Update for 2.6.6

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

.gitignore

@@ -28,7 +28,6 @@
 /ffserver
 /config.*
 /coverage.info
-/avversion.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -37,7 +36,6 @@
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
-/doc/examples/avio_dir_cmd
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
@@ -63,7 +61,6 @@
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
-/tests/checkasm/checkasm
 /tests/data/
 /tests/pixfmts.mak
 /tests/rotozoom

Changelog

@@ -1,74 +1,410 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 2.8:
-- colorkey video filter
-- BFSTM/BCSTM demuxer
-- little-endian ADPCM_THP decoder
-- Hap decoder and encoder
-- DirectDraw Surface image/texture decoder
-- ssim filter
-- optional new ASF demuxer
-- showvolume filter
-- Many improvements to the JPEG 2000 decoder
-- Go2Meeting decoding support
-- adrawgraph audio and drawgraph video filter
-- removegrain video filter
-- Intel QSV-accelerated MPEG-2 video and HEVC encoding
-- Intel QSV-accelerated MPEG-2 video and HEVC decoding
-- Intel QSV-accelerated VC-1 video decoding
-- libkvazaar HEVC encoder
-- erosion, dilation, deflate and inflate video filters
-- Dynamic Audio Normalizer as dynaudnorm filter
-- Reverse video and areverse audio filter
-- Random filter
-- deband filter
-- AAC fixed-point decoding
-- sidechaincompress audio filter
-- bitstream filter for converting HEVC from MP4 to Annex B
-- acrossfade audio filter
-- allyuv and allrgb video sources
-- atadenoise video filter
-- OS X VideoToolbox support
-- aphasemeter filter
-- showfreqs filter
-- vectorscope filter
-- waveform filter
-- hstack and vstack filter
-- Support DNx100 (1440x1080@8)
-- VAAPI hevc hwaccel
-- VDPAU hevc hwaccel
-- framerate filter
-- Switched default encoders for webm to VP9 and Opus
-- Removed experimental flag from the JPEG 2000 encoder
+
+version 2.6.6:
+- 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
+- avcodec/mpeg4videodec: also for empty partitioned slices
+- 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
+- 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: 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()
+- 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/h264_slice: Limit max_contexts when slice_context_count is initialized
+- avcodec/vp8: Do not use num_coeff_partitions in thread/buffer setup
+- avcodec/mpeg4videodec: Check available data before reading custom matrix
+- mjpegdec: consider chroma subsampling in size check
+- avcodec/dirac_parser: Check that there is a previous PU before accessing it
+- 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
+- golomb: always check for invalid UE golomb codes in get_ue_golomb
+- 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
+- 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
+- mxfdec: check edit_rate also for physical_track
+- mpegvideo: clear overread in clear_context
+- dvdsubdec: validate offset2 similar to offset1
+- aacdec: don't return frames without data from aac_decode_er_frame
+- avcodec/takdec: Use memove, avoid undefined memcpy() use
+- riffdec: prevent negative bit rate
+- Merge commit 'd80811c94e068085aab797f9ba35790529126f85'
+- 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.
 
 
-version 2.7:
-- FFT video filter
-- TDSC decoder
-- DTS lossless extension (XLL) decoding (not lossless, disabled by default)
-- showwavespic filter
-- DTS decoding through libdcadec
-- Drop support for nvenc API before 5.0
-- nvenc HEVC encoder
-- Detelecine filter
-- Intel QSV-accelerated H.264 encoding
-- MMAL-accelerated H.264 decoding
-- basic APNG encoder and muxer with default extension "apng"
-- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
-- WebM Live Chunk Muxer
-- nvenc level and tier options
-- chorus filter
-- Canopus HQ/HQA decoder
-- Automatically rotate videos based on metadata in ffmpeg
-- improved Quickdraw compatibility
-- VP9 high bit-depth and extended colorspaces decoding support
-- WebPAnimEncoder API when available for encoding and muxing WebP
-- Direct3D11-accelerated decoding
-- Support Secure Transport
-- Multipart JPEG demuxer
+version 2.6.5:
+- 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
+- doc/ffmpeg: Clarify that the sdp_file option requires an rtp output.
+- ffmpeg: Don't try and write sdp info if none of the outputs had an rtp format.
+- jvdec: avoid unsigned overflow in comparison
+- avcodec/jpeg2000dec: Clip all tile coordinates
+- avcodec/hevc_ps: Check chroma_format_idc
+- 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
+- avcodec/mpeg12dec: Do not call show_bits() with invalid bits
+- 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
+- avcodec/vp3: 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()
+- rawdec: fix mjpeg probing buffer size check
+- rawdec: fix mjpeg probing
+- videodsp: don't overread edges in vfix3 emu_edge.
+- lavf/matroskadec: Fully parse and repack MP3 packets
+- 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 2.6.4:
+- imc: use correct position for flcoeffs2 calculation
+- hevc: check slice address length
+- snow: remove an obsolete av_assert2
+- webp: fix infinite loop in webp_decode_frame
+- wavpack: limit extra_bits to 32 and use get_bits_long
+- ffmpeg: only count got_output/errors in decode_error_stat
+- ffmpeg: exit_on_error if decoding a packet failed
+- pthread_frame: forward error codes when flushing
+- huffyuvdec: validate image size
+- wavpack: use get_bits_long to read up to 32 bits
+- nutdec: check maxpos in read_sm_data before returning success
+- vc1dec: use get_bits_long and limit the read bits to 32
+- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
+- 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()
+- ffmpeg: Fix cleanup after failed allocation of output_files
+- 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
+- examples/demuxing_decoding: use properties from frame instead of video_dec_ctx
+- 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
+- avfilter/x86/vf_fspp: Fix invalid combination of opcode and operands
+- 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
+- nutdec: abort if EOF is reached in decode_info_header/read_sm_data
+- nutdec: stop skipping bytes at EOF
+- nutdec: fix infinite resync loops
+- avformat/nutdec: Check X in 2nd branch of index reading
+- avformat/nutdec: Fix recovery when immedeately after seeking a failure happens
+- avformat/nutdec: Return error on EOF from get_str()
+- rtsp: Make sure we don't write too many transport entries into a fixed-size array
+- rtpenc_jpeg: handle case of picture dimensions not dividing by 8
+- avcodec/golomb: get_ur_golomb_jpegls: Fix reading huge k values
+- avformat/swfdec: Do not error out on pixel format changes
+- avformat/mov: Mark avio context of decompressed atoms as seekable
+- avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer()
+- avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out
+- avfilter/x86/vf_hqdn3d: Fix register types
+- avcodec/exr: fix crash caused by merge
+- avcodec/x86/h264_weight: handle weight1=128
+- avcodec/dvbsubdec: Fix buf_size check in dvbsub_parse_display_definition_segment()
+- avcodec/hevc_ps: Only discard overread VPS if a previous is available
+- avcodec/flacenc: Fix Invalid Rice order
+- lavd/xcbgrab: fix comparison with screen size.
+
+version 2.6.3:
+- avcodec/libtheoraenc: Check for av_malloc failure
+- ffmpeg_opt: Fix -timestamp parsing
+- hevc: make avcodec_decode_video2() fail if get_format() fails
+- avcodec/cavsdec: Use ff_set_dimensions()
+- swr: fix alignment issue caused by 8ch sse functions
+- avcodec/mjpegdec: fix len computation in ff_mjpeg_decode_dqt()
+- avcodec/jpeg2000dec: fix boolean operator
+- avcodec/hevc_ps: Explicitly check num_tile_* for negative values
+- 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
+- avformat/matroskaenc: Check ff_vorbiscomment_length in put_flac_codecpriv()
+- avcodec/mpeg12dec: use the correct dimensions for checking SAR
+- 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
+- avformat/mov: Do not read ACLR into extradata for H.264
+- ffmpeg: remove incorrect network deinit
+- OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c
+- libvpxenc: only set noise reduction w/vp8
+- vp9: remove another optimization branch in iadst16 which causes overflows.
+- lavf: Reset global flag on deinit
+- network: Do not leave context locked on error
+- vp9: remove one optimization branch in iadst16 which causes overflows.
+- fate: Include branch information in the payload header
+- avformat/utils: Ensure that AVFMT_FLAG_CUSTOM_IO is set before use
+- avformat/img2dec: do not rewind custom io buffers
+- avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data
+- avcodec/atrac3plusdsp: fix on stack alignment
+- swresample/swresample-test: Randomly wipe out channel counts
+- swresample: Check channel layouts and channels against each other and print human readable error messages
+- swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07)
+- swresample: Allow reinitialization without ever setting channel counts
+- dashenc: replace attribute id with contentType for the AdaptationSet element
+- avformat/matroskaenc: Use avoid_negative_ts_use_pts if no stream writes dts
+- avformat/mux: Add avoid_negative_ts_use_pts
+- tests/fate-run: do not attempt to parse tiny_psnrs output if it failed
+- cafdec: free extradata before allocating it
+- imgutils: initialize palette padding bytes in av_image_alloc
+- aacdec: don't return frames without data
+- id3v2: catch avio_read errors in check_tag
+- avi: Validate sample_size
+- 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
+- avformat/matroskadec: Use tracks[k]->stream instead of s->streams[k]
+- matroskadec: use uint64_t instead of int for index_scale
+- pngdec: don't use AV_PIX_FMT_MONOBLACK for apng
+- pngdec: return correct error code from decode_frame_common
+- nutdec: fix illegal count check in decode_main_header
+- nutdec: fix memleaks on error in nut_read_header
+- apedec: prevent out of array writes in decode_array_0000
+- 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() it was wrong
+- alac: reject rice_limit 0 if compression is used
+- alsdec: only adapt order for positive max_order
+- bink: check vst->index_entries before using it
+- mpeg4videodec: only allow a positive length
+- aacpsy: correct calculation of minath in psy_3gpp_init
+- alsdec: validate time diff index
+- alsdec: ensure channel reordering is reversible
+- 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
+- matroskadec: export cover art correctly
+- mxfenc: don't try to write footer without header
+- mxfenc: fix memleaks in mxf_write_footer
+- rtpenc_mpegts: Set chain->rtp_ctx only after avformat_write_header succeeded
+- rtpenc_mpegts: Free the right ->pb in the error path in the init function
+
+version 2.6.2:
+- 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
+- avformat/utils: avoid discarded streams in av_find_default_stream_index()
+- ffmpeg: Fix extradata allocation
+- 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
+- Revert "avcodec/exr: fix memset first arg in reverse_lut()"
+- h264: avoid unnecessary calls to get_format
+- avutil/pca: Check for av_malloc* failures
+- avutil/cpu: add missing check for mmxext to av_force_cpu_flags
+- lavc/dnxhd: Fix pix_fmt change.
+- avformat/http: replace cookies with updated values instead of appending forever
+- avformat/hls: store cookies returned in HLS key response
+- avformat/rmdec: fix support for 0 sized mdpr
+- avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
+- avcodec/hevc_ps: Check cropping parameters more correctly
+- hevc: make the crop sizes unsigned
+- avcodec/dnxhddec: Reset is_444 if format is not 444
+- avcodec/dnxhddec: Check that the frame is interlaced before using cur_field
+- mips/float_dsp: fix vector_fmul_window_mips on mips64
+- doc: Remove non-existing decklink options.
+
+version 2.6.1:
+- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
+- avfilter/palettegen: make sure at least one frame was sent to the filter
+- avformat/mov: Check for string truncation in mov_open_dref()
+- ac3_fixed: fix out-of-bound read
+- mips/asmdefs: use _ABI64 as defined by gcc
+- hevc: delay ff_thread_finish_setup for hwaccel
+- avcodec/012v: Check dimensions more completely
+- asfenc: fix leaking asf->index_ptr on error
+- roqvideoenc: set enc->avctx in roq_encode_init
+- avcodec/options_table: remove extradata_size from the AVOptions table
+- ffmdec: limit the backward seek to the last resync position
+- Add dependencies to configure file for vf_fftfilt
+- ffmdec: make sure the time base is valid
+- ffmdec: fix infinite loop at EOF
+- ffmdec: initialize f_cprv, f_stvi and f_stau
+- arm: Suppress tags about used cpu arch and extensions
+- 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
+- lavc/hevcdsp: Fix compilation for arm with --disable-neon.
+- vp9: fix segmentation map retention with threading enabled.
+- Revert "avutil/opencl: is_compiled flag not being cleared in av_opencl_uninit"
+
 version 2.6:
 - nvenc encoder
 - 10bit spp filter
@@ -612,7 +948,7 @@ easier to use. The changes are:
           all the stream in the first input file, except for the second audio
           stream'.
     * There is a new option -c (or -codec) for choosing the decoder/encoder to
-      use, which makes it possible to precisely specify target stream(s) consistently with
+      use, which allows to precisely specify target stream(s) consistently with
       other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
       libvorbis sets the codec for the first audio stream and -c copy copies all
       the streams without reencoding. Old -vcodec/-acodec/-scodec options are now

LICENSE.md

@@ -16,7 +16,6 @@ Specifically, the GPL parts of FFmpeg are:
 - optional x86 optimizations in the files
   - `libavcodec/x86/flac_dsp_gpl.asm`
   - `libavcodec/x86/idct_mmx.c`
-  - `libavfilter/x86/vf_removegrain.asm`
 - libutvideo encoding/decoding wrappers in
   `libavcodec/libutvideo*.cpp`
 - the X11 grabber in `libavdevice/x11grab.c`
@@ -28,11 +27,9 @@ Specifically, the GPL parts of FFmpeg are:
     - `vf_blackframe.c`
     - `vf_boxblur.c`
     - `vf_colormatrix.c`
-    - `vf_cover_rect.c`
     - `vf_cropdetect.c`
     - `vf_delogo.c`
     - `vf_eq.c`
-    - `vf_find_rect.c`
     - `vf_fspp.c`
     - `vf_geq.c`
     - `vf_histeq.c`

MAINTAINERS

@@ -42,9 +42,9 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
+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, Lou Logan
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -58,7 +58,7 @@ 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, Reynaldo H. Verdejo Pinochet
+Twitter                                 Lou Logan
 Launchpad                               Timothy Gu
 
 
@@ -137,7 +137,6 @@ Codecs:
   4xm.c                                 Michael Niedermayer
   8bps.c                                Roberto Togni
   8svx.c                                Jaikrishnan Menon
-  aacenc*, aaccoder.c                   Rostislav Pehlivanov
   aasc.c                                Kostya Shishkov
   ac3*                                  Justin Ruggles
   alacenc.c                             Jaikrishnan Menon
@@ -171,7 +170,6 @@ Codecs:
   dvbsubdec.c                           Anshul Maheshwari
   dxa.c                                 Kostya Shishkov
   eacmv*, eaidct*, eat*                 Peter Ross
-  evrc*                                 Paul B Mahol
   exif.c, exif.h                        Thilo Borgmann
   ffv1*                                 Michael Niedermayer
   ffwavesynth.c                         Nicolas George
@@ -201,7 +199,6 @@ Codecs:
   libcelt_dec.c                         Nicolas George
   libdirac*                             David Conrad
   libgsm.c                              Michel Bardiaux
-  libkvazaar.c                          Arttu Ylä-Outinen
   libopenjpeg.c                         Jaikrishnan Menon
   libopenjpegenc.c                      Michael Bradshaw
   libschroedinger*                      David Conrad
@@ -239,7 +236,6 @@ Codecs:
   qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
   qdrw.c                                Kostya Shishkov
   qpeg.c                                Kostya Shishkov
-  qsv*                                  Ivan Uskov
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -301,12 +297,11 @@ Codecs:
 
 Hardware acceleration:
   crystalhd.c                           Philip Langdale
-  dxva2*                                Hendrik Leppkes, Laurent Aimar
+  dxva2*                                Laurent Aimar
   libstagefright.cpp                    Mohamed Naufal
   vaapi*                                Gwenole Beauchesne
   vda*                                  Sebastien Zwickert
-  vdpau*                                Philip Langdale, Carl Eugen Hoyos
-  videotoolbox*                         Sebastien Zwickert
+  vdpau*                                Carl Eugen Hoyos
 
 
 libavdevice
@@ -338,7 +333,6 @@ Generic parts:
   graphdump.c                           Nicolas George
 
 Filters:
-  f_drawgraph.c                         Paul B Mahol
   af_adelay.c                           Paul B Mahol
   af_aecho.c                            Paul B Mahol
   af_afade.c                            Paul B Mahol
@@ -349,21 +343,14 @@ Filters:
   af_astreamsync.c                      Nicolas George
   af_atempo.c                           Pavel Koshevoy
   af_biquads.c                          Paul B Mahol
-  af_chorus.c                           Paul B Mahol
   af_compand.c                          Paul B Mahol
   af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
-  af_sidechaincompress.c                Paul B Mahol
   af_silenceremove.c                    Paul B Mahol
-  avf_aphasemeter.c                     Paul B Mahol
   avf_avectorscope.c                    Paul B Mahol
   avf_showcqt.c                         Muhammad Faiz
   vf_blend.c                            Paul B Mahol
-  vf_colorchannelmixer.c                Paul B Mahol
   vf_colorbalance.c                     Paul B Mahol
-  vf_colorkey.c                         Timo Rothenpieler
-  vf_colorlevels.c                      Paul B Mahol
-  vf_deband.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
@@ -374,16 +361,12 @@ Filters:
   vf_il.c                               Paul B Mahol
   vf_lenscorrection.c                   Daniel Oberhoff
   vf_mergeplanes.c                      Paul B Mahol
-  vf_neighbor.c                         Paul B Mahol
   vf_psnr.c                             Paul B Mahol
-  vf_random.c                           Paul B Mahol
   vf_scale.c                            Michael Niedermayer
   vf_separatefields.c                   Paul B Mahol
-  vf_ssim.c                             Paul B Mahol
   vf_stereo3d.c                         Paul B Mahol
   vf_telecine.c                         Paul B Mahol
   vf_yadif.c                            Michael Niedermayer
-  vf_zoompan.c                          Paul B Mahol
 
 Sources:
   vsrc_mandelbrot.c                     Michael Niedermayer
@@ -400,7 +383,6 @@ Generic parts:
 
 Muxers/Demuxers:
   4xm.c                                 Mike Melanson
-  aadec.c                               Vesselin Bontchev (vesselin.bontchev at yandex dot com)
   adtsenc.c                             Robert Swain
   afc.c                                 Paul B Mahol
   aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
@@ -432,7 +414,6 @@ Muxers/Demuxers:
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
   hls.c                                 Anssi Hannula
-  hls encryption (hlsenc.c)             Christian Suloway
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
@@ -518,7 +499,6 @@ Muxers/Demuxers:
   wvenc.c                               Paul B Mahol
 
 Protocols:
-  async.c                               Zhang Rui
   bluray.c                              Petri Hintukainen
   ftp.c                                 Lukasz Marek
   http.c                                Ronald S. Bultje
@@ -554,7 +534,7 @@ Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver, Hendrik Leppkes
+Windows MSVC                            Matthew Oliver
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
@@ -564,7 +544,6 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 2.4                                     Michael Niedermayer
@@ -599,7 +578,6 @@ 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
-Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
 Reimar Doeffinger             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

Makefile

@@ -31,10 +31,7 @@ $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_o
 OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
 OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-ifndef CONFIG_VIDEOTOOLBOX
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
-endif
-OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
+OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
 OBJS-ffserver                 += ffserver_config.o
 
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
@@ -63,7 +60,6 @@ include $(SRC_PATH)/common.mak
 
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
-FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
 
 all: $(AVPROGS)
 
@@ -84,8 +80,8 @@ 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                           \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS       \
-               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS                \
+               OBJS SLIBOBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -179,7 +175,7 @@ clean::
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)

RELEASE

@@ -1 +1 @@
-2.8
+2.6.6

RELEASE_NOTES

@@ -1,12 +1,62 @@
 
-              ┌────────────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 2.8 "Feynman" │
-              └────────────────────────────────────────┘
+              ┌─────────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 2.6 "Grothendieck" │
+              └─────────────────────────────────────────────┘
 
-   The FFmpeg Project proudly presents FFmpeg 2.8 "Feynman", about 3
-   months after the release of FFmpeg 2.7.
+   The FFmpeg Project proudly presents FFmpeg 2.6 "Grothendieck", about 3
+   months after the release of FFmpeg 2.5.
 
-   A complete Changelog is available at the root of the project, and the
+   A lot of important work got in this time, so let's start talking about what
+   we like to brag the most about: features.
+
+   A lot of people will probably be happy to hear that we now have support for
+   NVENC — the Nvidia Video Encoder interface for H.264 encoding — thanks to
+   Timo Rothenpieler, with some little help from NVIDIA and Philip Langdale.
+
+   People in the broadcasting industry might also be interested in the first
+   steps of closed captions support with the introduction of a decoder by
+   Anshul Maheswhwari.
+
+   Regarding filters love, we improved and added many. We could talk about the
+   10-bit support in spp, but maybe it's more important to mention the addition
+   of colorlevels (yet another color handling filter), tblend (allowing you
+   to for example run a diff between successive frames of a video stream), or
+   the dcshift audio filter.
+
+   There are also two other important filters landing in libavfilter: palettegen
+   and paletteuse. Both submitted by the Stupeflix company. These filters will
+   be very useful in case you are looking for creating high quality GIFs, a
+   format that still bravely fights annihilation in 2015.
+
+   There are many other new features, but let's follow-up on one big cleanup
+   achievement: the libmpcodecs (MPlayer filters) wrapper is finally dead. The
+   last remaining filters (softpulldown/repeatfields, eq*, and various
+   postprocessing filters) were ported by Arwa Arif (OPW student) and Paul B
+   Mahol.
+
+   Concerning API changes, there are not many things to mention. Though, the
+   introduction of device inputs and outputs listing by Lukasz Marek is a
+   notable addition (try ffmpeg -sources or ffmpeg -sinks for an example of
+   the usage). As usual, see doc/APIchanges for more information.
+
+   Now let's talk about optimizations. Ronald S. Bultje made the VP9 decoder
+   usable on x86 32-bit systems and pre-ssse3 CPUs like Phenom (even dual core
+   Athlons can play 1080p 30fps VP9 content now), so we now secretly hope for
+   Google and Mozilla to use ffvp9 instead of libvpx. But VP9 is not the
+   center of attention anymore, and HEVC/H.265 is also getting many
+   improvements, which include C and x86 ASM optimizations, mainly from James
+   Almer, Christophe Gisquet and Pierre-Edouard Lepere.
+
+   Even though we had many x86 contributions, it is not the only architecture
+   getting some love, with Seppo Tomperi adding ARM NEON optimizations to the
+   HEVC stack, and James Cowgill adding MIPS64 assembly for all kind of audio
+   processing code in libavcodec.
+
+   And finally, Michael Niedermayer is still fixing many bugs, dealing with
+   most of the boring work such as making releases, applying tons of
+   contributors patches, and daily merging the changes from the Libav project.
+
+   A more complete Changelog is available at the root of the project, and the
    complete Git history on http://source.ffmpeg.org.
 
    We hope you will like this release as much as we enjoyed working on it, and

arch.mak

@@ -7,11 +7,8 @@ OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
-OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
-OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
 
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.c

@@ -41,10 +41,8 @@
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
-#include "libavutil/display.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/imgutils.h"
-#include "libavutil/libm.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/eval.h"
@@ -63,7 +61,7 @@
 
 static int init_report(const char *env);
 
-AVDictionary *sws_dict;
+struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;
 
@@ -73,13 +71,20 @@ int hide_banner = 0;
 
 void init_opts(void)
 {
-    av_dict_set(&sws_dict, "flags", "bicubic", 0);
+
+    if(CONFIG_SWSCALE)
+        sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
+                              NULL, NULL, NULL);
 }
 
 void uninit_opts(void)
 {
+#if CONFIG_SWSCALE
+    sws_freeContext(sws_opts);
+    sws_opts = NULL;
+#endif
+
     av_dict_free(&swr_opts);
-    av_dict_free(&sws_dict);
     av_dict_free(&format_opts);
     av_dict_free(&codec_opts);
     av_dict_free(&resample_opts);
@@ -475,22 +480,10 @@ static void dump_argument(const char *a)
     fputc('"', report_file);
 }
 
-static void check_options(const OptionDef *po)
-{
-    while (po->name) {
-        if (po->flags & OPT_PERFILE)
-            av_assert0(po->flags & (OPT_INPUT | OPT_OUTPUT));
-        po++;
-    }
-}
-
 void parse_loglevel(int argc, char **argv, const OptionDef *options)
 {
     int idx = locate_option(argc, argv, options, "loglevel");
     const char *env;
-
-    check_options(options);
-
     if (!idx)
         idx = locate_option(argc, argv, options, "v");
     if (idx && argv[idx + 1])
@@ -522,7 +515,7 @@ static const AVOption *opt_find(void *obj, const char *name, const char *unit,
     return o;
 }
 
-#define FLAGS (o->type == AV_OPT_TYPE_FLAGS && (arg[0]=='-' || arg[0]=='+')) ? AV_DICT_APPEND : 0
+#define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
 int opt_default(void *optctx, const char *opt, const char *arg)
 {
     const AVOption *o;
@@ -558,18 +551,14 @@ int opt_default(void *optctx, const char *opt, const char *arg)
     }
 #if CONFIG_SWSCALE
     sc = sws_get_class();
-    if (!consumed && (o = opt_find(&sc, opt, NULL, 0,
-                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
-        struct SwsContext *sws = sws_alloc_context();
-        int ret = av_opt_set(sws, opt, arg, 0);
-        sws_freeContext(sws);
+    if (!consumed && opt_find(&sc, opt, NULL, 0,
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
+        // XXX we only support sws_flags, not arbitrary sws options
+        int ret = av_opt_set(sws_opts, opt, arg, 0);
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
             return ret;
         }
-
-        av_dict_set(&sws_dict, opt, arg, FLAGS);
-
         consumed = 1;
     }
 #else
@@ -643,7 +632,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
     *g             = octx->cur_group;
     g->arg         = arg;
     g->group_def   = l->group_def;
-    g->sws_dict    = sws_dict;
+#if CONFIG_SWSCALE
+    g->sws_opts    = sws_opts;
+#endif
     g->swr_opts    = swr_opts;
     g->codec_opts  = codec_opts;
     g->format_opts = format_opts;
@@ -652,7 +643,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
     codec_opts  = NULL;
     format_opts = NULL;
     resample_opts = NULL;
-    sws_dict    = NULL;
+#if CONFIG_SWSCALE
+    sws_opts    = NULL;
+#endif
     swr_opts    = NULL;
     init_opts();
 
@@ -708,8 +701,9 @@ void uninit_parse_context(OptionParseContext *octx)
             av_dict_free(&l->groups[j].codec_opts);
             av_dict_free(&l->groups[j].format_opts);
             av_dict_free(&l->groups[j].resample_opts);
-
-            av_dict_free(&l->groups[j].sws_dict);
+#if CONFIG_SWSCALE
+            sws_freeContext(l->groups[j].sws_opts);
+#endif
             av_dict_free(&l->groups[j].swr_opts);
         }
         av_freep(&l->groups);
@@ -851,7 +845,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
         { "info"   , AV_LOG_INFO    },
         { "verbose", AV_LOG_VERBOSE },
         { "debug"  , AV_LOG_DEBUG   },
-        { "trace"  , AV_LOG_TRACE   },
     };
     char *tail;
     int level;
@@ -1316,12 +1309,12 @@ static void print_codec(const AVCodec *c)
     if (c->type == AVMEDIA_TYPE_VIDEO ||
         c->type == AVMEDIA_TYPE_AUDIO) {
         printf("    Threading capabilities: ");
-        switch (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
-                                   AV_CODEC_CAP_SLICE_THREADS)) {
-        case AV_CODEC_CAP_FRAME_THREADS |
-             AV_CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
-        case AV_CODEC_CAP_FRAME_THREADS: printf("frame");           break;
-        case AV_CODEC_CAP_SLICE_THREADS: printf("slice");           break;
+        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
+                                   CODEC_CAP_SLICE_THREADS)) {
+        case CODEC_CAP_FRAME_THREADS |
+             CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
+        case CODEC_CAP_FRAME_THREADS: printf("frame");           break;
+        case CODEC_CAP_SLICE_THREADS: printf("slice");           break;
         default:                      printf("no");              break;
         }
         printf("\n");
@@ -1495,11 +1488,11 @@ static void print_codecs(int encoder)
 
         while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
             printf(" %c", get_media_type_char(desc->type));
-            printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_DR1)           ? "D" : ".");
+            printf((codec->capabilities & CODEC_CAP_FRAME_THREADS) ? "F" : ".");
+            printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : ".");
+            printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
+            printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
+            printf((codec->capabilities & CODEC_CAP_DR1)           ? "D" : ".");
 
             printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : "");
             if (strcmp(codec->name, desc->name))
@@ -1542,10 +1535,10 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
     printf("Supported file protocols:\n"
            "Input:\n");
     while ((name = avio_enum_protocols(&opaque, 0)))
-        printf("  %s\n", name);
+        printf("%s\n", name);
     printf("Output:\n");
     while ((name = avio_enum_protocols(&opaque, 1)))
-        printf("  %s\n", name);
+        printf("%s\n", name);
     return 0;
 }
 
@@ -1560,7 +1553,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
     printf("Filters:\n"
            "  T.. = Timeline support\n"
            "  .S. = Slice threading\n"
-           "  ..C = Command support\n"
+           "  ..C = Commmand support\n"
            "  A = Audio input/output\n"
            "  V = Video input/output\n"
            "  N = Dynamic number and/or type of input/output\n"
@@ -1573,10 +1566,10 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                 *(descr_cur++) = '>';
             }
             pad = i ? filter->outputs : filter->inputs;
-            for (j = 0; pad && avfilter_pad_get_name(pad, j); j++) {
+            for (j = 0; pad && pad[j].name; j++) {
                 if (descr_cur >= descr + sizeof(descr) - 4)
                     break;
-                *(descr_cur++) = get_media_type_char(avfilter_pad_get_type(pad, j));
+                *(descr_cur++) = get_media_type_char(pad[j].type);
             }
             if (!j)
                 *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
@@ -1867,6 +1860,64 @@ int read_yesno(void)
     return yesno;
 }
 
+int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
+{
+    int64_t ret;
+    FILE *f = av_fopen_utf8(filename, "rb");
+
+    if (!f) {
+        ret = AVERROR(errno);
+        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
+               strerror(errno));
+        return ret;
+    }
+
+    ret = fseek(f, 0, SEEK_END);
+    if (ret == -1) {
+        ret = AVERROR(errno);
+        goto out;
+    }
+
+    ret = ftell(f);
+    if (ret < 0) {
+        ret = AVERROR(errno);
+        goto out;
+    }
+    *size = ret;
+
+    ret = fseek(f, 0, SEEK_SET);
+    if (ret == -1) {
+        ret = AVERROR(errno);
+        goto out;
+    }
+
+    *bufptr = av_malloc(*size + 1);
+    if (!*bufptr) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
+        ret = AVERROR(ENOMEM);
+        goto out;
+    }
+    ret = fread(*bufptr, 1, *size, f);
+    if (ret < *size) {
+        av_free(*bufptr);
+        if (ferror(f)) {
+            ret = AVERROR(errno);
+            av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
+                   filename, strerror(errno));
+        } else
+            ret = AVERROR_EOF;
+    } else {
+        ret = 0;
+        (*bufptr)[(*size)++] = '\0';
+    }
+
+out:
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", av_err2str(ret));
+    fclose(f);
+    return ret;
+}
+
 FILE *get_preset_file(char *filename, size_t filename_size,
                       const char *preset_name, int is_path,
                       const char *codec_name)
@@ -2021,33 +2072,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
     return array;
 }
 
-double get_rotation(AVStream *st)
-{
-    AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
-    uint8_t* displaymatrix = av_stream_get_side_data(st,
-                                                     AV_PKT_DATA_DISPLAYMATRIX, NULL);
-    double theta = 0;
-
-    if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
-        char *tail;
-        theta = av_strtod(rotate_tag->value, &tail);
-        if (*tail)
-            theta = 0;
-    }
-    if (displaymatrix && !theta)
-        theta = -av_display_rotation_get((int32_t*) displaymatrix);
-
-    theta -= 360*floor(theta/360 + 0.9/360);
-
-    if (fabs(theta - 90*round(theta/90)) > 2)
-        av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
-               "If you want to help, upload a sample "
-               "of this file to ftp://upload.ffmpeg.org/incoming/ "
-               "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
-
-    return theta;
-}
-
 #if CONFIG_AVDEVICE
 static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
 {
@@ -2204,5 +2228,4 @@ int show_sinks(void *optctx, const char *opt, const char *arg)
     av_log_set_level(error_level);
     return ret;
 }
-
 #endif

cmdutils.h

@@ -46,7 +46,7 @@ extern const int program_birth_year;
 
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
-extern AVDictionary *sws_dict;
+extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;
@@ -277,7 +277,7 @@ typedef struct OptionGroup {
     AVDictionary *codec_opts;
     AVDictionary *format_opts;
     AVDictionary *resample_opts;
-    AVDictionary *sws_dict;
+    struct SwsContext *sws_opts;
     AVDictionary *swr_opts;
 } OptionGroup;
 
@@ -529,6 +529,18 @@ int show_colors(void *optctx, const char *opt, const char *arg);
  */
 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
+ * AVERROR error code in case of failure.
+ */
+int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
+
 /**
  * Get a file corresponding to a preset file.
  *
@@ -585,6 +597,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
     char name[128];\
     av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
 
-double get_rotation(AVStream *st);
-
 #endif /* CMDUTILS_H */

common.mak

@@ -7,7 +7,7 @@ all: all-yes
 
 DEFAULT_YASMD=.dbg
 
-ifeq ($(DBG),1)
+ifeq (1, DBG)
 YASMD=$(DEFAULT_YASMD)
 else
 YASMD=
@@ -118,9 +118,8 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)
 
-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
-STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
 
 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))

compat/avisynth/avisynth_c.h

@@ -38,9 +38,40 @@
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__
 
-#include "avs/config.h"
-#include "avs/capi.h"
-#include "avs/types.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
+
+typedef unsigned char BYTE;
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
 
 
 /////////////////////////////////////////////////////////////////////
@@ -48,8 +79,8 @@
 // Constants
 //
 
-#ifndef __AVISYNTH_6_H__
-enum { AVISYNTH_INTERFACE_VERSION = 6 };
+#ifndef __AVISYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 4 };
 #endif
 
 enum {AVS_SAMPLE_INT8  = 1<<0,
@@ -81,8 +112,8 @@ enum {AVS_CS_BGR = 1<<28,
       AVS_CS_PLANAR = 1<<31,
 
       AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 8,
-      AVS_CS_SHIFT_SAMPLE_BITS = 16,
+      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
@@ -149,66 +180,15 @@ enum {  //SUBTYPES
   AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
 
 enum {
-  // New 2.6 explicitly defined cache hints.
-  AVS_CACHE_NOTHING=10, // Do not cache video.
-  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
-  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
-  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
-
-  AVS_CACHE_GET_POLICY=30, // Get the current policy.
-  AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
-  AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
-
-  AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
-  AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
-  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
-  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
-
-  AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
-  AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
-
-  AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
-  AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
-
-  AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
-  AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
-  AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
-  AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
-  AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
-  AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
-
-  AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
-  AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
-  AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
-  AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
-
-  AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
-    AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
-    AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
-    AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
-    AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
-    AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
-
-  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
-    AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
-    AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
-    AVS_CACHE_THREAD_SAFE=243, //  Allow all threads in any instance.
-    AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
-
-  AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
-    AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
-    AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
-    AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
+  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
   };
 
-#ifdef BUILDING_AVSCORE
-struct AVS_ScriptEnvironment {
-        IScriptEnvironment * env;
-        const char * error;
-        AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
-                : env(e), error(0) {}
-};
-#endif
+#define AVS_FRAME_ALIGN 16
 
 typedef struct AVS_Clip AVS_Clip;
 typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@@ -258,23 +238,29 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
         { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
 
-AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
+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_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
+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_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
+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_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
+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_API(int, avs_is_y8)(const AVS_VideoInfo * p);
+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->image_type & property)==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_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
+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); }
@@ -288,18 +274,25 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
         { return !!(p->image_type & AVS_IT_TFF); }
 
-AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
+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_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
+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_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
-
-AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
-
-AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
+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; }
@@ -357,13 +350,11 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
     p->fps_denominator = denominator/x;
 }
 
-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 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));
 }
-#endif
 
 /////////////////////////////////////////////////////////////////////
 //
@@ -400,38 +391,89 @@ typedef struct AVS_VideoFrame {
 } AVS_VideoFrame;
 
 // Access functions for AVS_VideoFrame
-AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
-
-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return avs_get_pitch_p(p, 0);}
-#endif
+        return p->pitch;}
 
-AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
+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_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
+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_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
+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;}
 
-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return avs_get_read_ptr_p(p, 0);}
-#endif
+        return p->vfb->data + p->offset;}
 
-AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
+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_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
+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;
+                }
+        }
+}
 
-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
-        return avs_get_write_ptr_p(p, 0);}
-#endif
 
 AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
 // makes a shallow copy of a video frame
@@ -616,16 +658,12 @@ enum {
   AVS_CPUF_SSSE3      = 0x200,   //  Core 2
   AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
   AVS_CPUF_SSE4_1     = 0x400,
-//AVS_CPUF_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
-  AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
-//AVS_CPUF_AVX2      = 0x2000,   //  Haswell
-//AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
+  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
 };
 
-
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
 
-AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+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);
@@ -662,12 +700,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                      const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
+  {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,FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 #endif
 
 
@@ -735,6 +773,7 @@ struct AVS_Library {
   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);
@@ -759,27 +798,6 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_subframe_planar);
   AVSC_DECLARE_FUNC(avs_take_clip);
   AVSC_DECLARE_FUNC(avs_vsprintf);
-
-  AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_is_yv24);
-  AVSC_DECLARE_FUNC(avs_is_yv16);
-  AVSC_DECLARE_FUNC(avs_is_yv12);
-  AVSC_DECLARE_FUNC(avs_is_yv411);
-  AVSC_DECLARE_FUNC(avs_is_y8);
-  AVSC_DECLARE_FUNC(avs_is_color_space);
-
-  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
-  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
-  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
-  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
-  AVSC_DECLARE_FUNC(avs_row_size);
-  AVSC_DECLARE_FUNC(avs_bmp_size);
-  AVSC_DECLARE_FUNC(avs_get_pitch_p);
-  AVSC_DECLARE_FUNC(avs_get_row_size_p);
-  AVSC_DECLARE_FUNC(avs_get_height_p);
-  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
-  AVSC_DECLARE_FUNC(avs_is_writable);
-  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
 };
 
 #undef AVSC_DECLARE_FUNC
@@ -787,7 +805,7 @@ struct AVS_Library {
 
 AVSC_INLINE AVS_Library * avs_load_library() {
   AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (library == NULL)
+  if (!library)
     return NULL;
   library->handle = LoadLibrary("avisynth");
   if (library->handle == NULL)
@@ -814,6 +832,7 @@ AVSC_INLINE AVS_Library * avs_load_library() {
   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);
@@ -839,27 +858,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
   AVSC_LOAD_FUNC(avs_take_clip);
   AVSC_LOAD_FUNC(avs_vsprintf);
 
-  AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_is_yv24);
-  AVSC_LOAD_FUNC(avs_is_yv16);
-  AVSC_LOAD_FUNC(avs_is_yv12);
-  AVSC_LOAD_FUNC(avs_is_yv411);
-  AVSC_LOAD_FUNC(avs_is_y8);
-  AVSC_LOAD_FUNC(avs_is_color_space);
-
-  AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
-  AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
-  AVSC_LOAD_FUNC(avs_bits_per_pixel);
-  AVSC_LOAD_FUNC(avs_bytes_from_pixels);
-  AVSC_LOAD_FUNC(avs_row_size);
-  AVSC_LOAD_FUNC(avs_bmp_size);
-  AVSC_LOAD_FUNC(avs_get_pitch_p);
-  AVSC_LOAD_FUNC(avs_get_row_size_p);
-  AVSC_LOAD_FUNC(avs_get_height_p);
-  AVSC_LOAD_FUNC(avs_get_read_ptr_p);
-  AVSC_LOAD_FUNC(avs_is_writable);
-  AVSC_LOAD_FUNC(avs_get_write_ptr_p);
-
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC
@@ -872,7 +870,7 @@ fail:
 }
 
 AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (library == NULL)
+  if (!library)
     return;
   FreeLibrary(library->handle);
   free(library);

compat/avisynth/avisynth_c_25.h

@@ -0,0 +1,68 @@
+//  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/avs/capi.h

@@ -1,62 +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., 675 Mass Ave, Cambridge, MA 02139, 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 AVS_CAPI_H
-#define AVS_CAPI_H
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef BUILDING_AVSCORE
-#  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
-
-#endif //AVS_CAPI_H

compat/avisynth/avs/config.h

@@ -1,55 +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., 675 Mass Ave, Cambridge, MA 02139, 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 AVS_CONFIG_H
-#define AVS_CONFIG_H
-
-// Undefine this to get cdecl calling convention
-#define AVSC_USE_STDCALL 1
-
-// NOTE TO PLUGIN AUTHORS:
-// Because FRAME_ALIGN can be substantially higher than the alignment
-// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
-// alignment. They should always request the exact alignment value they need.
-// This is to make sure that plugins work over the widest range of AviSynth
-// builds possible.
-#define FRAME_ALIGN 32
-
-#if   defined(_M_AMD64) || defined(__x86_64)
-#   define X86_64
-#elif defined(_M_IX86) || defined(__i386__)
-#   define X86_32
-#else
-#   error Unsupported CPU architecture.
-#endif
-
-#endif //AVS_CONFIG_H

compat/avisynth/avs/types.h

@@ -1,51 +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., 675 Mass Ave, Cambridge, MA 02139, 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 AVS_TYPES_H
-#define AVS_TYPES_H
-
-// Define all types necessary for interfacing with avisynth.dll
-
-// Raster types used by VirtualDub & Avisynth
-typedef unsigned int    Pixel32;
-typedef unsigned char   BYTE;
-
-// Audio Sample information
-typedef float SFLOAT;
-
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-#endif //AVS_TYPES_H

compat/os2threads.h

@@ -66,8 +66,6 @@ static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr
     struct thread_arg *thread_arg;
 
     thread_arg = av_mallocz(sizeof(struct thread_arg));
-    if (!thread_arg)
-        return ENOMEM;
 
     thread_arg->start_routine = start_routine;
     thread_arg->arg = arg;

compat/windows/mslink

@@ -1,9 +0,0 @@
-#!/bin/sh
-
-LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
-if [ -x "$LINK_EXE_PATH" ]; then
-    "$LINK_EXE_PATH" $@
-else
-    link $@
-fi
-exit $?

doc/APIchanges

@@ -15,98 +15,6 @@ libavutil:     2014-08-09
 
 API changes, most recent first:
 
--------- 8< --------- FFmpeg 2.8 was cut here -------- 8< ---------
-
-2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h
-  Deprecate old VA-API context (vaapi_context) fields that were only
-  set and used by libavcodec. They are all managed internally now.
-
-2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h
-  Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that
-  indicates the nature of the underlying storage: a VA surface. This
-  yields the same value as AV_PIX_FMT_VAAPI_VLD.
-  Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO,
-  AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD.
-
-2015-08-02 - lavu 54.30.100 / 54.17.0
-  9ed59f1 / 7a7df34c -  Add av_blowfish_alloc().
-  a130ec9 / ae365453 -  Add av_rc4_alloc().
-  9ca1997 / 5d8bea3b -  Add av_xtea_alloc().
-  3cf08e9 / d9e8b47e -  Add av_des_alloc().
-
-2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h
-  94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
-  444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
-  29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
-              to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE.
-
-2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h
-  Add AV_PKT_DATA_QUALITY_STATS to export the quality value, PSNR, and pict_type
-  of an AVPacket.
-
-2015-07-16 - 8dad213 - lavc 56.49.100
-  Add av_codec_get_codec_properties(), FF_CODEC_PROPERTY_LOSSLESS
-  and FF_CODEC_PROPERTY_CLOSED_CAPTIONS
-
-2015-07-03 - d563e13 / 83212943 - lavu 54.28.100 / 56.15.0
-  Add av_version_info().
-
--------- 8< --------- FFmpeg 2.7 was cut here -------- 8< ---------
-
-2015-06-04 - cc17b43 - lswr  1.2.100
-  Add swr_get_out_samples()
-
-2015-05-27 - c312bfa - lavu 54.26.100 - cpu.h
-  Add AV_CPU_FLAG_AVXSLOW.
-
-2015-05-26 - 1fb9b2a - lavu 54.25.100 - rational.h
-  Add av_q2intfloat().
-
-2015-05-13 - cc48409 / e7c5e17 - lavc 56.39.100 / 56.23.0
-  Add av_vda_default_init2.
-
-2015-05-11 - 541d75f - lavf 56.33.100 - avformat.h
-  Add AVOpenCallback AVFormatContext.open_cb
-
-2015-05-07 - a7dd933 - 56.38.100 - avcodec.h
-  Add av_packet_side_data_name().
-
-2015-05-07 - 01e59d4 - 56.37.102 - avcodec.h
-  Add FF_PROFILE_VP9_2 and FF_PROFILE_VP9_3.
-
-2015-05-04 - 079b7f6 - 56.37.100 - avcodec.h
-  Add FF_PROFILE_VP9_0 and FF_PROFILE_VP9_1.
-
-2015-04-22 - 748d481 - lavf 56.31.100 - avformat.h
-  Add AVFMT_FLAG_FAST_SEEK flag. Some formats (initially mp3) use it to enable
-  fast, but inaccurate seeking.
-
-2015-04-20 - 8e8219e / c253340 - lavu 54.23.100 / 54.12.0 - log.h
-  Add AV_LOG_TRACE for extremely verbose debugging.
-
-2015-04-02 - 26e0e393 - lavf 56.29.100 - avio.h
-  Add AVIODirEntryType.AVIO_ENTRY_SERVER.
-  Add AVIODirEntryType.AVIO_ENTRY_SHARE.
-  Add AVIODirEntryType.AVIO_ENTRY_WORKGROUP.
-
-2015-03-31 - 3188696 - lavu 54.22.100 - avstring.h
-  Add av_append_path_component()
-
-2015-03-27 - 184084c - lavf 56.27.100 - avio.h url.h
-  New directory listing API.
-
-  Add AVIODirEntryType enum.
-  Add AVIODirEntry, AVIODirContext structures.
-  Add avio_open_dir(), avio_read_dir(), avio_close_dir(), avio_free_directory_entry().
-  Add ff_alloc_dir_entry().
-  Extend URLProtocol with url_open_dir(), url_read_dir(), url_close_dir().
-
-2015-03-29 - 268ff17 / c484561 - lavu 54.21.100 / 54.10.0 - pixfmt.h
-  Add AV_PIX_FMT_MMAL for MMAL hardware acceleration.
-
-2015-03-19 - 11fe56c - 56.29.100 / lavc 56.22.0
-  Add FF_PROFILE_DTS_EXPRESS.
-
 -------- 8< --------- FFmpeg 2.6 was cut here -------- 8< ---------
 
 2015-03-04 - cca4476 - lavf 56.25.100
@@ -728,9 +636,6 @@ API changes, most recent first:
    av_ripemd_update()
    av_ripemd_final()
 
-2013-06-10 - 82ef670 - lavu 52.35.101 - hmac.h
-  Add AV_HMAC_SHA224, AV_HMAC_SHA256, AV_HMAC_SHA384, AV_HMAC_SHA512
-
 2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
   Add av_realloc_array and av_reallocp_array
 

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 2.8
+PROJECT_NUMBER         = 2.6.6
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55

doc/Makefile

@@ -36,7 +36,6 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
-DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
 DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding

doc/bitstream_filters.texi

@@ -139,26 +139,6 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section mp3_header_decompress
 
-@section mpeg4_unpack_bframes
-
-Unpack DivX-style packed B-frames.
-
-DivX-style packed B-frames are not valid MPEG-4 and were only a
-workaround for the broken Video for Windows subsystem.
-They use more space, can cause minor AV sync issues, require more
-CPU power to decode (unless the player has some decoded picture queue
-to compensate the 2,0,2,0 frame per packet style) and cause
-trouble if copied into a standard container like mp4 or mpeg-ps/ts,
-because MPEG-4 decoders may not be able to decode them, since they are
-not valid MPEG-4.
-
-For example to fix an AVI file containing an MPEG-4 stream with
-DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
-
-@example
-ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
-@end example
-
 @section noise
 
 Damages the contents of packets without damaging the container. Can be

doc/build_system.txt

@@ -16,9 +16,6 @@ DESTDIR
     Destination directory for the install targets, useful to prepare packages
     or install FFmpeg in cross-environments.
 
