Messenger webcam streams.
@@ -354,7 +274,6 @@ library:
     @tab SMPTE 386M, D-10/IMX Mapping.
 @item NC camera feed            @tab   @tab X
     @tab NC (AVIP NC4600) camera streams
-@item NIST SPeech HEader REsources @tab   @tab X
 @item NTT TwinVQ (VQF)          @tab   @tab X
     @tab Nippon Telegraph and Telephone Corporation TwinVQ.
 @item Nullsoft Streaming Video  @tab   @tab X
@@ -363,7 +282,6 @@ library:
     @tab NUT Open Container Format
 @item Ogg                       @tab X @tab X
 @item Playstation Portable PMP  @tab   @tab X
-@item Portable Voice Format     @tab   @tab X
 @item TechnoTrend PVA           @tab   @tab X
     @tab Used by TechnoTrend DVB PCI boards.
 @item QCP                       @tab   @tab X
@@ -374,14 +292,12 @@ library:
 @item raw Dirac                 @tab X @tab X
 @item raw DNxHD                 @tab X @tab X
 @item raw DTS                   @tab X @tab X
-@item raw DTS-HD                @tab   @tab X
 @item raw E-AC-3                @tab X @tab X
 @item raw FLAC                  @tab X @tab X
 @item raw GSM                   @tab   @tab X
 @item raw H.261                 @tab X @tab X
 @item raw H.263                 @tab X @tab X
 @item raw H.264                 @tab X @tab X
-@item raw HEVC                  @tab X @tab X
 @item raw Ingenient MJPEG       @tab   @tab X
 @item raw MJPEG                 @tab X @tab X
 @item raw MLP                   @tab   @tab X
@@ -393,9 +309,8 @@ library:
 @item raw video                 @tab X @tab X
 @item raw id RoQ                @tab X @tab
 @item raw Shorten               @tab   @tab X
-@item raw TAK                   @tab   @tab X
 @item raw TrueHD                @tab X @tab X
-@item raw VC-1                  @tab X @tab X
+@item raw VC-1                  @tab   @tab X
 @item raw PCM A-law             @tab X @tab X
 @item raw PCM mu-law            @tab X @tab X
 @item raw PCM signed 8 bit      @tab X @tab X
@@ -421,13 +336,11 @@ library:
     @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
 @item RealMedia                 @tab X @tab X
 @item Redirector                @tab   @tab X
-@item RedSpark                  @tab   @tab X
 @item Renderware TeXture Dictionary @tab   @tab X
 @item RL2                       @tab   @tab X
     @tab Audio and video format used in some games by Entertainment Software Partners.
 @item RPL/ARMovie               @tab   @tab X
 @item Lego Mindstorms RSO       @tab X @tab X
-@item RSD                       @tab   @tab X
 @item RTMP                      @tab X @tab X
     @tab Output is performed by publishing stream to RTMP server
 @item RTP                       @tab X @tab X
@@ -437,7 +350,6 @@ library:
 @item SDP                       @tab   @tab X
 @item Sega FILM/CPK             @tab   @tab X
     @tab Used in many Sega Saturn console games.
-@item Silicon Graphics Movie    @tab   @tab X
 @item Sierra SOL                @tab   @tab X
     @tab .sol files used in Sierra Online games.
 @item Sierra VMD                @tab   @tab X
@@ -451,10 +363,9 @@ library:
 @item Sony OpenMG (OMA)         @tab X @tab X
     @tab Audio format used in Sony Sonic Stage and Sony Vegas.
 @item Sony PlayStation STR      @tab   @tab X
-@item Sony Wave64 (W64)         @tab X @tab X
+@item Sony Wave64 (W64)         @tab   @tab X
 @item SoX native format         @tab X @tab X
 @item SUN AU format             @tab X @tab X
-@item SUP raw PGS subtitles     @tab   @tab X
 @item Text files                @tab   @tab X
 @item THP                       @tab   @tab X
     @tab Used on the Nintendo GameCube.