-GEN
-    Set to ‘1’ to generate the missing or mismatched references.
-
 Makefile targets:
 
 all

doc/codecs.texi

@@ -475,9 +475,6 @@ per-block quantization parameter (QP)
 motion vector
 @item dct_coeff
 
-@item green_metadata
-display complexity metadata for the upcoming frame, GoP or for a given duration.
-
 @item skip
 
 @item startcode
@@ -1045,11 +1042,7 @@ Possible values:
 @item color_primaries @var{integer} (@emph{decoding/encoding,video})
 @item color_trc @var{integer} (@emph{decoding/encoding,video})
 @item colorspace @var{integer} (@emph{decoding/encoding,video})
-
 @item color_range @var{integer} (@emph{decoding/encoding,video})
-If used as input parameter, it serves as a hint to the decoder, which
-color_range the input has.
-
 @item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
 
 @item log_level_offset @var{integer}

doc/decoders.texi

@@ -25,13 +25,6 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
 
-@section hevc
-
-HEVC / H.265 decoder.
-
-Note: the @option{skip_loop_filter} option has effect only at level
-@code{all}.
-
 @section rawvideo
 
 Raw video decoder.
@@ -90,23 +83,6 @@ Loud sounds are fully compressed.  Soft sounds are enhanced.
 
 @end table
 
-@section flac
-
-FLAC audio decoder.
-
-This decoder aims to implement the complete FLAC specification from Xiph.
-
-@subsection FLAC Decoder options
-
-@table @option
-
-@item -use_buggy_lpc
-The lavc FLAC encoder used to produce buggy streams with high lpc values
-(like the default value). This option makes it possible to decode such streams
-correctly by using lavc's old buggy lpc logic for decoding.
-
-@end table
-
 @section ffwavesynth
 
 Internal wave synthetizer.
@@ -195,25 +171,6 @@ without this library.
 @chapter Subtitles Decoders
 @c man begin SUBTILES DECODERS
 
-@section dvbsub
-
-@subsection Options
-
-@table @option
-@item compute_clut
-@table @option
-@item -1
-Compute clut if no matching CLUT is in the stream.
-@item 0
-Never compute CLUT
-@item 1
-Always compute CLUT and override the one provided in the stream.
-@end table
-@item dvb_substream
-Selects the dvb substream, or all substreams if -1 which is default.
-
-@end table
-
 @section dvdsub
 
 This codec decodes the bitmap subtitles used in DVDs; the same subtitles can

doc/demuxers.texi

@@ -18,12 +18,6 @@ enabled demuxers.
 
 The description of some of the currently available demuxers follows.
 
-@section aa
-
-Audible Format 2, 3, and 4 demuxer.
-
-This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
-
 @section applehttp
 
 Apple HTTP Live Streaming demuxer.
@@ -118,47 +112,6 @@ 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{inpoint @var{timestamp}}
-In point of the file. When the demuxer opens the file it instantly seeks to the
-specified timestamp. Seeking is done so that all streams can be presented
-successfully at In point.
-
-This directive works best with intra frame codecs, because for non-intra frame
-ones you will usually get extra packets before the actual In point and the
-decoded content will most likely contain frames before In point too.
-
-For each file, packets before the file In point will have timestamps less than
-the calculated start timestamp of the file (negative in case of the first
-file), and the duration of the files (if not specified by the @code{duration}
-directive) will be reduced based on their specified In point.
-
-Because of potential packets before the specified In point, packet timestamps
-may overlap between two concatenated files.
-
-@item @code{outpoint @var{timestamp}}
-Out point of the file. When the demuxer reaches the specified decoding
-timestamp in any of the streams, it handles it as an end of file condition and
-skips the current and all the remaining packets from all streams.
-
-Out point is exclusive, which means that the demuxer will not output packets
-with a decoding timestamp greater or equal to Out point.
-
-This directive works best with intra frame codecs and formats where all streams
-are tightly interleaved. For non-intra frame codecs you will usually get
-additional packets with presentation timestamp after Out point therefore the
-decoded content will most likely contain frames after Out point too. If your
-streams are not tightly interleaved you may not get all the packets from all
-streams before Out point and you may only will be able to decode the earliest
-stream until Out point.
-
-The duration of the files (if not specified by the @code{duration}
-directive) will be reduced based on their specified Out point.
-
-@item @code{file_packet_metadata @var{key=value}}
-Metadata of the packets of the file. The specified metadata will be set for
-each file packet. You can specify this directive multiple times to add multiple
-metadata entries.
-
 @item @code{stream}
 Introduce a stream in the virtual file.
 All subsequent stream-related directives apply to the last introduced
@@ -198,7 +151,6 @@ probed and 0 otherwise.
 @item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
-The default is 1.
 
 Currently, the only conversion is adding the h264_mp4toannexb bitstream
 filter to H.264 streams in MP4 format. This is necessary in particular if
@@ -253,11 +205,6 @@ It accepts the following options:
 Set the minimum valid delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 2.
 
-@item max_gif_delay
-Set the maximum valid delay between frames in hundredth of seconds.
-Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
-the maximum value allowed by the specification.
-
 @item default_delay
 Set the default delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 10.
@@ -306,10 +253,6 @@ Select the pattern type used to interpret the provided filename.
 
 @var{pattern_type} accepts one of the following values.
 @table @option
-@item none
-Disable pattern matching, therefore the video will only contain the specified
-image. You should use this option if you do not want to create sequences from
-multiple images and your filenames may contain special pattern characters.
 @item sequence
 Select a sequence pattern type, used to specify a sequence of files
 indexed by sequential numbers.
@@ -420,26 +363,13 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
 
 MPEG-2 transport stream demuxer.
 
-This demuxer accepts the following options:
 @table @option
-@item resync_size
-Set size limit for looking up a new synchronization. Default value is
-65536.
 
 @item fix_teletext_pts
-Override teletext packet PTS and DTS values with the timestamps calculated
+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.
-
-@item ts_packetsize
-Output option carrying the raw packet size in bytes.
-Show the detected raw packet size, cannot be set by the user.
-
-@item scan_all_pmts
-Scan and combine all PMTs. The value is an integer with value from -1
-to 1 (-1 means automatic setting, 1 means enabled, 0 means
-disabled). Default value is -1.
 @end table
 
 @section rawvideo

doc/developer.texi

@@ -228,7 +228,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
 @end example
 
 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@lisp
+@example
 (c-add-style "ffmpeg"
              '("k&r"
                (c-basic-offset . 4)
@@ -239,7 +239,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
                )
              )
 (setq c-default-style "ffmpeg")
-@end lisp
+@end example
 
 @section Development Policy
 
@@ -543,10 +543,6 @@ tools/trasher, the noise bitstream filter, and
 should not crash, end in a (near) infinite loop, or allocate ridiculous
 amounts of memory when fed damaged data.
 
-@item
-Did you test your decoder or demuxer against sample files?
-Samples may be obtained at @url{http://samples.ffmpeg.org}.
-
 @item
 Does the patch not mix functional and cosmetic changes?
 
@@ -637,10 +633,6 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.
 
-Everyone is welcome to review patches. Also if you are waiting for your patch
-to be reviewed, please consider helping to review other patches, that is a great
-way to get everyone's patches reviewed sooner.
-
 @anchor{Regression tests}
 @section Regression tests
 

doc/encoders.texi

@@ -494,85 +494,6 @@ Selected by Encoder (default)
 
 @end table
 
-@anchor{flac}
-@section flac
-
-FLAC (Free Lossless Audio Codec) Encoder
-
-@subsection Options
-
-The following options are supported by FFmpeg's flac encoder.
-
-@table @option
-@item compression_level
-Sets the compression level, which chooses defaults for many other options
-if they are not set explicitly.
-
-@item frame_size
-Sets the size of the frames in samples per channel.
-
-@item lpc_coeff_precision
-Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the
-default.
-
-@item lpc_type
-Sets the first stage LPC algorithm
-@table @samp
-@item none
-LPC is not used
-
-@item fixed
-fixed LPC coefficients
-
-@item levinson
-
-@item cholesky
-@end table
-
-@item lpc_passes
-Number of passes to use for Cholesky factorization during LPC analysis
-
-@item min_partition_order
-The minimum partition order
-
-@item max_partition_order
-The maximum partition order
-
-@item prediction_order_method
-@table @samp
-@item estimation
-@item 2level
-@item 4level
-@item 8level
-@item search
-Bruteforce search
-@item log
-@end table
-
-@item ch_mode
-Channel mode
-@table @samp
-@item auto
-The mode is chosen automatically for each frame
-@item indep
-Chanels are independently coded
-@item left_side
-@item right_side
-@item mid_side
-@end table
-
-@item exact_rice_parameters
-Chooses if rice parameters are calculated exactly or approximately.
-if set to 1 then they are chosen exactly, which slows the code down slightly and
-improves compression slightly.
-
-@item multi_dim_quant
-Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is
-applied after the first stage to finetune the coefficients. This is quite slow
-and slightly improves compression.
-
-@end table
-
 @anchor{libfaac}
 @section libfaac
 
@@ -1342,30 +1263,6 @@ disabled
 A description of some of the currently available video encoders
 follows.
 
-@section jpeg2000
-
-The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
-option can be used to set the encoding quality. Lossless encoding
-can be selected with @code{-pred 1}.
-
-@subsection Options
-
-@table @option
-@item format
-Can be set to either @code{j2k} or @code{jp2} (the default) that
-allows to store non-rgb pix_fmts.
-
-@end table
-
-@section snow
-
-@subsection Options
-
-@table @option
-@item iterative_dia_size
-dia size for the iterative motion estimation
-@end table
-
 @section libtheora
 
 libtheora Theora encoder wrapper.
@@ -1440,153 +1337,113 @@ You need to explicitly configure the build with @code{--enable-libvpx}.
 
 @subsection Options
 
-The following options are supported by the libvpx wrapper. The
-@command{vpxenc}-equivalent options or values are listed in parentheses
-for easy migration.
-
-To reduce the duplication of documentation, only the private options
-and some others requiring special attention are documented here. For
-the documentation of the undocumented generic options, see
-@ref{codec-options,,the Codec Options chapter}.
-
-To get more documentation of the libvpx options, invoke the command
-@command{ffmpeg -h encoder=libvpx}, @command{ffmpeg -h encoder=libvpx-vp9} or
-@command{vpxenc --help}. Further information is available in the libvpx API
-documentation.
+Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
 
 @table @option
 
-@item b (@emph{target-bitrate})
-Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
-expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in
-kilobits/s.
+@item threads
+g_threads
 
-@item g (@emph{kf-max-dist})
+@item profile
+g_profile
 
-@item keyint_min (@emph{kf-min-dist})
+@item vb
+rc_target_bitrate
 
-@item qmin (@emph{min-q})
+@item g
+kf_max_dist
 
-@item qmax (@emph{max-q})
+@item keyint_min
+kf_min_dist
 
-@item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz})
-Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
-specified in milliseconds, the libvpx wrapper converts this value as follows:
-@code{buf-sz = bufsize * 1000 / bitrate},
-@code{buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6}.
+@item qmin
+rc_min_quantizer
 
-@item rc_init_occupancy (@emph{buf-initial-sz})
-Set number of bits which should be loaded into the rc buffer before decoding
-starts. Note @command{vpxenc}'s option is specified in milliseconds, the libvpx
-wrapper converts this value as follows:
-@code{rc_init_occupancy * 1000 / bitrate}.
+@item qmax
+rc_max_quantizer
 
-@item undershoot-pct
-Set datarate undershoot (min) percentage of the target bitrate.
+@item bufsize, vb
+rc_buf_sz
+@code{(bufsize * 1000 / vb)}
 
-@item overshoot-pct
-Set datarate overshoot (max) percentage of the target bitrate.
+rc_buf_optimal_sz
+@code{(bufsize * 1000 / vb * 5 / 6)}
 
-@item skip_threshold (@emph{drop-frame})
+@item rc_init_occupancy, vb
+rc_buf_initial_sz
+@code{(rc_init_occupancy * 1000 / vb)}
 
-@item qcomp (@emph{bias-pct})
+@item rc_buffer_aggressivity
+rc_undershoot_pct
 
-@item maxrate (@emph{maxsection-pct})
-Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
-percentage of the target bitrate, the libvpx wrapper converts this value as
-follows: @code{(maxrate * 100 / bitrate)}.
+@item skip_threshold
+rc_dropframe_thresh
 
-@item minrate (@emph{minsection-pct})
-Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
-percentage of the target bitrate, the libvpx wrapper converts this value as
-follows: @code{(minrate * 100 / bitrate)}.
+@item qcomp
+rc_2pass_vbr_bias_pct
 
-@item minrate, maxrate, b @emph{end-usage=cbr}
-@code{(minrate == maxrate == bitrate)}.
+@item maxrate, vb
+rc_2pass_vbr_maxsection_pct
+@code{(maxrate * 100 / vb)}
 
-@item crf (@emph{end-usage=cq}, @emph{cq-level})
+@item minrate, vb
+rc_2pass_vbr_minsection_pct
+@code{(minrate * 100 / vb)}
 
-@item quality, deadline (@emph{deadline})
-@table @samp
-@item best
-Use best quality deadline. Poorly named and quite slow, this option should be
-avoided as it may give worse quality output than good.
-@item good
-Use good quality deadline. This is a good trade-off between speed and quality
-when used with the @option{cpu-used} option.
-@item realtime
-Use realtime quality deadline.
+@item minrate, maxrate, vb
+@code{VPX_CBR}
+@code{(minrate == maxrate == vb)}
+
+@item crf
+@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
+
+@item quality
+@table @option
+@item @var{best}
+@code{VPX_DL_BEST_QUALITY}
+@item @var{good}
+@code{VPX_DL_GOOD_QUALITY}
+@item @var{realtime}
+@code{VPX_DL_REALTIME}
 @end table
 
-@item speed, cpu-used (@emph{cpu-used})
-Set quality/speed ratio modifier. Higher values speed up the encode at the cost
-of quality.
+@item speed
+@code{VP8E_SET_CPUUSED}
 
-@item nr (@emph{noise-sensitivity})
+@item nr
+@code{VP8E_SET_NOISE_SENSITIVITY}
 
-@item static-thresh
-Set a change threshold on blocks below which they will be skipped by the
-encoder.
+@item mb_threshold
+@code{VP8E_SET_STATIC_THRESHOLD}
 
-@item slices (@emph{token-parts})
-Note that FFmpeg's @option{slices} option gives the total number of partitions,
-while @command{vpxenc}'s @option{token-parts} is given as
-@code{log2(partitions)}.
+@item slices
+@code{VP8E_SET_TOKEN_PARTITIONS}
 
 @item max-intra-rate
-Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0
-means unlimited.
+@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
 
 @item force_key_frames
 @code{VPX_EFLAG_FORCE_KF}
 
 @item Alternate reference frame related
 @table @option
-@item auto-alt-ref
-Enable use of alternate reference frames (2-pass only).
-@item arnr-max-frames
-Set altref noise reduction max frame count.
-@item arnr-type
-Set altref noise reduction filter type: backward, forward, centered.
-@item arnr-strength
-Set altref noise reduction filter strength.
-@item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
-Set number of frames to look ahead for frametype and ratecontrol.
+@item vp8flags altref
+@code{VP8E_SET_ENABLEAUTOALTREF}
+@item @var{arnr_max_frames}
+@code{VP8E_SET_ARNR_MAXFRAMES}
+@item @var{arnr_type}
+@code{VP8E_SET_ARNR_TYPE}
+@item @var{arnr_strength}
+@code{VP8E_SET_ARNR_STRENGTH}
+@item @var{rc_lookahead}
+g_lag_in_frames
 @end table
 
-@item error-resilient
-Enable error resiliency features.
+@item vp8flags error_resilient
+g_error_resilient
 
-@item VP9-specific options
-@table @option
-@item lossless
-Enable lossless mode.
-@item tile-columns
-Set number of tile columns to use. Note this is given as
-@code{log2(tile_columns)}. For example, 8 tile columns would be requested by
-setting the @option{tile-columns} option to 3.
-@item tile-rows
-Set number of tile rows to use. Note this is given as @code{log2(tile_rows)}.
-For example, 4 tile rows would be requested by setting the @option{tile-rows}
-option to 2.
-@item frame-parallel
-Enable frame parallel decodability features.
-@item aq-mode
-Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
-cyclic refresh).
-@item colorspace @emph{color-space}
-Set input color space. The VP9 bitstream supports signaling the following
-colorspaces:
-@table @option
-@item @samp{rgb} @emph{sRGB}
-@item @samp{bt709} @emph{bt709}
-@item @samp{unspecified} @emph{unknown}
-@item @samp{bt470bg} @emph{bt601}
-@item @samp{smpte170m} @emph{smpte170}
-@item @samp{smpte240m} @emph{smpte240}
-@item @samp{bt2020_ncl} @emph{bt2020}
-@end table
-@end table
+@item aq_mode
+@code{VP9E_SET_AQ_MODE}
 
 @end table
 
@@ -2324,30 +2181,6 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
-@section libkvazaar
-
-Kvazaar H.265/HEVC encoder.
-
-Requires the presence of the libkvazaar headers and library during
-configuration. You need to explicitly configure the build with
-@option{--enable-libkvazaar}.
-
-@subsection Options
-
-@table @option
-
-@item b
-Set target video bitrate in bit/s and enable rate control.
-
-@item threads
-Set number of encoding threads.
-
-@item kvazaar-params
-Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
-by commas (,). See kvazaar documentation for a list of options.
-
-@end table
-
 @c man end VIDEO ENCODERS
 
 @chapter Subtitles Encoders

doc/examples/Makefile

@@ -11,14 +11,12 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_dir_cmd                       \
-                avio_reading                       \
+EXAMPLES=       avio_reading                       \
                 decoding_encoding                  \
                 demuxing_decoding                  \
                 extract_mvs                        \
                 filtering_video                    \
                 filtering_audio                    \
-                http_multiclient                   \
                 metadata                           \
                 muxing                             \
                 remuxing                           \

doc/examples/avio_dir_cmd.c

@@ -1,180 +0,0 @@
-/*
- * Copyright (c) 2014 Lukasz Marek
- *
- * 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 <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavformat/avio.h>
-
-static const char *type_string(int type)
-{
-    switch (type) {
-    case AVIO_ENTRY_DIRECTORY:
-        return "<DIR>";
-    case AVIO_ENTRY_FILE:
-        return "<FILE>";
-    case AVIO_ENTRY_BLOCK_DEVICE:
-        return "<BLOCK DEVICE>";
-    case AVIO_ENTRY_CHARACTER_DEVICE:
-        return "<CHARACTER DEVICE>";
-    case AVIO_ENTRY_NAMED_PIPE:
-        return "<PIPE>";
-    case AVIO_ENTRY_SYMBOLIC_LINK:
-        return "<LINK>";
-    case AVIO_ENTRY_SOCKET:
-        return "<SOCKET>";
-    case AVIO_ENTRY_SERVER:
-        return "<SERVER>";
-    case AVIO_ENTRY_SHARE:
-        return "<SHARE>";
-    case AVIO_ENTRY_WORKGROUP:
-        return "<WORKGROUP>";
-    case AVIO_ENTRY_UNKNOWN:
-    default:
-        break;
-    }
-    return "<UNKNOWN>";
-}
-
-static int list_op(const char *input_dir)
-{
-    AVIODirEntry *entry = NULL;
-    AVIODirContext *ctx = NULL;
-    int cnt, ret;
-    char filemode[4], uid_and_gid[20];
-
-    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
-        goto fail;
-    }
-
-    cnt = 0;
-    for (;;) {
-        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
-            goto fail;
-        }
-        if (!entry)
-            break;
-        if (entry->filemode == -1) {
-            snprintf(filemode, 4, "???");
-        } else {
-            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
-        }
-        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
-        if (cnt == 0)
-            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
-                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
-                   "ACCESSED", "STATUS_CHANGED");
-        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
-               type_string(entry->type),
-               entry->size,
-               entry->name,
-               uid_and_gid,
-               filemode,
-               entry->modification_timestamp,
-               entry->access_timestamp,
-               entry->status_change_timestamp);
-        avio_free_directory_entry(&entry);
-        cnt++;
-    };
-
-  fail:
-    avio_close_dir(&ctx);
-    return ret;
-}
-
-static int del_op(const char *url)
-{
-    int ret = avpriv_io_delete(url);
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
-    return ret;
-}
-
-static int move_op(const char *src, const char *dst)
-{
-    int ret = avpriv_io_move(src, dst);
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
-    return ret;
-}
-
-
-static void usage(const char *program_name)
-{
-    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
-            "API example program to show how to manipulate resources "
-            "accessed through AVIOContext.\n"
-            "OPERATIONS:\n"
-            "list      list content of the directory\n"
-            "move      rename content in directory\n"
-            "del       delete content in directory\n",
-            program_name);
-}
-
-int main(int argc, char *argv[])
-{
-    const char *op = NULL;
-    int ret;
-
-    av_log_set_level(AV_LOG_DEBUG);
-
-    if (argc < 2) {
-        usage(argv[0]);
-        return 1;
-    }
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-    avformat_network_init();
-
-    op = argv[1];
-    if (strcmp(op, "list") == 0) {
-        if (argc < 3) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = list_op(argv[2]);
-        }
-    } else if (strcmp(op, "del") == 0) {
-        if (argc < 3) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = del_op(argv[2]);
-        }
-    } else if (strcmp(op, "move") == 0) {
-        if (argc < 4) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = move_op(argv[2], argv[3]);
-        }
-    } else {
-        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
-        ret = AVERROR(EINVAL);
-    }
-
-    avformat_network_deinit();
-
-    return ret < 0 ? 1 : 0;
-}

doc/examples/decoding_encoding.c

@@ -245,7 +245,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     AVCodecContext *c= NULL;
     int len;
     FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
     AVPacket avpkt;
     AVFrame *decoded_frame = NULL;
 
@@ -521,7 +521,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
         /* 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],
-                 frame->width, frame->height, buf);
+                 avctx->width, avctx->height, buf);
         (*frame_count)++;
     }
     if (pkt->data) {
@@ -538,13 +538,13 @@ static void video_decode_example(const char *outfilename, const char *filename)
     int frame_count;
     FILE *f;
     AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
     AVPacket avpkt;
 
     av_init_packet(&avpkt);
 
     /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
-    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
 
     printf("Decode video file %s to %s\n", filename, outfilename);
 
@@ -561,8 +561,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
-        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+    if(codec->capabilities&CODEC_CAP_TRUNCATED)
+        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
 
     /* For some codecs, such as msmpeg4 and mpeg4, width and height
        MUST be initialized there because this information is not

doc/examples/filtering_video.c

@@ -38,10 +38,7 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 
-const char *filter_descr = "scale=78:24,transpose=cclock";
-/* other way:
-   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
- */
+const char *filter_descr = "scale=78:24";
 
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;

doc/examples/http_multiclient.c

@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2015 Stephan Holljes
- *
- * 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 multi-client network API usage example.
- *
- * @example http_multiclient.c
- * This example will serve a file without decoding or demuxing it over http.
- * Multiple clients can connect and will receive the same file.
- */
-
-#include <libavformat/avformat.h>
-#include <libavutil/opt.h>
-#include <unistd.h>
-
-void process_client(AVIOContext *client, const char *in_uri)
-{
-    AVIOContext *input = NULL;
-    uint8_t buf[1024];
-    int ret, n, reply_code;
-    char *resource = NULL;
-    while ((ret = avio_handshake(client)) > 0) {
-        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
-        // check for strlen(resource) is necessary, because av_opt_get()
-        // may return empty string.
-        if (resource && strlen(resource))
-            break;
-    }
-    if (ret < 0)
-        goto end;
-    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
-    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
-        reply_code = 200;
-    } else {
-        reply_code = AVERROR_HTTP_NOT_FOUND;
-    }
-    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
-        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
-        goto end;
-    }
-    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
-
-    while ((ret = avio_handshake(client)) > 0);
-
-    if (ret < 0)
-        goto end;
-
-    fprintf(stderr, "Handshake performed.\n");
-    if (reply_code != 200)
-        goto end;
-    fprintf(stderr, "Opening input file.\n");
-    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
-        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
-               av_err2str(ret));
-        goto end;
-    }
-    for(;;) {
-        n = avio_read(input, buf, sizeof(buf));
-        if (n < 0) {
-            if (n == AVERROR_EOF)
-                break;
-            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
-                   av_err2str(n));
-            break;
-        }
-        avio_write(client, buf, n);
-        avio_flush(client);
-    }
-end:
-    fprintf(stderr, "Flushing client\n");
-    avio_flush(client);
-    fprintf(stderr, "Closing client\n");
-    avio_close(client);
-    fprintf(stderr, "Closing input\n");
-    avio_close(input);
-}
-
-int main(int argc, char **argv)
-{
-    av_log_set_level(AV_LOG_TRACE);
-    AVDictionary *options = NULL;
-    AVIOContext *client = NULL, *server = NULL;
-    const char *in_uri, *out_uri;
-    int ret, pid;
-    if (argc < 3) {
-        printf("usage: %s input http://hostname[:port]\n"
-               "API example program to serve http to multiple clients.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    in_uri = argv[1];
-    out_uri = argv[2];
-
-    av_register_all();
-    avformat_network_init();
-
-    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
-        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
-        return ret;
-    }
-    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
-        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
-        return ret;
-    }
-    fprintf(stderr, "Entering main loop.\n");
-    for(;;) {
-        if ((ret = avio_accept(server, &client)) < 0)
-            goto end;
-        fprintf(stderr, "Accepted client, forking process.\n");
-        // XXX: Since we don't reap our children and don't ignore signals
-        //      this produces zombie processes.
-        pid = fork();
-        if (pid < 0) {
-            perror("Fork failed");
-            ret = AVERROR(errno);
-            goto end;
-        }
-        if (pid == 0) {
-            fprintf(stderr, "In child.\n");
-            process_client(client, in_uri);
-            avio_close(server);
-            exit(0);
-        }
-        if (pid > 0)
-            avio_close(client);
-    }
-end:
-    avio_close(server);
-    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-    return 0;
-}

doc/examples/muxing.c

@@ -172,7 +172,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
 
     /* Some formats want stream headers to be separate. */
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 }
 
 /**************************************************************/
@@ -230,7 +230,7 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
     /* increment frequency by 110 Hz per second */
     ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
+    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
         nb_samples = 10000;
     else
         nb_samples = c->frame_size;

doc/examples/qsvdec.c

@@ -405,7 +405,7 @@ int main(int argc, char **argv)
     decoder_ctx->codec_id = AV_CODEC_ID_H264;
     if (video_st->codec->extradata_size) {
         decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
-                                            AV_INPUT_BUFFER_PADDING_SIZE);
+                                            FF_INPUT_BUFFER_PADDING_SIZE);
         if (!decoder_ctx->extradata) {
             ret = AVERROR(ENOMEM);
             goto finish;

doc/examples/remuxing.c

@@ -101,7 +101,7 @@ int main(int argc, char **argv)
         }
         out_stream->codec->codec_tag = 0;
         if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
     }
     av_dump_format(ofmt_ctx, 0, out_filename, 1);
 

doc/examples/transcode_aac.c

@@ -192,7 +192,7 @@ static int open_output_file(const char *filename,
      * Mark the encoder so that it behaves accordingly.
      */
     if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        (*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) {

doc/examples/transcoding.c

@@ -117,7 +117,7 @@ static int open_output_file(const char *filename)
             /* 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, "Necessary encoder not found\n");
+                av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
                 return AVERROR_INVALIDDATA;
             }
 
@@ -161,7 +161,7 @@ static int open_output_file(const char *filename)
         }
 
         if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
 
     }
     av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -449,7 +449,7 @@ static int flush_encoder(unsigned int stream_index)
     int got_frame;
 
     if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                AV_CODEC_CAP_DELAY))
+                CODEC_CAP_DELAY))
         return 0;
 
     while (1) {

doc/fate.texi

@@ -13,36 +13,36 @@
 
 @chapter Introduction
 
-FATE is an extended regression suite on the client-side and a means
+  FATE is an extended regression suite on the client-side and a means
 for results aggregation and presentation on the server-side.
 
-The first part of this document explains how you can use FATE from
+  The first part of this document explains how you can use FATE from
 your FFmpeg source directory to test your ffmpeg binary. The second
 part describes how you can run FATE to submit the results to FFmpeg's
 FATE server.
 
-In any way you can have a look at the publicly viewable FATE results
+  In any way you can have a look at the publicly viewable FATE results
 by visiting this website:
 
-@url{http://fate.ffmpeg.org/}
+  @url{http://fate.ffmpeg.org/}
 
-This is especially recommended for all people contributing source
+  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
 the developers could not test on.
 
-The second part of this document describes how you can run FATE to
+  The second part of this document describes how you can run FATE to
 submit your results to FFmpeg's FATE server. If you want to submit your
 results be sure to check that your combination of CPU, OS and compiler
 is not already listed on the above mentioned website.
 
-In the third part you can find a comprehensive listing of FATE makefile
+  In the third part you can find a comprehensive listing of FATE makefile
 targets and variables.
 
 
 @chapter Using FATE from your FFmpeg source directory
 
-If you want to run FATE on your machine you need to have the samples
+  If you want to run FATE on your machine you need to have the samples
 in place. You can get the samples via the build target fate-rsync.
 Use this command from the top-level source directory:
 
@@ -51,11 +51,11 @@ make fate-rsync SAMPLES=fate-suite/
 make fate       SAMPLES=fate-suite/
 @end example
 
-The above commands set the samples location by passing a makefile
+  The above commands set the samples location by passing a makefile
 variable via command line. It is also possible to set the samples
 location at source configuration time by invoking configure with
-@option{--samples=<path to the samples directory>}. Afterwards you can
-invoke the makefile targets without setting the @var{SAMPLES} makefile
+`--samples=<path to the samples directory>'. Afterwards you can
+invoke the makefile targets without setting the SAMPLES makefile
 variable. This is illustrated by the following commands:
 
 @example
@@ -64,7 +64,7 @@ make fate-rsync
 make fate
 @end example
 
-Yet another way to tell FATE about the location of the sample
+  Yet another way to tell FATE about the location of the sample
 directory is by making sure the environment variable FATE_SAMPLES
 contains the path to your samples directory. This can be achieved
 by e.g. putting that variable in your shell profile or by setting
@@ -85,7 +85,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to
 
 @chapter Submitting the results to the FFmpeg result aggregation server
 
-To submit your results to the server you should run fate through the
+  To submit your results to the server you should run fate through the
 shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.
 
@@ -93,23 +93,23 @@ to be invoked with a configuration file as its first argument.
 tests/fate.sh /path/to/fate_config
 @end example
 
-A configuration file template with comments describing the individual
+  A configuration file template with comments describing the individual
 configuration variables can be found at @file{doc/fate_config.sh.template}.
 
 @ifhtml
-The mentioned configuration template is also available here:
+  The mentioned configuration template is also available here:
 @verbatiminclude fate_config.sh.template
 @end ifhtml
 
-Create a configuration that suits your needs, based on the configuration
-template. The @env{slot} configuration variable can be any string that is not
+  Create a configuration that suits your needs, based on the configuration
+template. The `slot' configuration variable can be any string that is not
 yet used, but it is suggested that you name it adhering to the following
-pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
-configuration file itself will be sourced in a shell script, therefore all
-shell features may be used. This enables you to setup the environment as you
-need it for your build.
+pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
+itself will be sourced in a shell script, therefore all shell features may
+be used. This enables you to setup the environment as you need it for your
+build.
 
-For your first test runs the @env{fate_recv} variable should be empty or
+  For your first test runs the `fate_recv' variable should be empty or
 commented out. This will run everything as normal except that it will omit
 the submission of the results to the server. The following files should be
 present in $workdir as specified in the configuration file:
@@ -122,29 +122,29 @@ present in $workdir as specified in the configuration file:
     @item version
 @end itemize
 
-When you have everything working properly you can create an SSH key pair
+  When you have everything working properly you can create an SSH key pair
 and send the public key to the FATE server administrator who can be contacted
 at the email address @email{fate-admin@@ffmpeg.org}.
 
-Configure your SSH client to use public key authentication with that key
+  Configure your SSH client to use public key authentication with that key
 when connecting to the FATE server. Also do not forget to check the identity
 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 @samp
+@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
 
-If you have problems connecting to the FATE server, it may help to try out
+  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
 get detailed output concerning your SSH configuration and the authentication
 process.
 
-The only thing left is to automate the execution of the fate.sh script and
+  The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
 
 
@@ -165,7 +165,7 @@ Run the FATE test suite (requires the fate-suite dataset).
 
 @section Makefile variables
 
-@table @env
+@table @option
 @item V
 Verbosity level, can be set to 0, 1 or 2.
     @itemize
@@ -183,20 +183,20 @@ 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 @samp{slice} or @samp{frame},
-by default @samp{slice+frame}
+Specify which threading strategy test, either @var{slice} or @var{frame},
+by default @var{slice+frame}
 
 @item CPUFLAGS
 Specify CPU flags.
 
 @item TARGET_EXEC
 Specify or override the wrapper used to run the tests.
-The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
+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 @samp{1} to generate the missing or mismatched references.
+Set to @var{1} to generate the missing or mismatched references.
 @end table
 
 @section Examples

doc/fate_config.sh.template

@@ -1,6 +1,5 @@
 slot=                                    # some unique identifier
 repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
-#branch=release/2.6                       # the branch to test
 samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report

doc/ffmpeg.texi

@@ -80,7 +80,7 @@ The format option may be needed for raw input files.
 The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:
 
-@verbatim
+@example
  _______              ______________
 |       |            |              |
 | input |  demuxer   | encoded data |   decoder
@@ -99,7 +99,7 @@ the following diagram:
 |________|           |______________|
 
 
-@end verbatim
+@end example
 
 @command{ffmpeg} calls the libavformat library (containing demuxers) to read
 input files and get packets containing encoded data from them. When there are
@@ -124,7 +124,7 @@ Simple filtergraphs are those that have exactly one input and output, both of
 the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:
 
-@verbatim
+@example
  _________                        ______________
 |         |                      |              |
 | decoded |                      | encoded data |
@@ -136,19 +136,19 @@ an additional step between decoding and encoding:
                 | frames   |
                 |__________|
 
-@end verbatim
+@end example
 
 Simple filtergraphs are configured with the per-stream @option{-filter} option
 (with @option{-vf} and @option{-af} aliases for video and audio respectively).
 A simple filtergraph for video can look for example like this:
 
-@verbatim
+@example
  _______        _____________        _______        ________
 |       |      |             |      |       |      |        |
 | input | ---> | deinterlace | ---> | scale | ---> | output |
 |_______|      |_____________|      |_______|      |________|
 
-@end verbatim
+@end example
 
 Note that some filters change frame properties but not frame contents. E.g. the
 @code{fps} filter in the example above changes number of frames, but does not
@@ -161,7 +161,7 @@ processing chain applied to one stream. This is the case, for example, when the
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:
 
-@verbatim
+@example
  _________
 |         |
 | input 0 |\                    __________
@@ -179,7 +179,7 @@ input. They can be represented with the following diagram:
 | input 2 |/
 |_________|
 
-@end verbatim
+@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,
@@ -198,14 +198,14 @@ 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:
 
-@verbatim
+@example
  _______              ______________            ________
 |       |            |              |          |        |
 | input |  demuxer   | encoded data |  muxer   | output |
 | file  | ---------> | packets      | -------> | file   |
 |_______|            |______________|          |________|
 
-@end verbatim
+@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
@@ -280,15 +280,13 @@ 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}.
 
-@var{duration} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+@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} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+@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.
 
@@ -297,8 +295,8 @@ 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 that in most formats it is not possible to seek exactly,
-so @command{ffmpeg} will seek to the closest seek point before @var{position}.
+@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
@@ -307,13 +305,7 @@ 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} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-@item -sseof @var{position} (@emph{input/output})
-
-Like the @code{-ss} option but relative to the "end of file". That is negative
-values are earlier in the file, 0 is at EOF.
+@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.
@@ -328,7 +320,7 @@ the time duration specified in @var{offset}.
 @item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
 
-@var{date} must be a date specification,
+@var{date} must be a time duration specification,
 see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
 @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
@@ -470,9 +462,6 @@ Technical note -- attachments are implemented as codec extradata, so this
 option can actually be used to extract extradata from any stream, not just
 attachments.
 
-@item -noautorotate
-Disable automatically rotating video based on file metadata.
-
 @end table
 
 @section Video Options
@@ -698,10 +687,6 @@ is not specified, the value of the @var{DISPLAY} environment variable is used
 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
-
-@item -hwaccels
-List all hardware acceleration methods supported in this build of ffmpeg.
-
 @end table
 
 @section Audio Options
@@ -847,14 +832,6 @@ ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
 
 Note that using this option disables the default mappings for this output file.
 
-@item -ignore_unknown
-Ignore input streams with unknown type instead of failing if copying
-such streams is attempted.
-
-@item -copy_unknown
-Allow input streams with unknown type to be copied instead of failing if copying
-such streams is attempted.
-
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 Map an audio channel from a given input to an output. If
 @var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
@@ -1181,14 +1158,7 @@ This option enables or disables accurate seeking in input files with the
 transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
 e.g. when copying some streams and transcoding the others.
 
-@item -seek_timestamp (@emph{input})
-This option enables or disables seeking by timestamp in input files with the
-@option{-ss} option. It is disabled by default. If enabled, the argument
-to the @option{-ss} option is considered an actual timestamp, and is not
-offset by the start time of the file. This matters only for files which do
-not start from timestamp 0, such as transport streams.
-
-@item -thread_queue_size @var{size} (@emph{input})
+@item -thread_message_queue @var{size} (@emph{input})
 This option sets the maximum number of queued packets when reading from the
 file or device. With low latency / high rate live streams, packets may be
 discarded if they are not read in a timely manner; raising this value can
@@ -1205,9 +1175,9 @@ The option is intended for cases where features are needed that cannot be
 specified to @command{ffserver} but can be to @command{ffmpeg}.
 
 @item -sdp_file @var{file} (@emph{global})
-Print sdp information to @var{file}.
+Print sdp information for an output stream to @var{file}.
 This allows dumping sdp information when at least one output isn't an
-rtp stream.
+rtp stream. (Requires at least one of the output formats to be rtp).
 