@@ -462,7 +373,6 @@ library:
     @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
 @item True Audio                @tab   @tab X
 @item VC-1 test bitstream       @tab X @tab X
-@item Vivo                      @tab   @tab X
 @item WAV                       @tab X @tab X
 @item WavPack                   @tab X @tab X
 @item WebM                      @tab X @tab X
@@ -493,13 +403,10 @@ following image formats are supported:
 @item Name @tab Encoding @tab Decoding @tab Comments
 @item .Y.U.V       @tab X @tab X
     @tab one raw file per component
-@item Alias PIX    @tab X @tab X
-    @tab Alias/Wavefront PIX image format
 @item animated GIF @tab X @tab X
+    @tab Only uncompressed GIFs are generated.
 @item BMP          @tab X @tab X
     @tab Microsoft BMP image
-@item BRender PIX  @tab   @tab X
-    @tab Argonaut BRender 3D engine image format.
 @item DPX          @tab X @tab X
     @tab Digital Picture Exchange
 @item EXR          @tab   @tab X
@@ -535,12 +442,8 @@ following image formats are supported:
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item WebP         @tab E @tab X
-    @tab WebP image format, encoding supported through external library libwebp
 @item XBM  @tab X @tab X
     @tab X BitMap image format
-@item XFace @tab X @tab X
-    @tab X-Face image format
 @item XWD  @tab X @tab X
     @tab X Window Dump image format
 @end multitable
@@ -556,6 +459,8 @@ following image formats are supported:
 @item 4X Movie               @tab     @tab  X
     @tab Used in certain computer games.
 @item 8088flex TMV           @tab     @tab  X
+@item 8SVX exponential       @tab     @tab  X
+@item 8SVX fibonacci         @tab     @tab  X
 @item A64 multicolor         @tab  X  @tab
     @tab Creates video suitable to be played on a commodore 64 (multicolor mode).
 @item Amazing Studio PAF Video @tab     @tab  X
@@ -564,7 +469,6 @@ following image formats are supported:
 @item AMV Video              @tab  X  @tab  X
     @tab Used in Chinese MP3 players.
 @item ANSI/ASCII art         @tab     @tab  X
-@item Apple Intermediate Codec @tab     @tab  X
 @item Apple MJPEG-B          @tab     @tab  X
 @item Apple ProRes           @tab  X  @tab  X
 @item Apple QuickDraw        @tab     @tab  X
@@ -647,16 +551,12 @@ following image formats are supported:
     @tab Sorenson H.263 used in Flash
 @item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
-@item Go2Webinar             @tab     @tab  X
-    @tab fourcc: G2M4
 @item H.261                  @tab  X  @tab  X
 @item H.263 / H.263-1996     @tab  X  @tab  X
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
     @tab encoding supported through external library libx264
-@item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through the external library libx265
-@item HNM version 4          @tab     @tab  X
+@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration)  @tab  E  @tab  X
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
 @item IBM Ultimotion         @tab     @tab  X
@@ -687,8 +587,8 @@ following image formats are supported:
 @item LCL (LossLess Codec Library) MSZH  @tab     @tab  X
 @item LCL (LossLess Codec Library) ZLIB  @tab  E  @tab  E
 @item LOCO                   @tab     @tab  X
-@item LucasArts SANM/Smush   @tab     @tab  X
-    @tab Used in LucasArts games / SMUSH animations.
+@item LucasArts Smush        @tab     @tab  X
+    @tab Used in LucasArts games.
 @item lossless MJPEG         @tab  X  @tab  X
 @item Microsoft ATC Screen   @tab     @tab  X
     @tab Also known as Microsoft Screen 3.