 @item -discard (@emph{input})
 Allows discarding specific streams or frames of streams at the demuxer.
@@ -1233,9 +1203,6 @@ Discard all frames excepts keyframes.
 Discard all frames.
 @end table
 
-@item -xerror (@emph{global})
-Stop and exit on error
-
 @end table
 
 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1311,6 +1278,47 @@ If no such file is found, then ffmpeg will search for a file named
 
 @c man end OPTIONS
 
+@chapter Tips
+@c man begin TIPS
+
+@itemize
+@item
+For streaming at very low bitrates, 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:
+
+@example
+ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
+@end example
+
+@item
+The parameter 'q' which is displayed while encoding is the current
+quantizer. The value 1 indicates that a very good quality could
+be achieved. The value 31 indicates the worst quality. If q=31 appears
+too often, it means that the encoder cannot compress enough to meet
+your bitrate. You must either increase the bitrate, decrease the
+frame rate or decrease the frame size.
+
+@item
+If your computer is not fast enough, you can speed up the
+compression at the expense of the compression ratio. You can use
+'-me zero' to speed up motion estimation, and '-g 0' to disable
+motion estimation completely (you have only I-frames, which means it
+is about as good as JPEG compression).
+
+@item
+To have very low audio bitrates, reduce the sampling frequency
+(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
+
+@item
+To have a constant quality (but a variable bitrate), use the option
+'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
+quality).
+
+@end itemize
+@c man end TIPS
+
 @chapter Examples
 @c man begin EXAMPLES
 
@@ -1461,7 +1469,7 @@ combination with -ss to start extracting from a certain point in time.
 
 For creating a video from many images:
 @example
-ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
+ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
 @end example
 
 The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@@ -1476,7 +1484,7 @@ image2-specific @code{-pattern_type glob} option.
 For example, for creating a video from filenames matching the glob pattern
 @code{foo-*.jpeg}:
 @example
-ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
+ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
 @end example
 
 @item

doc/ffplay.texi

@@ -47,17 +47,9 @@ Disable video.
 @item -sn
 Disable subtitles.
 @item -ss @var{pos}
-Seek to @var{pos}. Note that in most formats it is not possible to seek
-exactly, so @command{ffplay} will seek to the nearest seek point to
-@var{pos}.
-
-@var{pos} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+Seek to a given position in seconds.
 @item -t @var{duration}
-Play @var{duration} seconds of audio/video.
-
-@var{duration} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+play <duration> seconds of audio/video
 @item -bytes
 Seek by bytes.
 @item -nodisp
@@ -169,7 +161,7 @@ Force a specific video decoder.
 Force a specific subtitle decoder.
 
 @item -autorotate
-Automatically rotate the video according to file metadata. Enabled by
+Automatically rotate the video according to presentation metadata. Enabled by
 default, use @option{-noautorotate} to disable it.
 
 @item -framedrop

doc/ffprobe.texi

@@ -447,17 +447,17 @@ writer).
 It can assume one of the following values:
 @table @option
 @item c
-Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
-return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
-character (@samp{\}) or the item separator character @var{SEP} are escaped
-using C-like fashioned escaping, so that a newline is converted to the
-sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
-the separator @var{SEP} is converted to @samp{\@var{SEP}}.
+Perform C-like escaping. Strings containing a newline ('\n'), carriage
+return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
+character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
+escaping, so that a newline is converted to the sequence "\n", a
+carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
+converted to "\@var{SEP}".
 
 @item csv
 Perform CSV-like escaping, as described in RFC4180.  Strings
-containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
-(@samp{"}), or @var{SEP} are enclosed in double-quotes.
+containing a newline ('\n'), a carriage return ('\r'), a double quote
+('"'), or @var{SEP} are enclosed in double-quotes.
 
 @item none
 Perform no escaping.
@@ -485,7 +485,7 @@ The description of the accepted options follows.
 Separator character used to separate the chapter, the section name, IDs and
 potential tags in the printed field key.
 
-Default value is @samp{.}.
+Default value is '.'.
 
 @item hierarchical, h
 Specify if the section name specification should be hierarchical. If
@@ -507,22 +507,21 @@ The following conventions are adopted:
 @item
 all key and values are UTF-8
 @item
-@samp{.} is the subgroup separator
+'.' is the subgroup separator
 @item
-newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
-escaped
+newline, '\t', '\f', '\b' and the following characters are escaped
 @item
-@samp{\} is the escape character
+'\' is the escape character
 @item
-@samp{#} is the comment indicator
+'#' is the comment indicator
 @item
-@samp{=} is the key/value separator
+'=' is the key/value separator
 @item
-@samp{:} is not used but usually parsed as key/value separator
+':' is not used but usually parsed as key/value separator
 @end itemize
 
 This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by @samp{:}.
+separated by ":".
 
 The description of the accepted options follows.
 

doc/ffprobe.xsd

@@ -48,11 +48,6 @@
     </xsd:complexType>
 
     <xsd:complexType name="packetType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
-      </xsd:sequence>
-
       <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
       <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
       <xsd:attribute name="pts"           type="xsd:long"  />
@@ -70,16 +65,6 @@
       <xsd:attribute name="data_hash"     type="xsd:string" />
     </xsd:complexType>
 
-    <xsd:complexType name="packetSideDataListType">
-        <xsd:sequence>
-            <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-    <xsd:complexType name="packetSideDataType">
-        <xsd:attribute name="side_data_type"              type="xsd:string"/>
-        <xsd:attribute name="side_data_size"              type="xsd:int"   />
-    </xsd:complexType>
-
     <xsd:complexType name="frameType">
       <xsd:sequence>
             <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -87,7 +72,6 @@
       </xsd:sequence>
 
       <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="stream_index"  type="xsd:int"  />
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
       <xsd:attribute name="pts_time"      type="xsd:float"/>
@@ -171,7 +155,6 @@
       <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:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
       </xsd:sequence>
 
       <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -188,8 +171,6 @@
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
       <xsd:attribute name="height"               type="xsd:int"/>
-      <xsd:attribute name="coded_width"          type="xsd:int"/>
-      <xsd:attribute name="coded_height"         type="xsd:int"/>
       <xsd:attribute name="has_b_frames"         type="xsd:int"/>
       <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -273,8 +254,6 @@
     <xsd:complexType name="programVersionType">
       <xsd:attribute name="version"          type="xsd:string" use="required"/>
       <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string"/>
-      <xsd:attribute name="build_time"       type="xsd:string"/>
       <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
       <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
     </xsd:complexType>

doc/ffserver.conf

@@ -82,7 +82,6 @@ Feed feed1.ffm
 # ra         : RealNetworks-compatible stream. Audio only.
 # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
 # jpeg       : Generate a single JPEG image.
-# mjpeg      : Generate a M-JPEG stream.
 # asf        : ASF compatible streaming (Windows Media Player format).
 # swf        : Macromedia Flash compatible stream
 # avi        : AVI format (MPEG-4 video, MPEG audio sound)

doc/ffserver.texi

@@ -118,8 +118,7 @@ Multiple streams can be connected to the same feed.
 
 For example, you can have a situation described by the following
 graph:
-
-@verbatim
+@example
                _________       __________
               |         |     |          |
 ffmpeg 1 -----| feed 1  |-----| stream 1 |
@@ -144,8 +143,7 @@ ffmpeg 2 -----| feed 3  |-----| stream 4 |
               |         |     |          |
               | file 1  |-----| stream 5 |
               |_________|     |__________|
-
-@end verbatim
+@end example
 
 @anchor{FFM}
 @section FFM, FFM2 formats

doc/fftools-common-opts.texi

@@ -36,10 +36,8 @@ Possible forms of stream specifiers are:
 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' or 'V' for video, 'a' for audio, 's'
-for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
-streams, 'V' only matches video streams which are not attached pictures, video
-thumbnails or cover arts.  If @var{stream_index} is given, then it matches
+@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}]
@@ -52,9 +50,6 @@ Match the stream by stream id (e.g. PID in MPEG-TS container).
 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.
-@item u
-Matches streams with usable configuration, the codec must be defined and the
-essential information such as video dimension or audio sample rate must be present.
 
 Note that in @command{ffmpeg}, matching by metadata will only work properly for
 input files.
@@ -192,7 +187,6 @@ warnings and errors. This is the default value.
 Same as @code{info}, except more verbose.
 @item debug, 48
 Show everything, including debugging information.
-@item trace, 56
 @end table
 
 By default the program logs to stderr, if coloring is supported by the

doc/filter_design.txt

@@ -98,7 +98,7 @@ Buffer references ownership and permissions
     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 and partial_buf are used by libavfilter internally
+    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
     and must not be accessed by filters.
 
   Reference permissions

doc/formats.texi

@@ -37,8 +37,6 @@ Possible values:
 @table @samp
 @item ignidx
 Ignore index.
-@item fastseek
-Enable fast, but inaccurate seeks for some formats.
 @item genpts
 Generate PTS.
 @item nofillin
@@ -127,25 +125,6 @@ Consider all spec non compliancies as errors.
 Consider things that a sane encoder should not do as an error.
 @end table
 
-@item max_interleave_delta @var{integer} (@emph{output})
-Set maximum buffering duration for interleaving. The duration is
-expressed in microseconds, and defaults to 1000000 (1 second).
-
-To ensure all the streams are interleaved correctly, libavformat will
-wait until it has at least one packet for each stream before actually
-writing any packets to the output file. When some streams are
-"sparse" (i.e. there are large gaps between successive packets), this
-can result in excessive buffering.
-
-This field specifies the maximum difference between the timestamps of the
-first and the last packet in the muxing queue, above which libavformat
-will output a packet regardless of whether it has queued a packet for all
-the streams.
-
-If set to 0, libavformat will continue buffering packets until it has
-a packet for each stream, regardless of the maximum timestamp
-difference between the buffered packets.
-
 @item use_wallclock_as_timestamps @var{integer} (@emph{input})
 Use wallclock as timestamps.
 

doc/general.texi

@@ -145,14 +145,6 @@ x265 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 kvazaar
-
-FFmpeg can make use of the kvazaar library for HEVC encoding.
-
-Go to @url{https://github.com/ultravideo/kvazaar} and follow the
-instructions for installing the library. Then pass
-@code{--enable-libkvazaar} to configure to enable it.
-
 @section libilbc
 
 iLBC is a narrowband speech codec that has been made freely available
@@ -187,8 +179,8 @@ 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 2.6 RC1 or higher} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
+@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}.
@@ -200,17 +192,6 @@ end user having AviSynth or AvxSynth installed - they'll only need to be
 installed to use AviSynth scripts (obviously).
 @end float
 
-@section Intel QuickSync Video
-
-FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
-of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
-dispatcher, which loads the actual decoding libraries.
-
-The dispatcher is open source and can be downloaded from
-@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
-with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
-locate the dispatcher's @code{.pc} files.
-
 
 @chapter Supported File Formats, Codecs or Features
 
@@ -226,10 +207,6 @@ library:
 @item 4xm                       @tab   @tab X
     @tab 4X Technologies format, used in some games.
 @item 8088flex TMV              @tab   @tab X
-@item AAX                       @tab   @tab X
-    @tab Audible Enhanced Audio format, used in audiobooks.
-@item AA                        @tab   @tab X
-    @tab Audible Format 2, 3, and 4, used in audiobooks.
 @item ACT Voice                 @tab   @tab X
     @tab contains G.729 audio
 @item Adobe Filmstrip           @tab X @tab X
@@ -245,7 +222,6 @@ library:
     @tab Audio format used on the Nintendo Gamecube.
 @item AFC                       @tab   @tab X
     @tab Audio format used on the Nintendo Gamecube.
-@item APNG                      @tab X @tab X
 @item ASF                       @tab X @tab X
 @item AST                       @tab X @tab X
     @tab Audio format used on the Nintendo Wii.
@@ -266,8 +242,6 @@ 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 BFSTM                     @tab   @tab X
-    @tab Audio format used on the Nintendo WiiU (based on BRSTM).
 @item BRSTM                     @tab   @tab X
     @tab Audio format used on the Nintendo Wii.
 @item BWF                       @tab X @tab X
@@ -279,8 +253,6 @@ library:
 @item Delphine Software International CIN @tab   @tab X
     @tab Multimedia format used by Delphine Software games.
 @item Digital Speech Standard (DSS) @tab   @tab X
-@item Canopus HQ                @tab   @tab X
-@item Canopus HQA               @tab   @tab X
 @item Canopus HQX               @tab   @tab X
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
@@ -298,7 +270,6 @@ library:
 @item Deluxe Paint Animation    @tab   @tab X
 @item DFA                       @tab   @tab X
     @tab This format is used in Chronomaster game
-@item DirectDraw Surface        @tab   @tab X
 @item DSD Stream File (DSF)     @tab   @tab X
 @item DV video                  @tab X @tab X
 @item DXA                       @tab   @tab X
@@ -495,7 +466,6 @@ library:
 @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 TDSC                      @tab   @tab X
 @item Text files                @tab   @tab X
 @item THP                       @tab   @tab X
     @tab Used on the Nintendo GameCube.
@@ -503,7 +473,6 @@ 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 Vidvox Hap                @tab X @tab X
 @item Vivo                      @tab   @tab X
 @item WAV                       @tab X @tab X
 @item WavPack                   @tab X @tab X
@@ -538,7 +507,6 @@ following image formats are supported:
 @item Alias PIX    @tab X @tab X
     @tab Alias/Wavefront PIX image format
 @item animated GIF @tab X @tab X
-@item APNG         @tab X @tab X
 @item BMP          @tab X @tab X
     @tab Microsoft BMP image
 @item BRender PIX  @tab   @tab X
@@ -690,8 +658,6 @@ following image formats are supported:
     @tab Sorenson H.263 used in Flash
 @item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
-@item Go2Meeting             @tab     @tab  X
-    @tab fourcc: G2M2, G2M3
 @item Go2Webinar             @tab     @tab  X
     @tab fourcc: G2M4
 @item H.261                  @tab  X  @tab  X
@@ -700,7 +666,7 @@ following image formats are supported:
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
     @tab encoding supported through external library libx264 and OpenH264
 @item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through external library libx265 and libkvazaar
+    @tab encoding supported through the external library libx265
 @item HNM version 4          @tab     @tab  X
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
@@ -865,7 +831,7 @@ following image formats are supported:
 @item Name @tab Encoding @tab Decoding @tab Comments
 @item 8SVX exponential       @tab     @tab  X
 @item 8SVX fibonacci         @tab     @tab  X
-@item AAC+                   @tab  E  @tab  IX
+@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
@@ -905,7 +871,7 @@ following image formats are supported:
 @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 THP  @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.
@@ -939,7 +905,6 @@ following image formats are supported:
 @item COOK                   @tab     @tab  X
     @tab All versions except 5.1 are supported.
 @item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X
-    @tab supported extensions: XCh, XLL (partially)
 @item DPCM id RoQ            @tab  X  @tab  X
     @tab Used in Quake III, Jedi Knight 2 and other computer games.
 @item DPCM Interplay         @tab     @tab  X
@@ -983,8 +948,8 @@ following image formats are supported:
 @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  E
-    @tab supported through external library libopus
+@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

doc/git-howto.texi

@@ -1,10 +1,10 @@
 \input texinfo @c -*- texinfo -*-
 @documentencoding UTF-8
 
-@settitle Using Git to develop FFmpeg
+@settitle Using git to develop FFmpeg
 
 @titlepage
-@center @titlefont{Using Git to develop FFmpeg}
+@center @titlefont{Using git to develop FFmpeg}
 @end titlepage
 
 @top
@@ -13,9 +13,9 @@
 
 @chapter Introduction
 
-This document aims in giving some quick references on a set of useful Git
+This document aims in giving some quick references on a set of useful git
 commands. You should always use the extensive and detailed documentation
-provided directly by Git:
+provided directly by git:
 
 @example
 git --help
@@ -32,21 +32,22 @@ man git-<command>
 shows information about the subcommand <command>.
 
 Additional information could be found on the
-@url{http://gitref.org, Git Reference} website.
+@url{http://gitref.org, Git Reference} website
 
 For more information about the Git project, visit the
-@url{http://git-scm.com/, Git website}.
+
+@url{http://git-scm.com/, Git website}
 
 Consult these resources whenever you have problems, they are quite exhaustive.
 
 What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines to ease the contribution to the project.
+guidelines to ease the contribution to the project
 
 @chapter Basics Usage
 
-@section Get Git
+@section Get GIT
 
-You can get Git from @url{http://git-scm.com/}
+You can get git from @url{http://git-scm.com/}
 Most distribution and operating system provide a package for it.
 
 
@@ -74,7 +75,6 @@ git config --global core.autocrlf false
 @end example
 
 
-@anchor{Updating the source tree to the latest revision}
 @section Updating the source tree to the latest revision
 
 @example
@@ -107,7 +107,7 @@ git add [-A] <filename/dirname>
 git rm [-r] <filename/dirname>
 @end example
 
-Git needs to get notified of all changes you make to your working
+GIT needs to get notified of all changes you make to your working
 directory that makes files appear or disappear.
 Line moves across files are automatically tracked.
 
@@ -127,8 +127,8 @@ will show all local modifications in your working directory as unified diff.
 git log <filename(s)>
 @end example
 
-You may also use the graphical tools like @command{gitview} or @command{gitk}
-or the web interface available at @url{http://source.ffmpeg.org/}.
+You may also use the graphical tools like gitview or gitk or the web
+interface available at http://source.ffmpeg.org/
 
 @section Checking source tree status
 
@@ -149,7 +149,6 @@ git diff --check
 to double check your changes before committing them to avoid trouble later
 on. All experienced developers do this on each and every commit, no matter
 how small.
-
 Every one of them has been saved from looking like a fool by this many times.
 It's very easy for stray debug output or cosmetic modifications to slip in,
 please avoid problems through this extra level of scrutiny.
@@ -172,14 +171,14 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
 @end example
 
-Make sure you have told Git your name and email address
+Make sure you have told git your name and email address
 
 @example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
 @end example
 
-Use @option{--global} to set the global configuration for all your Git checkouts.
+Use @var{--global} to set the global configuration for all your git checkouts.
 
 Git will select the changes to the files for commit. Optionally you can use
 the interactive or the patch mode to select hunk by hunk what should be
@@ -210,7 +209,7 @@ include filenames in log messages, Git provides that information.
 
 Possibly make the commit message have a terse, descriptive first line, an
 empty line and then a full description. The first line will be used to name
-the patch by @command{git format-patch}.
+the patch by git format-patch.
 
 @section Preparing a patchset
 
@@ -326,14 +325,12 @@ faulty commit disappear from the history.
 @section Pushing changes to remote trees
 
 @example
-git push origin master --dry-run
+git push
 @end example
 
-Will simulate a push of the local master branch to the default remote
-(@var{origin}). And list which branches and ranges or commits would have been
-pushed.
+Will push the changes to the default remote (@var{origin}).
 Git will prevent you from pushing changes if the local and remote trees are
-out of sync. Refer to @ref{Updating the source tree to the latest revision}.
+out of sync. Refer to and to sync the local tree.
 
 @example
 git remote add <name> <url>
@@ -352,24 +349,23 @@ branches matching the local ones.
 
 @section Finding a specific svn revision
 
-Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits
+Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
 based on a regular expression. see man gitrevisions
 
 @example
 git show :/'as revision 23456'
 @end example
 
-will show the svn changeset @samp{r23456}. With older Git versions searching in
+will show the svn changeset @var{r23456}. With older git versions searching in
 the @command{git log} output is the easiest option (especially if a pager with
 search capabilities is used).
-
 This commit can be checked out with
 
 @example
 git checkout -b svn_23456 :/'as revision 23456'
 @end example
 
-or for Git < 1.7.1 with
+or for git < 1.7.1 with
 
 @example
 git checkout -b svn_23456 $SHA1
@@ -378,7 +374,7 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
 
-@chapter Pre-push checklist
+@chapter pre-push checklist
 
 Once you have a set of commits that you feel are ready for pushing,
 work through the following checklist to doublecheck everything is in
@@ -389,7 +385,7 @@ Apply your common sense, but if in doubt, err on the side of caution.
 First, make sure that the commits and branches you are going to push
 match what you want pushed and that nothing is missing, extraneous or
 wrong. You can see what will be pushed by running the git push command
-with @option{--dry-run} first. And then inspecting the commits listed with
+with --dry-run first. And then inspecting the commits listed with
 @command{git log -p 1234567..987654}. The @command{git status} command
 may help in finding local changes that have been forgotten to be added.
 
@@ -398,7 +394,7 @@ Next let the code pass through a full run of our testsuite.
 @itemize
 @item @command{make distclean}
 @item @command{/path/to/ffmpeg/configure}
-@item @command{make fate}
+@item @command{make check}
 @item if fate fails due to missing samples run @command{make fate-rsync} and retry
 @end itemize
 
@@ -416,5 +412,5 @@ recommended.
 
 @chapter Server Issues
 
-Contact the project admins at @email{root@@ffmpeg.org} if you have technical
-problems with the Git server.
+Contact the project admins @email{root@@ffmpeg.org} if you have technical
+problems with the GIT server.

doc/indevs.texi

@@ -51,18 +51,6 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
 For more information see:
 @url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
 
-@subsection Options
-
-@table @option
-
-@item sample_rate
-Set the sample rate in Hz. Default is 48000.
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
 @section avfoundation
 
 AVFoundation input device.
@@ -126,19 +114,6 @@ und the first one in this list is used instead. Available pixel formats are:
  bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
  yuv420p, nv12, yuyv422, gray}
 
-@item -framerate
-Set the grabbing frame rate. Default is @code{ntsc}, corresponding to a
-frame rate of @code{30000/1001}.
-
-@item -video_size
-Set the video frame size.
-
-@item -capture_cursor
-Capture the mouse pointer. Default is 0.
-
-@item -capture_mouse_clicks
-Capture the screen mouse clicks. Default is 0.
-
 @end table
 
 @subsection Examples
@@ -175,36 +150,6 @@ $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
 
 BSD video input device.
 
-@subsection Options
-
-@table @option
-
-@item framerate
-Set the frame rate.
-
-@item video_size
-Set the video frame size. Default is @code{vga}.
-
-@item standard
-
-Available values are:
-@table @samp
-@item pal
-
-@item ntsc
-
-@item secam
-
-@item paln
-
-@item palm
-
-@item ntscj
-
-@end table
-
-@end table
-
 @section decklink
 
 The decklink input device provides capture capabilities for Blackmagic
@@ -266,6 +211,18 @@ Capture video clip at 1080i50 10 bit:
 ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
 @end example
 
+@item
+Capture video clip at 720p50 with 32bit audio:
+@example
+ffmpeg -bm_audiodepth 32 -f decklink -i 'UltraStudio Mini Recorder@@14' -acodec copy -vcodec copy output.avi
+@end example
+
+@item
+Capture video clip at 576i50 with 8 audio channels:
+@example
+ffmpeg -bm_channels 8 -f decklink -i 'UltraStudio Mini Recorder@@3' -acodec copy -vcodec copy output.avi
+@end example
+
 @end itemize
 
 @section dshow
@@ -318,11 +275,11 @@ If set to @option{true}, print a list of selected device's options
 and exit.
 
 @item video_device_number
-Set video device number for devices with the same name (starts at 0,
+Set video device number for devices with same name (starts at 0,
 defaults to 0).
 
 @item audio_device_number
-Set audio device number for devices with the same name (starts at 0,
+Set audio device number for devices with same name (starts at 0,
 defaults to 0).
 
 @item pixel_format
@@ -393,30 +350,6 @@ If set to @option{true}, before capture starts, popup a display
 dialog to the end user, allowing them to manually
 modify TV audio (like mono vs. stereo, Language A,B or C).
 
-@item audio_device_load
-Load an audio capture filter device from file instead of searching
-it by name. It may load additional parameters too, if the filter
-supports the serialization of its properties to.
-To use this an audio capture source has to be specified, but it can
-be anything even fake one.
-
-@item audio_device_save
-Save the currently used audio capture filter device and its
-parameters (if the filter supports it) to a file.
-If a file with the same name exists it will be overwritten.
-
-@item video_device_load
-Load a video capture filter device from file instead of searching
-it by name. It may load additional parameters too, if the filter
-supports the serialization of its properties to.
-To use this a video capture source has to be specified, but it can
-be anything even fake one.
-
-@item video_device_save
-Save the currently used video capture filter device and its
-parameters (if the filter supports it) to a file.
-If a file with the same name exists it will be overwritten.
-
 @end table
 
 @subsection Examples
@@ -472,27 +405,6 @@ $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_numbe
 
 Linux DV 1394 input device.
 
-@subsection Options
-
-@table @option
-
-@item framerate
-Set the frame rate. Default is 25.
-
-@item standard
-
-Available values are:
-@table @samp
-@item pal
-
-@item ntsc
-
-@end table
-
-Default value is @code{ntsc}.
-
-@end table
-
 @section fbdev
 
 Linux framebuffer input device.
@@ -505,27 +417,18 @@ console. It is accessed through a file device node, usually
 For more detailed information read the file
 Documentation/fb/framebuffer.txt included in the Linux source tree.
 
-See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
-
 To record from the framebuffer device @file{/dev/fb0} with
 @command{ffmpeg}:
 @example
-ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
+ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
 @end example
 
 You can take a single screenshot image with the command:
 @example
-ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
+ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
 @end example
 
-@subsection Options
-
-@table @option
-
-@item framerate
-Set the frame rate. Default is 25.
-
-@end table
+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
 @section gdigrab
 
@@ -711,15 +614,6 @@ $ jack_connect metro:120_bpm ffmpeg:input_1
 For more information read:
 @url{http://jackaudio.org/}
 
-@subsection Options
-
-@table @option
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
 @section lavfi
 
 Libavfilter input virtual device.
@@ -760,9 +654,6 @@ Set the filename of the filtergraph to be read and sent to the other
 filters. Syntax of the filtergraph is the same as the one specified by
 the option @var{graph}.
 
-@item dumpgraph
-Dump graph to stderr.
-
 @end table
 
 @subsection Examples
@@ -964,19 +855,6 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
 @url{http://manuals.opensound.com/usersguide/dsp.html}
 
-@subsection Options
-
-@table @option
-
-@item sample_rate
-Set the sample rate in Hz. Default is 48000.
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
-
 @section pulse
 
 PulseAudio input device.
@@ -1017,10 +895,6 @@ Specify the number of bytes per frame, by default it is set to 1024.
 @item fragment_size
 Specify the minimal buffering fragment in PulseAudio, it will affect the
 audio latency. By default it is unset.
-
-@item wallclock
-Set the initial PTS using the current time. Default is 1.
-
 @end table
 
 @subsection Examples
@@ -1056,22 +930,6 @@ ffmpeg -f qtkit -i "default" out.mpg
 ffmpeg -f qtkit -list_devices true -i ""
 @end example
 
-@subsection Options
-
-@table @option
-
-@item frame_rate
-Set frame rate. Default is 30.
-
-@item list_devices
-If set to @code{true}, print a list of devices and exit. Default is
-@code{false}.
-
-@item video_device_index
-Select the video device by index for devices with the same name (starts at 0).
-
-@end table
-
 @section sndio
 
 sndio input device.
@@ -1089,18 +947,6 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
 @end example
 
-@subsection Options
-
-@table @option
-
-@item sample_rate
-Set the sample rate in Hz. Default is 48000.
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
 @section video4linux2, v4l2
 
 Video4Linux2 input video device.
@@ -1133,12 +979,6 @@ conversion into the real time clock.
 Some usage examples of the video4linux2 device with @command{ffmpeg}
 and @command{ffplay}:
 @itemize
-@item
-List supported formats for a video4linux2 device:
-@example
-ffplay -f video4linux2 -list_formats all /dev/video0
-@end example
-
 @item
 Grab and show the input of a video4linux2 device:
 @example
@@ -1223,10 +1063,6 @@ Force conversion from monotonic to absolute timestamps.
 @end table
 
 Default value is @code{default}.
-
-@item use_libv4l2
-Use libv4l2 (v4l-utils) conversion functions. Default is 0.
-
 @end table
 
 @section vfwcap
@@ -1237,19 +1073,6 @@ The filename passed as input is the capture driver number, ranging from
 0 to 9. You may use "list" as filename to print a list of drivers. Any
 other filename will be interpreted as device number 0.
 
-@subsection Options
-
-@table @option
-
-@item video_size
-Set the video frame size.
-
-@item framerate
-Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a frame rate of @code{30000/1001}.
-
-@end table
-
 @section x11grab
 
 X11 video input device.

doc/metadata.texi

@@ -12,10 +12,10 @@ A file consists of a header and a number of metadata tags divided into sections,
 each on its own line.
 
 @item
-The header is a @samp{;FFMETADATA} string, followed by a version number (now 1).
+The header is a ';FFMETADATA' string, followed by a version number (now 1).
 
 @item
-Metadata tags are of the form @samp{key=value}
+Metadata tags are of the form 'key=value'
 
 @item
 Immediately after header follows global metadata
@@ -26,30 +26,26 @@ metadata.
 
 @item
 A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
-brackets (@samp{[}, @samp{]}) and ends with next section or end of file.
+brackets ('[', ']') and ends with next section or end of file.
 
 @item
 At the beginning of a chapter section there may be an optional timebase to be
-used for start/end values. It must be in form
-@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
-integers. If the timebase is missing then start/end times are assumed to
+used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
+den are integers. If the timebase is missing then start/end times are assumed to
 be in milliseconds.
-
 Next a chapter section must contain chapter start and end times in form
-@samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive
-integer.
+'START=num', 'END=num', where num is a positive integer.
 
 @item
-Empty lines and lines starting with @samp{;} or @samp{#} are ignored.
+Empty lines and lines starting with ';' or '#' are ignored.
 
 @item
-Metadata keys or values containing special characters (@samp{=}, @samp{;},
-@samp{#}, @samp{\} and a newline) must be escaped with a backslash @samp{\}.
+Metadata keys or values containing special characters ('=', ';', '#', '\' and a
+newline) must be escaped with a backslash '\'.
 
 @item
-Note that whitespace in metadata (e.g. @samp{foo = bar}) is considered to be
-a part of the tag (in the example above key is @samp{foo }, value is
-@samp{ bar}).
+Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
+the tag (in the example above key is 'foo ', value is ' bar').
 @end enumerate
 
 A ffmetadata file might look like this:

doc/mips.txt

@@ -47,16 +47,12 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
       float_dsp_mips.c
       libm_mips.h
-      softfloat_tables.h
 * libavcodec/
       fft_fixed_32.c
       fft_init_table.c
       fft_table.h
       mdct_fixed_32.c
 * libavcodec/mips/
-      aacdec_fixed.c
-      aacsbr_fixed.c
-      aacsbr_template.c
       aaccoder_mips.c
       aacpsy_mips.h
       ac3dsp_mips.c

doc/multithreading.txt

@@ -54,7 +54,7 @@ thread.
 If the codec allocates writable tables in its init(), add an init_thread_copy()
 which re-allocates them for other threads.
 
-Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
+Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.
 
 If there are inter-frame dependencies, so the codec calls

doc/muxers.texi

@@ -263,62 +263,6 @@ ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
 This example will produce the playlist, @file{out.m3u8}, and segment files:
 @file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc.
 
-@item hls_key_info_file @var{key_info_file}
-Use the information in @var{key_info_file} for segment encryption. The first
-line of @var{key_info_file} specifies the key URI written to the playlist. The
-key URL is used to access the encryption key during playback. The second line
-specifies the path to the key file used to obtain the key during the encryption
-process. The key file is read as a single packed array of 16 octets in binary
-format. The optional third line specifies the initialization vector (IV) as a
-hexadecimal string to be used instead of the segment sequence number (default)
-for encryption. Changes to @var{key_info_file} will result in segment
-encryption with the new key/IV and an entry in the playlist for the new key
-URI/IV.
-
-Key info file format:
-@example
-@var{key URI}
-@var{key file path}
-@var{IV} (optional)
-@end example
-
-Example key URIs:
-@example
-http://server/file.key
-/path/to/file.key
-file.key
-@end example
-
-Example key file paths:
-@example
-file.key
-/path/to/file.key
-@end example
-
-Example IV:
-@example
-0123456789ABCDEF0123456789ABCDEF
-@end example
-
-Key info file example:
-@example
-http://server/file.key
-/path/to/file.key
-0123456789ABCDEF0123456789ABCDEF
-@end example
-
-Example shell script:
-@example
-#!/bin/sh
-BASE_URL=$@{1:-'.'@}
-openssl rand 16 > file.key
-echo $BASE_URL/file.key > file.keyinfo
-echo file.key >> file.keyinfo
-echo $(openssl rand -hex 16) >> file.keyinfo
-ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
-  -hls_key_info_file file.keyinfo out.m3u8
-@end example
-
 @item hls_flags single_file
 If this flag is set, the muxer will store all segments in a single MPEG-TS
 file, and will use byte ranges in the playlist. HLS playlists generated with
@@ -667,13 +611,6 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
 @end example
 
-@subsection Audible AAX
-
-Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
-@example
-ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
-@end example
-
 @section mp3
 
 The MP3 muxer writes a raw MP3 stream with the following optional features:
@@ -766,10 +703,6 @@ Set a constant muxrate (default VBR).
 @item -pcr_period @var{numer}
 Override the default PCR retransmission time (default 20ms), ignored
 if variable muxrate is selected.
-@item pat_period @var{number}
-Maximal time in seconds between PAT/PMT tables.
-@item sdt_period @var{number}
-Maximal time in seconds between SDT tables.
 @item -pes_payload_size @var{number}
 Set minimum PES packet payload in bytes.
 @item -mpegts_flags @var{flags}
@@ -821,8 +754,6 @@ Option mpegts_flags may take a set of such flags:
 Reemit PAT/PMT before writing the next packet.
 @item latm
 Use LATM packetization for AAC.
-@item pat_pmt_at_frames
-Reemit PAT and PMT at each video frame.
 @end table
 
 @subsection Example
@@ -977,6 +908,13 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
 @end table
 
+@item segment_list_type @var{type}
+Select the listing format.
+@table @option
+@item @var{flat} use a simple flat list of entries.
+@item @var{hls} use a m3u8-like structure.
+@end table
+
 @item segment_list_size @var{size}
 Update the list file so that it contains at most @var{size}
 segments. If 0 the list file will contain all the segments. Default
@@ -986,9 +924,6 @@ value is 0.
 Prepend @var{prefix} to each entry. Useful to generate absolute paths.
 By default no prefix is applied.
 
-@item segment_list_type @var{type}
-Select the listing format.
-
 The following values are recognized:
 @table @samp
 @item flat
@@ -1095,12 +1030,6 @@ segments to write. If this is selected, the output segment name must
 contain a @code{strftime} function template. Default value is
 @code{0}.
 
-@item break_non_keyframes @var{1|0}
-If enabled, allow segments to start on frames other than keyframes. This
-improves behavior on some players when the time between keyframes is
-inconsistent, but may make things worse on others, and can cause some oddities
-during seeking. Defaults to @code{0}.
-
 @item reset_timestamps @var{1|0}
 Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
@@ -1281,17 +1210,7 @@ is the @option{global_header} flag.
 
 WebM DASH Manifest muxer.
 
-This muxer implements the WebM DASH Manifest specification to generate the DASH
-manifest XML. It also supports manifest generation for DASH live streams.
-
-For more information see:
-
-@itemize @bullet
-@item
-WebM DASH Specification: @url{https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification}
-@item
-ISO DASH Specification: @url{http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip}
-@end itemize
+This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
 
 @subsection Options
 
@@ -1302,32 +1221,6 @@ This muxer supports the following options:
 This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
 unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
 audio and video streams. Any number of adaptation sets can be added using this option.
-
-@item live
-Set this to 1 to create a live stream DASH Manifest. Default: 0.
-
-@item chunk_start_index
-Start index of the first chunk. This will go in the @samp{startNumber} attribute
-of the @samp{SegmentTemplate} element in the manifest. Default: 0.
-
-@item chunk_duration_ms
-Duration of each chunk in milliseconds. This will go in the @samp{duration}
-attribute of the @samp{SegmentTemplate} element in the manifest. Default: 1000.
-
-@item utc_timing_url
-URL of the page that will return the UTC timestamp in ISO format. This will go
-in the @samp{value} attribute of the @samp{UTCTiming} element in the manifest.
-Default: None.
-
-@item time_shift_buffer_depth
-Smallest time (in seconds) shifting buffer for which any Representation is
-guaranteed to be available. This will go in the @samp{timeShiftBufferDepth}
-attribute of the @samp{MPD} element. Default: 60.
-
-@item minimum_update_period
-Minimum update period (in seconds) of the manifest. This will go in the
-@samp{minimumUpdatePeriod} attribute of the @samp{MPD} element. Default: 0.
-
 @end table
 
 @subsection Example
@@ -1343,47 +1236,4 @@ ffmpeg -f webm_dash_manifest -i video1.webm \
        manifest.xml
 @end example
 
-@section webm_chunk
-
-WebM Live Chunk Muxer.
-
-This muxer writes out WebM headers and chunks as separate files which can be
-consumed by clients that support WebM Live streams via DASH.
-
-@subsection Options
-
-This muxer supports the following options:
-
-@table @option
-@item chunk_start_index
-Index of the first chunk (defaults to 0).
-
-@item header
-Filename of the header where the initialization data will be written.
-
-@item audio_chunk_duration
-Duration of each audio chunk in milliseconds (defaults to 5000).
-@end table
-
-@subsection Example
-@example
-ffmpeg -f v4l2 -i /dev/video0 \
-       -f alsa -i hw:0 \
-       -map 0:0 \
-       -c:v libvpx-vp9 \
-       -s 640x360 -keyint_min 30 -g 30 \
-       -f webm_chunk \
-       -header webm_live_video_360.hdr \
-       -chunk_start_index 1 \
-       webm_live_video_360_%d.chk \
-       -map 1:0 \
-       -c:a libvorbis \
-       -b:a 128k \
-       -f webm_chunk \
-       -header webm_live_audio_128.hdr \
-       -chunk_start_index 1 \
-       -audio_chunk_duration 1000 \
-       webm_live_audio_128_%d.chk
-@end example
-
 @c man end MUXERS

doc/platform.texi

@@ -175,6 +175,12 @@ Notes:
 
 @itemize
 
+@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
+You can find out by running @code{which link} to see which @code{link.exe} you
+are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
+in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
+@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
+
 @item If you wish to build with zlib support, you will have to grab a compatible
 zlib binary from somewhere, with an MSVC import lib, or if you wish to link
 statically, you can follow the instructions below to build a compatible

doc/protocols.texi

@@ -19,18 +19,6 @@ supported protocols.
 
 A description of the currently available protocols follows.
 
-@section async
-
-Asynchronous data filling wrapper for input stream.
-
-Fill data in a background thread, to decouple I/O operation from demux thread.
-
-@example
-async:@var{URL}
-async:http://host/resource
-async:cache:http://host/resource
-@end example
-
 @section bluray
 
 Read BluRay playlist.
@@ -289,43 +277,6 @@ Set initial byte offset.
 
 @item end_offset
 Try to limit the request to bytes preceding this offset.
-
-@item method
-When used as a client option it sets the HTTP method for the request.
-
-When used as a server option it sets the HTTP method that is going to be
-expected from the client(s).
-If the expected and the received HTTP method do not match the client will
-be given a Bad Request response.
-When unset the HTTP method is not checked for now. This will be replaced by
-autodetection in the future.
-
-@item listen
-If set to 1 enables experimental HTTP server. This can be used to send data when
-used as an output option, or read data from a client with HTTP POST when used as
-an input option.
-If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
-in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
-@example
-# Server side (sending):
-ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
-
-# Client side (receiving):
-ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
-
-# Client can also be done with wget:
-wget http://@var{server}:@var{port} -O somefile.ogg
-
-# Server side (receiving):
-ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
-
-# Client side (sending):
-ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
-
-# Client can also be done with wget:
-wget --post-file=somefile.ogg http://@var{server}:@var{port}
-@end example
-
 @end table
 
 @subsection HTTP Cookies
@@ -1104,9 +1055,7 @@ subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
 @end example
 
 Play an AVI file directly from a TAR archive:
-@example
 subfile,,start,183241728,end,366490624,,:archive.tar
-@end example
 
 @section tcp
 

doc/scaler.texi

@@ -122,22 +122,6 @@ a_dither).
 
 @end table
 
-@item alphablend
-Set the alpha blending to use when the input has alpha but the output does not.
-Default value is @samp{none}.
-
-@table @samp
-@item uniform_color
-Blend onto a uniform background color
-
-@item checkerboard
-Blend onto a checkerboard
-
-@item none
-No blending
-
-@end table
-
 @end table
 
 @c man end SCALER OPTIONS

doc/texi2pod.pl

@@ -166,7 +166,7 @@ INF: while(<$inf>) {
         if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) {
             $skipping = pop @skstack;
             next;
-        } elsif ($ended =~ /^(?:example|smallexample|verbatim|display)$/) {
+        } elsif ($ended =~ /^(?:example|smallexample|display)$/) {
             $shift = "";
             $_ = "";        # need a paragraph break
         } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
@@ -290,7 +290,7 @@ INF: while(<$inf>) {
         $_ = "\n=over 4\n";
     };
 
-    /^\@((?:small)?example|verbatim|display)/ and do {
+    /^\@((?:small)?example|display)/ and do {
         push @endwstack, $endw;
         $endw = $1;
         $shift = "\t";
@@ -384,7 +384,7 @@ sub postprocess
     # @* is also impossible in .pod; we discard it and any newline that
     # follows it.  Similarly, our macro @gol must be discarded.
 
-    s/\@anchor\{(?:[^\}]*)\}//g;
+    s/\@anchor{(?:[^\}]*)\}//g;
     s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
     s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
     s/;\s+\@pxref\{(?:[^\}]*)\}//g;

doc/utils.texi

@@ -12,17 +12,17 @@ explicitly specified. The following rules are applied:
 
 @itemize
 @item
-@samp{'} and @samp{\} are special characters (respectively used for
+@code{'} and @code{\} are special characters (respectively used for
 quoting and escaping). In addition to them, there might be other
 special characters depending on the specific syntax where the escaping
 and quoting are employed.
 
 @item
-A special character is escaped by prefixing it with a @samp{\}.
+A special character is escaped by prefixing it with a '\'.
 
 @item
-All characters enclosed between @samp{''} are included literally in the
-parsed string. The quote character @samp{'} itself cannot be quoted,
+All characters enclosed between '' are included literally in the
+parsed string. The quote character @code{'} itself cannot be quoted,
 so you may need to close the quote and escape it.
 
 @item
@@ -71,7 +71,7 @@ Escaping and quoting can be mixed together:
 @end example
 
 @item
-To include a literal @samp{\} you can use either escaping or quoting:
+To include a literal @code{\} you can use either escaping or quoting:
 @example
 'c:\foo' can be written as c:\\foo
 @end example
@@ -238,14 +238,6 @@ The following abbreviations are recognized:
 480x320
 @item qhd
 960x540
-@item 2kdci
-2048x1080
-@item 4kdci
-4096x2160
-@item uhd2160
-3840x2160
-@item uhd4320
-7680x4320
 @end table
 
 @anchor{video rate syntax}

ffmpeg.h

@@ -63,7 +63,6 @@ enum HWAccelID {
     HWACCEL_VDPAU,
     HWACCEL_DXVA2,
     HWACCEL_VDA,
-    HWACCEL_VIDEOTOOLBOX,
 };
 
 typedef struct HWAccel {
@@ -93,8 +92,6 @@ typedef struct OptionsContext {
 
     /* input/output options */
     int64_t start_time;
-    int64_t start_time_eof;
-    int seek_timestamp;
     const char *format;
 
     SpecifierOpt *codec_names;
@@ -124,8 +121,6 @@ typedef struct OptionsContext {
     int        nb_hwaccels;
     SpecifierOpt *hwaccel_devices;
     int        nb_hwaccel_devices;
-    SpecifierOpt *autorotate;
-    int        nb_autorotate;
 
     /* output options */
     StreamMap *stream_maps;
@@ -231,7 +226,6 @@ typedef struct OutputFilter {
 
     /* temporary storage until stream maps are processed */
     AVFilterInOut       *out_tmp;
-    enum AVMediaType     type;
 } OutputFilter;
 
 typedef struct FilterGraph {
@@ -281,7 +275,6 @@ typedef struct InputStream {
     int top_field_first;
     int guess_layout_max;
 
-    int autorotate;
     int resample_height;
     int resample_width;
     int resample_pix_fmt;
@@ -346,7 +339,6 @@ typedef struct InputFile {
     int64_t ts_offset;
     int64_t last_ts;
     int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
-    int seek_timestamp;
     int64_t recording_time;
     int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
                              from ctx.nb_streams if new streams appear during av_read_frame() */
@@ -402,13 +394,11 @@ typedef struct OutputStream {
     AVFrame *filtered_frame;
     AVFrame *last_frame;
     int last_droped;
-    int last_nb0_frames[3];
 
     /* video only */
     AVRational frame_rate;
     int force_fps;
     int top_field_first;
-    int rotate_overridden;
 
     AVRational frame_aspect_ratio;
 
@@ -432,8 +422,8 @@ typedef struct OutputStream {
     char *filters;         ///< filtergraph associated to the -filter option
     char *filters_script;  ///< filtergraph script associated to the -filter_script option
 
+    int64_t sws_flags;
     AVDictionary *encoder_opts;
-    AVDictionary *sws_dict;
     AVDictionary *swr_opts;
     AVDictionary *resample_opts;
     AVDictionary *bsf_args;
@@ -458,15 +448,6 @@ typedef struct OutputStream {
     // number of frames/samples sent to the encoder
     uint64_t frames_encoded;
     uint64_t samples_encoded;
-
-    /* packet quality factor */
-    int quality;
-
-    /* packet picture type */
-    int pict_type;
-
-    /* frame encode sum of squared error values */
-    int64_t error[4];
 } OutputStream;
 
 typedef struct OutputFile {
@@ -521,7 +502,6 @@ extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
 extern float max_error_rate;
 extern int vdpau_api_ver;
-extern char *videotoolbox_pixfmt;
 
 extern const AVIOInterruptCB int_cb;
 
@@ -549,13 +529,11 @@ int configure_filtergraph(FilterGraph *fg);
 int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out);
 int ist_in_filtergraph(FilterGraph *fg, InputStream *ist);
 FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
-int init_complex_filtergraph(FilterGraph *fg);
 
 int ffmpeg_parse_options(int argc, char **argv);
 
 int vdpau_init(AVCodecContext *s);
 int dxva2_init(AVCodecContext *s);
 int vda_init(AVCodecContext *s);
-int videotoolbox_init(AVCodecContext *s);
 
 #endif /* FFMPEG_H */

ffmpeg_dxva2.c

@@ -364,7 +364,7 @@ static int dxva2_alloc(AVCodecContext *s)
     d3dpp.SwapEffect       = D3DSWAPEFFECT_DISCARD;
     d3dpp.Flags            = D3DPRESENTFLAG_VIDEO;
 
-    hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetDesktopWindow(),
+    hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetShellWindow(),
                                  D3DCREATE_SOFTWARE_VERTEXPROCESSING | D3DCREATE_MULTITHREADED | D3DCREATE_FPU_PRESERVE,
                                  &d3dpp, &ctx->d3d9device);
     if (FAILED(hr)) {

ffmpeg_filter.c

@@ -31,7 +31,6 @@
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
 #include "libavutil/channel_layout.h"
-#include "libavutil/display.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/pixfmt.h"
@@ -85,7 +84,7 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
                 break;
         }
         if (*p == -1) {
-            if((codec->capabilities & AV_CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
+            if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
                 av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
             if(av_get_sample_fmt_name(st->codec->sample_fmt))
             av_log(NULL, AV_LOG_WARNING,
@@ -289,45 +288,6 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
     ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
 }
 
-int init_complex_filtergraph(FilterGraph *fg)
-{
-    AVFilterInOut *inputs, *outputs, *cur;
-    AVFilterGraph *graph;
-    int ret = 0;
-
-    /* this graph is only used for determining the kinds of inputs
-     * and outputs we have, and is discarded on exit from this function */
-    graph = avfilter_graph_alloc();
-    if (!graph)
-        return AVERROR(ENOMEM);
-
-    ret = avfilter_graph_parse2(graph, fg->graph_desc, &inputs, &outputs);
-    if (ret < 0)
-        goto fail;
-
-    for (cur = inputs; cur; cur = cur->next)
-        init_input_filter(fg, cur);
-
-    for (cur = outputs; cur;) {
-        GROW_ARRAY(fg->outputs, fg->nb_outputs);
-        fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]));
-        if (!fg->outputs[fg->nb_outputs - 1])
-            exit_program(1);
-
-        fg->outputs[fg->nb_outputs - 1]->graph   = fg;
-        fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
-        fg->outputs[fg->nb_outputs - 1]->type    = avfilter_pad_get_type(cur->filter_ctx->output_pads,
-                                                                         cur->pad_idx);
-        cur = cur->next;
-        fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
-    }
-
-fail:
-    avfilter_inout_free(&inputs);
-    avfilter_graph_free(&graph);
-    return ret;
-}
-
 static int insert_trim(int64_t start_time, int64_t duration,
                        AVFilterContext **last_filter, int *pad_idx,
                        const char *filter_name)
@@ -379,28 +339,6 @@ static int insert_trim(int64_t start_time, int64_t duration,
     return 0;
 }
 
-static int insert_filter(AVFilterContext **last_filter, int *pad_idx,
-                         const char *filter_name, const char *args)
-{
-    AVFilterGraph *graph = (*last_filter)->graph;
-    AVFilterContext *ctx;
-    int ret;
-
-    ret = avfilter_graph_create_filter(&ctx,
-                                       avfilter_get_by_name(filter_name),
-                                       filter_name, args, NULL, graph);
-    if (ret < 0)
-        return ret;
-
-    ret = avfilter_link(*last_filter, *pad_idx, ctx, 0);
-    if (ret < 0)
-        return ret;
-
-    *last_filter = ctx;
-    *pad_idx     = 0;
-    return 0;
-}
-
 static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
     char *pix_fmts;
@@ -423,17 +361,11 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
     if (codec->width || codec->height) {
         char args[255];
         AVFilterContext *filter;
-        AVDictionaryEntry *e = NULL;
 
-        snprintf(args, sizeof(args), "%d:%d",
+        snprintf(args, sizeof(args), "%d:%d:0x%X",
                  codec->width,
-                 codec->height);
-
-        while ((e = av_dict_get(ost->sws_dict, "", e,
-                                AV_DICT_IGNORE_SUFFIX))) {
-            av_strlcatf(args, sizeof(args), ":%s=%s", e->key, e->value);
-        }
-
+                 codec->height,
+                 (unsigned)ost->sws_flags);
         snprintf(name, sizeof(name), "scaler for output stream %d:%d",
                  ost->file_index, ost->index);
         if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
@@ -544,7 +476,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
                    av_get_default_channel_layout(ost->audio_channels_mapped));
         for (i = 0; i < ost->audio_channels_mapped; i++)
             if (ost->audio_channels_map[i] != -1)
-                av_bprintf(&pan_buf, "|c%d=c%d", i, ost->audio_channels_map[i]);
+                av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
 
         AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str);
         av_bprint_finalize(&pan_buf, NULL);
@@ -647,11 +579,6 @@ int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOu
     av_freep(&ofilter->name);
     DESCRIBE_FILTER_LINK(ofilter, out, 0);
 
-    if (!ofilter->ost) {
-        av_log(NULL, AV_LOG_FATAL, "Filter %s has a unconnected output\n", ofilter->name);
-        exit_program(1);
-    }
-
     switch (avfilter_pad_get_type(out->filter_ctx->output_pads, out->pad_idx)) {
     case AVMEDIA_TYPE_VIDEO: return configure_output_video_filter(fg, ofilter, out);
     case AVMEDIA_TYPE_AUDIO: return configure_output_audio_filter(fg, ofilter, out);
@@ -682,8 +609,8 @@ static int sub2video_prepare(InputStream *ist)
         }
         av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
     }
-    ist->sub2video.w = ist->resample_width  = w;
-    ist->sub2video.h = ist->resample_height = h;
+    ist->sub2video.w = ist->dec_ctx->width  = ist->resample_width  = w;
+    ist->sub2video.h = ist->dec_ctx->height = ist->resample_height = h;
 
     /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
        palettes for all rectangles are identical or compatible */
@@ -738,7 +665,7 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
              ist->resample_height,
              ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->resample_pix_fmt,
              tb.num, tb.den, sar.num, sar.den,
-             SWS_BILINEAR + ((ist->dec_ctx->flags&AV_CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
+             SWS_BILINEAR + ((ist->dec_ctx->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
     if (fr.num && fr.den)
         av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
     snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
@@ -749,27 +676,6 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
         return ret;
     last_filter = ifilter->filter;
 
-    if (ist->autorotate) {
-        double theta = get_rotation(ist->st);
-
-        if (fabs(theta - 90) < 1.0) {
-            ret = insert_filter(&last_filter, &pad_idx, "transpose", "clock");
-        } else if (fabs(theta - 180) < 1.0) {
-            ret = insert_filter(&last_filter, &pad_idx, "hflip", NULL);
-            if (ret < 0)
-                return ret;
-            ret = insert_filter(&last_filter, &pad_idx, "vflip", NULL);
-        } else if (fabs(theta - 270) < 1.0) {
-            ret = insert_filter(&last_filter, &pad_idx, "transpose", "cclock");
-        } else if (fabs(theta) > 1.0) {
-            char rotate_buf[64];
-            snprintf(rotate_buf, sizeof(rotate_buf), "%f*PI/180", theta);
-            ret = insert_filter(&last_filter, &pad_idx, "rotate", rotate_buf);
-        }
-        if (ret < 0)
-            return ret;
-    }
-
     if (ist->framerate.num) {
         AVFilterContext *setpts;
 
@@ -954,7 +860,7 @@ static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
 int configure_filtergraph(FilterGraph *fg)
 {
     AVFilterInOut *inputs, *outputs, *cur;
-    int ret, i, simple = !fg->graph_desc;
+    int ret, i, init = !fg->graph, simple = !fg->graph_desc;
     const char *graph_desc = simple ? fg->outputs[0]->ost->avfilter :
                                       fg->graph_desc;
 
@@ -967,13 +873,7 @@ int configure_filtergraph(FilterGraph *fg)
         char args[512];
         AVDictionaryEntry *e = NULL;
 
-        args[0] = 0;
-        while ((e = av_dict_get(ost->sws_dict, "", e,
-                                AV_DICT_IGNORE_SUFFIX))) {
-            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
-        }
-        if (strlen(args))
-            args[strlen(args)-1] = 0;
+        snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
         fg->graph->scale_sws_opts = av_strdup(args);
 
         args[0] = 0;
@@ -1003,30 +903,14 @@ int configure_filtergraph(FilterGraph *fg)
         return ret;
 
     if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
-        const char *num_inputs;
-        const char *num_outputs;
-        if (!outputs) {
-            num_outputs = "0";
-        } else if (outputs->next) {
-            num_outputs = ">1";
-        } else {
-            num_outputs = "1";
-        }
-        if (!inputs) {
-            num_inputs = "0";
-        } else if (inputs->next) {
-            num_inputs = ">1";
-        } else {
-            num_inputs = "1";
-        }
-        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' was expected "
-               "to have exactly 1 input and 1 output."
-               " However, it had %s input(s) and %s output(s)."
-               " Please adjust, or use a complex filtergraph (-filter_complex) instead.\n",
-               graph_desc, num_inputs, num_outputs);
+        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' does not have "
+               "exactly one input and output.\n", graph_desc);
         return AVERROR(EINVAL);
     }
 
+    for (cur = inputs; !simple && init && cur; cur = cur->next)
+        init_input_filter(fg, cur);
+
     for (cur = inputs, i = 0; cur; cur = cur->next, i++)
         if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0) {
             avfilter_inout_free(&inputs);
@@ -1035,12 +919,27 @@ int configure_filtergraph(FilterGraph *fg)
         }
     avfilter_inout_free(&inputs);
 
-    for (cur = outputs, i = 0; cur; cur = cur->next, i++)
-        configure_output_filter(fg, fg->outputs[i], cur);
-    avfilter_inout_free(&outputs);
+    if (!init || simple) {
+        /* we already know the mappings between lavfi outputs and output streams,
+         * so we can finish the setup */
+        for (cur = outputs, i = 0; cur; cur = cur->next, i++)
+            configure_output_filter(fg, fg->outputs[i], cur);
+        avfilter_inout_free(&outputs);
 
-    if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
-        return ret;
+        if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
+            return ret;
+    } else {
+        /* wait until output mappings are processed */
+        for (cur = outputs; cur;) {
+            GROW_ARRAY(fg->outputs, fg->nb_outputs);
+            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
+                exit_program(1);
+            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
+            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
+            cur = cur->next;
+            fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
+        }
+    }
 
     fg->reconfiguration = 1;
 
@@ -1048,7 +947,7 @@ int configure_filtergraph(FilterGraph *fg)
         OutputStream *ost = fg->outputs[i]->ost;
         if (ost &&
             ost->enc->type == AVMEDIA_TYPE_AUDIO &&
-            !(ost->enc->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
+            !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
             av_buffersink_set_frame_size(ost->filter->filter,
                                          ost->enc_ctx->frame_size);
     }

ffmpeg_opt.c

@@ -40,9 +40,6 @@
 #include "libavutil/parseutils.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/pixfmt.h"
-#include "libavutil/time_internal.h"
-
-#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
 
 #define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
 {\
@@ -74,10 +71,7 @@ const HWAccel hwaccels[] = {
     { "dxva2", dxva2_init, HWACCEL_DXVA2, AV_PIX_FMT_DXVA2_VLD },
 #endif
 #if CONFIG_VDA
-    { "vda",   videotoolbox_init,   HWACCEL_VDA,   AV_PIX_FMT_VDA },
-#endif
-#if CONFIG_VIDEOTOOLBOX
-    { "videotoolbox",   videotoolbox_init,   HWACCEL_VIDEOTOOLBOX,   AV_PIX_FMT_VIDEOTOOLBOX },
+    { "vda",   vda_init,   HWACCEL_VDA,   AV_PIX_FMT_VDA },
 #endif
     { 0 },
 };
@@ -116,9 +110,6 @@ static int no_file_overwrite  = 0;
 static int do_psnr            = 0;
 static int input_sync;
 static int override_ffserver  = 0;
-static int input_stream_potentially_available = 0;
-static int ignore_unknown_streams = 0;
-static int copy_unknown_streams = 0;
 
 static void uninit_options(OptionsContext *o)
 {
@@ -159,25 +150,12 @@ static void init_options(OptionsContext *o)
     o->stop_time = INT64_MAX;
     o->mux_max_delay  = 0.7;
     o->start_time     = AV_NOPTS_VALUE;
-    o->start_time_eof = AV_NOPTS_VALUE;
     o->recording_time = INT64_MAX;
     o->limit_filesize = UINT64_MAX;
     o->chapters_input_file = INT_MAX;
     o->accurate_seek  = 1;
 }
 
-static int show_hwaccels(void *optctx, const char *opt, const char *arg)
-{
-    int i;
-
-    printf("Hardware acceleration methods:\n");
-    for (i = 0; i < FF_ARRAY_ELEMS(hwaccels) - 1; i++) {
-        printf("%s\n", hwaccels[i].name);
-    }
-    printf("\n");
-    return 0;
-}
-
 /* return a copy of the input with the stream specifiers removed from the keys */
 static AVDictionary *strip_specifiers(AVDictionary *dict)
 {
@@ -249,7 +227,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
     int sync_file_idx = -1, sync_stream_idx = 0;
     char *p, *sync;
     char *map;
-    char *allow_unused;
 
     if (*arg == '-') {
         negative = 1;
@@ -294,8 +271,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
             exit_program(1);
         }
     } else {
-        if (allow_unused = strchr(map, '?'))
-            *allow_unused = 0;
         file_idx = strtol(map, &p, 0);
         if (file_idx >= nb_input_files || file_idx < 0) {
             av_log(NULL, AV_LOG_FATAL, "Invalid input file index: %d.\n", file_idx);
@@ -333,13 +308,8 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
     }
 
     if (!m) {
-        if (allow_unused) {
-            av_log(NULL, AV_LOG_VERBOSE, "Stream map '%s' matches no streams; ignoring.\n", arg);
-        } else {
-            av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n"
-                                       "To ignore this, add a trailing '?' to the map.\n", arg);
-            exit_program(1);
-        }
+        av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n", arg);
+        exit_program(1);
     }
 
     av_freep(&map);
@@ -631,9 +601,6 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
         ist->ts_scale = 1.0;
         MATCH_PER_STREAM_OPT(ts_scale, dbl, ist->ts_scale, ic, st);
 
-        ist->autorotate = 1;
-        MATCH_PER_STREAM_OPT(autorotate, i, ist->autorotate, ic, st);
-
         MATCH_PER_STREAM_OPT(codec_tags, str, codec_tag, ic, st);
         if (codec_tag) {
             uint32_t tag = strtol(codec_tag, &next, 0);
@@ -674,11 +641,9 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
         case AVMEDIA_TYPE_VIDEO:
             if(!ist->dec)
                 ist->dec = avcodec_find_decoder(dec->codec_id);
-#if FF_API_EMU_EDGE
             if (av_codec_get_lowres(dec)) {
                 dec->flags |= CODEC_FLAG_EMU_EDGE;
             }
-#endif
 
             ist->resample_height  = ist->dec_ctx->height;
             ist->resample_width   = ist->dec_ctx->width;
@@ -951,33 +916,14 @@ static int open_input_file(OptionsContext *o, const char *filename)
         }
     }
 
-    if (o->start_time_eof != AV_NOPTS_VALUE) {
-        if (ic->duration>0) {
-            o->start_time = o->start_time_eof + ic->duration;
-        } else
-            av_log(NULL, AV_LOG_WARNING, "Cannot use -sseof, duration of %s not known\n", filename);
-    }
     timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
     /* add the stream start time */
-    if (!o->seek_timestamp && ic->start_time != AV_NOPTS_VALUE)
+    if (ic->start_time != AV_NOPTS_VALUE)
         timestamp += ic->start_time;
 
     /* if seeking requested, we execute it */
     if (o->start_time != AV_NOPTS_VALUE) {
-        int64_t seek_timestamp = timestamp;
-
-        if (!(ic->iformat->flags & AVFMT_SEEK_TO_PTS)) {
-            int dts_heuristic = 0;
-            for (i=0; i<ic->nb_streams; i++) {
-                AVCodecContext *avctx = ic->streams[i]->codec;
-                if (avctx->has_b_frames)
-                    dts_heuristic = 1;
-            }
-            if (dts_heuristic) {
-                seek_timestamp -= 3*AV_TIME_BASE / 23;
-            }
-        }
-        ret = avformat_seek_file(ic, -1, INT64_MIN, seek_timestamp, seek_timestamp, 0);
+        ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0);
         if (ret < 0) {
             av_log(NULL, AV_LOG_WARNING, "%s: could not seek to position %0.3f\n",
                    filename, (double)timestamp / AV_TIME_BASE);
@@ -1062,8 +1008,6 @@ static int open_input_file(OptionsContext *o, const char *filename)
         av_dict_free(&opts[i]);
     av_freep(&opts);
 
-    input_stream_potentially_available = 1;
-
     return 0;
 }
 
@@ -1238,13 +1182,12 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
         uint32_t tag = strtol(codec_tag, &next, 0);
         if (*next)
             tag = AV_RL32(codec_tag);
-        ost->st->codec->codec_tag =
         ost->enc_ctx->codec_tag = tag;
     }
 
     MATCH_PER_STREAM_OPT(qscale, dbl, qscale, oc, st);
     if (qscale >= 0) {
-        ost->enc_ctx->flags |= AV_CODEC_FLAG_QSCALE;
+        ost->enc_ctx->flags |= CODEC_FLAG_QSCALE;
         ost->enc_ctx->global_quality = FF_QP2LAMBDA * qscale;
     }
 
@@ -1252,9 +1195,9 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
     ost->disposition = av_strdup(ost->disposition);
 
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        ost->enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        ost->enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
 
-    av_dict_copy(&ost->sws_dict, o->g->sws_dict, 0);
+    av_opt_get_int(o->g->sws_opts, "sws_flags", 0, &ost->sws_flags);
 
     av_dict_copy(&ost->swr_opts, o->g->swr_opts, 0);
     if (ost->enc && av_get_exact_bits_per_sample(ost->enc->id) == 24)
@@ -1474,17 +1417,17 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
         video_enc->rc_override_count = i;
 
         if (do_psnr)
-            video_enc->flags|= AV_CODEC_FLAG_PSNR;
+            video_enc->flags|= CODEC_FLAG_PSNR;
 
         /* two pass mode */
         MATCH_PER_STREAM_OPT(pass, i, do_pass, oc, st);
         if (do_pass) {
             if (do_pass & 1) {
-                video_enc->flags |= AV_CODEC_FLAG_PASS1;
+                video_enc->flags |= CODEC_FLAG_PASS1;
                 av_dict_set(&ost->encoder_opts, "flags", "+pass1", AV_DICT_APPEND);
             }
             if (do_pass & 2) {
-                video_enc->flags |= AV_CODEC_FLAG_PASS2;
+                video_enc->flags |= CODEC_FLAG_PASS2;
                 av_dict_set(&ost->encoder_opts, "flags", "+pass2", AV_DICT_APPEND);
             }
         }
@@ -1494,40 +1437,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
             !(ost->logfile_prefix = av_strdup(ost->logfile_prefix)))
             exit_program(1);
 
-        if (do_pass) {
-            char logfilename[1024];
-            FILE *f;
-
-            snprintf(logfilename, sizeof(logfilename), "%s-%d.log",
-                     ost->logfile_prefix ? ost->logfile_prefix :
-                                           DEFAULT_PASS_LOGFILENAME_PREFIX,
-                     i);
-            if (!strcmp(ost->enc->name, "libx264")) {
-                av_dict_set(&ost->encoder_opts, "stats", logfilename, AV_DICT_DONT_OVERWRITE);
-            } else {
-                if (video_enc->flags & AV_CODEC_FLAG_PASS2) {
-                    char  *logbuffer = read_file(logfilename);
-
-                    if (!logbuffer) {
-                        av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n",
-                               logfilename);
-                        exit_program(1);
-                    }
-                    video_enc->stats_in = logbuffer;
-                }
-                if (video_enc->flags & AV_CODEC_FLAG_PASS1) {
-                    f = av_fopen_utf8(logfilename, "wb");
-                    if (!f) {
-                        av_log(NULL, AV_LOG_FATAL,
-                               "Cannot write log file '%s' for pass-1 encoding: %s\n",
-                               logfilename, strerror(errno));
-                        exit_program(1);
-                    }
-                    ost->logfile = f;
-                }
-            }
-        }
-
         MATCH_PER_STREAM_OPT(forced_key_frames, str, ost->forced_keyframes, oc, st);
         if (ost->forced_keyframes)
             ost->forced_keyframes = av_strdup(ost->forced_keyframes);
@@ -1637,19 +1546,6 @@ static OutputStream *new_data_stream(OptionsContext *o, AVFormatContext *oc, int
     return ost;
 }
 
-static OutputStream *new_unknown_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
-{
-    OutputStream *ost;
-
-    ost = new_output_stream(o, oc, AVMEDIA_TYPE_UNKNOWN, source_index);
-    if (!ost->stream_copy) {
-        av_log(NULL, AV_LOG_FATAL, "Unknown stream encoding not supported yet (only streamcopy)\n");
-        exit_program(1);
-    }
-
-    return ost;
-}
-
 static OutputStream *new_attachment_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
 {
     OutputStream *ost = new_output_stream(o, oc, AVMEDIA_TYPE_ATTACHMENT, source_index);
@@ -1806,7 +1702,8 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
 {
     OutputStream *ost;
 
-    switch (ofilter->type) {
+    switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads,
+                                  ofilter->out_tmp->pad_idx)) {
     case AVMEDIA_TYPE_VIDEO: ost = new_video_stream(o, oc, -1); break;
     case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream(o, oc, -1); break;
     default:
@@ -1839,19 +1736,11 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
         exit_program(1);
     }
 
-    avfilter_inout_free(&ofilter->out_tmp);
-}
-
-static int init_complex_filters(void)
-{
-    int i, ret = 0;
-
-    for (i = 0; i < nb_filtergraphs; i++) {
-        ret = init_complex_filtergraph(filtergraphs[i]);
-        if (ret < 0)
-            return ret;
+    if (configure_output_filter(ofilter->graph, ofilter, ofilter->out_tmp) < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Error configuring filter.\n");
+        exit_program(1);
     }
-    return 0;
+    avfilter_inout_free(&ofilter->out_tmp);
 }
 
 static int configure_complex_filters(void)
@@ -1876,6 +1765,10 @@ static int open_output_file(OptionsContext *o, const char *filename)
     AVDictionary *unused_opts = NULL;
     AVDictionaryEntry *e = NULL;
 
+    if (configure_complex_filters() < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Error configuring filters.\n");
+        exit_program(1);
+    }
 
     if (o->stop_time != INT64_MAX && o->recording_time != INT64_MAX) {
         o->stop_time = INT64_MAX;
@@ -1930,7 +1823,8 @@ static int open_output_file(OptionsContext *o, const char *filename)
             if (!ofilter->out_tmp || ofilter->out_tmp->name)
                 continue;
 
-            switch (ofilter->type) {
+            switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads,
+                                          ofilter->out_tmp->pad_idx)) {
             case AVMEDIA_TYPE_VIDEO:    o->video_disable    = 1; break;
             case AVMEDIA_TYPE_AUDIO:    o->audio_disable    = 1; break;
             case AVMEDIA_TYPE_SUBTITLE: o->subtitle_disable = 1; break;
@@ -1989,7 +1883,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
             for (i = 0; i < nb_input_streams; i++) {
                 int new_area;
                 ist = input_streams[i];
-                new_area = ist->st->codec->width * ist->st->codec->height + 100000000*!!ist->st->codec_info_nb_frames;
+                new_area = ist->st->codec->width * ist->st->codec->height;
                 if((qcr!=MKTAG('A', 'P', 'I', 'C')) && (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC))
                     new_area = 1;
                 if (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO &&
@@ -2006,14 +1900,12 @@ static int open_output_file(OptionsContext *o, const char *filename)
 
         /* audio: most channels */
         if (!o->audio_disable && av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_AUDIO) != AV_CODEC_ID_NONE) {
-            int best_score = 0, idx = -1;
+            int channels = 0, idx = -1;
             for (i = 0; i < nb_input_streams; i++) {
-                int score;
                 ist = input_streams[i];
-                score = ist->st->codec->channels + 100000000*!!ist->st->codec_info_nb_frames;
                 if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
-                    score > best_score) {
-                    best_score = score;
+                    ist->st->codec->channels > channels) {
+                    channels = ist->st->codec->channels;
                     idx = i;
                 }
             }
@@ -2100,33 +1992,17 @@ loop_end:
                 if(o->    data_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_DATA)
                     continue;
 
-                ost = NULL;
                 switch (ist->st->codec->codec_type) {
                 case AVMEDIA_TYPE_VIDEO:      ost = new_video_stream     (o, oc, src_idx); break;
                 case AVMEDIA_TYPE_AUDIO:      ost = new_audio_stream     (o, oc, src_idx); break;
                 case AVMEDIA_TYPE_SUBTITLE:   ost = new_subtitle_stream  (o, oc, src_idx); break;
                 case AVMEDIA_TYPE_DATA:       ost = new_data_stream      (o, oc, src_idx); break;
                 case AVMEDIA_TYPE_ATTACHMENT: ost = new_attachment_stream(o, oc, src_idx); break;
-                case AVMEDIA_TYPE_UNKNOWN:
-                    if (copy_unknown_streams) {
-                        ost = new_unknown_stream   (o, oc, src_idx);
-                        break;
-                    }
                 default:
-                    av_log(NULL, ignore_unknown_streams ? AV_LOG_WARNING : AV_LOG_FATAL,
-                           "Cannot map stream #%d:%d - unsupported type.\n",
+                    av_log(NULL, AV_LOG_FATAL, "Cannot map stream #%d:%d - unsupported type.\n",
                            map->file_index, map->stream_index);
-                    if (!ignore_unknown_streams) {
-                        av_log(NULL, AV_LOG_FATAL,
-                               "If you want unsupported types ignored instead "
-                               "of failing, please use the -ignore_unknown option\n"
-                               "If you want them copied, please use -copy_unknown\n");
-                        exit_program(1);
-                    }
+                    exit_program(1);
                 }
-                if (ost)
-                    ost->sync_ist = input_streams[  input_files[map->sync_file_index]->ist_index
-                                                  + map->sync_stream_index];
             }
         }
     }
@@ -2156,11 +2032,11 @@ loop_end:
         avio_read(pb, attachment, len);
 
         ost = new_attachment_stream(o, oc, -1);
-        ost->stream_copy               = 1;
+        ost->stream_copy               = 0;
         ost->attachment_filename       = o->attachments[i];
         ost->finished                  = 1;
-        ost->st->codec->extradata      = attachment;
-        ost->st->codec->extradata_size = len;
+        ost->enc_ctx->extradata      = attachment;
+        ost->enc_ctx->extradata_size = len;
 
         p = strrchr(o->attachments[i], '/');
         av_dict_set(&ost->st->metadata, "filename", (p && *p) ? p + 1 : o->attachments[i], AV_DICT_DONT_OVERWRITE);
@@ -2178,12 +2054,6 @@ loop_end:
                 exit_program(1);
     }
 
-    if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) {
-        av_dump_format(oc, nb_output_files - 1, oc->filename, 1);
-        av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", nb_output_files - 1);
-        exit_program(1);
-    }
-
     /* check if all codec options have been used */
     unused_opts = strip_specifiers(o->g->codec_opts);
     for (i = of->ost_index; i < nb_output_streams; i++) {
@@ -2226,17 +2096,6 @@ loop_end:
     }
     av_dict_free(&unused_opts);
 
-    /* set the encoding/decoding_needed flags */
-    for (i = of->ost_index; i < nb_output_streams; i++) {
-        OutputStream *ost = output_streams[i];
-
-        ost->encoding_needed = !ost->stream_copy;
-        if (ost->encoding_needed && ost->source_index >= 0) {
-            InputStream *ist = input_streams[ost->source_index];
-            ist->decoding_needed |= DECODING_FOR_OST;
-        }
-    }
-
     /* check filename in case of an image number is expected */
     if (oc->oformat->flags & AVFMT_NEEDNUMBER) {
         if (!av_filename_number_test(oc->filename)) {
@@ -2245,12 +2104,6 @@ loop_end:
         }
     }
 
-    if (!(oc->oformat->flags & AVFMT_NOSTREAMS) && !input_stream_potentially_available) {
-        av_log(NULL, AV_LOG_ERROR,
-               "No input streams but output needs an input stream\n");
-        exit_program(1);
-    }
-
     if (!(oc->oformat->flags & AVFMT_NOFILE)) {
         /* test if it already exists to avoid losing precious files */
         assert_file_overwrite(filename);
@@ -2319,11 +2172,8 @@ loop_end:
                 continue;
             ist = input_streams[output_streams[i]->source_index];
             av_dict_copy(&output_streams[i]->st->metadata, ist->st->metadata, AV_DICT_DONT_OVERWRITE);
-            if (!output_streams[i]->stream_copy) {
+            if (!output_streams[i]->stream_copy)
                 av_dict_set(&output_streams[i]->st->metadata, "encoder", NULL, 0);
-                if (ist->autorotate)
-                    av_dict_set(&output_streams[i]->st->metadata, "rotate", NULL, 0);
-            }
         }
 
     /* process manually set metadata */
@@ -2332,7 +2182,6 @@ loop_end:
         char type, *val;
         const char *stream_spec;
         int index = 0, j, ret = 0;
-        char now_time[256];
 
         val = strchr(o->metadata[i].u.str, '=');
         if (!val) {
@@ -2342,26 +2191,11 @@ loop_end:
         }
         *val++ = 0;
 
-        if (!strcmp(o->metadata[i].u.str, "creation_time") &&
-            !strcmp(val, "now")) {
-            time_t now = time(0);
-            struct tm *ptm, tmbuf;
-            ptm = localtime_r(&now, &tmbuf);
-            if (ptm) {
-                if (strftime(now_time, sizeof(now_time), "%Y-%m-%d %H:%M:%S", ptm))
-                    val = now_time;
-            }
-        }
-
         parse_meta_type(o->metadata[i].specifier, &type, &index, &stream_spec);
         if (type == 's') {
             for (j = 0; j < oc->nb_streams; j++) {
-                ost = output_streams[nb_output_streams - oc->nb_streams + j];
                 if ((ret = check_stream_specifier(oc, oc->streams[j], stream_spec)) > 0) {
                     av_dict_set(&oc->streams[j]->metadata, o->metadata[i].u.str, *val ? val : NULL, 0);
-                    if (!strcmp(o->metadata[i].u.str, "rotate")) {
-                        ost->rotate_overridden = 1;
-                    }
                 } else if (ret < 0)
                     exit_program(1);
             }
@@ -2545,10 +2379,8 @@ static int opt_vstats(void *optctx, const char *opt, const char *arg)
     time_t today2 = time(NULL);
     struct tm *today = localtime(&today2);
 
-    if (!today) { // maybe tomorrow
-        av_log(NULL, AV_LOG_FATAL, "Unable to get current time: %s\n", strerror(errno));
-        exit_program(1);
-    }
+    if (!today)
+        return AVERROR(errno);
 
     snprintf(filename, sizeof(filename), "vstats_%02d%02d%02d.log", today->tm_hour, today->tm_min,
              today->tm_sec);
@@ -2779,9 +2611,6 @@ static int opt_filter_complex(void *optctx, const char *opt, const char *arg)
     filtergraphs[nb_filtergraphs - 1]->graph_desc = av_strdup(arg);
     if (!filtergraphs[nb_filtergraphs - 1]->graph_desc)
         return AVERROR(ENOMEM);
-
-    input_stream_potentially_available = 1;
-
     return 0;
 }
 
@@ -2796,9 +2625,6 @@ static int opt_filter_complex_script(void *optctx, const char *opt, const char *
         return AVERROR(ENOMEM);
     filtergraphs[nb_filtergraphs - 1]->index      = nb_filtergraphs - 1;
     filtergraphs[nb_filtergraphs - 1]->graph_desc = graph_desc;
-
-    input_stream_potentially_available = 1;
-
     return 0;
 }
 
@@ -2950,13 +2776,6 @@ int ffmpeg_parse_options(int argc, char **argv)
         goto fail;
     }
 
-    /* create the complex filtergraphs */
-    ret = init_complex_filters();
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_FATAL, "Error initializing complex filters.\n");
-        goto fail;
-    }
-
     /* open output files */
     ret = open_files(&octx.groups[GROUP_OUTFILE], "output", open_output_file);
     if (ret < 0) {
@@ -2964,13 +2783,6 @@ int ffmpeg_parse_options(int argc, char **argv)
         goto fail;
     }
 
-    /* configure the complex filtergraphs */
-    ret = configure_complex_filters();
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_FATAL, "Error configuring complex filters.\n");
-        goto fail;
-    }
-
 fail:
     uninit_parse_context(&octx);
     if (ret < 0) {
@@ -3008,10 +2820,6 @@ const OptionDef options[] = {
         "overwrite output files" },
     { "n",              OPT_BOOL,                                    {              &no_file_overwrite },
         "never overwrite output files" },
-    { "ignore_unknown", OPT_BOOL,                                    {              &ignore_unknown_streams },
-        "Ignore unknown stream types" },
-    { "copy_unknown",   OPT_BOOL | OPT_EXPERT,                       {              &copy_unknown_streams },
-        "Copy unknown stream types" },
     { "c",              HAS_ARG | OPT_STRING | OPT_SPEC |
                         OPT_INPUT | OPT_OUTPUT,                      { .off       = OFFSET(codec_names) },
         "codec name", "codec" },
@@ -3045,12 +2853,6 @@ const OptionDef options[] = {
     { "ss",             HAS_ARG | OPT_TIME | OPT_OFFSET |
                         OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time) },
         "set the start time offset", "time_off" },
-    { "sseof",          HAS_ARG | OPT_TIME | OPT_OFFSET |
-                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time_eof) },
-        "set the start time offset relative to EOF", "time_off" },
-    { "seek_timestamp", HAS_ARG | OPT_INT | OPT_OFFSET |
-                        OPT_INPUT,                                   { .off = OFFSET(seek_timestamp) },
-        "enable/disable seeking by timestamp with -ss" },
     { "accurate_seek",  OPT_BOOL | OPT_OFFSET | OPT_EXPERT |
                         OPT_INPUT,                                   { .off = OFFSET(accurate_seek) },
         "enable/disable accurate seeking with -ss" },
@@ -3085,8 +2887,8 @@ const OptionDef options[] = {
                         OPT_INPUT,                                   { .off = OFFSET(rate_emu) },
         "read input at native frame rate", "" },
     { "target",         HAS_ARG | OPT_PERFILE | OPT_OUTPUT,          { .func_arg = opt_target },
-        "specify target file type (\"vcd\", \"svcd\", \"dvd\", \"dv\" or \"dv50\" "
-        "with optional prefixes \"pal-\", \"ntsc-\" or \"film-\")", "type" },
+        "specify target file type (\"vcd\", \"svcd\", \"dvd\","
+        " \"dv\", \"dv50\", \"pal-vcd\", \"ntsc-svcd\", ...)", "type" },
     { "vsync",          HAS_ARG | OPT_EXPERT,                        { opt_vsync },
         "video sync method", "" },
     { "frame_drop_threshold", HAS_ARG | OPT_FLOAT | OPT_EXPERT,      { &frame_drop_threshold },
@@ -3226,7 +3028,7 @@ const OptionDef options[] = {
                       OPT_INPUT | OPT_OUTPUT,                                    { .off = OFFSET(top_field_first) },
         "top=1/bottom=0/auto=-1 field first", "" },
     { "vtag",         OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_PERFILE |
-                      OPT_INPUT | OPT_OUTPUT,                                    { .func_arg = opt_old2new },
+                      OPT_OUTPUT,                                                { .func_arg = opt_old2new },
         "force video tag/fourcc", "fourcc/tag" },
     { "qphist",       OPT_VIDEO | OPT_BOOL | OPT_EXPERT ,                        { &qp_hist },
         "show QP histogram" },
@@ -3248,18 +3050,10 @@ const OptionDef options[] = {
         "use HW accelerated decoding", "hwaccel name" },
     { "hwaccel_device",   OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
                           OPT_SPEC | OPT_INPUT,                                  { .off = OFFSET(hwaccel_devices) },
-        "select a device for HW acceleration", "devicename" },
+        "select a device for HW acceleration" "devicename" },
 #if HAVE_VDPAU_X11
     { "vdpau_api_ver", HAS_ARG | OPT_INT | OPT_EXPERT, { &vdpau_api_ver }, "" },
 #endif
-#if CONFIG_VDA || CONFIG_VIDEOTOOLBOX
-    { "videotoolbox_pixfmt", HAS_ARG | OPT_STRING | OPT_EXPERT, { &videotoolbox_pixfmt}, "" },
-#endif
-    { "hwaccels",         OPT_EXIT,                                              { .func_arg = show_hwaccels },
-        "show available HW acceleration methods" },
-    { "autorotate",       HAS_ARG | OPT_BOOL | OPT_SPEC |
-                          OPT_EXPERT | OPT_INPUT,                                { .off = OFFSET(autorotate) },
-        "automatically insert correct rotate filters" },
 