@@ -708,6 +608,8 @@ following image formats are supported:
 @item Mobotix MxPEG video    @tab     @tab  X
 @item Motion Pixels video    @tab     @tab  X
 @item MPEG-1 video           @tab  X  @tab  X
+@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
+@item MPEG-1/2 video (VDPAU acceleration)  @tab     @tab  X
 @item MPEG-2 video           @tab  X  @tab  X
 @item MPEG-4 part 2          @tab  X  @tab  X
     @tab libxvidcore can be used alternatively for encoding.
@@ -723,14 +625,8 @@ following image formats are supported:
     @tab fourcc: VP50
 @item On2 VP6                @tab     @tab  X
     @tab fourcc: VP60,VP61,VP62
-@item On2 VP7                @tab     @tab  X
-    @tab fourcc: VP70,VP71
 @item VP8                    @tab  E  @tab  X
     @tab fourcc: VP80, encoding supported through external library libvpx
-@item VP9                    @tab  E  @tab  X
-    @tab encoding supported through external library libvpx
-@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
-    @tab fourcc: Y216
 @item Prores                 @tab     @tab  X
     @tab fourcc: apch,apcn,apcs,apco
 @item Q-team QPEG            @tab     @tab  X
@@ -756,9 +652,6 @@ following image formats are supported:
     @tab used in some games by Entertainment Software Partners
 @item Sierra VMD video       @tab     @tab  X
     @tab Used in Sierra VMD files.
-@item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
-@item Silicon Graphics Motion Video Compressor 2 (MVC2)  @tab     @tab  X
-@item Silicon Graphics RLE 8-bit video  @tab     @tab  X
 @item Smacker video          @tab     @tab  X
     @tab Video encoding used in Smacker.
 @item SMPTE VC-1             @tab     @tab  X
@@ -818,13 +711,12 @@ following image formats are supported:
 
 @multitable @columnfractions .4 .1 .1 .4
 @item Name @tab Encoding @tab Decoding @tab Comments
-@item 8SVX exponential       @tab     @tab  X
-@item 8SVX fibonacci         @tab     @tab  X
+@item 8SVX audio             @tab     @tab  X
 @item AAC+                   @tab  E  @tab  X
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
     @tab encoding supported through external library libfaac and libvo-aacenc
-@item AC-3                   @tab IX  @tab  IX
+@item AC-3                   @tab IX  @tab  X
 @item ADPCM 4X Movie         @tab     @tab  X
 @item ADPCM CDROM XA         @tab     @tab  X
 @item ADPCM Creative Technology @tab     @tab  X
@@ -850,26 +742,22 @@ following image formats are supported:
 @item ADPCM IMA Westwood     @tab     @tab  X
 @item ADPCM ISS IMA          @tab     @tab  X
     @tab Used in FunCom games.
-@item ADPCM IMA Dialogic     @tab     @tab  X
 @item ADPCM IMA Duck DK3     @tab     @tab  X
     @tab Used in some Sega Saturn console games.
 @item ADPCM IMA Duck DK4     @tab     @tab  X
     @tab Used in some Sega Saturn console games.
-@item ADPCM IMA Radical      @tab     @tab  X
 @item ADPCM Microsoft        @tab  X  @tab  X
 @item ADPCM MS IMA           @tab  X  @tab  X
-@item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
-@item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
 @item ADPCM Nintendo Gamecube THP  @tab     @tab  X
 @item ADPCM QT IMA           @tab  X  @tab  X
 @item ADPCM SEGA CRI ADX     @tab  X  @tab  X
     @tab Used in Sega Dreamcast games.
 @item ADPCM Shockwave Flash  @tab  X  @tab  X
+@item ADPCM SMJPEG IMA       @tab     @tab  X
+    @tab Used in certain Loki game ports.
 @item ADPCM Sound Blaster Pro 2-bit  @tab     @tab  X
 @item ADPCM Sound Blaster Pro 2.6-bit  @tab     @tab  X
 @item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
-@item ADPCM VIMA
-    @tab Used in LucasArts SMUSH animations.
 @item ADPCM Westwood Studios IMA @tab     @tab  X
     @tab Used in Westwood Studios games like Command and Conquer.