     /* audio options */
     { "aframes",        OPT_AUDIO | HAS_ARG  | OPT_PERFILE | OPT_OUTPUT,           { .func_arg = opt_audio_frames },

ffmpeg_vda.c

@@ -0,0 +1,136 @@
+/*
+ * 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 "libavcodec/avcodec.h"
+#include "libavcodec/vda.h"
+#include "libavutil/imgutils.h"
+
+#include "ffmpeg.h"
+
+typedef struct VDAContext {
+    AVFrame *tmp_frame;
+} VDAContext;
+
+static int vda_retrieve_data(AVCodecContext *s, AVFrame *frame)
+{
+    InputStream *ist = s->opaque;
+    VDAContext  *vda = ist->hwaccel_ctx;
+    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
+    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
+    CVReturn err;
+    uint8_t *data[4] = { 0 };
+    int linesize[4] = { 0 };
+    int planes, ret, i;
+
+    av_frame_unref(vda->tmp_frame);
+
+    switch (pixel_format) {
+    case kCVPixelFormatType_420YpCbCr8Planar: vda->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
+    case kCVPixelFormatType_422YpCbCr8:       vda->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
+    default:
+        av_log(NULL, AV_LOG_ERROR,
+               "Unsupported pixel format: %u\n", pixel_format);
+        return AVERROR(ENOSYS);
+    }
+
+    vda->tmp_frame->width  = frame->width;
+    vda->tmp_frame->height = frame->height;
+    ret = av_frame_get_buffer(vda->tmp_frame, 32);
+    if (ret < 0)
+        return ret;
+
+    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+    if (err != kCVReturnSuccess) {
+        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
+        return AVERROR_UNKNOWN;
+    }
+
+    if (CVPixelBufferIsPlanar(pixbuf)) {
+
+        planes = CVPixelBufferGetPlaneCount(pixbuf);
+        for (i = 0; i < planes; i++) {
+            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
+            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
+        }
+    } else {
+        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
+        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
+    }
+
+    av_image_copy(vda->tmp_frame->data, vda->tmp_frame->linesize,
+                  data, linesize, vda->tmp_frame->format,
+                  frame->width, frame->height);
+
+    ret = av_frame_copy_props(vda->tmp_frame, frame);
+    CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+
+    if (ret < 0)
+        return ret;
+
+    av_frame_unref(frame);
+    av_frame_move_ref(frame, vda->tmp_frame);
+
+    return 0;
+}
+
+static void vda_uninit(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    VDAContext  *vda = ist->hwaccel_ctx;
+
+    ist->hwaccel_uninit        = NULL;
+    ist->hwaccel_retrieve_data = NULL;
+
+    av_frame_free(&vda->tmp_frame);
+
+    av_vda_default_free(s);
+    av_freep(&ist->hwaccel_ctx);
+}
+
+int vda_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    VDAContext *vda;
+    int ret;
+
+    vda = av_mallocz(sizeof(*vda));
+    if (!vda)
+        return AVERROR(ENOMEM);
+
+    ist->hwaccel_ctx           = vda;
+    ist->hwaccel_uninit        = vda_uninit;
+    ist->hwaccel_retrieve_data = vda_retrieve_data;
+
+    vda->tmp_frame = av_frame_alloc();
+    if (!vda->tmp_frame) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    ret = av_vda_default_init(s);
+    if (ret < 0) {
+        av_log(NULL, loglevel, "Error creating VDA decoder.\n");
+        goto fail;
+    }
+
+    return 0;
+fail:
+    vda_uninit(s);
+    return ret;
+}

ffmpeg_vdpau.c

@@ -289,8 +289,7 @@ do {
 
         s->hwaccel_context = vdpau_ctx;
     } else
-    if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address,
-                              AV_HWACCEL_FLAG_IGNORE_LEVEL))
+    if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address, 0))
         goto fail;
 
     ctx->get_information_string(&vendor);

ffmpeg_videotoolbox.c

@@ -1,187 +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
- */
-
-#include <CoreServices/CoreServices.h>
-
-#include "config.h"
-#include "libavcodec/avcodec.h"
-#if CONFIG_VDA
-#  include "libavcodec/vda.h"
-#endif
-#if CONFIG_VIDEOTOOLBOX
-#  include "libavcodec/videotoolbox.h"
-#endif
-#include "libavutil/imgutils.h"
-#include "ffmpeg.h"
-
-typedef struct VTContext {
-    AVFrame *tmp_frame;
-} VTContext;
-
-char *videotoolbox_pixfmt;
-
-static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
-{
-    InputStream *ist = s->opaque;
-    VTContext  *vt = ist->hwaccel_ctx;
-    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
-    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
-    CVReturn err;
-    uint8_t *data[4] = { 0 };
-    int linesize[4] = { 0 };
-    int planes, ret, i;
-    char codec_str[32];
-
-    av_frame_unref(vt->tmp_frame);
-
-    switch (pixel_format) {
-    case kCVPixelFormatType_420YpCbCr8Planar: vt->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
-    case kCVPixelFormatType_422YpCbCr8:       vt->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
-    case kCVPixelFormatType_32BGRA:           vt->tmp_frame->format = AV_PIX_FMT_BGRA; break;
-#ifdef kCFCoreFoundationVersionNumber10_7
-    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
-#endif
-    default:
-        av_get_codec_tag_string(codec_str, sizeof(codec_str), s->codec_tag);
-        av_log(NULL, AV_LOG_ERROR,
-               "%s: Unsupported pixel format: %s\n", codec_str, videotoolbox_pixfmt);
-        return AVERROR(ENOSYS);
-    }
-
-    vt->tmp_frame->width  = frame->width;
-    vt->tmp_frame->height = frame->height;
-    ret = av_frame_get_buffer(vt->tmp_frame, 32);
-    if (ret < 0)
-        return ret;
-
-    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
-    if (err != kCVReturnSuccess) {
-        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
-        return AVERROR_UNKNOWN;
-    }
-
-    if (CVPixelBufferIsPlanar(pixbuf)) {
-
-        planes = CVPixelBufferGetPlaneCount(pixbuf);
-        for (i = 0; i < planes; i++) {
-            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
-            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
-        }
-    } else {
-        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
-        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
-    }
-
-    av_image_copy(vt->tmp_frame->data, vt->tmp_frame->linesize,
-                  (const uint8_t **)data, linesize, vt->tmp_frame->format,
-                  frame->width, frame->height);
-
-    ret = av_frame_copy_props(vt->tmp_frame, frame);
-    CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
-    if (ret < 0)
-        return ret;
-
-    av_frame_unref(frame);
-    av_frame_move_ref(frame, vt->tmp_frame);
-
-    return 0;
-}
-
-static void videotoolbox_uninit(AVCodecContext *s)
-{
-    InputStream *ist = s->opaque;
-    VTContext  *vt = ist->hwaccel_ctx;
-
-    ist->hwaccel_uninit        = NULL;
-    ist->hwaccel_retrieve_data = NULL;
-
-    av_frame_free(&vt->tmp_frame);
-
-    if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
-#if CONFIG_VIDEOTOOLBOX
-        av_videotoolbox_default_free(s);
-#endif
-    } else {
-#if CONFIG_VDA
-        av_vda_default_free(s);
-#endif
-    }
-    av_freep(&ist->hwaccel_ctx);
-}
-
-int videotoolbox_init(AVCodecContext *s)
-{
-    InputStream *ist = s->opaque;
-    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
-    int ret = 0;
-    VTContext *vt;
-
-    vt = av_mallocz(sizeof(*vt));
-    if (!vt)
-        return AVERROR(ENOMEM);
-
-    ist->hwaccel_ctx           = vt;
-    ist->hwaccel_uninit        = videotoolbox_uninit;
-    ist->hwaccel_retrieve_data = videotoolbox_retrieve_data;
-
-    vt->tmp_frame = av_frame_alloc();
-    if (!vt->tmp_frame) {
-        ret = AVERROR(ENOMEM);
-        goto fail;
-    }
-
-    if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
-#if CONFIG_VIDEOTOOLBOX
-        if (!videotoolbox_pixfmt) {
-            ret = av_videotoolbox_default_init(s);
-        } else {
-            AVVideotoolboxContext *vtctx = av_videotoolbox_alloc_context();
-            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
-                                                               videotoolbox_pixfmt,
-                                                               kCFStringEncodingUTF8);
-            vtctx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
-            ret = av_videotoolbox_default_init2(s, vtctx);
-            CFRelease(pixfmt_str);
-        }
-#endif
-    } else {
-#if CONFIG_VDA
-        if (!videotoolbox_pixfmt) {
-            ret = av_vda_default_init(s);
-        } else {
-            AVVDAContext *vdactx = av_vda_alloc_context();
-            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
-                                                               videotoolbox_pixfmt,
-                                                               kCFStringEncodingUTF8);
-            vdactx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
-            ret = av_vda_default_init2(s, vdactx);
-            CFRelease(pixfmt_str);
-        }
-#endif
-    }
-    if (ret < 0) {
-        av_log(NULL, loglevel,
-               "Error creating %s decoder.\n", ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX ? "Videotoolbox" : "VDA");
-        goto fail;
-    }
-
-    return 0;
-fail:
-    videotoolbox_uninit(s);
-    return ret;
-}

ffplay.c

@@ -31,7 +31,7 @@
 #include <stdint.h>
 
 #include "libavutil/avstring.h"
-#include "libavutil/eval.h"
+#include "libavutil/colorspace.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/imgutils.h"
@@ -65,9 +65,7 @@ const char program_name[] = "ffplay";
 const int program_birth_year = 2003;
 
 #define MAX_QUEUE_SIZE (15 * 1024 * 1024)
-#define MIN_FRAMES 25
-#define EXTERNAL_CLOCK_MIN_FRAMES 2
-#define EXTERNAL_CLOCK_MAX_FRAMES 10
+#define MIN_FRAMES 5
 
 /* Minimum SDL audio buffer size, in samples. */
 #define SDL_AUDIO_MIN_BUFFER_SIZE 512
@@ -103,7 +101,7 @@ const int program_birth_year = 2003;
 
 #define CURSOR_HIDE_DELAY 1000000
 
-static unsigned sws_flags = SWS_BICUBIC;
+static int64_t sws_flags = SWS_BICUBIC;
 
 typedef struct MyAVPacketList {
     AVPacket pkt;
@@ -224,9 +222,6 @@ typedef struct VideoState {
     Decoder viddec;
     Decoder subdec;
 
-    int viddec_width;
-    int viddec_height;
-
     int audio_stream;
 
     int av_sync_type;
@@ -282,7 +277,6 @@ typedef struct VideoState {
 #if !CONFIG_AVFILTER
     struct SwsContext *img_convert_ctx;
 #endif
-    struct SwsContext *sub_convert_ctx;
     SDL_Rect last_display_rect;
     int eof;
 
@@ -834,50 +828,229 @@ static void fill_border(int xleft, int ytop, int width, int height, int x, int y
 #define ALPHA_BLEND(a, oldp, newp, s)\
 ((((oldp << s) * (255 - (a))) + (newp * (a))) / (255 << s))
 
+#define RGBA_IN(r, g, b, a, s)\
+{\
+    unsigned int v = ((const uint32_t *)(s))[0];\
+    a = (v >> 24) & 0xff;\
+    r = (v >> 16) & 0xff;\
+    g = (v >> 8) & 0xff;\
+    b = v & 0xff;\
+}
+
+#define YUVA_IN(y, u, v, a, s, pal)\
+{\
+    unsigned int val = ((const uint32_t *)(pal))[*(const uint8_t*)(s)];\
+    a = (val >> 24) & 0xff;\
+    y = (val >> 16) & 0xff;\
+    u = (val >> 8) & 0xff;\
+    v = val & 0xff;\
+}
+
+#define YUVA_OUT(d, y, u, v, a)\
+{\
+    ((uint32_t *)(d))[0] = (a << 24) | (y << 16) | (u << 8) | v;\
+}
 
 
 #define BPP 1
 
 static void blend_subrect(AVPicture *dst, const AVSubtitleRect *rect, int imgw, int imgh)
 {
-    int x, y, Y, U, V, A;
+    int wrap, wrap3, width2, skip2;
+    int y, u, v, a, u1, v1, a1, w, h;
     uint8_t *lum, *cb, *cr;
+    const uint8_t *p;
+    const uint32_t *pal;
     int dstx, dsty, dstw, dsth;
-    const AVPicture *src = &rect->pict;
 
     dstw = av_clip(rect->w, 0, imgw);
     dsth = av_clip(rect->h, 0, imgh);
     dstx = av_clip(rect->x, 0, imgw - dstw);
     dsty = av_clip(rect->y, 0, imgh - dsth);
-    lum = dst->data[0] + dstx + dsty * dst->linesize[0];
-    cb  = dst->data[1] + dstx/2 + (dsty >> 1) * dst->linesize[1];
-    cr  = dst->data[2] + dstx/2 + (dsty >> 1) * dst->linesize[2];
+    lum = dst->data[0] + dsty * dst->linesize[0];
+    cb  = dst->data[1] + (dsty >> 1) * dst->linesize[1];
+    cr  = dst->data[2] + (dsty >> 1) * dst->linesize[2];
 
-    for (y = 0; y<dsth; y++) {
-        for (x = 0; x<dstw; x++) {
-            Y = src->data[0][x + y*src->linesize[0]];
-            A = src->data[3][x + y*src->linesize[3]];
-            lum[0] = ALPHA_BLEND(A, lum[0], Y, 0);
-            lum++;
-        }
-        lum += dst->linesize[0] - dstw;
-    }
+    width2 = ((dstw + 1) >> 1) + (dstx & ~dstw & 1);
+    skip2 = dstx >> 1;
+    wrap = dst->linesize[0];
+    wrap3 = rect->pict.linesize[0];
+    p = rect->pict.data[0];
+    pal = (const uint32_t *)rect->pict.data[1];  /* Now in YCrCb! */
 
-    for (y = 0; y<dsth/2; y++) {
-        for (x = 0; x<dstw/2; x++) {
-            U = src->data[1][x + y*src->linesize[1]];
-            V = src->data[2][x + y*src->linesize[2]];
-            A = src->data[3][2*x     +  2*y   *src->linesize[3]]
-              + src->data[3][2*x + 1 +  2*y   *src->linesize[3]]
-              + src->data[3][2*x + 1 + (2*y+1)*src->linesize[3]]
-              + src->data[3][2*x     + (2*y+1)*src->linesize[3]];
-            cb[0] = ALPHA_BLEND(A>>2, cb[0], U, 0);
-            cr[0] = ALPHA_BLEND(A>>2, cr[0], V, 0);
+    if (dsty & 1) {
+        lum += dstx;
+        cb += skip2;
+        cr += skip2;
+
+        if (dstx & 1) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
             cb++;
             cr++;
+            lum++;
+            p += BPP;
+        }
+        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+            cb++;
+            cr++;
+            p += 2 * BPP;
+            lum += 2;
+        }
+        if (w) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
+            p++;
+            lum++;
+        }
+        p += wrap3 - dstw * BPP;
+        lum += wrap - dstw - dstx;
+        cb += dst->linesize[1] - width2 - skip2;
+        cr += dst->linesize[2] - width2 - skip2;
+    }
+    for (h = dsth - (dsty & 1); h >= 2; h -= 2) {
+        lum += dstx;
+        cb += skip2;
+        cr += skip2;
+
+        if (dstx & 1) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            p += wrap3;
+            lum += wrap;
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+            cb++;
+            cr++;
+            p += -wrap3 + BPP;
+            lum += -wrap + 1;
+        }
+        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+            p += wrap3;
+            lum += wrap;
+
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 2);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 2);
+
+            cb++;
+            cr++;
+            p += -wrap3 + 2 * BPP;
+            lum += -wrap + 2;
+        }
+        if (w) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            p += wrap3;
+            lum += wrap;
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+            cb++;
+            cr++;
+            p += -wrap3 + BPP;
+            lum += -wrap + 1;
+        }
+        p += wrap3 + (wrap3 - dstw * BPP);
+        lum += wrap + (wrap - dstw - dstx);
+        cb += dst->linesize[1] - width2 - skip2;
+        cr += dst->linesize[2] - width2 - skip2;
+    }
+    /* handle odd height */
+    if (h) {
+        lum += dstx;
+        cb += skip2;
+        cr += skip2;
+
+        if (dstx & 1) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
+            cb++;
+            cr++;
+            lum++;
+            p += BPP;
+        }
+        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v, 1);
+            cb++;
+            cr++;
+            p += 2 * BPP;
+            lum += 2;
+        }
+        if (w) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
         }
-        cb += dst->linesize[1] - dstw/2;
-        cr += dst->linesize[2] - dstw/2;
     }
 }
 
@@ -1132,7 +1305,6 @@ static void stream_close(VideoState *is)
 #if !CONFIG_AVFILTER
     sws_freeContext(is->img_convert_ctx);
 #endif
-    sws_freeContext(is->sub_convert_ctx);
     av_free(is);
 }
 
@@ -1302,11 +1474,11 @@ static double get_master_clock(VideoState *is)
 }
 
 static void check_external_clock_speed(VideoState *is) {
-   if (is->video_stream >= 0 && is->videoq.nb_packets <= EXTERNAL_CLOCK_MIN_FRAMES ||
-       is->audio_stream >= 0 && is->audioq.nb_packets <= EXTERNAL_CLOCK_MIN_FRAMES) {
+   if (is->video_stream >= 0 && is->videoq.nb_packets <= MIN_FRAMES / 2 ||
+       is->audio_stream >= 0 && is->audioq.nb_packets <= MIN_FRAMES / 2) {
        set_clock_speed(&is->extclk, FFMAX(EXTERNAL_CLOCK_SPEED_MIN, is->extclk.speed - EXTERNAL_CLOCK_SPEED_STEP));
-   } else if ((is->video_stream < 0 || is->videoq.nb_packets > EXTERNAL_CLOCK_MAX_FRAMES) &&
-              (is->audio_stream < 0 || is->audioq.nb_packets > EXTERNAL_CLOCK_MAX_FRAMES)) {
+   } else if ((is->video_stream < 0 || is->videoq.nb_packets > MIN_FRAMES * 2) &&
+              (is->audio_stream < 0 || is->audioq.nb_packets > MIN_FRAMES * 2)) {
        set_clock_speed(&is->extclk, FFMIN(EXTERNAL_CLOCK_SPEED_MAX, is->extclk.speed + EXTERNAL_CLOCK_SPEED_STEP));
    } else {
        double speed = is->extclk.speed;
@@ -1359,7 +1531,7 @@ static void step_to_next_frame(VideoState *is)
 
 static double compute_target_delay(double delay, VideoState *is)
 {
-    double sync_threshold, diff = 0;
+    double sync_threshold, diff;
 
     /* update delay to follow master synchronisation source */
     if (get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER) {
@@ -1381,7 +1553,7 @@ static double compute_target_delay(double delay, VideoState *is)
         }
     }
 
-    av_log(NULL, AV_LOG_TRACE, "video: delay=%0.3f A-V=%f\n",
+    av_dlog(NULL, "video: delay=%0.3f A-V=%f\n",
             delay, -diff);
 
     return delay;
@@ -1678,18 +1850,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
         av_picture_copy(&pict, (AVPicture *)src_frame,
                         src_frame->format, vp->width, vp->height);
 #else
-        {
-            AVDictionaryEntry *e = av_dict_get(sws_dict, "sws_flags", NULL, 0);
-            if (e) {
-                const AVClass *class = sws_get_class();
-                const AVOption    *o = av_opt_find(&class, "sws_flags", NULL, 0,
-                                                   AV_OPT_SEARCH_FAKE_OBJ);
-                int ret = av_opt_eval_flags(&class, o, e->value, &sws_flags);
-                if (ret < 0)
-                    exit(1);
-            }
-        }
-
+        av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
         is->img_convert_ctx = sws_getCachedContext(is->img_convert_ctx,
             vp->width, vp->height, src_frame->format, vp->width, vp->height,
             AV_PIX_FMT_YUV420P, sws_flags, NULL, NULL, NULL);
@@ -1731,9 +1892,6 @@ static int get_video_frame(VideoState *is, AVFrame *frame)
 
         frame->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, frame);
 
-        is->viddec_width  = frame->width;
-        is->viddec_height = frame->height;
-
         if (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) {
             if (frame->pts != AV_NOPTS_VALUE) {
                 double diff = dpts - get_master_clock(is);
@@ -1799,23 +1957,15 @@ fail:
 static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const char *vfilters, AVFrame *frame)
 {
     static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_YUV420P, AV_PIX_FMT_NONE };
-    char sws_flags_str[512] = "";
+    char sws_flags_str[128];
     char buffersrc_args[256];
     int ret;
     AVFilterContext *filt_src = NULL, *filt_out = NULL, *last_filter = NULL;
     AVCodecContext *codec = is->video_st->codec;
     AVRational fr = av_guess_frame_rate(is->ic, is->video_st, NULL);
-    AVDictionaryEntry *e = NULL;
-
-    while ((e = av_dict_get(sws_dict, "", e, AV_DICT_IGNORE_SUFFIX))) {
-        if (!strcmp(e->key, "sws_flags")) {
-            av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", "flags", e->value);
-        } else
-            av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", e->key, e->value);
-    }
-    if (strlen(sws_flags_str))
-        sws_flags_str[strlen(sws_flags_str)-1] = '\0';
 
+    av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
+    snprintf(sws_flags_str, sizeof(sws_flags_str), "flags=%"PRId64, sws_flags);
     graph->scale_sws_opts = av_strdup(sws_flags_str);
 
     snprintf(buffersrc_args, sizeof(buffersrc_args),
@@ -1845,20 +1995,20 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
 
 /* Note: this macro adds a filter before the lastly added filter, so the
  * processing order of the filters is in reverse */
-#define INSERT_FILT(name, arg) do {                                          \
-    AVFilterContext *filt_ctx;                                               \
-                                                                             \
-    ret = avfilter_graph_create_filter(&filt_ctx,                            \
-                                       avfilter_get_by_name(name),           \
-                                       "ffplay_" name, arg, NULL, graph);    \
-    if (ret < 0)                                                             \
-        goto fail;                                                           \
-                                                                             \
-    ret = avfilter_link(filt_ctx, 0, last_filter, 0);                        \
-    if (ret < 0)                                                             \
-        goto fail;                                                           \
-                                                                             \
-    last_filter = filt_ctx;                                                  \
+#define INSERT_FILT(name, arg) do {                                         \
+    AVFilterContext *filt_ctx;                                              \
+                                                                            \
+    ret = avfilter_graph_create_filter(&filt_ctx,                           \
+                                       avfilter_get_by_name(name),          \
+                                       "ffplay_" name, arg, NULL, graph);   \
+    if (ret < 0)                                                            \
+        goto fail;                                                          \
+                                                                            \
+    ret = avfilter_link(filt_ctx, 0, last_filter, 0);                       \
+    if (ret < 0)                                                            \
+        goto fail;                                                          \
+                                                                            \
+    last_filter = filt_ctx;                                                 \
 } while (0)
 
     /* SDL YUV code is not handling odd width/height for some driver
@@ -1866,19 +2016,20 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
     INSERT_FILT("crop", "floor(in_w/2)*2:floor(in_h/2)*2");
 
     if (autorotate) {
-        double theta  = get_rotation(is->video_st);
-
-        if (fabs(theta - 90) < 1.0) {
-            INSERT_FILT("transpose", "clock");
-        } else if (fabs(theta - 180) < 1.0) {
-            INSERT_FILT("hflip", NULL);
-            INSERT_FILT("vflip", NULL);
-        } else if (fabs(theta - 270) < 1.0) {
-            INSERT_FILT("transpose", "cclock");
-        } else if (fabs(theta) > 1.0) {
-            char rotate_buf[64];
-            snprintf(rotate_buf, sizeof(rotate_buf), "%f*PI/180", theta);
-            INSERT_FILT("rotate", rotate_buf);
+        AVDictionaryEntry *rotate_tag = av_dict_get(is->video_st->metadata, "rotate", NULL, 0);
+        if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
+            if (!strcmp(rotate_tag->value, "90")) {
+                INSERT_FILT("transpose", "clock");
+            } else if (!strcmp(rotate_tag->value, "180")) {
+                INSERT_FILT("hflip", NULL);
+                INSERT_FILT("vflip", NULL);
+            } else if (!strcmp(rotate_tag->value, "270")) {
+                INSERT_FILT("transpose", "cclock");
+            } else {
+                char rotate_buf[64];
+                snprintf(rotate_buf, sizeof(rotate_buf), "%s*PI/180", rotate_tag->value);
+                INSERT_FILT("rotate", rotate_buf);
+            }
         }
     }
 
@@ -2080,19 +2231,10 @@ static int video_thread(void *arg)
     enum AVPixelFormat last_format = -2;
     int last_serial = -1;
     int last_vfilter_idx = 0;
-    if (!graph) {
-        av_frame_free(&frame);
-        return AVERROR(ENOMEM);
-    }
-
 #endif
 
-    if (!frame) {
-#if CONFIG_AVFILTER
-        avfilter_graph_free(&graph);
-#endif
+    if (!frame)
         return AVERROR(ENOMEM);
-    }
 
     for (;;) {
         ret = get_video_frame(is, frame);
@@ -2177,7 +2319,8 @@ static int subtitle_thread(void *arg)
     Frame *sp;
     int got_subtitle;
     double pts;
-    int i;
+    int i, j;
+    int r, g, b, y, u, v, a;
 
     for (;;) {
         if (!(sp = frame_queue_peek_writable(&is->subpq)))
@@ -2196,41 +2339,14 @@ static int subtitle_thread(void *arg)
 
             for (i = 0; i < sp->sub.num_rects; i++)
             {
-                int in_w = sp->sub.rects[i]->w;
-                int in_h = sp->sub.rects[i]->h;
-                int subw = is->subdec.avctx->width  ? is->subdec.avctx->width  : is->viddec_width;
-                int subh = is->subdec.avctx->height ? is->subdec.avctx->height : is->viddec_height;
-                int out_w = is->viddec_width  ? in_w * is->viddec_width  / subw : in_w;
-                int out_h = is->viddec_height ? in_h * is->viddec_height / subh : in_h;
-                AVPicture newpic;
-
-                //can not use avpicture_alloc as it is not compatible with avsubtitle_free()
-                av_image_fill_linesizes(newpic.linesize, AV_PIX_FMT_YUVA420P, out_w);
-                newpic.data[0] = av_malloc(newpic.linesize[0] * out_h);
-                newpic.data[3] = av_malloc(newpic.linesize[3] * out_h);
-                newpic.data[1] = av_malloc(newpic.linesize[1] * ((out_h+1)/2));
-                newpic.data[2] = av_malloc(newpic.linesize[2] * ((out_h+1)/2));
-
-                is->sub_convert_ctx = sws_getCachedContext(is->sub_convert_ctx,
-                    in_w, in_h, AV_PIX_FMT_PAL8, out_w, out_h,
-                    AV_PIX_FMT_YUVA420P, sws_flags, NULL, NULL, NULL);
-                if (!is->sub_convert_ctx || !newpic.data[0] || !newpic.data[3] ||
-                    !newpic.data[1] || !newpic.data[2]
-                ) {
-                    av_log(NULL, AV_LOG_FATAL, "Cannot initialize the sub conversion context\n");
-                    exit(1);
+                for (j = 0; j < sp->sub.rects[i]->nb_colors; j++)
+                {
+                    RGBA_IN(r, g, b, a, (uint32_t*)sp->sub.rects[i]->pict.data[1] + j);
+                    y = RGB_TO_Y_CCIR(r, g, b);
+                    u = RGB_TO_U_CCIR(r, g, b, 0);
+                    v = RGB_TO_V_CCIR(r, g, b, 0);
+                    YUVA_OUT((uint32_t*)sp->sub.rects[i]->pict.data[1] + j, y, u, v, a);
                 }
-                sws_scale(is->sub_convert_ctx,
-                          (void*)sp->sub.rects[i]->pict.data, sp->sub.rects[i]->pict.linesize,
-                          0, in_h, newpic.data, newpic.linesize);
-
-                av_free(sp->sub.rects[i]->pict.data[0]);
-                av_free(sp->sub.rects[i]->pict.data[1]);
-                sp->sub.rects[i]->pict = newpic;
-                sp->sub.rects[i]->w = out_w;
-                sp->sub.rects[i]->h = out_h;
-                sp->sub.rects[i]->x = sp->sub.rects[i]->x * out_w / in_w;
-                sp->sub.rects[i]->y = sp->sub.rects[i]->y * out_h / in_h;
             }
 
             /* now we can update the picture count */
@@ -2287,9 +2403,9 @@ static int synchronize_audio(VideoState *is, int nb_samples)
                     wanted_nb_samples = nb_samples + (int)(diff * is->audio_src.freq);
                     min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100));
                     max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
-                    wanted_nb_samples = av_clip(wanted_nb_samples, min_nb_samples, max_nb_samples);
+                    wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
                 }
-                av_log(NULL, AV_LOG_TRACE, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
+                av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
                         diff, avg_diff, wanted_nb_samples - nb_samples,
                         is->audio_clock, is->audio_diff_threshold);
             }
@@ -2323,13 +2439,6 @@ static int audio_decode_frame(VideoState *is)
         return -1;
 
     do {
-#if defined(_WIN32)
-        while (frame_queue_nb_remaining(&is->sampq) == 0) {
-            if ((av_gettime_relative() - audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)
-                return -1;
-            av_usleep (1000);
-        }
-#endif
         if (!(af = frame_queue_peek_readable(&is->sampq)))
             return -1;
         frame_queue_next(&is->sampq);
@@ -2577,15 +2686,10 @@ static int stream_component_open(VideoState *is, int stream_index)
     }
     av_codec_set_lowres(avctx, stream_lowres);
 
-#if FF_API_EMU_EDGE
     if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
-#endif
-    if (fast)
-        avctx->flags2 |= AV_CODEC_FLAG2_FAST;
-#if FF_API_EMU_EDGE
-    if(codec->capabilities & AV_CODEC_CAP_DR1)
+    if (fast)   avctx->flags2 |= CODEC_FLAG2_FAST;
+    if(codec->capabilities & CODEC_CAP_DR1)
         avctx->flags |= CODEC_FLAG_EMU_EDGE;
-#endif
 
     opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
     if (!av_dict_get(opts, "threads", NULL, 0))
@@ -2658,9 +2762,6 @@ static int stream_component_open(VideoState *is, int stream_index)
         is->video_stream = stream_index;
         is->video_st = ic->streams[stream_index];
 
-        is->viddec_width  = avctx->width;
-        is->viddec_height = avctx->height;
-
         decoder_init(&is->viddec, avctx, &is->videoq, is->continue_read_thread);
         decoder_start(&is->viddec, video_thread, is);
         is->queue_attachments_req = 1;

ffprobe.c

@@ -33,12 +33,10 @@
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
-#include "libavutil/display.h"
 #include "libavutil/hash.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/dict.h"
-#include "libavutil/intreadwrite.h"
 #include "libavutil/libm.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/timecode.h"
@@ -137,8 +135,6 @@ typedef enum {
     SECTION_ID_PACKET,
     SECTION_ID_PACKETS,
     SECTION_ID_PACKETS_AND_FRAMES,
-    SECTION_ID_PACKET_SIDE_DATA_LIST,
-    SECTION_ID_PACKET_SIDE_DATA,
     SECTION_ID_PIXEL_FORMAT,
     SECTION_ID_PIXEL_FORMAT_FLAGS,
     SECTION_ID_PIXEL_FORMAT_COMPONENT,
@@ -157,8 +153,6 @@ typedef enum {
     SECTION_ID_STREAM_DISPOSITION,
     SECTION_ID_STREAMS,
     SECTION_ID_STREAM_TAGS,
-    SECTION_ID_STREAM_SIDE_DATA_LIST,
-    SECTION_ID_STREAM_SIDE_DATA,
     SECTION_ID_SUBTITLE,
 } SectionID;
 
@@ -178,9 +172,7 @@ static struct section sections[] = {
     [SECTION_ID_LIBRARY_VERSION] =    { SECTION_ID_LIBRARY_VERSION, "library_version", 0, { -1 } },
     [SECTION_ID_PACKETS] =            { SECTION_ID_PACKETS, "packets", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
     [SECTION_ID_PACKETS_AND_FRAMES] = { SECTION_ID_PACKETS_AND_FRAMES, "packets_and_frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
-    [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { SECTION_ID_PACKET_SIDE_DATA_LIST, -1 } },
-    [SECTION_ID_PACKET_SIDE_DATA_LIST] ={ SECTION_ID_PACKET_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET_SIDE_DATA, -1 } },
-    [SECTION_ID_PACKET_SIDE_DATA] =     { SECTION_ID_PACKET_SIDE_DATA, "side_data", 0, { -1 } },
+    [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { -1 } },
     [SECTION_ID_PIXEL_FORMATS] =      { SECTION_ID_PIXEL_FORMATS, "pixel_formats", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PIXEL_FORMAT, -1 } },
     [SECTION_ID_PIXEL_FORMAT] =       { SECTION_ID_PIXEL_FORMAT, "pixel_format", 0, { SECTION_ID_PIXEL_FORMAT_FLAGS, SECTION_ID_PIXEL_FORMAT_COMPONENTS, -1 } },
     [SECTION_ID_PIXEL_FORMAT_FLAGS] = { SECTION_ID_PIXEL_FORMAT_FLAGS, "flags", 0, { -1 }, .unique_name = "pixel_format_flags" },
@@ -199,11 +191,9 @@ static struct section sections[] = {
                                           SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS,
                                           SECTION_ID_PIXEL_FORMATS, -1} },
     [SECTION_ID_STREAMS] =            { SECTION_ID_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM, -1 } },
-    [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, SECTION_ID_STREAM_SIDE_DATA_LIST, -1 } },
+    [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
     [SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
     [SECTION_ID_STREAM_TAGS] =        { SECTION_ID_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "stream_tags" },
-    [SECTION_ID_STREAM_SIDE_DATA_LIST] ={ SECTION_ID_STREAM_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_SIDE_DATA, -1 } },
-    [SECTION_ID_STREAM_SIDE_DATA] =     { SECTION_ID_STREAM_SIDE_DATA, "side_data", 0, { -1 } },
     [SECTION_ID_SUBTITLE] =           { SECTION_ID_SUBTITLE, "subtitle", 0, { -1 } },
 };
 
@@ -728,32 +718,6 @@ static void writer_print_data_hash(WriterContext *wctx, const char *name,
     writer_print_string(wctx, name, buf, 0);
 }
 
-static void writer_print_integers(WriterContext *wctx, const char *name,
-                                  uint8_t *data, int size, const char *format,
-                                  int columns, int bytes, int offset_add)
-{
-    AVBPrint bp;
-    int offset = 0, l, i;
-
-    av_bprint_init(&bp, 0, AV_BPRINT_SIZE_UNLIMITED);
-    av_bprintf(&bp, "\n");
-    while (size) {
-        av_bprintf(&bp, "%08x: ", offset);
-        l = FFMIN(size, columns);
-        for (i = 0; i < l; i++) {
-            if      (bytes == 1) av_bprintf(&bp, format, *data);
-            else if (bytes == 2) av_bprintf(&bp, format, AV_RN16(data));
-            else if (bytes == 4) av_bprintf(&bp, format, AV_RN32(data));
-            data += bytes;
-            size --;
-        }
-        av_bprintf(&bp, "\n");
-        offset += offset_add;
-    }
-    writer_print_string(wctx, name, bp.str, 0);
-    av_bprint_finalize(&bp, NULL);
-}
-
 #define MAX_REGISTERED_WRITERS_NB 64
 
 static const Writer *registered_writers[MAX_REGISTERED_WRITERS_NB + 1];
@@ -1759,25 +1723,6 @@ static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pk
     if (pkt->pos != -1) print_fmt    ("pos", "%"PRId64, pkt->pos);
     else                print_str_opt("pos", "N/A");
     print_fmt("flags", "%c",      pkt->flags & AV_PKT_FLAG_KEY ? 'K' : '_');
-
-    if (pkt->side_data_elems) {
-        int i;
-        writer_print_section_header(w, SECTION_ID_PACKET_SIDE_DATA_LIST);
-        for (i = 0; i < pkt->side_data_elems; i++) {
-            AVPacketSideData *sd = &pkt->side_data[i];
-            const char *name = av_packet_side_data_name(sd->type);
-            writer_print_section_header(w, SECTION_ID_PACKET_SIDE_DATA);
-            print_str("side_data_type", name ? name : "unknown");
-            print_int("side_data_size", sd->size);
-            if (sd->type == AV_PKT_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
-                writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
-                print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
-            }
-            writer_print_section_footer(w);
-        }
-        writer_print_section_footer(w);
-    }
-
     if (do_show_data)
         writer_print_data(w, "data", pkt->data, pkt->size);
     writer_print_data_hash(w, "data_hash", pkt->data, pkt->size);
@@ -1824,7 +1769,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
     s = av_get_media_type_string(stream->codec->codec_type);
     if (s) print_str    ("media_type", s);
     else   print_str_opt("media_type", "unknown");
-    print_int("stream_index",           stream->index);
     print_int("key_frame",              frame->key_frame);
     print_ts  ("pkt_pts",               frame->pkt_pts);
     print_time("pkt_pts_time",          frame->pkt_pts, &stream->time_base);
@@ -1889,11 +1833,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
             name = av_frame_side_data_name(sd->type);
             print_str("side_data_type", name ? name : "unknown");
             print_int("side_data_size", sd->size);
-            if (sd->type == AV_FRAME_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
-                abort();
-                writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
-                print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
-            }
             writer_print_section_footer(w);
         }
         writer_print_section_footer(w);
@@ -2155,8 +2094,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
         case AVMEDIA_TYPE_VIDEO:
             print_int("width",        dec_ctx->width);
             print_int("height",       dec_ctx->height);
-            print_int("coded_width",  dec_ctx->coded_width);
-            print_int("coded_height", dec_ctx->coded_height);
             print_int("has_b_frames", dec_ctx->has_b_frames);
             sar = av_guess_sample_aspect_ratio(fmt_ctx, stream, NULL);
             if (sar.den) {
@@ -2302,25 +2239,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
     if (do_show_stream_tags)
         ret = show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
 
-    if (stream->nb_side_data) {
-        int i;
-        writer_print_section_header(w, SECTION_ID_STREAM_SIDE_DATA_LIST);
-        for (i = 0; i < stream->nb_side_data; i++) {
-            AVPacketSideData *sd = &stream->side_data[i];
-            const char *name = av_packet_side_data_name(sd->type);
-
-            writer_print_section_header(w, SECTION_ID_STREAM_SIDE_DATA);
-            print_str("side_data_type", name ? name : "unknown");
-            print_int("side_data_size", sd->size);
-            if (sd->type == AV_PKT_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
-                writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
-                print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
-            }
-            writer_print_section_footer(w);
-        }
-        writer_print_section_footer(w);
-    }
-
     writer_print_section_footer(w);
     av_bprint_finalize(&pbuf, NULL);
     fflush(stdout);
@@ -2831,9 +2749,6 @@ static int opt_show_format_entry(void *optctx, const char *opt, const char *arg)
     char *buf = av_asprintf("format=%s", arg);
     int ret;
 
-    if (!buf)
-        return AVERROR(ENOMEM);
-
     av_log(NULL, AV_LOG_WARNING,
            "Option '%s' is deprecated, use '-show_entries format=%s' instead\n",
            opt, arg);

ffserver.c

@@ -31,7 +31,7 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include "libavformat/avformat.h"
-/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */
+// FIXME those are internal headers, ffserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
@@ -209,7 +209,6 @@ static void close_connection(HTTPContext *c);
 
 /* HTTP handling */
 static int handle_connection(HTTPContext *c);
-static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream);
 static void compute_status(HTTPContext *c);
 static int open_input_stream(HTTPContext *c, const char *info);
 static int http_parse_request(HTTPContext *c);
@@ -251,8 +250,7 @@ static unsigned int nb_connections;
 
 static uint64_t current_bandwidth;
 
-/* Making this global saves on passing it around everywhere */
-static int64_t cur_time;
+static int64_t cur_time;           // Making this global saves on passing it around everywhere
 
 static AVLFG random_state;
 
@@ -316,12 +314,12 @@ static char *ctime1(char *buf2, int buf_size)
 static void http_vlog(const char *fmt, va_list vargs)
 {
     static int print_prefix = 1;
-    char buf[32];
 
     if (!logfile)
         return;
 
     if (print_prefix) {
+        char buf[32];
         ctime1(buf, sizeof(buf));
         fprintf(logfile, "%s ", buf);
     }
@@ -506,7 +504,8 @@ static void start_multicast(void)
         random1 = av_lfg_get(&random_state);
 
         /* open the RTP connection */
-        snprintf(session_id, sizeof(session_id), "%08x%08x", random0, random1);
+        snprintf(session_id, sizeof(session_id), "%08x%08x",
+                 random0, random1);
 
         /* choose a port if none given */
         if (stream->multicast_port == 0) {
@@ -631,8 +630,9 @@ static int http_server(void)
                     poll_entry++;
                 } else {
                     /* when ffserver is doing the timing, we work by
-                     * looking at which packet needs to be sent every
-                     * 10 ms (one tick wait XXX: 10 ms assumed) */
+                       looking at which packet needs to be sent every
+                       10 ms */
+                    /* one tick wait XXX: 10 ms assumed */
                     if (delay > 10)
                         delay = 10;
                 }
@@ -655,7 +655,7 @@ static int http_server(void)
         }
 
         /* wait for an event on one connection. We poll at least every
-         * second to handle timeouts */
+           second to handle timeouts */
         do {
             ret = poll(poll_table, poll_entry - poll_table, delay);
             if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) &&
@@ -703,9 +703,13 @@ static void start_wait_request(HTTPContext *c, int is_rtsp)
     c->buffer_ptr = c->buffer;
     c->buffer_end = c->buffer + c->buffer_size - 1; /* leave room for '\0' */
 
-    c->state = is_rtsp ? RTSPSTATE_WAIT_REQUEST : HTTPSTATE_WAIT_REQUEST;
-    c->timeout = cur_time +
-                 (is_rtsp ? RTSP_REQUEST_TIMEOUT : HTTP_REQUEST_TIMEOUT);
+    if (is_rtsp) {
+        c->timeout = cur_time + RTSP_REQUEST_TIMEOUT;
+        c->state = RTSPSTATE_WAIT_REQUEST;
+    } else {
+        c->timeout = cur_time + HTTP_REQUEST_TIMEOUT;
+        c->state = HTTPSTATE_WAIT_REQUEST;
+    }
 }
 
 static void http_send_too_busy_reply(int fd)
@@ -783,6 +787,7 @@ static void close_connection(HTTPContext *c)
     HTTPContext **cp, *c1;
     int i, nb_streams;
     AVFormatContext *ctx;
+    URLContext *h;
     AVStream *st;
 
     /* remove connection from list */
@@ -827,7 +832,9 @@ static void close_connection(HTTPContext *c)
             av_freep(&ctx->streams[0]);
             av_freep(&ctx);
         }
-        ffurl_close(c->rtp_handles[i]);
+        h = c->rtp_handles[i];
+        if (h)
+            ffurl_close(h);
     }
 
     ctx = &c->fmt_ctx;
@@ -896,11 +903,11 @@ static int handle_connection(HTTPContext *c)
         if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
             (ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
             /* request found : parse it and reply */
-            if (c->state == HTTPSTATE_WAIT_REQUEST)
+            if (c->state == HTTPSTATE_WAIT_REQUEST) {
                 ret = http_parse_request(c);
-            else
+            } else {
                 ret = rtsp_parse_request(c);
-
+            }
             if (ret < 0)
                 return -1;
         } else if (ptr >= c->buffer_end) {
@@ -945,8 +952,8 @@ static int handle_connection(HTTPContext *c)
     case HTTPSTATE_SEND_DATA_HEADER:
     case HTTPSTATE_SEND_DATA_TRAILER:
         /* for packetized output, we consider we can always write (the
-         * input streams set the speed). It may be better to verify
-         * that we do not rely too much on the kernel queues */
+           input streams set the speed). It may be better to verify
+           that we do not rely too much on the kernel queues */
         if (!c->is_packetized) {
             if (c->poll_entry->revents & (POLLERR | POLLHUP))
                 return -1;
@@ -960,10 +967,6 @@ static int handle_connection(HTTPContext *c)
         /* close connection if trailer sent */
         if (c->state == HTTPSTATE_SEND_DATA_TRAILER)
             return -1;
-        /* Check if it is a single jpeg frame 123 */
-        if (c->stream->single_frame && c->data_count > c->cur_frame_bytes && c->cur_frame_bytes > 0) {
-            close_connection(c);
-        }
         break;
     case HTTPSTATE_RECEIVE_DATA:
         /* no need to read if no events */
@@ -1159,10 +1162,8 @@ static int modify_current_stream(HTTPContext *c, char *rates)
                 break;
         }
 
-        if (c->switch_feed_streams[i] >= 0 &&
-            c->switch_feed_streams[i] != c->feed_streams[i]) {
+        if (c->switch_feed_streams[i] >= 0 && c->switch_feed_streams[i] != c->feed_streams[i])
             action_required = 1;
-        }
     }
 
     return action_required;
@@ -1203,10 +1204,6 @@ static FFServerIPAddressACL* parse_dynamic_acl(FFServerStream *stream,
     }
 
     acl = av_mallocz(sizeof(FFServerIPAddressACL));
-    if (!acl) {
-        fclose(f);
-        return NULL;
-    }
 
     /* Build ACL */
     while (fgets(line, sizeof(line), f)) {
@@ -1266,17 +1263,17 @@ static int validate_acl(FFServerStream *stream, HTTPContext *c)
 
     if (stream->dynamic_acl[0]) {
         acl = parse_dynamic_acl(stream, c);
+
         ret = validate_acl_list(acl, c);
+
         free_acl_list(acl);
     }
 
     return ret;
 }
 
-/**
- * compute the real filename of a file by matching it without its
- * extensions to all the stream's filenames
- */
+/* compute the real filename of a file by matching it without its
+   extensions to all the stream's filenames */
 static void compute_real_filename(char *filename, int max_size)
 {
     char file1[1024];
@@ -1394,7 +1391,7 @@ static int http_parse_request(HTTPContext *c)
         compute_real_filename(filename, sizeof(filename) - 1);
     }
 
-    /* "redirect" request to index.html */
+    // "redirect" / request to index.html
     if (!strlen(filename))
         av_strlcpy(filename, "index.html", sizeof(filename) - 1);
 
@@ -1733,9 +1730,8 @@ static int http_parse_request(HTTPContext *c)
     return 0;
  send_status:
     compute_status(c);
-    /* horrible: we use this value to avoid
-     * going to the send data state */
-    c->http_error = 200;
+    c->http_error = 200; /* horrible : we use this value to avoid
+                            going to the send data state */
     c->state = HTTPSTATE_SEND_HEADER;
     return 0;
 }
@@ -1750,52 +1746,6 @@ static void fmt_bytecount(AVIOContext *pb, int64_t count)
     avio_printf(pb, "%"PRId64"%c", count, *s);
 }
 