 @item ADPCM Yamaha           @tab  X  @tab  X
@@ -880,9 +768,8 @@ following image formats are supported:
 @item Amazing Studio PAF Audio @tab     @tab  X
 @item Apple lossless audio   @tab  X  @tab  X
     @tab QuickTime fourcc 'alac'
-@item ATRAC1                 @tab     @tab  X
-@item ATRAC3                 @tab     @tab  X
-@item ATRAC3+                @tab     @tab  X
+@item Atrac 1                @tab     @tab  X
+@item Atrac 3                @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
@@ -902,14 +789,9 @@ following image formats are supported:
 @item DPCM Sol               @tab     @tab  X
 @item DPCM Xan               @tab     @tab  X
     @tab Used in Origin's Wing Commander IV AVI files.
-@item DSD (Direct Stream Digitial), least significant bit first  @tab  @tab  X
-@item DSD (Direct Stream Digitial), most significant bit first   @tab  @tab  X
-@item DSD (Direct Stream Digitial), least significant bit first, planar  @tab  @tab  X
-@item DSD (Direct Stream Digitial), most significant bit first, planar   @tab  @tab  X
 @item DSP Group TrueSpeech   @tab     @tab  X
 @item DV audio               @tab     @tab  X
 @item Enhanced AC-3          @tab  X  @tab  X
-@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
 @item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
 @item G.723.1                @tab X @tab X
 @item G.729                  @tab     @tab  X
@@ -926,25 +808,19 @@ following image formats are supported:
 @item MLP (Meridian Lossless Packing)  @tab     @tab  X
     @tab Used in DVD-Audio discs.
 @item Monkey's Audio         @tab     @tab  X
+    @tab Only versions 3.97-3.99 are supported.
 @item MP1 (MPEG audio layer 1)  @tab     @tab IX
 @item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
-    @tab encoding supported also through external library TwoLAME
+    @tab libtwolame can be used alternatively for encoding.
 @item MP3 (MPEG audio layer 3)  @tab  E  @tab IX
     @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
 @item MPEG-4 Audio Lossless Coding (ALS)  @tab     @tab  X
 @item Musepack SV7           @tab     @tab  X
 @item Musepack SV8           @tab     @tab  X
 @item Nellymoser Asao        @tab  X  @tab  X
-@item On2 AVC (Audio for Video Codec) @tab     @tab  X
-@item Opus                   @tab  E  @tab  X
-    @tab encoding supported through external library libopus
 @item PCM A-law              @tab  X  @tab  X
 @item PCM mu-law             @tab  X  @tab  X
-@item PCM signed 8-bit planar  @tab  X  @tab  X
-@item PCM signed 16-bit big-endian planar  @tab  X  @tab  X
-@item PCM signed 16-bit little-endian planar  @tab  X  @tab  X
-@item PCM signed 24-bit little-endian planar  @tab  X  @tab  X
-@item PCM signed 32-bit little-endian planar  @tab  X  @tab  X
+@item PCM 16-bit little-endian planar  @tab     @tab  X
 @item PCM 32-bit floating point big-endian  @tab  X  @tab  X
 @item PCM 32-bit floating point little-endian  @tab  X  @tab  X
 @item PCM 64-bit floating point big-endian  @tab  X  @tab  X
@@ -981,15 +857,14 @@ following image formats are supported:
 @item Sierra VMD audio       @tab     @tab  X
     @tab Used in Sierra VMD files.
 @item Smacker audio          @tab     @tab  X
-@item SMPTE 302M AES3 audio  @tab  X  @tab  X
+@item SMPTE 302M AES3 audio  @tab     @tab  X
 @item Sonic                  @tab  X  @tab  X
     @tab experimental codec
 @item Sonic lossless         @tab  X  @tab  X
     @tab experimental codec
 @item Speex                  @tab  E  @tab  E
     @tab supported through external library libspeex
-@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
-@item True Audio (TTA)       @tab  X  @tab  X
+@item True Audio (TTA)       @tab     @tab  X
 @item TrueHD                 @tab     @tab  X
     @tab Used in HD-DVD and Blu-Ray discs.