-static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream)
-{
-    int i, stream_no;
-    const char *type = "unknown";
-    char parameters[64];
-    AVStream *st;
-    AVCodec *codec;
-
-    stream_no = stream->nb_streams;
-
-    avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
-                    "type<th>kbits/s<th align=left>codec<th align=left>"
-                    "Parameters\n");
-
-    for (i = 0; i < stream_no; i++) {
-        st = stream->streams[i];
-        codec = avcodec_find_encoder(st->codec->codec_id);
-
-        parameters[0] = 0;
-
-        switch(st->codec->codec_type) {
-        case AVMEDIA_TYPE_AUDIO:
-            type = "audio";
-            snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
-                     st->codec->channels, st->codec->sample_rate);
-            break;
-        case AVMEDIA_TYPE_VIDEO:
-            type = "video";
-            snprintf(parameters, sizeof(parameters),
-                     "%dx%d, q=%d-%d, fps=%d", st->codec->width,
-                     st->codec->height, st->codec->qmin, st->codec->qmax,
-                     st->codec->time_base.den / st->codec->time_base.num);
-            break;
-        default:
-            abort();
-        }
-
-        avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d"
-                        "<td>%s<td>%s\n",
-                    i, type, st->codec->bit_rate/1000,
-                    codec ? codec->name : "", parameters);
-     }
-
-     avio_printf(pb, "</table>\n");
-}
-
 static void compute_status(HTTPContext *c)
 {
     HTTPContext *c1;
@@ -1846,8 +1796,8 @@ static void compute_status(HTTPContext *c)
                 strcpy(eosf - 3, ".ram");
             else if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
                 /* generate a sample RTSP director if
-                 * unicast. Generate an SDP redirector if
-                 * multicast */
+                   unicast. Generate an SDP redirector if
+                   multicast */
                 eosf = strrchr(sfilename, '.');
                 if (!eosf)
                     eosf = sfilename + strlen(sfilename);
@@ -1936,7 +1886,7 @@ static void compute_status(HTTPContext *c)
 
         avio_printf(pb, "<h2>Feed %s</h2>", stream->filename);
         if (stream->pid) {
-            avio_printf(pb, "Running as pid %"PRId64".\n", (int64_t) stream->pid);
+            avio_printf(pb, "Running as pid %d.\n", stream->pid);
 
 #if defined(linux)
             {
@@ -1945,8 +1895,8 @@ static void compute_status(HTTPContext *c)
 
                 /* This is somewhat linux specific I guess */
                 snprintf(ps_cmd, sizeof(ps_cmd),
-                         "ps -o \"%%cpu,cputime\" --no-headers %"PRId64"",
-                         (int64_t) stream->pid);
+                         "ps -o \"%%cpu,cputime\" --no-headers %d",
+                         stream->pid);
 
                  pid_stat = popen(ps_cmd, "r");
                  if (pid_stat) {
@@ -1966,7 +1916,42 @@ static void compute_status(HTTPContext *c)
             avio_printf(pb, "<p>");
         }
 
-        print_stream_params(pb, stream);
+        avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
+                        "type<th>kbits/s<th align=left>codec<th align=left>"
+                        "Parameters\n");
+
+        for (i = 0; i < stream->nb_streams; i++) {
+            AVStream *st = stream->streams[i];
+            AVCodec *codec = avcodec_find_encoder(st->codec->codec_id);
+            const char *type = "unknown";
+            char parameters[64];
+
+            parameters[0] = 0;
+
+            switch(st->codec->codec_type) {
+            case AVMEDIA_TYPE_AUDIO:
+                type = "audio";
+                snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
+                         st->codec->channels, st->codec->sample_rate);
+                break;
+            case AVMEDIA_TYPE_VIDEO:
+                type = "video";
+                snprintf(parameters, sizeof(parameters),
+                         "%dx%d, q=%d-%d, fps=%d", st->codec->width,
+                         st->codec->height, st->codec->qmin, st->codec->qmax,
+                         st->codec->time_base.den / st->codec->time_base.num);
+                break;
+            default:
+                abort();
+            }
+
+            avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d"
+                            "<td>%s<td>%s\n",
+                        i, type, st->codec->bit_rate/1000,
+                        codec ? codec->name : "", parameters);
+        }
+
+        avio_printf(pb, "</table>\n");
         stream = stream->next;
     }
 
@@ -2070,19 +2055,12 @@ static int open_input_stream(HTTPContext *c, const char *info)
     ret = avformat_open_input(&s, input_filename, c->stream->ifmt,
                               &c->stream->in_opts);
     if (ret < 0) {
-        http_log("Could not open input '%s': %s\n",
-                 input_filename, av_err2str(ret));
+        http_log("Could not open input '%s': %s\n", input_filename, av_err2str(ret));
         return ret;
     }
 
     /* set buffer size */
-    if (buf_size > 0) {
-        ret = ffio_set_buf_size(s->pb, buf_size);
-        if (ret < 0) {
-            http_log("Failed to set buffer size\n");
-            return ret;
-        }
-    }
+    if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size);
 
     s->flags |= AVFMT_FLAG_GENPTS;
     c->fmt_in = s;
@@ -2118,7 +2096,8 @@ static int64_t get_server_clock(HTTPContext *c)
     return (cur_time - c->start_time) * 1000;
 }
 
-/* return the estimated time (in us) at which the current packet must be sent */
+/* return the estimated time at which the current packet must be sent
+   (in us) */
 static int64_t get_packet_send_clock(HTTPContext *c)
 {
     int bytes_left, bytes_sent, frame_bytes;
@@ -2126,10 +2105,11 @@ static int64_t get_packet_send_clock(HTTPContext *c)
     frame_bytes = c->cur_frame_bytes;
     if (frame_bytes <= 0)
         return c->cur_pts;
-
-    bytes_left = c->buffer_end - c->buffer_ptr;
-    bytes_sent = frame_bytes - bytes_left;
-    return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
+    else {
+        bytes_left = c->buffer_end - c->buffer_ptr;
+        bytes_sent = frame_bytes - bytes_left;
+        return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
+    }
 }
 
 
@@ -2142,22 +2122,15 @@ static int http_prepare_data(HTTPContext *c)
     switch(c->state) {
     case HTTPSTATE_SEND_DATA_HEADER:
         ctx = avformat_alloc_context();
-        if (!ctx)
-            return AVERROR(ENOMEM);
         c->fmt_ctx = *ctx;
         av_freep(&ctx);
         av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0);
-        c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams,
-                                              sizeof(AVStream *));
-        if (!c->fmt_ctx.streams)
-            return AVERROR(ENOMEM);
+        c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams, sizeof(AVStream *));
 
         for(i=0;i<c->stream->nb_streams;i++) {
             AVStream *src;
             c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));
-
-            /* if file or feed, then just take streams from FFServerStream
-             * struct */
+            /* if file or feed, then just take streams from FFServerStream struct */
             if (!c->stream->feed ||
                 c->stream->feed == c->stream)
                 src = c->stream->streams[i];
@@ -2222,23 +2195,23 @@ static int http_prepare_data(HTTPContext *c)
             if (ret < 0) {
                 if (c->stream->feed) {
                     /* if coming from feed, it means we reached the end of the
-                     * ffm file, so must wait for more data */
+                       ffm file, so must wait for more data */
                     c->state = HTTPSTATE_WAIT_FEED;
                     return 1; /* state changed */
-                }
-                if (ret == AVERROR(EAGAIN)) {
+                } else if (ret == AVERROR(EAGAIN)) {
                     /* input not ready, come back later */
                     return 0;
-                }
-                if (c->stream->loop) {
-                    avformat_close_input(&c->fmt_in);
-                    if (open_input_stream(c, "") < 0)
-                        goto no_loop;
-                    goto redo;
                 } else {
+                    if (c->stream->loop) {
+                        avformat_close_input(&c->fmt_in);
+                        if (open_input_stream(c, "") < 0)
+                            goto no_loop;
+                        goto redo;
+                    } else {
                     no_loop:
                         /* must send trailer now because EOF or error */
                         c->state = HTTPSTATE_SEND_DATA_TRAILER;
+                    }
                 }
             } else {
                 int source_index = pkt.stream_index;
@@ -2307,11 +2280,10 @@ static int http_prepare_data(HTTPContext *c)
                             max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
                         else
                             max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
-                        ret = ffio_open_dyn_packet_buf(&ctx->pb,
-                                                       max_packet_size);
-                    } else
+                        ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
+                    } else {
                         ret = avio_open_dyn_buf(&ctx->pb);
-
+                    }
                     if (ret < 0) {
                         /* XXX: potential leak */
                         return -1;
@@ -2320,13 +2292,10 @@ static int http_prepare_data(HTTPContext *c)
 
                     ctx->pb->seekable = 0;
                     if (pkt.dts != AV_NOPTS_VALUE)
-                        pkt.dts = av_rescale_q(pkt.dts, ist->time_base,
-                                               ost->time_base);
+                        pkt.dts = av_rescale_q(pkt.dts, ist->time_base, ost->time_base);
                     if (pkt.pts != AV_NOPTS_VALUE)
-                        pkt.pts = av_rescale_q(pkt.pts, ist->time_base,
-                                               ost->time_base);
-                    pkt.duration = av_rescale_q(pkt.duration, ist->time_base,
-                                                ost->time_base);
+                        pkt.pts = av_rescale_q(pkt.pts, ist->time_base, ost->time_base);
+                    pkt.duration = av_rescale_q(pkt.duration, ist->time_base, ost->time_base);
                     if ((ret = av_write_frame(ctx, &pkt)) < 0) {
                         http_log("Error writing frame to output for stream '%s': %s\n",
                                  c->stream->filename, av_err2str(ret));
@@ -2374,8 +2343,7 @@ static int http_prepare_data(HTTPContext *c)
 
 /* should convert the format at the same time */
 /* send data starting at c->buffer_ptr to the output connection
- * (either UDP or TCP)
- */
+ * (either UDP or TCP) */
 static int http_send_data(HTTPContext *c)
 {
     int len, ret;
@@ -2456,8 +2424,8 @@ static int http_send_data(HTTPContext *c)
                         rtsp_c->packet_buffer_ptr += len;
                     if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
                         /* if we could not send all the data, we will
-                         * send it later, so a new state is needed to
-                         * "lock" the RTSP TCP connection */
+                           send it later, so a new state is needed to
+                           "lock" the RTSP TCP connection */
                         rtsp_c->state = RTSPSTATE_SEND_PACKET;
                         break;
                     } else
@@ -2469,8 +2437,7 @@ static int http_send_data(HTTPContext *c)
                     ffurl_write(c->rtp_handles[c->packet_stream_index],
                                 c->buffer_ptr, len);
                     c->buffer_ptr += len;
-                    /* here we continue as we can send several packets
-                     * per 10 ms slot */
+                    /* here we continue as we can send several packets per 10 ms slot */
                 }
             } else {
                 /* TCP data output */
@@ -2541,8 +2508,9 @@ static int http_start_receive_data(HTTPContext *c)
             http_log("Error reading write index from feed file '%s': %s\n",
                      c->stream->feed_filename, strerror(errno));
             return ret;
+        } else {
+            c->stream->feed_write_index = ret;
         }
-        c->stream->feed_write_index = ret;
     }
 
     c->stream->feed_write_index = FFMAX(ffm_read_write_index(fd),
@@ -2584,11 +2552,12 @@ static int http_receive_data(HTTPContext *c)
                 goto fail;
             c->buffer_ptr = c->buffer;
             break;
-        } else if (++loop_run > 10)
+        } else if (++loop_run > 10) {
             /* no chunk header, abort */
             goto fail;
-        else
+        } else {
             c->buffer_ptr++;
+        }
     }
 
     if (c->buffer_end > c->buffer_ptr) {
@@ -2621,7 +2590,7 @@ static int http_receive_data(HTTPContext *c)
     if (c->buffer_ptr >= c->buffer_end) {
         FFServerStream *feed = c->stream;
         /* a packet has been received : write it in the store, except
-         * if header */
+           if header */
         if (c->data_count > FFM_PACKET_SIZE) {
             /* XXX: use llseek or url_seek
              * XXX: Should probably fail? */
@@ -2673,9 +2642,6 @@ static int http_receive_data(HTTPContext *c)
 
             pb = avio_alloc_context(c->buffer, c->buffer_end - c->buffer,
                                     0, NULL, NULL, NULL, NULL);
-            if (!pb)
-                goto fail;
-
             pb->seekable = 0;
 
             s->pb = pb;
@@ -2827,10 +2793,10 @@ static int rtsp_parse_request(HTTPContext *c)
  the_end:
     len = avio_close_dyn_buf(c->pb, &c->pb_buffer);
     c->pb = NULL; /* safety */
-    if (len < 0)
+    if (len < 0) {
         /* XXX: cannot do more */
         return -1;
-
+    }
     c->buffer_ptr = c->pb_buffer;
     c->buffer_end = c->pb_buffer + len;
     c->state = RTSPSTATE_SEND_REPLY;
@@ -2849,9 +2815,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
     *pbuffer = NULL;
 
     avc =  avformat_alloc_context();
-    if (!avc || !rtp_format)
+    if (!avc || !rtp_format) {
         return -1;
-
+    }
     avc->oformat = rtp_format;
     av_dict_set(&avc->metadata, "title",
                 entry ? entry->value : "No Title", 0);
@@ -2860,15 +2826,13 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
         snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
                  inet_ntoa(stream->multicast_ip),
                  stream->multicast_port, stream->multicast_ttl);
-    } else
+    } else {
         snprintf(avc->filename, 1024, "rtp://0.0.0.0");
+    }
 
-    avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
-    if (!avc->streams)
+    if (!(avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams))))
         goto sdp_done;
-
-    avs = av_malloc_array(avc->nb_streams, sizeof(*avs));
-    if (!avs)
+    if (!(avs = av_malloc_array(avc->nb_streams, sizeof(*avs))))
         goto sdp_done;
 
     for(i = 0; i < stream->nb_streams; i++) {
@@ -2876,8 +2840,6 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
         avc->streams[i]->codec = stream->streams[i]->codec;
     }
     *pbuffer = av_mallocz(2048);
-    if (!*pbuffer)
-        goto sdp_done;
     av_sdp_create(&avc, 1, *pbuffer, 2048);
 
  sdp_done:
@@ -2891,7 +2853,7 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
 
 static void rtsp_cmd_options(HTTPContext *c, const char *url)
 {
-    /* rtsp_reply_header(c, RTSP_STATUS_OK); */
+//    rtsp_reply_header(c, RTSP_STATUS_OK);
     avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
     avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
     avio_printf(c->pb, "Public: %s\r\n",
@@ -2932,8 +2894,7 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url)
     /* get the host IP */
     len = sizeof(my_addr);
     getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
-    content_length = prepare_sdp_description(stream, &content,
-                                             my_addr.sin_addr);
+    content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
     if (content_length < 0) {
         rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
         return;
@@ -3058,7 +3019,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
     }
 
     /* test if stream is OK (test needed because several SETUP needs
-     * to be done for a given file) */
+       to be done for a given file) */
     if (rtp_c->stream != stream) {
         rtsp_reply_error(c, RTSP_STATUS_SERVICE);
         return;
@@ -3119,10 +3080,8 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
 }
 
 
-/**
- * find an RTP connection by using the session ID. Check consistency
- * with filename
- */
+/* find an RTP connection by using the session ID. Check consistency
+   with filename */
 static HTTPContext *find_rtp_session_with_url(const char *url,
                                               const char *session_id)
 {
@@ -3145,10 +3104,10 @@ static HTTPContext *find_rtp_session_with_url(const char *url,
     for(s=0; s<rtp_c->stream->nb_streams; ++s) {
       snprintf(buf, sizeof(buf), "%s/streamid=%d",
         rtp_c->stream->filename, s);
-      if(!strncmp(path, buf, sizeof(buf)))
-        /* XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE
-         * if nb_streams>1? */
+      if(!strncmp(path, buf, sizeof(buf))) {
+    // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
         return rtp_c;
+      }
     }
     len = strlen(path);
     if (len > 0 && path[len - 1] == '/' &&
@@ -3226,7 +3185,7 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
     const char *proto_str;
 
     /* XXX: should output a warning page when coming
-     * close to the connection limit */
+       close to the connection limit */
     if (nb_connections >= config.nb_max_connections)
         goto fail;
 
@@ -3281,11 +3240,9 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
     return NULL;
 }
 
-/**
- * add a new RTP stream in an RTP connection (used in RTSP SETUP
- * command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
- * used.
- */
+/* add a new RTP stream in an RTP connection (used in RTSP SETUP
+   command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
+   used. */
 static int rtp_new_av_stream(HTTPContext *c,
                              int stream_index, struct sockaddr_in *dest_addr,
                              HTTPContext *rtsp_c)
@@ -3363,10 +3320,10 @@ static int rtp_new_av_stream(HTTPContext *c,
 
     /* normally, no packets should be output here, but the packet size may
      * be checked */
-    if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0)
+    if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
         /* XXX: close stream */
         goto fail;
-
+    }
     if (avformat_write_header(ctx, NULL) < 0) {
     fail:
         if (h)
@@ -3398,17 +3355,13 @@ static AVStream *add_av_stream1(FFServerStream *stream,
         return NULL;
     if (copy) {
         fst->codec = avcodec_alloc_context3(codec->codec);
-        if (!fst->codec) {
-            av_free(fst);
-            return NULL;
-        }
         avcodec_copy_context(fst->codec, codec);
-    } else
+    } else {
         /* live streams must use the actual feed's codec since it may be
          * updated later to carry extradata needed by them.
          */
         fst->codec = codec;
-
+    }
     fst->priv_data = av_mallocz(sizeof(FeedData));
     fst->index = stream->nb_streams;
     avpriv_set_pts_info(fst, 33, 1, 90000);
@@ -3510,7 +3463,8 @@ static void extract_mpeg4_header(AVFormatContext *infile)
                 if (p[0] == 0x00 && p[1] == 0x00 &&
                     p[2] == 0x01 && p[3] == 0xb6) {
                     size = p - pkt.data;
-                    st->codec->extradata = av_mallocz(size + AV_INPUT_BUFFER_PADDING_SIZE);
+                    //                    av_hex_dump_log(infile, AV_LOG_DEBUG, pkt.data, size);
+                    st->codec->extradata = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE);
                     st->codec->extradata_size = size;
                     memcpy(st->codec->extradata, pkt.data, size);
                     break;
@@ -3540,7 +3494,7 @@ static void build_file_streams(void)
             /* open stream */
             if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
                 /* specific case : if transport stream output to RTP,
-                 * we use a raw transport stream reader */
+                   we use a raw transport stream reader */
                 av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
             }
 
@@ -3562,7 +3516,7 @@ static void build_file_streams(void)
                 remove_stream(stream);
             } else {
                 /* find all the AVStreams inside and reference them in
-                 * 'stream' */
+                   'stream' */
                 if (avformat_find_stream_info(infile, NULL) < 0) {
                     http_log("Could not find codec parameters from '%s'\n",
                              stream->feed_filename);
@@ -3589,17 +3543,15 @@ static void build_feed_streams(void)
     /* gather all streams */
     for(stream = config.first_stream; stream; stream = stream->next) {
         feed = stream->feed;
-        if (!feed)
-            continue;
-
-        if (stream->is_feed) {
-            for(i=0;i<stream->nb_streams;i++)
-                stream->feed_streams[i] = i;
-        } else {
-            /* we handle a stream coming from a feed */
-            for(i=0;i<stream->nb_streams;i++)
-                stream->feed_streams[i] = add_av_stream(feed,
-                                                        stream->streams[i]);
+        if (feed) {
+            if (stream->is_feed) {
+                for(i=0;i<stream->nb_streams;i++)
+                    stream->feed_streams[i] = i;
+            } else {
+                /* we handle a stream coming from a feed */
+                for(i=0;i<stream->nb_streams;i++)
+                    stream->feed_streams[i] = add_av_stream(feed, stream->streams[i]);
+            }
         }
     }
 
@@ -3614,12 +3566,7 @@ static void build_feed_streams(void)
 
             if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) {
                 /* set buffer size */
-                int ret = ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
-                if (ret < 0) {
-                    http_log("Failed to set buffer size\n");
-                    exit(1);
-                }
-
+                ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
                 /* Now see if it matches */
                 if (s->nb_streams == feed->nb_streams) {
                     matches = 1;
@@ -3690,11 +3637,6 @@ static void build_feed_streams(void)
         if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) {
             AVFormatContext *s = avformat_alloc_context();
 
-            if (!s) {
-                http_log("Failed to allocate context\n");
-                exit(1);
-            }
-
             if (feed->readonly) {
                 http_log("Unable to create feed file '%s' as it is marked readonly\n",
                     feed->feed_filename);
@@ -3766,24 +3708,24 @@ static void compute_bandwidth(void)
 static void handle_child_exit(int sig)
 {
     pid_t pid;
-    int status, uptime;
+    int status;
 
     while ((pid = waitpid(-1, &status, WNOHANG)) > 0) {
         FFServerStream *feed;
 
         for (feed = config.first_feed; feed; feed = feed->next) {
-            if (feed->pid != pid)
-                continue;
+            if (feed->pid == pid) {
+                int uptime = time(0) - feed->pid_start;
 
-            uptime = time(0) - feed->pid_start;
-            feed->pid = 0;
-            fprintf(stderr,
-                    "%s: Pid %"PRId64" exited with status %d after %d seconds\n",
-                    feed->filename, (int64_t) pid, status, uptime);
+                feed->pid = 0;
+                fprintf(stderr,
+                        "%s: Pid %d exited with status %d after %d seconds\n",
+                        feed->filename, pid, status, uptime);
 
-            if (uptime < 30)
-                /* Turn off any more restarts */
-                ffserver_free_child_args(&feed->child_argv);
+                if (uptime < 30)
+                    /* Turn off any more restarts */
+                    ffserver_free_child_args(&feed->child_argv);
+            }
         }
     }
 

ffserver_config.c

@@ -230,9 +230,9 @@ static void add_codec(FFServerStream *stream, AVCodecContext *av,
     /* compute default parameters */
     switch(av->codec_type) {
     case AVMEDIA_TYPE_AUDIO:
-        if (!av_dict_get(recommended, "b", NULL, 0)) {
+        if (!av_dict_get(recommended, "ab", NULL, 0)) {
             av->bit_rate = 64000;
-            av_dict_set_int(&recommended, "b", av->bit_rate, 0);
+            av_dict_set_int(&recommended, "ab", av->bit_rate, 0);
             WARNING("Setting default value for audio bit rate = %d. "
                     "Use NoDefaults to disable it.\n",
                     av->bit_rate);
@@ -862,10 +862,8 @@ static int ffserver_parse_config_stream(FFServerConfig *config, const char *cmd,
         } else {
             stream->stream_type = STREAM_TYPE_LIVE;
             /* JPEG cannot be used here, so use single frame MJPEG */
-            if (!strcmp(arg, "jpeg")) {
-                strcpy(arg, "singlejpeg");
-                stream->single_frame=1;
-            }
+            if (!strcmp(arg, "jpeg"))
+                strcpy(arg, "mjpeg");
             stream->fmt = ffserver_guess_format(arg, NULL, NULL);
             if (!stream->fmt)
                 ERROR("Unknown Format: '%s'\n", arg);
@@ -923,7 +921,7 @@ static int ffserver_parse_config_stream(FFServerConfig *config, const char *cmd,
         ffserver_get_arg(arg, sizeof(arg), p);
         ffserver_set_float_param(&f, arg, 1000, -FLT_MAX, FLT_MAX, config,
                 "Invalid %s: '%s'\n", cmd, arg);
-        if (ffserver_save_avoption_int("b", (int64_t)lrintf(f),
+        if (ffserver_save_avoption_int("ab", (int64_t)lrintf(f),
                                        AV_OPT_FLAG_AUDIO_PARAM, config) < 0)
             goto nomem;
     } else if (!av_strcasecmp(cmd, "AudioChannels")) {

ffserver_config.h

@@ -79,7 +79,6 @@ typedef struct FFServerStream {
     int multicast_port;           /* first port used for multicast */
     int multicast_ttl;
     int loop;                     /* if true, send the stream in loops (only meaningful if file) */
-    char single_frame;            /* only single frame */
 
     /* feed specific */
     int feed_opened;              /* true if someone is writing to the feed */

libavcodec/012v.c

@@ -151,5 +151,5 @@ AVCodec ff_zero12v_decoder = {
     .id             = AV_CODEC_ID_012V,
     .init           = zero12v_decode_init,
     .decode         = zero12v_decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = CODEC_CAP_DR1,
 };

libavcodec/4xm.c

@@ -559,7 +559,7 @@ static inline void idct_put(FourXContext *f, int x, int y)
         idct(block[i]);
     }
 
-    if (!(f->avctx->flags & AV_CODEC_FLAG_GRAY)) {
+    if (!(f->avctx->flags & CODEC_FLAG_GRAY)) {
         for (i = 4; i < 6; i++)
             idct(block[i]);
     }
@@ -883,11 +883,11 @@ static int decode_frame(AVCodecContext *avctx, void *data,
         }
         cfrm = &f->cfrm[i];
 
-        if (data_size > UINT_MAX -  cfrm->size - AV_INPUT_BUFFER_PADDING_SIZE)
+        if (data_size > UINT_MAX -  cfrm->size - FF_INPUT_BUFFER_PADDING_SIZE)
             return AVERROR_INVALIDDATA;
 
         cfrm->data = av_fast_realloc(cfrm->data, &cfrm->allocated_size,
-                                     cfrm->size + data_size + AV_INPUT_BUFFER_PADDING_SIZE);
+                                     cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
         // explicit check needed as memcpy below might not catch a NULL
         if (!cfrm->data) {
             av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
@@ -1026,5 +1026,5 @@ AVCodec ff_fourxm_decoder = {
     .init           = decode_init,
     .close          = decode_end,
     .decode         = decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = CODEC_CAP_DR1,
 };

libavcodec/8bps.c

@@ -184,5 +184,5 @@ AVCodec ff_eightbps_decoder = {
     .priv_data_size = sizeof(EightBpsContext),
     .init           = decode_init,
     .decode         = decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = CODEC_CAP_DR1,
 };

libavcodec/8svx.c

@@ -194,7 +194,7 @@ AVCodec ff_eightsvx_fib_decoder = {
   .init           = eightsvx_decode_init,
   .decode         = eightsvx_decode_frame,
   .close          = eightsvx_decode_close,
-  .capabilities   = AV_CODEC_CAP_DR1,
+  .capabilities   = CODEC_CAP_DR1,
   .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                     AV_SAMPLE_FMT_NONE },
 };
@@ -209,7 +209,7 @@ AVCodec ff_eightsvx_exp_decoder = {
   .init           = eightsvx_decode_init,
   .decode         = eightsvx_decode_frame,
   .close          = eightsvx_decode_close,
-  .capabilities   = AV_CODEC_CAP_DR1,
+  .capabilities   = CODEC_CAP_DR1,
   .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                     AV_SAMPLE_FMT_NONE },
 };

libavcodec/Makefile

@@ -5,7 +5,6 @@ NAME = avcodec
 HEADERS = avcodec.h                                                     \
           avfft.h                                                       \
           dv_profile.h                                                  \
-          d3d11va.h                                                     \
           dxva2.h                                                       \
           old_codec_ids.h                                               \
           qsv.h                                                         \
@@ -13,7 +12,6 @@ HEADERS = avcodec.h                                                     \
           vda.h                                                         \
           vdpau.h                                                       \
           version.h                                                     \
-          videotoolbox.h                                                \
           vorbis_parser.h                                               \
           xvmc.h                                                        \
 
@@ -57,7 +55,6 @@ FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
 OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
                                           fft_fixed_32.o fft_init_table.o \
                                           $(FFT-OBJS-yes)
-OBJS-$(CONFIG_FLACDSP)                 += flacdsp.o
 OBJS-$(CONFIG_FMTCONVERT)              += fmtconvert.o
 OBJS-$(CONFIG_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H263DSP)                 += h263dsp.o
@@ -73,8 +70,6 @@ OBJS-$(CONFIG_IDCTDSP)                 += idctdsp.o simple_idct.o jrevdct.o
 OBJS-$(CONFIG_IIRFILTER)               += iirfilter.o
 OBJS-$(CONFIG_IMDCT15)                 += imdct15.o
 OBJS-$(CONFIG_INTRAX8)                 += intrax8.o intrax8dsp.o
-OBJS-$(CONFIG_IVIDSP)                  += ivi_dsp.o
-OBJS-$(CONFIG_JPEGTABLES)              += jpegtables.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LLAUDDSP)                += lossless_audiodsp.o
 OBJS-$(CONFIG_LLVIDDSP)                += lossless_videodsp.o
@@ -89,52 +84,34 @@ OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
                                           mpegaudiodsp_data.o           \
                                           mpegaudiodsp_fixed.o          \
                                           mpegaudiodsp_float.o
-OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o rl.o \
-                                          mpegvideo_motion.o mpegutils.o \
-                                          mpegvideodata.o mpegpicture.o
+OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o    \
+                                          mpegvideo_motion.o mpegutils.o
 OBJS-$(CONFIG_MPEGVIDEOENC)            += mpegvideo_enc.o mpeg12data.o  \
                                           motion_est.o ratecontrol.o    \
                                           mpegvideoencdsp.o
-OBJS-$(CONFIG_MSS34DSP)                += mss34dsp.o
-OBJS-$(CONFIG_NVENC)                   += nvenc.o
 OBJS-$(CONFIG_PIXBLOCKDSP)             += pixblockdsp.o
 OBJS-$(CONFIG_QPELDSP)                 += qpeldsp.o
 OBJS-$(CONFIG_QSV)                     += qsv.o
-OBJS-$(CONFIG_QSVDEC)                  += qsvdec.o
-OBJS-$(CONFIG_QSVENC)                  += qsvenc.o
 OBJS-$(CONFIG_RANGECODER)              += rangecoder.o
 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o
 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)
-OBJS-$(CONFIG_RV34DSP)                 += rv34dsp.o
-OBJS-$(CONFIG_SHARED)                  += log2_tab.o reverse.o
-OBJS-$(CONFIG_SINEWIN)                 += sinewin.o sinewin_fixed.o
-OBJS-$(CONFIG_SNAPPY)                  += snappy.o
+OBJS-$(CONFIG_SHARED)                  += log2_tab.o
+OBJS-$(CONFIG_SINEWIN)                 += sinewin.o
 OBJS-$(CONFIG_STARTCODE)               += startcode.o
-OBJS-$(CONFIG_TEXTUREDSP)              += texturedsp.o
-OBJS-$(CONFIG_TEXTUREDSPENC)           += texturedspenc.o
 OBJS-$(CONFIG_TPELDSP)                 += tpeldsp.o
 OBJS-$(CONFIG_VIDEODSP)                += videodsp.o
 OBJS-$(CONFIG_VP3DSP)                  += vp3dsp.o
-OBJS-$(CONFIG_VP56DSP)                 += vp56dsp.o
-OBJS-$(CONFIG_VP8DSP)                  += vp8dsp.o
 OBJS-$(CONFIG_WMA_FREQS)               += wma_freqs.o
-OBJS-$(CONFIG_WMV2DSP)                 += wmv2dsp.o
 
 # decoders/encoders
 OBJS-$(CONFIG_ZERO12V_DECODER)         += 012v.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
-OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps_float.o \
+OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \
                                           aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp.o aacpsdsp_float.o
-OBJS-$(CONFIG_AAC_FIXED_DECODER)       += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
-                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp_fixed.o aacpsdsp_fixed.o
-OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o aacenctab.o    \
+                                          sbrdsp.o aacpsdsp.o
+OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                           aacpsy.o aactab.o      \
-                                          aacenc_is.o \
-                                          aacenc_tns.o \
-                                          aacenc_pred.o \
                                           psymodel.o mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
@@ -156,7 +133,7 @@ OBJS-$(CONFIG_AMRWB_DECODER)           += amrwbdec.o celp_filters.o   \
                                           celp_math.o acelp_filters.o \
                                           acelp_vectors.o             \
                                           acelp_pitch_delay.o
-OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpegenc_common.o \
+OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpeg.o mjpegenc_common.o \
                                           mpegvideo_enc.o motion_est.o \
                                           ratecontrol.o mpeg12data.o   \
                                           mpegvideo.o
@@ -164,7 +141,6 @@ OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
 OBJS-$(CONFIG_APNG_DECODER)            += png.o pngdec.o pngdsp.o
-OBJS-$(CONFIG_APNG_ENCODER)            += png.o pngenc.o
 OBJS-$(CONFIG_SSA_DECODER)             += assdec.o ass.o ass_split.o
 OBJS-$(CONFIG_SSA_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASS_DECODER)             += assdec.o ass.o ass_split.o
@@ -179,7 +155,7 @@ OBJS-$(CONFIG_ATRAC3P_DECODER)         += atrac3plusdec.o atrac3plus.o \
                                           atrac3plusdsp.o atrac.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
-OBJS-$(CONFIG_AVRN_DECODER)            += avrndec.o mjpegdec.o
+OBJS-$(CONFIG_AVRN_DECODER)            += avrndec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_AVRP_DECODER)            += r210dec.o
 OBJS-$(CONFIG_AVRP_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
@@ -208,7 +184,7 @@ OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
 OBJS-$(CONFIG_CINEPAK_ENCODER)         += cinepakenc.o elbg.o
 OBJS-$(CONFIG_CLJR_DECODER)            += cljrdec.o
 OBJS-$(CONFIG_CLJR_ENCODER)            += cljrenc.o
-OBJS-$(CONFIG_CLLC_DECODER)            += cllc.o canopus.o
+OBJS-$(CONFIG_CLLC_DECODER)            += cllc.o
 OBJS-$(CONFIG_COOK_DECODER)            += cook.o
 OBJS-$(CONFIG_COMFORTNOISE_DECODER)    += cngdec.o celp_filters.o
 OBJS-$(CONFIG_COMFORTNOISE_ENCODER)    += cngenc.o
@@ -217,9 +193,8 @@ OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
 OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
                                           dcadata.o dca_exss.o         \
-                                          dca_xll.o synth_filter.o
+                                          synth_filter.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o dcadata.o
-OBJS-$(CONFIG_DDS_DECODER)             += dds.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                           dirac_arith.o mpeg12data.o dirac_dwt.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
@@ -261,8 +236,8 @@ OBJS-$(CONFIG_FFV1_DECODER)            += ffv1dec.o ffv1.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
 OBJS-$(CONFIG_FIC_DECODER)             += fic.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o
-OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
+OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o flacdsp.o
+OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o flacdsp.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
 OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
 OBJS-$(CONFIG_FLASHSV2_ENCODER)        += flashsv2enc.o
@@ -271,7 +246,7 @@ OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o
 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
-OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o elsdec.o
+OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o mjpeg.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                           celp_filters.o celp_math.o
 OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
@@ -284,28 +259,20 @@ OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261data.o h261.o
 OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261data.o h261.o
 OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                           mpeg4video.o mpeg4videodec.o flvdec.o\
-                                          intelh263dec.o h263data.o
+                                          intelh263dec.o
 OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
-                                          h263.o h263data.o ituh263enc.o flvenc.o
+                                          h263.o ituh263enc.o flvenc.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o h264_cabac.o h264_cavlc.o \
                                           h264_direct.o h264_loopfilter.o  \
                                           h264_mb.o h264_picture.o h264_ps.o \
                                           h264_refs.o h264_sei.o h264_slice.o
-OBJS-$(CONFIG_H264_MMAL_DECODER)       += mmaldec.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
-OBJS-$(CONFIG_H264_QSV_DECODER)        += qsvdec_h2645.o
-OBJS-$(CONFIG_H264_QSV_ENCODER)        += qsvenc_h264.o
-OBJS-$(CONFIG_HAP_DECODER)             += hapdec.o hap.o
-OBJS-$(CONFIG_HAP_ENCODER)             += hapenc.o hap.o
+OBJS-$(CONFIG_H264_QSV_DECODER)        += qsv_h264.o
 OBJS-$(CONFIG_HEVC_DECODER)            += hevc.o hevc_mvs.o hevc_ps.o hevc_sei.o \
                                           hevc_cabac.o hevc_refs.o hevcpred.o    \
-                                          hevcdsp.o hevc_filter.o hevc_parse.o hevc_data.o
-OBJS-$(CONFIG_HEVC_QSV_DECODER)        += qsvdec_h2645.o
-OBJS-$(CONFIG_HEVC_QSV_ENCODER)        += qsvenc_hevc.o hevc_ps_enc.o hevc_parse.o
+                                          hevcdsp.o hevc_filter.o
 OBJS-$(CONFIG_HNM4_VIDEO_DECODER)      += hnm4video.o
-OBJS-$(CONFIG_HQ_HQA_DECODER)          += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
-                                          canopus.o
-OBJS-$(CONFIG_HQX_DECODER)             += hqx.o hqxvlc.o hqxdsp.o canopus.o
+OBJS-$(CONFIG_HQX_DECODER)             += hqx.o hqxvlc.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IDCIN_DECODER)           += idcinvideo.o
@@ -315,8 +282,8 @@ OBJS-$(CONFIG_IFF_ILBM_DECODER)        += iff.o
 OBJS-$(CONFIG_IMC_DECODER)             += imc.o
 OBJS-$(CONFIG_INDEO2_DECODER)          += indeo2.o
 OBJS-$(CONFIG_INDEO3_DECODER)          += indeo3.o
-OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi.o
-OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi.o
+OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi_common.o ivi_dsp.o
+OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
 OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
@@ -330,7 +297,7 @@ OBJS-$(CONFIG_JV_DECODER)              += jvdec.o
 OBJS-$(CONFIG_KGV1_DECODER)            += kgv1dec.o
 OBJS-$(CONFIG_KMVC_DECODER)            += kmvc.o
 OBJS-$(CONFIG_LAGARITH_DECODER)        += lagarith.o lagarithrac.o
-OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpegenc_common.o
+OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpeg.o mjpegenc_common.o
 OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
 OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
@@ -339,8 +306,8 @@ OBJS-$(CONFIG_METASOUND_DECODER)       += metasound.o metasound_data.o \
                                           twinvq.o
 OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
-OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o
-OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpegenc_common.o
+OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
+OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o mjpegenc_common.o
 OBJS-$(CONFIG_MJPEGB_DECODER)          += mjpegbdec.o
 OBJS-$(CONFIG_MLP_DECODER)             += mlpdec.o mlpdsp.o
 OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
@@ -368,8 +335,6 @@ OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
 OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
-OBJS-$(CONFIG_MPEG2_QSV_DECODER)       += qsvdec_mpeg2.o
-OBJS-$(CONFIG_MPEG2_QSV_ENCODER)       += qsvenc_mpeg2.o
 OBJS-$(CONFIG_MPEG4_DECODER)           += xvididct.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
@@ -378,19 +343,20 @@ OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
-OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o
+OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
 OBJS-$(CONFIG_MSS2_DECODER)            += mss2.o mss12.o mss2dsp.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
-OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o
+OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o mss34dsp.o
 OBJS-$(CONFIG_MVC1_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MVC2_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
 OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
 OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
+OBJS-$(CONFIG_NVENC_ENCODER)           += nvenc.o
 OBJS-$(CONFIG_ON2AVC_DECODER)          += on2avc.o on2avcdata.o
 OBJS-$(CONFIG_OPUS_DECODER)            += opusdec.o opus.o opus_celt.o \
                                           opus_silk.o vorbis_data.o
@@ -448,8 +414,8 @@ OBJS-$(CONFIG_RV10_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV10_ENCODER)            += rv10enc.o
 OBJS-$(CONFIG_RV20_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV20_ENCODER)            += rv20enc.o
-OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o
-OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv40dsp.o
+OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o
+OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o
 OBJS-$(CONFIG_SAMI_DECODER)            += samidec.o ass.o
 OBJS-$(CONFIG_S302M_DECODER)           += s302m.o
 OBJS-$(CONFIG_S302M_ENCODER)           += s302menc.o
@@ -492,7 +458,6 @@ OBJS-$(CONFIG_TAK_DECODER)             += takdec.o tak.o
 OBJS-$(CONFIG_TARGA_DECODER)           += targa.o
 OBJS-$(CONFIG_TARGA_ENCODER)           += targaenc.o rle.o
 OBJS-$(CONFIG_TARGA_Y216_DECODER)      += targa_y216dec.o
-OBJS-$(CONFIG_TDSC_DECODER)            += tdsc.o
 OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
 OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
 OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o tiff_data.o
@@ -506,7 +471,7 @@ OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
 OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o ttadsp.o
 OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
 OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvqdec.o twinvq.o
-OBJS-$(CONFIG_TXD_DECODER)             += txd.o
+OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideodec.o utvideo.o
 OBJS-$(CONFIG_UTVIDEO_ENCODER)         += utvideoenc.o utvideo.o
@@ -526,7 +491,6 @@ OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1_block.o vc1_loopfilter.o
                                           vc1dsp.o \
                                           msmpeg4dec.o msmpeg4.o msmpeg4data.o \
                                           wmv2dsp.o
-OBJS-$(CONFIG_VC1_QSV_DECODER)         += qsvdec_vc1.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdaudio.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdvideo.o
@@ -536,13 +500,13 @@ OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbisdsp.o vorbis.o \
 OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                           vorbis_data.o
 OBJS-$(CONFIG_VP3_DECODER)             += vp3.o
-OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56rac.o
-OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o \
+OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
+                                          vp56rac.o
+OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
                                           vp6dsp.o vp56rac.o
-OBJS-$(CONFIG_VP7_DECODER)             += vp8.o vp56rac.o
-OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp56rac.o
-OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o vp9dsp_8bpp.o \
-                                          vp9dsp_10bpp.o vp9dsp_12bpp.o
+OBJS-$(CONFIG_VP7_DECODER)             += vp8.o vp8dsp.o vp56rac.o
+OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
+OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
@@ -562,9 +526,9 @@ OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                           acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV1_ENCODER)            += msmpeg4enc.o
-OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o \
+OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
                                           msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
+OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
                                           msmpeg4.o msmpeg4enc.o msmpeg4data.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
@@ -697,39 +661,26 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o
 
 # hardware accelerators
-OBJS-$(CONFIG_D3D11VA)                    += dxva2.o
 OBJS-$(CONFIG_DXVA2)                      += dxva2.o
 OBJS-$(CONFIG_VAAPI)                      += vaapi.o
-OBJS-$(CONFIG_VDA)                        += vda.o videotoolbox.o
-OBJS-$(CONFIG_VIDEOTOOLBOX)               += videotoolbox.o
+OBJS-$(CONFIG_VDA)                        += vda.o
 OBJS-$(CONFIG_VDPAU)                      += vdpau.o
 
 OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o
 OBJS-$(CONFIG_H263_VDPAU_HWACCEL)         += vdpau_mpeg4.o
-OBJS-$(CONFIG_H263_VIDEOTOOLBOX_HWACCEL)  += videotoolbox.o
-OBJS-$(CONFIG_H264_D3D11VA_HWACCEL)       += dxva2_h264.o
 OBJS-$(CONFIG_H264_DXVA2_HWACCEL)         += dxva2_h264.o
 OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
 OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
 OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
-OBJS-$(CONFIG_H264_VIDEOTOOLBOX_HWACCEL)  += videotoolbox.o
-OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL)       += dxva2_hevc.o
 OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL)         += dxva2_hevc.o
-OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL)         += vaapi_hevc.o
-OBJS-$(CONFIG_HEVC_VDPAU_HWACCEL)         += vdpau_hevc.o
 OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG1_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_MPEG1_XVMC_HWACCEL)         += mpegvideo_xvmc.o
-OBJS-$(CONFIG_MPEG2_D3D11VA_HWACCEL)      += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG2_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_MPEG2_XVMC_HWACCEL)         += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o
 OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
-OBJS-$(CONFIG_MPEG4_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
-OBJS-$(CONFIG_VC1_D3D11VA_HWACCEL)        += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o
 OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)          += vdpau_vc1.o
@@ -765,6 +716,7 @@ OBJS-$(CONFIG_OGG_DEMUXER)             += mpeg12data.o \
 OBJS-$(CONFIG_OGG_MUXER)               += flac.o flacdata.o \
                                           vorbis_data.o
 OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o
+OBJS-$(CONFIG_RTPDEC)                  += mjpeg.o
 OBJS-$(CONFIG_SPDIF_DEMUXER)           += aacadtsdec.o mpeg4audio.o
 OBJS-$(CONFIG_SPDIF_MUXER)             += dca.o
 OBJS-$(CONFIG_TAK_DEMUXER)             += tak.o
@@ -779,7 +731,6 @@ OBJS-$(CONFIG_ELBG_FILTER)             += elbg.o
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
-OBJS-$(CONFIG_LIBDCADEC_DECODER)          += libdcadec.o dca.o
 OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
 OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
@@ -789,7 +740,6 @@ OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsmdec.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsmenc.o
 OBJS-$(CONFIG_LIBILBC_DECODER)            += libilbc.o
 OBJS-$(CONFIG_LIBILBC_ENCODER)            += libilbc.o
-OBJS-$(CONFIG_LIBKVAZAAR_ENCODER)         += libkvazaar.o
 OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o
@@ -823,9 +773,7 @@ OBJS-$(CONFIG_LIBVPX_VP8_ENCODER)         += libvpxenc.o
 OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o libvpx.o
 OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o libvpx.o
 OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
-OBJS-$(CONFIG_LIBWEBP_ENCODER)            += libwebpenc_common.o libwebpenc.o
-OBJS-$(CONFIG_LIBWEBP_ANIM_ENCODER)       += libwebpenc_common.o libwebpenc_animencoder.o
-OBJS-$(CONFIG_LIBX262_ENCODER)            += libx264.o
+OBJS-$(CONFIG_LIBWEBP_ENCODER)            += libwebpenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX265_ENCODER)            += libx265.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
@@ -851,12 +799,11 @@ OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                           vorbis_data.o
-OBJS-$(CONFIG_G729_PARSER)             += g729_parser.o
 OBJS-$(CONFIG_GSM_PARSER)              += gsm_parser.o
 OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o
-OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o hevc_parse.o hevc_ps.o hevc_data.o
+OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
@@ -884,11 +831,9 @@ OBJS-$(CONFIG_AAC_ADTSTOASC_BSF)          += aac_adtstoasc_bsf.o aacadtsdec.o \
 OBJS-$(CONFIG_CHOMP_BSF)                  += chomp_bsf.o
 OBJS-$(CONFIG_DUMP_EXTRADATA_BSF)         += dump_extradata_bsf.o
 OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF)       += h264_mp4toannexb_bsf.o
-OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF)       += hevc_mp4toannexb_bsf.o
 OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF)        += imx_dump_header_bsf.o
-OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o
+OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o mjpeg.o
 OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF)     += mjpega_dump_header_bsf.o
-OBJS-$(CONFIG_MPEG4_UNPACK_BFRAMES_BSF)   += mpeg4_unpack_bframes_bsf.o
 OBJS-$(CONFIG_MOV2TEXTSUB_BSF)            += movsub_bsf.o
 OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF)  += mp3_header_decompress_bsf.o \
                                              mpegaudiodata.o
@@ -909,31 +854,26 @@ SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           aac_tablegen_decl.h           \
                                           fft-internal.h                \
+                                          libutvideo.h                  \
                                           old_codec_ids.h               \
                                           tableprint.h                  \
                                           tableprint_vlc.h              \
                                           $(ARCH)/vp56_arith.h          \
 
-SKIPHEADERS-$(CONFIG_D3D11VA)          += d3d11va.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
 SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
-SKIPHEADERS-$(CONFIG_LIBWEBP_ENCODER)  += libwebpenc_common.h
 SKIPHEADERS-$(CONFIG_QSV)              += qsv.h qsv_internal.h
-SKIPHEADERS-$(CONFIG_QSVDEC)           += qsvdec.h
-SKIPHEADERS-$(CONFIG_QSVENC)           += qsvenc.h
 SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
-SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_vt_internal.h
+SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h vdpau_internal.h
-SKIPHEADERS-$(CONFIG_VIDEOTOOLBOX)     += videotoolbox.h vda_vt_internal.h
 
 TESTPROGS = imgconvert                                                  \
-            jpeg2000dwt                                                 \
-            mathops                                                    \
             options                                                     \
             avfft                                                       \
 
+
 TESTPROGS-$(CONFIG_CABAC)                 += cabac
 TESTPROGS-$(CONFIG_FFT)                   += fft fft-fixed fft-fixed32
 TESTPROGS-$(CONFIG_IDCTDSP)               += dct
@@ -949,12 +889,9 @@ TOOLS = fourcc2pixfmt
 
 HOSTPROGS = aac_tablegen                                                \
             aacps_tablegen                                              \
-            aacps_fixed_tablegen                                        \
             aacsbr_tablegen                                             \
-            aacsbr_fixed_tablegen                                       \
             cabac_tablegen                                              \
             cbrt_tablegen                                               \
-            cbrt_fixed_tablegen                                         \
             cos_tablegen                                                \
             dsd_tablegen                                                \
             dv_tablegen                                                 \
@@ -963,7 +900,6 @@ HOSTPROGS = aac_tablegen                                                \
             pcm_tablegen                                                \
             qdm2_tablegen                                               \
             sinewin_tablegen                                            \
-            sinewin_fixed_tablegen                                      \
 
 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 
@@ -982,9 +918,8 @@ else
 $(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
 endif
 
-GEN_HEADERS = cabac_tables.h cbrt_tables.h cbrt_fixed_tables.h aacps_tables.h aacps_fixed_tables.h aacsbr_tables.h \
-              aacsbr_fixed_tables.h aac_tables.h dsd_tables.h dv_tables.h     \
-              sinewin_tables.h sinewin_fixed_tables.h mpegaudio_tables.h motionpixels_tables.h \
+GEN_HEADERS = cabac_tables.h cbrt_tables.h aacps_tables.h aacsbr_tables.h aac_tables.h dsd_tables.h dv_tables.h     \
+              sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
               pcm_tables.h qdm2_tables.h
 GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))
 
@@ -993,18 +928,13 @@ $(GEN_HEADERS): $(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)
 
 ifdef CONFIG_HARDCODED_TABLES
 $(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
-$(SUBDIR)aacdec_fixed.o: $(SUBDIR)cbrt_fixed_tables.h
-$(SUBDIR)aacps_float.o: $(SUBDIR)aacps_tables.h
-$(SUBDIR)aacps_fixed.o: $(SUBDIR)aacps_fixed_tables.h
+$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
 $(SUBDIR)aacsbr.o: $(SUBDIR)aacsbr_tables.h
-$(SUBDIR)aacsbr_fixed.o: $(SUBDIR)aacsbr_fixed_tables.h
 $(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
-$(SUBDIR)aactab_fixed.o: $(SUBDIR)aac_fixed_tables.h
 $(SUBDIR)cabac.o: $(SUBDIR)cabac_tables.h
 $(SUBDIR)dsddec.o: $(SUBDIR)dsd_tables.h
 $(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
 $(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
-$(SUBDIR)sinewin_fixed.o: $(SUBDIR)sinewin_fixed_tables.h
 $(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h

libavcodec/a64multienc.c

@@ -66,8 +66,7 @@ static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
 //static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
 //static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};
 
-static void to_meta_with_crop(AVCodecContext *avctx,
-                              const AVFrame *p, int *dest)
+static void to_meta_with_crop(AVCodecContext *avctx, const AVFrame *p, int *dest)
 {
     int blockx, blocky, x, y;
     int luma = 0;
@@ -235,7 +234,7 @@ static av_cold int a64multi_encode_init(AVCodecContext *avctx)
     }
 
     /* set up extradata */
-    if (!(avctx->extradata = av_mallocz(8 * 4 + AV_INPUT_BUFFER_PADDING_SIZE))) {
+    if (!(avctx->extradata = av_mallocz(8 * 4 + FF_INPUT_BUFFER_PADDING_SIZE))) {
         av_log(avctx, AV_LOG_ERROR, "Failed to allocate memory for extradata.\n");
         return AVERROR(ENOMEM);
     }
@@ -328,7 +327,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
         /* any frames to encode? */
         if (c->mc_lifetime) {
             int alloc_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
-            if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size, 0)) < 0)
+            if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size)) < 0)
                 return ret;
             buf = pkt->data;
 
@@ -406,7 +405,7 @@ AVCodec ff_a64multi_encoder = {
     .encode2        = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
-    .capabilities   = AV_CODEC_CAP_DELAY,
+    .capabilities   = CODEC_CAP_DELAY,
 };
 #endif
 #if CONFIG_A64MULTI5_ENCODER
@@ -420,6 +419,6 @@ AVCodec ff_a64multi5_encoder = {
     .encode2        = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
-    .capabilities   = AV_CODEC_CAP_DELAY,
+    .capabilities   = CODEC_CAP_DELAY,
 };
 #endif

libavcodec/aac.h

@@ -30,14 +30,9 @@
 #ifndef AVCODEC_AAC_H
 #define AVCODEC_AAC_H
 
-
-#include "aac_defines.h"
 #include "libavutil/float_dsp.h"
-#include "libavutil/fixed_dsp.h"
 #include "avcodec.h"
-#if !USE_FIXED
 #include "imdct15.h"
-#endif
 #include "fft.h"
 #include "mpeg4audio.h"
 #include "sbr.h"
@@ -50,8 +45,6 @@
 #define TNS_MAX_ORDER 20
 #define MAX_LTP_LONG_SFB 40
 
-#define CLIP_AVOIDANCE_FACTOR 0.95f
-
 enum RawDataBlockType {
     TYPE_SCE,
     TYPE_CPE,
@@ -83,10 +76,9 @@ enum BandType {
     ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
     FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
     ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
-    RESERVED_BT    = 12,    ///< Band types following are encoded differently from others.
     NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
-    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions (out of phase).
-    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions (in phase).
+    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
+    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
 };
 
 #define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
@@ -133,14 +125,12 @@ typedef struct OutputConfiguration {
  * Predictor State
  */
 typedef struct PredictorState {
-    AAC_FLOAT cor0;
-    AAC_FLOAT cor1;
-    AAC_FLOAT var0;
-    AAC_FLOAT var1;
-    AAC_FLOAT r0;
-    AAC_FLOAT r1;
-    AAC_FLOAT k1;
-    AAC_FLOAT x_est;
+    float cor0;
+    float cor1;
+    float var0;
+    float var1;
+    float r0;
+    float r1;
 } PredictorState;
 
 #define MAX_PREDICTORS 672
@@ -151,17 +141,13 @@ typedef struct PredictorState {
 #define SCALE_MAX_DIFF   60    ///< maximum scalefactor difference allowed by standard
 #define SCALE_DIFF_ZERO  60    ///< codebook index corresponding to zero scalefactor indices difference
 
-#define NOISE_PRE       256    ///< preamble for NOISE_BT, put in bitstream with the first noise band
-#define NOISE_PRE_BITS    9    ///< length of preamble
-#define NOISE_OFFSET     90    ///< subtracted from global gain, used as offset for the preamble
-
 /**
  * Long Term Prediction
  */
 typedef struct LongTermPrediction {
     int8_t present;
     int16_t lag;
-    INTFLOAT coef;
+    float coef;
     int8_t used[MAX_LTP_LONG_SFB];
 } LongTermPrediction;
 
@@ -183,10 +169,7 @@ typedef struct IndividualChannelStream {
     int predictor_present;
     int predictor_initialized;
     int predictor_reset_group;
-    int predictor_reset_count[31];  ///< used by encoder to count prediction resets
     uint8_t prediction_used[41];
-    uint8_t window_clipping[8]; ///< set if a certain window is near clipping
-    float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
 } IndividualChannelStream;
 
 /**
@@ -198,8 +181,7 @@ typedef struct TemporalNoiseShaping {
     int length[8][4];
     int direction[8][4];
     int order[8][4];
-    int coef_idx[8][4][TNS_MAX_ORDER];
-    INTFLOAT coef[8][4][TNS_MAX_ORDER];
+    float coef[8][4][TNS_MAX_ORDER];
 } TemporalNoiseShaping;
 
 /**
@@ -236,7 +218,7 @@ typedef struct ChannelCoupling {
     int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
                             *   [2] list of gains for left channel; [3] lists of gains for both channels
                             */
-    INTFLOAT gain[16][120];
+    float gain[16][120];
 } ChannelCoupling;
 
 /**
@@ -247,21 +229,17 @@ typedef struct SingleChannelElement {
     TemporalNoiseShaping tns;
     Pulse pulse;
     enum BandType band_type[128];                   ///< band types
-    enum BandType band_alt[128];                    ///< alternative band type (used by encoder)
     int band_type_run_end[120];                     ///< band type run end points
-    INTFLOAT sf[120];                               ///< scalefactors
+    float sf[120];                                  ///< scalefactors
     int sf_idx[128];                                ///< scalefactor indices (used by encoder)
     uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
-    float  is_ener[128];                            ///< Intensity stereo pos (used by encoder)
-    float pns_ener[128];                            ///< Noise energy values (used by encoder)
-    DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024];   ///< coefficients for IMDCT, pristine
-    DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024];    ///< coefficients for IMDCT, maybe processed
-    DECLARE_ALIGNED(32, INTFLOAT, saved)[1536];     ///< overlap
-    DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048];   ///< PCM output buffer
-    DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
-    DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
+    DECLARE_ALIGNED(32, float,   pcoeffs)[1024];    ///< coefficients for IMDCT, pristine
+    DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT, maybe processed
+    DECLARE_ALIGNED(32, float,   saved)[1536];      ///< overlap
+    DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
+    DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
     PredictorState predictor_state[MAX_PREDICTORS];
-    INTFLOAT *ret;                                  ///< PCM output
+    float *ret;                                     ///< PCM output
 } SingleChannelElement;
 
 /**
@@ -272,9 +250,7 @@ typedef struct ChannelElement {
     // CPE specific
     int common_window;        ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
     int     ms_mode;          ///< Signals mid/side stereo flags coding mode (used by encoder)
-    uint8_t is_mode;          ///< Set if any bands have been encoded using intensity stereo (used by encoder)
     uint8_t ms_mask[128];     ///< Set if mid/side stereo is used for each scalefactor window band
-    uint8_t is_mask[128];     ///< Set if intensity stereo is used (used by encoder)
     // shared
     SingleChannelElement ch[2];
     // CCE specific
@@ -308,7 +284,7 @@ struct AACContext {
      * (We do not want to have these on the stack.)
      * @{
      */
-    DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
+    DECLARE_ALIGNED(32, float, buf_mdct)[1024];
     /** @} */
 
     /**
@@ -319,12 +295,8 @@ struct AACContext {
     FFTContext mdct_small;
     FFTContext mdct_ld;
     FFTContext mdct_ltp;
-#if USE_FIXED
-    AVFixedDSPContext *fdsp;
-#else
     IMDCT15Context *mdct480;
     AVFloatDSPContext *fdsp;
-#endif /* USE_FIXED */
     int random_state;
     /** @} */
 
@@ -344,7 +316,7 @@ struct AACContext {
     int dmono_mode;      ///< 0->not dmono, 1->use first channel, 2->use second channel
     /** @} */
 
-    DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
+    DECLARE_ALIGNED(32, float, temp)[128];
 
     OutputConfiguration oc[2];
     int warned_num_aac_frames;
@@ -352,13 +324,11 @@ struct AACContext {
     /* aacdec functions pointers */
     void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
     void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
-    void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
+    void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
                       IndividualChannelStream *ics, int decode);
-    void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
-                                   INTFLOAT *in, IndividualChannelStream *ics);
+    void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
+                                   float *in, IndividualChannelStream *ics);
     void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
-    void (*vector_pow43)(int *coefs, int len);
-    void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);
 
 };
 

libavcodec/aac_adtstoasc_bsf.c

@@ -89,7 +89,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
         }
         av_free(avctx->extradata);
         avctx->extradata_size = 2 + pce_size;
-        avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
+        avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
         if (!avctx->extradata) {
             avctx->extradata_size = 0;
             return AVERROR(ENOMEM);

libavcodec/aac_defines.h

@@ -1,114 +0,0 @@
-/*
- * AAC defines
- *
- * 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 AVCODEC_AAC_DEFINES_H
-#define AVCODEC_AAC_DEFINES_H
-
-#ifndef USE_FIXED
-#define USE_FIXED 0
-#endif
-
-#if USE_FIXED
-
-#include "libavutil/softfloat.h"
-
-#define FFT_FLOAT    0
-#define FFT_FIXED_32 1
-
-#define AAC_RENAME(x)       x ## _fixed
-#define AAC_RENAME_32(x)    x ## _fixed_32
-#define INTFLOAT int
-#define INT64FLOAT          int64_t
-#define SHORTFLOAT int16_t
-#define AAC_FLOAT SoftFloat
-#define AAC_SIGNE           int
-#define FIXR(a)             ((int)((a) * 1 + 0.5))
-#define FIXR10(a)           ((int)((a) * 1024.0 + 0.5))
-#define Q23(a)              (int)((a) * 8388608.0 + 0.5)
-#define Q30(x)              (int)((x)*1073741824.0 + 0.5)
-#define Q31(x)              (int)((x)*2147483648.0 + 0.5)
-#define RANGE15(x)          x
-#define GET_GAIN(x, y)      (-(y) << (x)) + 1024
-#define AAC_MUL16(x, y)     (int)(((int64_t)(x) * (y) + 0x8000) >> 16)
-#define AAC_MUL26(x, y)     (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
-#define AAC_MUL30(x, y)     (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
-#define AAC_MUL31(x, y)     (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
-#define AAC_MADD28(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
-                                      ((int64_t)(a) * (b)) + \
-                                        0x8000000) >> 28)
-#define AAC_MADD30(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
-                                      ((int64_t)(a) * (b)) + \
-                                        0x20000000) >> 30)
-#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
-                                                     ((int64_t)(a) * (b)) + \
-                                                     ((int64_t)(c) * (d)) + \
-                                                     ((int64_t)(e) * (f)) + \
-                                                       0x20000000) >> 30)
-#define AAC_MSUB30(x, y, a, b) (int)((((int64_t)(x) * (y)) - \
-                                      ((int64_t)(a) * (b)) + \
-                                        0x20000000) >> 30)
-#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
-                                                     ((int64_t)(a) * (b)) - \
-                                                     ((int64_t)(c) * (d)) - \
-                                                     ((int64_t)(e) * (f)) + \
-                                                       0x20000000) >> 30)
-#define AAC_MSUB31_V3(x, y, z)    (int)((((int64_t)(x) * (z)) - \
-                                      ((int64_t)(y) * (z)) + \
-                                        0x40000000) >> 31)
-#define AAC_HALF_SUM(x, y)  (x) >> 1 + (y) >> 1
-#define AAC_SRA_R(x, y)     (int)(((x) + (1 << ((y) - 1))) >> (y))
-
-#else
-
-#define FFT_FLOAT    1
-#define FFT_FIXED_32 0
-
-#define AAC_RENAME(x)       x
-#define AAC_RENAME_32(x)    x
-#define INTFLOAT float
-#define INT64FLOAT          float
-#define SHORTFLOAT float
-#define AAC_FLOAT float
-#define AAC_SIGNE           unsigned
-#define FIXR(x)             ((float)(x))
-#define FIXR10(x)           ((float)(x))
-#define Q23(x)              x
-#define Q30(x)              x
-#define Q31(x)              x
-#define RANGE15(x)          (32768.0 * (x))
-#define GET_GAIN(x, y)      powf((x), -(y))
-#define AAC_MUL16(x, y)     ((x) * (y))
-#define AAC_MUL26(x, y)     ((x) * (y))
-#define AAC_MUL30(x, y)     ((x) * (y))
-#define AAC_MUL31(x, y)     ((x) * (y))
-#define AAC_MADD28(x, y, a, b) ((x) * (y) + (a) * (b))
-#define AAC_MADD30(x, y, a, b) ((x) * (y) + (a) * (b))
-#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) + \
-                                               (c) * (d) + (e) * (f))
-#define AAC_MSUB30(x, y, a, b) ((x) * (y) - (a) * (b))
-#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) - \
-                                               (c) * (d) - (e) * (f))
-#define AAC_MSUB31_V3(x, y, z)    ((x) - (y)) * (z)
-#define AAC_HALF_SUM(x, y)  ((x) + (y)) * 0.5f
-#define AAC_SRA_R(x, y)     (x)
-
-#endif /* USE_FIXED */
-
-#endif /* AVCODEC_AAC_DEFINES_H */

libavcodec/aac_parser.c

@@ -34,7 +34,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
     int size;
     union {
         uint64_t u64;
-        uint8_t  u8[8 + AV_INPUT_BUFFER_PADDING_SIZE];
+        uint8_t  u8[8 + FF_INPUT_BUFFER_PADDING_SIZE];
     } tmp;
 
     tmp.u64 = av_be2ne64(state);

libavcodec/aac_tablegen.c

@@ -33,7 +33,5 @@ int main(void)
 
     WRITE_ARRAY("const", float, ff_aac_pow2sf_tab);
 
-    WRITE_ARRAY("const", float, ff_aac_pow34sf_tab);
-
     return 0;
 }

libavcodec/aac_tablegen.h

@@ -30,15 +30,12 @@
 #else
 #include "libavutil/mathematics.h"
 float ff_aac_pow2sf_tab[428];
-float ff_aac_pow34sf_tab[428];
 
 av_cold void ff_aac_tableinit(void)
 {
     int i;
-    for (i = 0; i < 428; i++) {
+    for (i = 0; i < 428; i++)
         ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
-        ff_aac_pow34sf_tab[i] = pow(ff_aac_pow2sf_tab[i], 3.0/4.0);
-    }
 }
 #endif /* CONFIG_HARDCODED_TABLES */
 

libavcodec/aac_tablegen_decl.h

@@ -28,11 +28,9 @@
 #if CONFIG_HARDCODED_TABLES
 #define ff_aac_tableinit()
 extern const float ff_aac_pow2sf_tab[428];
-extern const float ff_aac_pow34sf_tab[428];
 #else
 void ff_aac_tableinit(void);
 extern       float ff_aac_pow2sf_tab[428];
-extern       float ff_aac_pow34sf_tab[428];
 #endif /* CONFIG_HARDCODED_TABLES */
 
 #endif /* AVCODEC_AAC_TABLEGEN_DECL_H */

libavcodec/aaccoder.c

@@ -39,26 +39,263 @@
 #include "aac.h"
 #include "aacenc.h"
 #include "aactab.h"
-#include "aacenctab.h"
-#include "aacenc_utils.h"
-#include "aacenc_quantization.h"
-#include "aac_tablegen_decl.h"
 
-#include "aacenc_is.h"
-#include "aacenc_tns.h"
-#include "aacenc_pred.h"
+/** bits needed to code codebook run value for long windows */
+static const uint8_t run_value_bits_long[64] = {
+     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
+     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
+    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
+    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
+};
 
-/** Frequency in Hz for lower limit of noise substitution **/
-#define NOISE_LOW_LIMIT 4500
+/** bits needed to code codebook run value for short windows */
+static const uint8_t run_value_bits_short[16] = {
+    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
+};
 
-/* Energy spread threshold value below which no PNS is used, this corresponds to
- * typically around 17Khz, after which PNS usage decays ending at 19Khz */
-#define NOISE_SPREAD_THRESHOLD 0.5f
+static const uint8_t * const run_value_bits[2] = {
+    run_value_bits_long, run_value_bits_short
+};
 
-/* This constant gets divided by lambda to return ~1.65 which when multiplied
- * by the band->threshold and compared to band->energy is the boundary between
- * excessive PNS and little PNS usage. */
-#define NOISE_LAMBDA_NUMERATOR 252.1f
+
+/**
+ * Quantize one coefficient.
+ * @return absolute value of the quantized coefficient
+ * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
+ */
+static av_always_inline int quant(float coef, const float Q)
+{
+    float a = coef * Q;
+    return sqrtf(a * sqrtf(a)) + 0.4054;
+}
+
+static void quantize_bands(int *out, const float *in, const float *scaled,
+                           int size, float Q34, int is_signed, int maxval)
+{
+    int i;
+    double qc;
+    for (i = 0; i < size; i++) {
+        qc = scaled[i] * Q34;
+        out[i] = (int)FFMIN(qc + 0.4054, (double)maxval);
+        if (is_signed && in[i] < 0.0f) {
+            out[i] = -out[i];
+        }
+    }
+}
+
+static void abs_pow34_v(float *out, const float *in, const int size)
+{
+#ifndef USE_REALLY_FULL_SEARCH
+    int i;
+    for (i = 0; i < size; i++) {
+        float a = fabsf(in[i]);
+        out[i] = sqrtf(a * sqrtf(a));
+    }
+#endif /* USE_REALLY_FULL_SEARCH */
+}
+
+static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
+static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
+
+/**
+ * Calculate rate distortion cost for quantizing with given codebook
+ *
+ * @return quantization distortion
+ */
+static av_always_inline float quantize_and_encode_band_cost_template(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, int BT_ZERO, int BT_UNSIGNED,
+                                int BT_PAIR, int BT_ESC)
+{
+    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
+    const float Q   = ff_aac_pow2sf_tab [q_idx];
+    const float Q34 = ff_aac_pow34sf_tab[q_idx];
+    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float CLIPPED_ESCAPE = 165140.0f*IQ;
+    int i, j;
+    float cost = 0;
+    const int dim = BT_PAIR ? 2 : 4;
+    int resbits = 0;
+    const int range  = aac_cb_range[cb];
+    const int maxval = aac_cb_maxval[cb];
+    int off;
+
+    if (BT_ZERO) {
+        for (i = 0; i < size; i++)
+            cost += in[i]*in[i];
+        if (bits)
+            *bits = 0;
+        return cost * lambda;
+    }
+    if (!scaled) {
+        abs_pow34_v(s->scoefs, in, size);
+        scaled = s->scoefs;
+    }
+    quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, maxval);
+    if (BT_UNSIGNED) {
+        off = 0;
+    } else {
+        off = maxval;
+    }
+    for (i = 0; i < size; i += dim) {
+        const float *vec;
+        int *quants = s->qcoefs + i;
+        int curidx = 0;
+        int curbits;
+        float rd = 0.0f;
+        for (j = 0; j < dim; j++) {
+            curidx *= range;
+            curidx += quants[j] + off;
+        }
+        curbits =  ff_aac_spectral_bits[cb-1][curidx];
+        vec     = &ff_aac_codebook_vectors[cb-1][curidx*dim];
+        if (BT_UNSIGNED) {
+            for (j = 0; j < dim; j++) {
+                float t = fabsf(in[i+j]);
+                float di;
+                if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow
+                    if (t >= CLIPPED_ESCAPE) {
+                        di = t - CLIPPED_ESCAPE;
+                        curbits += 21;
+                    } else {
+                        int c = av_clip_uintp2(quant(t, Q), 13);
+                        di = t - c*cbrtf(c)*IQ;
+                        curbits += av_log2(c)*2 - 4 + 1;
+                    }
+                } else {
+                    di = t - vec[j]*IQ;
+                }
+                if (vec[j] != 0.0f)
+                    curbits++;
+                rd += di*di;
+            }
+        } else {
+            for (j = 0; j < dim; j++) {
+                float di = in[i+j] - vec[j]*IQ;
+                rd += di*di;
+            }
+        }
+        cost    += rd * lambda + curbits;
+        resbits += curbits;
+        if (cost >= uplim)
+            return uplim;
+        if (pb) {
+            put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]);
+            if (BT_UNSIGNED)
+                for (j = 0; j < dim; j++)
+                    if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f)
+                        put_bits(pb, 1, in[i+j] < 0.0f);
+            if (BT_ESC) {
+                for (j = 0; j < 2; j++) {
+                    if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) {
+                        int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q), 13);
+                        int len = av_log2(coef);
+
+                        put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
+                        put_bits(pb, len, coef & ((1 << len) - 1));
+                    }
+                }
+            }
+        }
+    }
+
+    if (bits)
+        *bits = resbits;
+    return cost;
+}
+
+#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC) \
+static float quantize_and_encode_band_cost_ ## NAME(                                        \
+                                struct AACEncContext *s,                                \
+                                PutBitContext *pb, const float *in,                     \
+                                const float *scaled, int size, int scale_idx,           \
+                                int cb, const float lambda, const float uplim,          \
+                                int *bits) {                                            \
+    return quantize_and_encode_band_cost_template(                                      \
+                                s, pb, in, scaled, size, scale_idx,                     \
+                                BT_ESC ? ESC_BT : cb, lambda, uplim, bits,              \
+                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC);                 \
+}
+
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1)
+
+static float (*const quantize_and_encode_band_cost_arr[])(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits) = {
+    quantize_and_encode_band_cost_ZERO,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_ESC,
+};
+
+#define quantize_and_encode_band_cost(                                  \
+                                s, pb, in, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits)                    \
+    quantize_and_encode_band_cost_arr[cb](                              \
+                                s, pb, in, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits)
+
+static float quantize_band_cost(struct AACEncContext *s, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits)
+{
+    return quantize_and_encode_band_cost(s, NULL, in, scaled, size, scale_idx,
+                                         cb, lambda, uplim, bits);
+}
+
+static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
+                                     const float *in, int size, int scale_idx,
+                                     int cb, const float lambda)
+{
+    quantize_and_encode_band_cost(s, pb, in, NULL, size, scale_idx, cb, lambda,
+                                  INFINITY, NULL);
+}
+
+static float find_max_val(int group_len, int swb_size, const float *scaled) {
+    float maxval = 0.0f;
+    int w2, i;
+    for (w2 = 0; w2 < group_len; w2++) {
+        for (i = 0; i < swb_size; i++) {
+            maxval = FFMAX(maxval, scaled[w2*128+i]);
+        }
+    }
+    return maxval;
+}
+
+static int find_min_book(float maxval, int sf) {
+    float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
+    float Q34 = sqrtf(Q * sqrtf(Q));
+    int qmaxval, cb;
+    qmaxval = maxval * Q34 + 0.4054f;
+    if      (qmaxval ==  0) cb = 0;
+    else if (qmaxval ==  1) cb = 1;
+    else if (qmaxval ==  2) cb = 3;
+    else if (qmaxval <=  4) cb = 5;
+    else if (qmaxval <=  7) cb = 7;
+    else if (qmaxval <= 12) cb = 9;
+    else                    cb = 11;
+    return cb;
+}
 
 /**
  * structure used in optimal codebook search
@@ -75,7 +312,7 @@ typedef struct BandCodingPath {
 static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
                                      int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][CB_TOT_ALL];
+    BandCodingPath path[120][12];
     int w, swb, cb, start, size;
     int i, j;
     const int max_sfb  = sce->ics.max_sfb;
@@ -88,7 +325,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
 
     abs_pow34_v(s->scoefs, sce->coeffs, 1024);
     start = win*128;
-    for (cb = 0; cb < CB_TOT_ALL; cb++) {
+    for (cb = 0; cb < 12; cb++) {
         path[0][cb].cost     = 0.0f;
         path[0][cb].prev_idx = -1;
         path[0][cb].run      = 0;
@@ -96,7 +333,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
     for (swb = 0; swb < max_sfb; swb++) {
         size = sce->ics.swb_sizes[swb];
         if (sce->zeroes[win*16 + swb]) {
-            for (cb = 0; cb < CB_TOT_ALL; cb++) {
+            for (cb = 0; cb < 12; cb++) {
                 path[swb+1][cb].prev_idx = cb;
                 path[swb+1][cb].cost     = path[swb][cb].cost;
                 path[swb+1][cb].run      = path[swb][cb].run + 1;
@@ -106,22 +343,15 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
             int mincb = next_mincb;
             next_minrd = INFINITY;
             next_mincb = 0;
-            for (cb = 0; cb < CB_TOT_ALL; cb++) {
+            for (cb = 0; cb < 12; cb++) {
                 float cost_stay_here, cost_get_here;
                 float rd = 0.0f;
-                if (cb >= 12 && sce->band_type[win*16+swb] < aac_cb_out_map[cb] ||
-                    cb  < aac_cb_in_map[sce->band_type[win*16+swb]] && sce->band_type[win*16+swb] > aac_cb_out_map[cb]) {
-                    path[swb+1][cb].prev_idx = -1;
-                    path[swb+1][cb].cost     = INFINITY;
-                    path[swb+1][cb].run      = path[swb][cb].run + 1;
-                    continue;
-                }
                 for (w = 0; w < group_len; w++) {
                     FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
-                    rd += quantize_band_cost(s, &sce->coeffs[start + w*128],
-                                             &s->scoefs[start + w*128], size,
-                                             sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
-                                             lambda / band->threshold, INFINITY, NULL, 0);
+                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                             s->scoefs + start + w*128, size,
+                                             sce->sf_idx[(win+w)*16+swb], cb,
+                                             lambda / band->threshold, INFINITY, NULL);
                 }
                 cost_stay_here = path[swb][cb].cost + rd;
                 cost_get_here  = minrd              + rd + run_bits + 4;
@@ -149,12 +379,11 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
     //convert resulting path from backward-linked list
     stack_len = 0;
     idx       = 0;
-    for (cb = 1; cb < CB_TOT_ALL; cb++)
+    for (cb = 1; cb < 12; cb++)
         if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
             idx = cb;
     ppos = max_sfb;
     while (ppos > 0) {
-        av_assert1(idx >= 0);
         cb = idx;
         stackrun[stack_len] = path[ppos][cb].run;
         stackcb [stack_len] = cb;
@@ -165,13 +394,12 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
     //perform actual band info encoding
     start = 0;
     for (i = stack_len - 1; i >= 0; i--) {
-        cb = aac_cb_out_map[stackcb[i]];
-        put_bits(&s->pb, 4, cb);
+        put_bits(&s->pb, 4, stackcb[i]);
         count = stackrun[i];
-        memset(sce->zeroes + win*16 + start, !cb, count);
+        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
         //XXX: memset when band_type is also uint8_t
         for (j = 0; j < count; j++) {
-            sce->band_type[win*16 + start] = cb;
+            sce->band_type[win*16 + start] =  stackcb[i];
             start++;
         }
         while (count >= run_esc) {
@@ -185,7 +413,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
 static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                                   int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][CB_TOT_ALL];
+    BandCodingPath path[120][12];
     int w, swb, cb, start, size;
     int i, j;
     const int max_sfb  = sce->ics.max_sfb;
@@ -198,7 +426,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
 
     abs_pow34_v(s->scoefs, sce->coeffs, 1024);
     start = win*128;
-    for (cb = 0; cb < CB_TOT_ALL; cb++) {
+    for (cb = 0; cb < 12; cb++) {
         path[0][cb].cost     = run_bits+4;
         path[0][cb].prev_idx = -1;
         path[0][cb].run      = 0;
@@ -222,7 +450,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             }
             next_minbits = path[swb+1][0].cost;
             next_mincb = 0;
-            for (cb = 1; cb < CB_TOT_ALL; cb++) {
+            for (cb = 1; cb < 12; cb++) {
                 path[swb+1][cb].cost = 61450;
                 path[swb+1][cb].prev_idx = -1;
                 path[swb+1][cb].run = 0;
@@ -231,7 +459,6 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             float minbits = next_minbits;
             int mincb = next_mincb;
             int startcb = sce->band_type[win*16+swb];
-            startcb = aac_cb_in_map[startcb];
             next_minbits = INFINITY;
             next_mincb = 0;
             for (cb = 0; cb < startcb; cb++) {
@@ -239,21 +466,14 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                 path[swb+1][cb].prev_idx = -1;
                 path[swb+1][cb].run = 0;
             }
-            for (cb = startcb; cb < CB_TOT_ALL; cb++) {
+            for (cb = startcb; cb < 12; cb++) {
                 float cost_stay_here, cost_get_here;
                 float bits = 0.0f;
-                if (cb >= 12 && sce->band_type[win*16+swb] != aac_cb_out_map[cb]) {
-                    path[swb+1][cb].cost = 61450;
-                    path[swb+1][cb].prev_idx = -1;
-                    path[swb+1][cb].run = 0;
-                    continue;
-                }
                 for (w = 0; w < group_len; w++) {
-                    bits += quantize_band_cost(s, &sce->coeffs[start + w*128],
-                                               &s->scoefs[start + w*128], size,
-                                               sce->sf_idx[win*16+swb],
-                                               aac_cb_out_map[cb],
-                                               0, INFINITY, NULL, 0);
+                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                               s->scoefs + start + w*128, size,
+                                               sce->sf_idx[(win+w)*16+swb], cb,
+                                               0, INFINITY, NULL);
                 }
                 cost_stay_here = path[swb][cb].cost + bits;
                 cost_get_here  = minbits            + bits + run_bits + 4;
@@ -281,7 +501,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     //convert resulting path from backward-linked list
     stack_len = 0;
     idx       = 0;
-    for (cb = 1; cb < CB_TOT_ALL; cb++)
+    for (cb = 1; cb < 12; cb++)
         if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
             idx = cb;
     ppos = max_sfb;
@@ -297,13 +517,12 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     //perform actual band info encoding
     start = 0;
     for (i = stack_len - 1; i >= 0; i--) {
-        cb = aac_cb_out_map[stackcb[i]];
-        put_bits(&s->pb, 4, cb);
+        put_bits(&s->pb, 4, stackcb[i]);
         count = stackrun[i];
-        memset(sce->zeroes + win*16 + start, !cb, count);
+        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
         //XXX: memset when band_type is also uint8_t
         for (j = 0; j < count; j++) {
-            sce->band_type[win*16 + start] = cb;
+            sce->band_type[win*16 + start] =  stackcb[i];
             start++;
         }
         while (count >= run_esc) {
@@ -314,6 +533,16 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     }
 }
 
+/** Return the minimum scalefactor where the quantized coef does not clip. */
+static av_always_inline uint8_t coef2minsf(float coef) {
+    return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
+/** Return the maximum scalefactor where the quantized coef is not zero. */
+static av_always_inline uint8_t coef2maxsf(float coef) {
+    return av_clip_uint8(log2f(coef)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
 typedef struct TrellisPath {
     float cost;
     int prev;
@@ -322,43 +551,6 @@ typedef struct TrellisPath {
 #define TRELLIS_STAGES 121
 #define TRELLIS_STATES (SCALE_MAX_DIFF+1)
 
-static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce)
-{
-    int w, g, start = 0;
-    int minscaler_n = sce->sf_idx[0], minscaler_i = sce->sf_idx[0];
-    int bands = 0;
-
-    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        start = 0;
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
-                sce->sf_idx[w*16+g] = av_clip(ceilf(log2f(sce->is_ener[w*16+g])*2), -155, 100);
-                minscaler_i = FFMIN(minscaler_i, sce->sf_idx[w*16+g]);
-                bands++;
-            } else if (sce->band_type[w*16+g] == NOISE_BT) {
-                sce->sf_idx[w*16+g] = av_clip(4+log2f(sce->pns_ener[w*16+g])*2, -100, 155);
-                minscaler_n = FFMIN(minscaler_n, sce->sf_idx[w*16+g]);
-                bands++;
-            }
-            start += sce->ics.swb_sizes[g];
-        }
-    }
-
-    if (!bands)
-        return;
-
-    /* Clip the scalefactor indices */
-    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
-                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_i, minscaler_i + SCALE_MAX_DIFF);
-            } else if (sce->band_type[w*16+g] == NOISE_BT) {
-                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
-            }
-        }
-    }
-}
-
 static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
                                        SingleChannelElement *sce,
                                        const float lambda)
@@ -424,7 +616,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         start = w*128;
         for (g = 0; g < sce->ics.num_swb; g++) {
-            const float *coefs = &sce->coeffs[start];
+            const float *coefs = sce->coeffs + start;
             float qmin, qmax;
             int nz = 0;
 
@@ -463,7 +655,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
                     for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                         FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
                         dist += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
-                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL, 0);
+                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL);
                     }
                     minrd = FFMIN(minrd, dist);
 
@@ -499,7 +691,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
     }
     while (idx) {
         sce->sf_idx[bandaddr[idx]] = minq + q0;
-        minq = paths[idx][minq].prev;
+        minq = FFMAX(paths[idx][minq].prev, 0);
         idx--;
     }
     //set the same quantizers inside window groups
@@ -519,7 +711,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 {
     int start = 0, i, w, w2, g;
     int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
-    float dists[128] = { 0 }, uplims[128] = { 0 };
+    float dists[128] = { 0 }, uplims[128];
     float maxvals[128];
     int fflag, minscaler;
     int its  = 0;
@@ -534,11 +726,10 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         for (g = 0;  g < sce->ics.num_swb; g++) {
             int nz = 0;
-            float uplim = 0.0f, energy = 0.0f;
+            float uplim = 0.0f;
             for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                 FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
-                uplim  += band->threshold;
-                energy += band->energy;
+                uplim += band->threshold;
                 if (band->energy <= band->threshold || band->threshold == 0.0f) {
                     sce->zeroes[(w+w2)*16+g] = 1;
                     continue;
@@ -588,8 +779,8 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
             for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
                 start = w*128;
                 for (g = 0;  g < sce->ics.num_swb; g++) {
-                    const float *coefs = &sce->coeffs[start];
-                    const float *scaled = &s->scoefs[start];
+                    const float *coefs = sce->coeffs + start;
+                    const float *scaled = s->scoefs + start;
                     int bits = 0;
                     int cb;
                     float dist = 0.0f;
@@ -609,8 +800,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                                                    cb,
                                                    1.0f,
                                                    INFINITY,
-                                                   &b,
-                                                   0);
+                                                   &b);
                         bits += b;
                     }
                     dists[w*16+g] = dist - bits;
@@ -638,7 +828,6 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 
         fflag = 0;
         minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
-
         for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
             for (g = 0; g < sce->ics.num_swb; g++) {
                 int prevsc = sce->sf_idx[w*16+g];
@@ -684,7 +873,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
         }
     } else {
         for (w = 0; w < 8; w++) {
-            const float *coeffs = &sce->coeffs[w*128];
+            const float *coeffs = sce->coeffs + w*128;
             curband = start = 0;
             for (i = 0; i < 128; i++) {
                 if (i - start >= sce->ics.swb_sizes[curband]) {
@@ -709,7 +898,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         start = w*128;
         for (g = 0; g < sce->ics.num_swb; g++) {
-            float *coefs   = &sce->coeffs[start];
+            float *coefs   = sce->coeffs + start;
             const int size = sce->ics.swb_sizes[g];
             int start2 = start, end2 = start + size, peakpos = start;
             float maxval = -1, thr = 0.0f, t;
@@ -750,8 +939,8 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         start = w*128;
         for (g = 0;  g < sce->ics.num_swb; g++) {
-            const float *coefs  = &sce->coeffs[start];
-            const float *scaled = &s->scoefs[start];
+            const float *coefs  = sce->coeffs + start;
+            const float *scaled = s->scoefs   + start;
             const int size      = sce->ics.swb_sizes[g];
             int scf, prev_scf, step;
             int min_scf = -1, max_scf = 256;
@@ -776,12 +965,11 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
                                                ESC_BT,
                                                lambda,
                                                INFINITY,
-                                               &b,
-                                               0);
+                                               &b);
                     dist -= b;
                 }
                 dist *= 1.0f / 512.0f / lambda;
-                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512], ROUND_STANDARD);
+                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
                 if (quant_max >= 8191) { // too much, return to the previous quantizer
                     sce->sf_idx[w*16+g] = prev_scf;
                     break;
@@ -861,50 +1049,17 @@ static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
                 sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
 }
 