 @item TwinVQ (VQF flavor)    @tab     @tab  X
@@ -997,8 +872,7 @@ following image formats are supported:
     @tab Used in LucasArts SMUSH animations.
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
-@item Voxware MetaSound      @tab     @tab  X
-@item WavPack                @tab  X  @tab  X
+@item WavPack                @tab     @tab  X
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
 @item Windows Media Audio 2  @tab  X  @tab  X
@@ -1018,46 +892,32 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1 .1 .1 .1
 @item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
-@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
-@item AQTitle          @tab   @tab X @tab   @tab X
+@item SSA/ASS          @tab X @tab X @tab X @tab X
 @item DVB              @tab X @tab X @tab X @tab X
-@item DVB teletext     @tab   @tab X @tab   @tab E
 @item DVD              @tab X @tab X @tab X @tab X
 @item JACOsub          @tab X @tab X @tab   @tab X
 @item MicroDVD         @tab X @tab X @tab   @tab X
-@item MPL2             @tab   @tab X @tab   @tab X
-@item MPsub (MPlayer)  @tab   @tab X @tab   @tab X
 @item PGS              @tab   @tab   @tab   @tab X
-@item PJS (Phoenix)    @tab   @tab X @tab   @tab X
 @item RealText         @tab   @tab X @tab   @tab X
 @item SAMI             @tab   @tab X @tab   @tab X
-@item Spruce format (STL) @tab   @tab X @tab   @tab X
-@item SSA/ASS          @tab X @tab X @tab X @tab X
 @item SubRip (SRT)     @tab X @tab X @tab X @tab X
-@item SubViewer v1     @tab   @tab X @tab   @tab X
 @item SubViewer        @tab   @tab X @tab   @tab X
-@item TED Talks captions @tab @tab X @tab   @tab X
-@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
-@item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab X @tab X @tab X @tab X
+@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
+@item WebVTT           @tab   @tab X @tab   @tab X
 @item XSUB             @tab   @tab   @tab X @tab X
 @end multitable
 
 @code{X} means that the feature is supported.
 
-@code{E} means that support is provided through an external library.
-
 @section Network Protocols
 
 @multitable @columnfractions .4 .1
 @item Name         @tab Support
 @item file         @tab X
-@item FTP          @tab X
 @item Gopher       @tab X
 @item HLS          @tab X
 @item HTTP         @tab X
 @item HTTPS        @tab X
-@item Icecast      @tab X
 @item MMSH         @tab X
 @item MMST         @tab X
 @item pipe         @tab X
@@ -1068,9 +928,7 @@ performance on systems without hardware floating point support).
 @item RTMPTE       @tab X
 @item RTMPTS       @tab X
 @item RTP          @tab X
-@item SAMBA        @tab E
 @item SCTP         @tab X
-@item SFTP         @tab E
 @item TCP          @tab X
 @item TLS          @tab X
 @item UDP          @tab X
@@ -1090,19 +948,17 @@ performance on systems without hardware floating point support).
 @item caca              @tab        @tab X
 @item DV1394            @tab X      @tab
 @item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab X
+@item Linux framebuffer @tab X      @tab
 @item JACK              @tab X      @tab
 @item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
 @item OpenAL            @tab X
-@item OpenGL            @tab        @tab X
 @item OSS               @tab X      @tab X
-@item PulseAudio        @tab X      @tab X
+@item Pulseaudio        @tab X      @tab
 @item SDL               @tab        @tab X
-@item Video4Linux2      @tab X      @tab X
+@item Video4Linux2      @tab X      @tab
 @item VfW capture       @tab X      @tab
 @item X11 grabbing      @tab X      @tab
-@item Win32 grabbing    @tab X      @tab
 @end multitable
 
 @code{X} means that input/output is supported.