-static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
-{
-    int start = 0, w, w2, g;
-    const float lambda = s->lambda;
-    const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
-    const float spread_threshold = NOISE_SPREAD_THRESHOLD*(lambda/120.f);
-    const float thr_mult = NOISE_LAMBDA_NUMERATOR/lambda;
-
-    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        start = 0;
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (start*freq_mult > NOISE_LOW_LIMIT*(lambda/170.0f)) {
-                float energy = 0.0f, threshold = 0.0f, spread = 0.0f;
-                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
-                    FFPsyBand *band = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-                    energy += band->energy;
-                    threshold += band->threshold;
-                    spread += band->spread;
-                }
-                if (spread > spread_threshold*sce->ics.group_len[w] &&
-                    ((sce->zeroes[w*16+g] && energy >= threshold) ||
-                    energy < threshold*thr_mult*sce->ics.group_len[w])) {
-                    sce->band_type[w*16+g] = NOISE_BT;
-                    sce->pns_ener[w*16+g] = energy / sce->ics.group_len[w];
-                    sce->zeroes[w*16+g] = 0;
-                }
-            }
-            start += sce->ics.swb_sizes[g];
-        }
-    }
-}
-
-static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
+static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
+                          const float lambda)
 {
     int start = 0, i, w, w2, g;
     float M[128], S[128];
     float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
-    const float lambda = s->lambda;
     SingleChannelElement *sce0 = &cpe->ch[0];
     SingleChannelElement *sce1 = &cpe->ch[1];
     if (!cpe->common_window)
         return;
     for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
-        start = 0;
         for (g = 0;  g < sce0->ics.num_swb; g++) {
             if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
                 float dist1 = 0.0f, dist2 = 0.0f;
@@ -914,39 +1069,39 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
                     float minthr = FFMIN(band0->threshold, band1->threshold);
                     float maxthr = FFMAX(band0->threshold, band1->threshold);
                     for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                        M[i] = (sce0->coeffs[start+(w+w2)*128+i]
-                              + sce1->coeffs[start+(w+w2)*128+i]) * 0.5;
+                        M[i] = (sce0->pcoeffs[start+w2*128+i]
+                              + sce1->pcoeffs[start+w2*128+i]) * 0.5;
                         S[i] =  M[i]
-                              - sce1->coeffs[start+(w+w2)*128+i];
+                              - sce1->pcoeffs[start+w2*128+i];
                     }
-                    abs_pow34_v(L34, sce0->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
-                    abs_pow34_v(R34, sce1->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
+                    abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
+                    abs_pow34_v(R34, sce1->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
                     abs_pow34_v(M34, M,                         sce0->ics.swb_sizes[g]);
                     abs_pow34_v(S34, S,                         sce0->ics.swb_sizes[g]);
-                    dist1 += quantize_band_cost(s, &sce0->coeffs[start + (w+w2)*128],
+                    dist1 += quantize_band_cost(s, sce0->coeffs + start + w2*128,
                                                 L34,
                                                 sce0->ics.swb_sizes[g],
                                                 sce0->sf_idx[(w+w2)*16+g],
                                                 sce0->band_type[(w+w2)*16+g],
-                                                lambda / band0->threshold, INFINITY, NULL, 0);
-                    dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
+                                                lambda / band0->threshold, INFINITY, NULL);
+                    dist1 += quantize_band_cost(s, sce1->coeffs + start + w2*128,
                                                 R34,
                                                 sce1->ics.swb_sizes[g],
                                                 sce1->sf_idx[(w+w2)*16+g],
                                                 sce1->band_type[(w+w2)*16+g],
-                                                lambda / band1->threshold, INFINITY, NULL, 0);
+                                                lambda / band1->threshold, INFINITY, NULL);
                     dist2 += quantize_band_cost(s, M,
                                                 M34,
                                                 sce0->ics.swb_sizes[g],
                                                 sce0->sf_idx[(w+w2)*16+g],
                                                 sce0->band_type[(w+w2)*16+g],
-                                                lambda / maxthr, INFINITY, NULL, 0);
+                                                lambda / maxthr, INFINITY, NULL);
                     dist2 += quantize_band_cost(s, S,
                                                 S34,
                                                 sce1->ics.swb_sizes[g],
                                                 sce1->sf_idx[(w+w2)*16+g],
                                                 sce1->band_type[(w+w2)*16+g],
-                                                lambda / minthr, INFINITY, NULL, 0);
+                                                lambda / minthr, INFINITY, NULL);
                 }
                 cpe->ms_mask[w*16+g] = dist2 < dist1;
             }
@@ -960,64 +1115,24 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
         search_for_quantizers_faac,
         encode_window_bands_info,
         quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
         search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
     },
     [AAC_CODER_ANMR] = {
         search_for_quantizers_anmr,
         encode_window_bands_info,
         quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
         search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
     },
     [AAC_CODER_TWOLOOP] = {
         search_for_quantizers_twoloop,
         codebook_trellis_rate,
         quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
         search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
     },
     [AAC_CODER_FAST] = {
         search_for_quantizers_fast,
         encode_window_bands_info,
         quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
         search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
     },
 };

libavcodec/aacdec_fixed.c

@@ -1,443 +0,0 @@
-/*
- * Copyright (c) 2013
- *      MIPS Technologies, Inc., California.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
- * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
- *    contributors may be used to endorse or promote products derived from
- *    this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
- * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
- * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
- * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
- * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
- * SUCH DAMAGE.
- *
- * AAC decoder fixed-point implementation
- *
- * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
- * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail 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
- * AAC decoder
- * @author Oded Shimon  ( ods15 ods15 dyndns org )
- * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
- *
- * Fixed point implementation
- * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
- */
-
-#define FFT_FLOAT 0
-#define FFT_FIXED_32 1
-#define USE_FIXED 1
-
-#include "libavutil/fixed_dsp.h"
-#include "libavutil/opt.h"
-#include "avcodec.h"
-#include "internal.h"
-#include "get_bits.h"
-#include "fft.h"
-#include "lpc.h"
-#include "kbdwin.h"
-#include "sinewin.h"
-
-#include "aac.h"
-#include "aactab.h"
-#include "aacdectab.h"
-#include "cbrt_tablegen.h"
-#include "sbr.h"
-#include "aacsbr.h"
-#include "mpeg4audio.h"
-#include "aacadtsdec.h"
-#include "libavutil/intfloat.h"
-
-#include <math.h>
-#include <string.h>
-
-static av_always_inline void reset_predict_state(PredictorState *ps)
-{
-    ps->r0.mant   = 0;
-    ps->r0.exp   = 0;
-    ps->r1.mant   = 0;
-    ps->r1.exp   = 0;
-    ps->cor0.mant = 0;
-    ps->cor0.exp = 0;
-    ps->cor1.mant = 0;
-    ps->cor1.exp = 0;
-    ps->var0.mant = 0x20000000;
-    ps->var0.exp = 1;
-    ps->var1.mant = 0x20000000;
-    ps->var1.exp = 1;
-}
-
-static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) };  // 2^0, 2^0.25, 2^0.5, 2^0.75
-
-static inline int *DEC_SPAIR(int *dst, unsigned idx)
-{
-    dst[0] = (idx & 15) - 4;
-    dst[1] = (idx >> 4 & 15) - 4;
-
-    return dst + 2;
-}
-
-static inline int *DEC_SQUAD(int *dst, unsigned idx)
-{
-    dst[0] = (idx & 3) - 1;
-    dst[1] = (idx >> 2 & 3) - 1;
-    dst[2] = (idx >> 4 & 3) - 1;
-    dst[3] = (idx >> 6 & 3) - 1;
-
-    return dst + 4;
-}
-
-static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
-{
-    dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
-    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) << 1));
-
-    return dst + 2;
-}
-
-static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
-{
-    unsigned nz = idx >> 12;
-
-    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) << 1));
-    sign <<= nz & 1;
-    nz >>= 1;
-    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
-    sign <<= nz & 1;
-    nz >>= 1;
-    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
-    sign <<= nz & 1;
-    nz >>= 1;
-    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));
-
-    return dst + 4;
-}
-
-static void vector_pow43(int *coefs, int len)
-{
-    int i, coef;
-
-    for (i=0; i<len; i++) {
-        coef = coefs[i];
-        if (coef < 0)
-            coef = -(int)cbrt_tab[-coef];
-        else
-            coef = (int)cbrt_tab[coef];
-        coefs[i] = coef;
-    }
-}
-
-static void subband_scale(int *dst, int *src, int scale, int offset, int len)
-{
-    int ssign = scale < 0 ? -1 : 1;
-    int s = FFABS(scale);
-    unsigned int round;
-    int i, out, c = exp2tab[s & 3];
-
-    s = offset - (s >> 2);
-
-    if (s > 0) {
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)(((int64_t)src[i] * c) >> 32);
-            dst[i] = ((int)(out+round) >> s) * ssign;
-        }
-    }
-    else {
-        s = s + 32;
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
-            dst[i] = out * ssign;
-        }
-    }
-}
-
-static void noise_scale(int *coefs, int scale, int band_energy, int len)
-{
-    int ssign = scale < 0 ? -1 : 1;
-    int s = FFABS(scale);
-    unsigned int round;
-    int i, out, c = exp2tab[s & 3];
-    int nlz = 0;
-
-    while (band_energy > 0x7fff) {
-        band_energy >>= 1;
-        nlz++;
-    }
-    c /= band_energy;
-    s = 21 + nlz - (s >> 2);
-
-    if (s > 0) {
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)(((int64_t)coefs[i] * c) >> 32);
-            coefs[i] = ((int)(out+round) >> s) * ssign;
-        }
-    }
-    else {
-        s = s + 32;
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
-            coefs[i] = out * ssign;
-        }
-    }
-}
-
-static av_always_inline SoftFloat flt16_round(SoftFloat pf)
-{
-    SoftFloat tmp;
-    int s;
-
-    tmp.exp = pf.exp;
-    s = pf.mant >> 31;
-    tmp.mant = (pf.mant ^ s) - s;
-    tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
-    tmp.mant = (tmp.mant ^ s) - s;
-
-    return tmp;
-}
-
-static av_always_inline SoftFloat flt16_even(SoftFloat pf)
-{
-    SoftFloat tmp;
-    int s;
-
-    tmp.exp = pf.exp;
-    s = pf.mant >> 31;
-    tmp.mant = (pf.mant ^ s) - s;
-    tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
-    tmp.mant = (tmp.mant ^ s) - s;
-
-    return tmp;
-}
-
-static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
-{
-    SoftFloat pun;
-    int s;
-
-    pun.exp = pf.exp;
-    s = pf.mant >> 31;
-    pun.mant = (pf.mant ^ s) - s;
-    pun.mant = pun.mant & 0xFFC00000U;
-    pun.mant = (pun.mant ^ s) - s;
-
-    return pun;
-}
-
-static av_always_inline void predict(PredictorState *ps, int *coef,
-                                     int output_enable)
-{
-    const SoftFloat a     = { 1023410176, 0 };  // 61.0 / 64
-    const SoftFloat alpha = {  973078528, 0 };  // 29.0 / 32
-    SoftFloat e0, e1;
-    SoftFloat pv;
-    SoftFloat k1, k2;
-    SoftFloat   r0 = ps->r0,     r1 = ps->r1;
-    SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
-    SoftFloat var0 = ps->var0, var1 = ps->var1;
-    SoftFloat tmp;
-
-    if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
-        k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
-    }
-    else {
-        k1.mant = 0;
-        k1.exp = 0;
-    }
-
-    if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
-        k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
-    }
-    else {
-        k2.mant = 0;
-        k2.exp = 0;
-    }
-
-    tmp = av_mul_sf(k1, r0);
-    pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
-    if (output_enable) {
-        int shift = 28 - pv.exp;
-
-        if (shift < 31)
-            *coef += (pv.mant + (1 << (shift - 1))) >> shift;
-    }
-
-    e0 = av_int2sf(*coef, 2);
-    e1 = av_sub_sf(e0, tmp);
-
-    ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
-    tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
-    tmp.exp--;
-    ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
-    ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
-    tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
-    tmp.exp--;
-    ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
-
-    ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
-    ps->r0 = flt16_trunc(av_mul_sf(a, e0));
-}
-
-
-static const int cce_scale_fixed[8] = {
-    Q30(1.0),          //2^(0/8)
-    Q30(1.0905077327), //2^(1/8)
-    Q30(1.1892071150), //2^(2/8)
-    Q30(1.2968395547), //2^(3/8)
-    Q30(1.4142135624), //2^(4/8)
-    Q30(1.5422108254), //2^(5/8)
-    Q30(1.6817928305), //2^(6/8)
-    Q30(1.8340080864), //2^(7/8)
-};
-
-/**
- * Apply dependent channel coupling (applied before IMDCT).
- *
- * @param   index   index into coupling gain array
- */
-static void apply_dependent_coupling_fixed(AACContext *ac,
-                                     SingleChannelElement *target,
-                                     ChannelElement *cce, int index)
-{
-    IndividualChannelStream *ics = &cce->ch[0].ics;
-    const uint16_t *offsets = ics->swb_offset;
-    int *dest = target->coeffs;
-    const int *src = cce->ch[0].coeffs;
-    int g, i, group, k, idx = 0;
-    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
-        av_log(ac->avctx, AV_LOG_ERROR,
-               "Dependent coupling is not supported together with LTP\n");
-        return;
-    }
-    for (g = 0; g < ics->num_window_groups; g++) {
-        for (i = 0; i < ics->max_sfb; i++, idx++) {
-            if (cce->ch[0].band_type[idx] != ZERO_BT) {
-                const int gain = cce->coup.gain[index][idx];
-                int shift, round, c, tmp;
-
-                if (gain < 0) {
-                    c = -cce_scale_fixed[-gain & 7];
-                    shift = (-gain-1024) >> 3;
-                }
-                else {
-                    c = cce_scale_fixed[gain & 7];
-                    shift = (gain-1024) >> 3;
-                }
-
-                if (shift < 0) {
-                    shift = -shift;
-                    round = 1 << (shift - 1);
-
-                    for (group = 0; group < ics->group_len[g]; group++) {
-                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
-                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
-                                       (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += (tmp + round) >> shift;
-                        }
-                    }
-                }
-                else {
-                    for (group = 0; group < ics->group_len[g]; group++) {
-                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
-                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
-                                        (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += tmp << shift;
-                        }
-                    }
-                }
-            }
-        }
-        dest += ics->group_len[g] * 128;
-        src  += ics->group_len[g] * 128;
-    }
-}
-
-/**
- * Apply independent channel coupling (applied after IMDCT).
- *
- * @param   index   index into coupling gain array
- */
-static void apply_independent_coupling_fixed(AACContext *ac,
-                                       SingleChannelElement *target,
-                                       ChannelElement *cce, int index)
-{
-    int i, c, shift, round, tmp;
-    const int gain = cce->coup.gain[index][0];
-    const int *src = cce->ch[0].ret;
-    int *dest = target->ret;
-    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
-
-    c = cce_scale_fixed[gain & 7];
-    shift = (gain-1024) >> 3;
-    if (shift < 0) {
-        shift = -shift;
-        round = 1 << (shift - 1);
-
-        for (i = 0; i < len; i++) {
-            tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
-            dest[i] += (tmp + round) >> shift;
-        }
-    }
-    else {
-      for (i = 0; i < len; i++) {
-          tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
-          dest[i] += tmp << shift;
-      }
-    }
-}
-
-#include "aacdec_template.c"
-
-AVCodec ff_aac_fixed_decoder = {
-    .name            = "aac_fixed",
-    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
-    .type            = AVMEDIA_TYPE_AUDIO,
-    .id              = AV_CODEC_ID_AAC,
-    .priv_data_size  = sizeof(AACContext),
-    .init            = aac_decode_init,
-    .close           = aac_decode_close,
-    .decode          = aac_decode_frame,
-    .sample_fmts     = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
-    },
-    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
-    .channel_layouts = aac_channel_layout,
-    .flush = flush,
-};

libavcodec/aacdectab.h

@@ -38,14 +38,49 @@
 /* @name ltp_coef
  * Table of the LTP coefficients
  */
-static const INTFLOAT ltp_coef[8] = {
-    Q30(0.570829f), Q30(0.696616f), Q30(0.813004f), Q30(0.911304f),
-    Q30(0.984900f), Q30(1.067894f), Q30(1.194601f), Q30(1.369533f),
+static const float ltp_coef[8] = {
+    0.570829, 0.696616, 0.813004, 0.911304,
+    0.984900, 1.067894, 1.194601, 1.369533,
 };
 
-static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 4, 5, 0, 5, 0 };
+/* @name tns_tmp2_map
+ * Tables of the tmp2[] arrays of LPC coefficients used for TNS.
+ * The suffix _M_N[] indicate the values of coef_compress and coef_res
+ * respectively.
+ * @{
+ */
+static const float tns_tmp2_map_1_3[4] = {
+     0.00000000, -0.43388373,  0.64278758,  0.34202015,
+};
 
-static const uint8_t aac_channel_layout_map[16][5][3] = {
+static const float tns_tmp2_map_0_3[8] = {
+     0.00000000, -0.43388373, -0.78183150, -0.97492790,
+     0.98480773,  0.86602539,  0.64278758,  0.34202015,
+};
+
+static const float tns_tmp2_map_1_4[8] = {
+     0.00000000, -0.20791170, -0.40673664, -0.58778524,
+     0.67369562,  0.52643216,  0.36124167,  0.18374951,
+};
+
+static const float tns_tmp2_map_0_4[16] = {
+     0.00000000, -0.20791170, -0.40673664, -0.58778524,
+    -0.74314481, -0.86602539, -0.95105654, -0.99452192,
+     0.99573416,  0.96182561,  0.89516330,  0.79801720,
+     0.67369562,  0.52643216,  0.36124167,  0.18374951,
+};
+
+static const float * const tns_tmp2_map[4] = {
+    tns_tmp2_map_0_3,
+    tns_tmp2_map_0_4,
+    tns_tmp2_map_1_3,
+    tns_tmp2_map_1_4
+};
+// @}
+
+static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
+
+static const uint8_t aac_channel_layout_map[7][5][3] = {
     { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
     { { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
     { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
@@ -53,16 +88,9 @@ static const uint8_t aac_channel_layout_map[16][5][3] = {
     { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
     { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
     { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
-    { { 0, } },
-    { { 0, } },
-    { { 0, } },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_SIDE }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
-    { { 0, } },
-    /* TODO: Add 7+1 TOP configuration */
 };
 
-static const uint64_t aac_channel_layout[16] = {
+static const uint64_t aac_channel_layout[8] = {
     AV_CH_LAYOUT_MONO,
     AV_CH_LAYOUT_STEREO,
     AV_CH_LAYOUT_SURROUND,
@@ -71,12 +99,6 @@ static const uint64_t aac_channel_layout[16] = {
     AV_CH_LAYOUT_5POINT1_BACK,
     AV_CH_LAYOUT_7POINT1_WIDE_BACK,
     0,
-    0,
-    0,
-    AV_CH_LAYOUT_6POINT1,
-    AV_CH_LAYOUT_7POINT1,
-    0,
-    /* AV_CH_LAYOUT_7POINT1_TOP, */
 };
 
 #endif /* AVCODEC_AACDECTAB_H */

libavcodec/aacenc.c

@@ -27,6 +27,7 @@
 /***********************************
  *              TODOs:
  * add sane pulse detection
+ * add temporal noise shaping
  ***********************************/
 
 #include "libavutil/float_dsp.h"
@@ -41,11 +42,127 @@
 #include "aac.h"
 #include "aactab.h"
 #include "aacenc.h"
-#include "aacenctab.h"
-#include "aacenc_utils.h"
 
 #include "psymodel.h"
 
+#define AAC_MAX_CHANNELS 6
+
+#define ERROR_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
+        return AVERROR(EINVAL); \
+    }
+
+#define WARN_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
+    }
+
+float ff_aac_pow34sf_tab[428];
+
+static const uint8_t swb_size_1024_96[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
+    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_64[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
+    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
+    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
+};
+
+static const uint8_t swb_size_1024_48[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
+    96
+};
+
+static const uint8_t swb_size_1024_32[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
+};
+
+static const uint8_t swb_size_1024_24[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_16[] = {
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
+    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_8[] = {
+    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
+    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
+    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
+};
+
+static const uint8_t *swb_size_1024[] = {
+    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
+    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
+    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
+    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
+    swb_size_1024_8
+};
+
+static const uint8_t swb_size_128_96[] = {
+    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
+};
+
+static const uint8_t swb_size_128_48[] = {
+    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
+};
+
+static const uint8_t swb_size_128_24[] = {
+    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
+};
+
+static const uint8_t swb_size_128_16[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
+};
+
+static const uint8_t swb_size_128_8[] = {
+    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
+};
+
+static const uint8_t *swb_size_128[] = {
+    /* the last entry on the following row is swb_size_128_64 but is a
+       duplicate of swb_size_128_96 */
+    swb_size_128_96, swb_size_128_96, swb_size_128_96,
+    swb_size_128_48, swb_size_128_48, swb_size_128_48,
+    swb_size_128_24, swb_size_128_24, swb_size_128_16,
+    swb_size_128_16, swb_size_128_16, swb_size_128_8,
+    swb_size_128_8
+};
+
+/** default channel configurations */
+static const uint8_t aac_chan_configs[6][5] = {
+ {1, TYPE_SCE},                               // 1 channel  - single channel element
+ {1, TYPE_CPE},                               // 2 channels - channel pair
+ {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
+ {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
+ {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
+ {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
+};
+
+/**
+ * Table to remap channels from libavcodec's default order to AAC order.
+ */
+static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+    { 0 },
+    { 0, 1 },
+    { 2, 0, 1 },
+    { 2, 0, 1, 3 },
+    { 2, 0, 1, 3, 4 },
+    { 2, 0, 1, 4, 5, 3 },
+};
+
 /**
  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -56,7 +173,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
     AACEncContext *s = avctx->priv_data;
 
     init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
-    put_bits(&pb, 5, s->profile+1); //profile
+    put_bits(&pb, 5, 2); //object type - AAC-LC
     put_bits(&pb, 4, s->samplerate_index); //sample rate index
     put_bits(&pb, 4, s->channels);
     //GASpecificConfig
@@ -148,7 +265,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
     else
         for (i = 0; i < 1024; i += 128)
-            s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
     memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
     memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
 }
@@ -166,7 +283,7 @@ static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
     put_bits(&s->pb, 1, info->use_kb_window[0]);
     if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
         put_bits(&s->pb, 6, info->max_sfb);
-        put_bits(&s->pb, 1, !!info->predictor_present);
+        put_bits(&s->pb, 1, 0);            // no prediction
     } else {
         put_bits(&s->pb, 4, info->max_sfb);
         for (w = 1; w < 8; w++)
@@ -195,14 +312,26 @@ static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
 static void adjust_frame_information(ChannelElement *cpe, int chans)
 {
     int i, w, w2, g, ch;
-    int maxsfb, cmaxsfb;
+    int start, maxsfb, cmaxsfb;
 
     for (ch = 0; ch < chans; ch++) {
         IndividualChannelStream *ics = &cpe->ch[ch].ics;
+        start = 0;
         maxsfb = 0;
         cpe->ch[ch].pulse.num_pulse = 0;
         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-            for (w2 =  0; w2 < ics->group_len[w]; w2++) {
+            for (w2 = 0; w2 < ics->group_len[w]; w2++) {
+                start = (w+w2) * 128;
+                for (g = 0; g < ics->num_swb; g++) {
+                    //apply M/S
+                    if (cpe->common_window && !ch && cpe->ms_mask[w*16 + g]) {
+                        for (i = 0; i < ics->swb_sizes[g]; i++) {
+                            cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f;
+                            cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i];
+                        }
+                    }
+                    start += ics->swb_sizes[g];
+                }
                 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
                     ;
                 maxsfb = FFMAX(maxsfb, cmaxsfb);
@@ -242,59 +371,6 @@ static void adjust_frame_information(ChannelElement *cpe, int chans)
     }
 }
 
-static void apply_intensity_stereo(ChannelElement *cpe)
-{
-    int w, w2, g, i;
-    IndividualChannelStream *ics = &cpe->ch[0].ics;
-    if (!cpe->common_window)
-        return;
-    for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-        for (w2 =  0; w2 < ics->group_len[w]; w2++) {
-            int start = (w+w2) * 128;
-            for (g = 0; g < ics->num_swb; g++) {
-                int p  = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
-                float scale = cpe->ch[0].is_ener[w*16+g];
-                if (!cpe->is_mask[w*16 + g]) {
-                    start += ics->swb_sizes[g];
-                    continue;
-                }
-                for (i = 0; i < ics->swb_sizes[g]; i++) {
-                    float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
-                    cpe->ch[0].coeffs[start+i] = sum;
-                    cpe->ch[1].coeffs[start+i] = 0.0f;
-                }
-                start += ics->swb_sizes[g];
-            }
-        }
-    }
-}
-
-static void apply_mid_side_stereo(ChannelElement *cpe)
-{
-    int w, w2, g, i;
-    IndividualChannelStream *ics = &cpe->ch[0].ics;
-    if (!cpe->common_window)
-        return;
-    for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-        for (w2 =  0; w2 < ics->group_len[w]; w2++) {
-            int start = (w+w2) * 128;
-            for (g = 0; g < ics->num_swb; g++) {
-                if (!cpe->ms_mask[w*16 + g]) {
-                    start += ics->swb_sizes[g];
-                    continue;
-                }
-                for (i = 0; i < ics->swb_sizes[g]; i++) {
-                    float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
-                    float R = L - cpe->ch[1].coeffs[start+i];
-                    cpe->ch[0].coeffs[start+i] = L;
-                    cpe->ch[1].coeffs[start+i] = R;
-                }
-                start += ics->swb_sizes[g];
-            }
-        }
-    }
-}
-
 /**
  * Encode scalefactor band coding type.
  */
@@ -302,9 +378,6 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 {
     int w;
 
-    if (s->coder->set_special_band_scalefactors)
-        s->coder->set_special_band_scalefactors(s, sce);
-
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
         s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
 }
@@ -315,30 +388,15 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                                  SingleChannelElement *sce)
 {
-    int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
-    int off_is = 0, noise_flag = 1;
+    int off = sce->sf_idx[0], diff;
     int i, w;
 
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         for (i = 0; i < sce->ics.max_sfb; i++) {
             if (!sce->zeroes[w*16 + i]) {
-                if (sce->band_type[w*16 + i] == NOISE_BT) {
-                    diff = sce->sf_idx[w*16 + i] - off_pns;
-                    off_pns = sce->sf_idx[w*16 + i];
-                    if (noise_flag-- > 0) {
-                        put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
-                        continue;
-                    }
-                } else if (sce->band_type[w*16 + i] == INTENSITY_BT  ||
-                           sce->band_type[w*16 + i] == INTENSITY_BT2) {
-                    diff = sce->sf_idx[w*16 + i] - off_is;
-                    off_is = sce->sf_idx[w*16 + i];
-                } else {
-                    diff = sce->sf_idx[w*16 + i] - off_sf;
-                    off_sf = sce->sf_idx[w*16 + i];
-                }
-                diff += SCALE_DIFF_ZERO;
+                diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
                 av_assert0(diff >= 0 && diff <= 120);
+                off = sce->sf_idx[w*16 + i];
                 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
             }
         }
@@ -378,40 +436,17 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
                 start += sce->ics.swb_sizes[i];
                 continue;
             }
-            for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
-                s->coder->quantize_and_encode_band(s, &s->pb,
-                                                   &sce->coeffs[start + w2*128],
-                                                   NULL, sce->ics.swb_sizes[i],
+            for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
+                s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
+                                                   sce->ics.swb_sizes[i],
                                                    sce->sf_idx[w*16 + i],
                                                    sce->band_type[w*16 + i],
-                                                   s->lambda,
-                                                   sce->ics.window_clipping[w]);
-            }
+                                                   s->lambda);
             start += sce->ics.swb_sizes[i];
         }
     }
 }
 
-/**
- * Downscale spectral coefficients for near-clipping windows to avoid artifacts
- */
-static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
-{
-    int start, i, j, w;
-
-    if (sce->ics.clip_avoidance_factor < 1.0f) {
-        for (w = 0; w < sce->ics.num_windows; w++) {
-            start = 0;
-            for (i = 0; i < sce->ics.max_sfb; i++) {
-                float *swb_coeffs = &sce->coeffs[start + w*128];
-                for (j = 0; j < sce->ics.swb_sizes[i]; j++)
-                    swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
-                start += sce->ics.swb_sizes[i];
-            }
-        }
-    }
-}
-
 /**
  * Encode one channel of audio data.
  */
@@ -420,17 +455,12 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
                                      int common_window)
 {
     put_bits(&s->pb, 8, sce->sf_idx[0]);
-    if (!common_window) {
+    if (!common_window)
         put_ics_info(s, &sce->ics);
-        if (s->coder->encode_main_pred)
-            s->coder->encode_main_pred(s, sce);
-    }
     encode_band_info(s, sce);
     encode_scale_factors(avctx, s, sce);
     encode_pulses(s, &sce->pulse);
-    put_bits(&s->pb, 1, !!sce->tns.present);
-    if (s->coder->encode_tns_info)
-        s->coder->encode_tns_info(s, sce);
+    put_bits(&s->pb, 1, 0); //tns
     put_bits(&s->pb, 1, 0); //ssr
     encode_spectral_coeffs(s, sce);
     return 0;
@@ -488,9 +518,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     AACEncContext *s = avctx->priv_data;
     float **samples = s->planar_samples, *samples2, *la, *overlap;
     ChannelElement *cpe;
-    SingleChannelElement *sce;
-    int i, ch, w, chans, tag, start_ch, ret;
-    int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
+    int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0;
     int chan_el_counter[4];
     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
 
@@ -519,7 +547,6 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         for (ch = 0; ch < chans; ch++) {
             IndividualChannelStream *ics = &cpe->ch[ch].ics;
             int cur_channel = start_ch + ch;
-            float clip_avoidance_factor;
             overlap  = &samples[cur_channel][0];
             samples2 = overlap + 1024;
             la       = samples2 + (448+64);
@@ -547,46 +574,25 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             ics->num_windows        = wi[ch].num_windows;
             ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
             ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
-            ics->swb_offset         = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
-                                        ff_swb_offset_128 [s->samplerate_index]:
-                                        ff_swb_offset_1024[s->samplerate_index];
-            ics->tns_max_bands      = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
-                                        ff_tns_max_bands_128 [s->samplerate_index]:
-                                        ff_tns_max_bands_1024[s->samplerate_index];
-            clip_avoidance_factor = 0.0f;
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];
-            for (w = 0; w < ics->num_windows; w++) {
-                if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
-                    ics->window_clipping[w] = 1;
-                    clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
-                } else {
-                    ics->window_clipping[w] = 0;
-                }
-            }
-            if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
-                ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
-            } else {
-                ics->clip_avoidance_factor = 1.0f;
-            }
 
             apply_window_and_mdct(s, &cpe->ch[ch], overlap);
             if (isnan(cpe->ch->coeffs[0])) {
                 av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
                 return AVERROR(EINVAL);
             }
-            avoid_clipping(s, &cpe->ch[ch]);
         }
         start_ch += chans;
     }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
         return ret;
     do {
         int frame_bits;
 
         init_put_bits(&s->pb, avpkt->data, avpkt->size);
 
-        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
+        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
             put_bitstream_info(s, LIBAVCODEC_IDENT);
         start_ch = 0;
         memset(chan_el_counter, 0, sizeof(chan_el_counter));
@@ -596,26 +602,16 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             tag      = s->chan_map[i+1];
             chans    = tag == TYPE_CPE ? 2 : 1;
             cpe      = &s->cpe[i];
-            cpe->common_window = 0;
-            memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
-            memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
             put_bits(&s->pb, 3, tag);
             put_bits(&s->pb, 4, chan_el_counter[tag]++);
-            for (ch = 0; ch < chans; ch++) {
-                sce = &cpe->ch[ch];
-                coeffs[ch] = sce->coeffs;
-                sce->ics.predictor_present = 0;
-                memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
-                memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
-                for (w = 0; w < 128; w++)
-                    if (sce->band_type[w] > RESERVED_BT)
-                        sce->band_type[w] = 0;
-            }
+            for (ch = 0; ch < chans; ch++)
+                coeffs[ch] = cpe->ch[ch].coeffs;
             s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
             for (ch = 0; ch < chans; ch++) {
                 s->cur_channel = start_ch + ch;
                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
             }
+            cpe->common_window = 0;
             if (chans > 1
                 && wi[0].window_type[0] == wi[1].window_type[0]
                 && wi[0].window_shape   == wi[1].window_shape) {
@@ -628,59 +624,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                     }
                 }
             }
-            for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
-                sce = &cpe->ch[ch];
-                s->cur_channel = start_ch + ch;
-                if (s->options.pns && s->coder->search_for_pns)
-                    s->coder->search_for_pns(s, avctx, sce);
-                if (s->options.tns && s->coder->search_for_tns)
-                    s->coder->search_for_tns(s, sce);
-                if (s->options.tns && s->coder->apply_tns_filt)
-                    s->coder->apply_tns_filt(s, sce);
-                if (sce->tns.present)
-                    tns_mode = 1;
-            }
             s->cur_channel = start_ch;
-            if (s->options.intensity_stereo) { /* Intensity Stereo */
-                if (s->coder->search_for_is)
-                    s->coder->search_for_is(s, avctx, cpe);
-                if (cpe->is_mode) is_mode = 1;
-                apply_intensity_stereo(cpe);
-            }
-            if (s->options.pred) { /* Prediction */
-                for (ch = 0; ch < chans; ch++) {
-                    sce = &cpe->ch[ch];
-                    s->cur_channel = start_ch + ch;
-                    if (s->options.pred && s->coder->search_for_pred)
-                        s->coder->search_for_pred(s, sce);
-                    if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
+            if (s->options.stereo_mode && cpe->common_window) {
+                if (s->options.stereo_mode > 0) {
+                    IndividualChannelStream *ics = &cpe->ch[0].ics;
+                    for (w = 0; w < ics->num_windows; w += ics->group_len[w])
+                        for (g = 0;  g < ics->num_swb; g++)
+                            cpe->ms_mask[w*16+g] = 1;
+                } else if (s->coder->search_for_ms) {
+                    s->coder->search_for_ms(s, cpe, s->lambda);
                 }
-                if (s->coder->adjust_common_prediction)
-                    s->coder->adjust_common_prediction(s, cpe);
-                for (ch = 0; ch < chans; ch++) {
-                    sce = &cpe->ch[ch];
-                    s->cur_channel = start_ch + ch;
-                    if (s->options.pred && s->coder->apply_main_pred)
-                        s->coder->apply_main_pred(s, sce);
-                }
-                s->cur_channel = start_ch;
-            }
-            if (s->options.stereo_mode) { /* Mid/Side stereo */
-                if (s->options.stereo_mode == -1 && s->coder->search_for_ms)
-                    s->coder->search_for_ms(s, cpe);
-                else if (cpe->common_window)
-                    memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
-                for (w = 0; w < 128; w++)
-                    cpe->ms_mask[w] = cpe->is_mask[w] ? 0 : cpe->ms_mask[w];
-                apply_mid_side_stereo(cpe);
             }
             adjust_frame_information(cpe, chans);
             if (chans == 2) {
                 put_bits(&s->pb, 1, cpe->common_window);
                 if (cpe->common_window) {
                     put_ics_info(s, &cpe->ch[0].ics);
-                    if (s->coder->encode_main_pred)
-                        s->coder->encode_main_pred(s, &cpe->ch[0]);
                     encode_ms_info(&s->pb, cpe);
                     if (cpe->ms_mode) ms_mode = 1;
                 }
@@ -697,7 +656,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             s->psy.bitres.bits = frame_bits / s->channels;
             break;
         }
-        if (is_mode || ms_mode || tns_mode || pred_mode) {
+        if (ms_mode) {
             for (i = 0; i < s->chan_map[0]; i++) {
                 // Must restore coeffs
                 chans = tag == TYPE_CPE ? 2 : 1;
@@ -716,7 +675,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     avctx->frame_bits = put_bits_count(&s->pb);
 
     // rate control stuff
-    if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
+    if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
         float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
         s->lambda *= ratio;
         s->lambda = FFMIN(s->lambda, 65536.f);
@@ -740,7 +699,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);
     ff_psy_end(&s->psy);
-    ff_lpc_end(&s->lpc);
     if (s->psypp)
         ff_psy_preprocess_end(s->psypp);
     av_freep(&s->buffer.samples);
@@ -754,7 +712,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 
-    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
     if (!s->fdsp)
         return AVERROR(ENOMEM);
 
@@ -764,9 +722,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
     ff_init_ff_sine_windows(10);
     ff_init_ff_sine_windows(7);
 
-    if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
+    if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
         return ret;
-    if ((ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0)) < 0)
+    if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))
         return ret;
 
     return 0;
@@ -777,7 +735,7 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
     int ch;
     FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
     FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
 
     for(ch = 0; ch < s->channels; ch++)
         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
@@ -803,29 +761,16 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
 
     s->channels = avctx->channels;
 
-    ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
+    ERROR_IF(i == 16
+                || i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))
+                || i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),
              "Unsupported sample rate %d\n", avctx->sample_rate);
     ERROR_IF(s->channels > AAC_MAX_CHANNELS,
              "Unsupported number of channels: %d\n", s->channels);
+    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
+             "Unsupported profile %d\n", avctx->profile);
     WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
              "Too many bits per frame requested, clamping to max\n");
-    if (avctx->profile == FF_PROFILE_AAC_MAIN) {
-        s->options.pred = 1;
-    } else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
-                avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
-        s->profile = 0; /* Main */
-        WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
-    } else if (avctx->profile == FF_PROFILE_AAC_LOW ||
-               avctx->profile == FF_PROFILE_UNKNOWN) {
-        s->profile = 1; /* Low */
-    } else {
-        ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
-    }
-
-    if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
-        s->options.intensity_stereo = 0;
-        s->options.pns = 0;
-    }
 
     avctx->bit_rate = (int)FFMIN(
         6144 * s->channels / 1024.0 * avctx->sample_rate,
@@ -844,8 +789,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     avctx->extradata_size = 5;
     put_audio_specific_config(avctx);
 
-    sizes[0]   = ff_aac_swb_size_1024[i];
-    sizes[1]   = ff_aac_swb_size_128[i];
+    sizes[0]   = swb_size_1024[i];
+    sizes[1]   = swb_size_128[i];
     lengths[0] = ff_aac_num_swb_1024[i];
     lengths[1] = ff_aac_num_swb_128[i];
     for (i = 0; i < s->chan_map[0]; i++)
@@ -855,7 +800,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
         goto fail;
     s->psypp = ff_psy_preprocess_init(avctx);
     s->coder = &ff_aac_coders[s->options.aac_coder];
-    ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
 
     if (HAVE_MIPSDSPR1)
         ff_aac_coder_init_mips(s);
@@ -864,6 +808,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
 
     ff_aac_tableinit();
 
+    for (i = 0; i < 428; i++)
+        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
+
     avctx->initial_padding = 1024;
     ff_af_queue_init(avctx, &s->afq);
 
@@ -879,23 +826,11 @@ static const AVOption aacenc_options[] = {
         {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
+    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
         {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
         {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
         {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
-    {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "aac_pns"},
-        {"disable",  "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-        {"enable",   "Enable perceptual noise substitution",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-    {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
-        {"disable",  "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
-        {"enable",   "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
-    {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_tns"},
-        {"disable",  "Disable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
-        {"enable",   "Enable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
-    {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pred"},
-        {"disable",  "Disable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
-        {"enable",   "Enable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
     {NULL}
 };
 
@@ -906,6 +841,13 @@ static const AVClass aacenc_class = {
     LIBAVUTIL_VERSION_INT,
 };
 
+/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
+ * failures */
+static const int mpeg4audio_sample_rates[16] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+
 AVCodec ff_aac_encoder = {
     .name           = "aac",
     .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
@@ -916,8 +858,8 @@ AVCodec ff_aac_encoder = {
     .encode2        = aac_encode_frame,
     .close          = aac_encode_end,
     .supported_samplerates = mpeg4audio_sample_rates,
-    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
-                      AV_CODEC_CAP_EXPERIMENTAL,
+    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
+                      CODEC_CAP_EXPERIMENTAL,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
     .priv_class     = &aacenc_class,

libavcodec/aacenc.h

@@ -30,8 +30,6 @@
 #include "audio_frame_queue.h"
 #include "psymodel.h"
 
-#include "lpc.h"
-
 typedef enum AACCoder {
     AAC_CODER_FAAC = 0,
     AAC_CODER_ANMR,
@@ -44,10 +42,6 @@ typedef enum AACCoder {
 typedef struct AACEncOptions {
     int stereo_mode;
     int aac_coder;
-    int pns;
-    int tns;
-    int pred;
-    int intensity_stereo;
 } AACEncOptions;
 
 struct AACEncContext;
@@ -57,19 +51,9 @@ typedef struct AACCoefficientsEncoder {
                                   SingleChannelElement *sce, const float lambda);
     void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
                                      int win, int group_len, const float lambda);
-    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
-                                     int scale_idx, int cb, const float lambda, int rtz);
-    void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*adjust_common_prediction)(struct AACEncContext *s, ChannelElement *cpe);
-    void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
-    void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
-    void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
-    void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
+                                     int scale_idx, int cb, const float lambda);
+    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
 } AACCoefficientsEncoder;
 
 extern AACCoefficientsEncoder ff_aac_coders[];
@@ -86,8 +70,6 @@ typedef struct AACEncContext {
     AVFloatDSPContext *fdsp;
     float *planar_samples[6];                    ///< saved preprocessed input
 
-    int profile;                                 ///< copied from avctx
-    LPCContext lpc;                              ///< used by TNS
     int samplerate_index;                        ///< MPEG-4 samplerate index
     int channels;                                ///< channel count
     const uint8_t *chan_map;                     ///< channel configuration map
@@ -108,6 +90,8 @@ typedef struct AACEncContext {
     } buffer;
 } AACEncContext;
 
+extern float ff_aac_pow34sf_tab[428];
+
 void ff_aac_coder_init_mips(AACEncContext *c);
 
 #endif /* AVCODEC_AACENC_H */

libavcodec/aacenc_is.c

@@ -1,136 +0,0 @@
-/*
- * AAC encoder intensity stereo
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * 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
- * AAC encoder Intensity Stereo
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#include "aacenc.h"
-#include "aacenc_utils.h"
-#include "aacenc_is.h"
-#include "aacenc_quantization.h"
-
-struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
-                                         int start, int w, int g, float ener0,
-                                         float ener1, float ener01,
-                                         int use_pcoeffs, int phase)
-{
-    int i, w2;
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    float *L = use_pcoeffs ? sce0->pcoeffs : sce0->coeffs;
-    float *R = use_pcoeffs ? sce1->pcoeffs : sce1->coeffs;
-    float *L34 = &s->scoefs[256*0], *R34 = &s->scoefs[256*1];
-    float *IS  = &s->scoefs[256*2], *I34 = &s->scoefs[256*3];
-    float dist1 = 0.0f, dist2 = 0.0f;
-    struct AACISError is_error = {0};
-
-    for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
-        FFPsyBand *band0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-        FFPsyBand *band1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
-        int is_band_type, is_sf_idx = FFMAX(1, sce0->sf_idx[(w+w2)*16+g]-4);
-        float e01_34 = phase*pow(ener1/ener0, 3.0/4.0);
-        float maxval, dist_spec_err = 0.0f;
-        float minthr = FFMIN(band0->threshold, band1->threshold);
-        for (i = 0; i < sce0->ics.swb_sizes[g]; i++)
-            IS[i] = (L[start+(w+w2)*128+i] + phase*R[start+(w+w2)*128+i])*sqrt(ener0/ener01);
-        abs_pow34_v(L34, &L[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
-        abs_pow34_v(R34, &R[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
-        abs_pow34_v(I34, IS,                   sce0->ics.swb_sizes[g]);
-        maxval = find_max_val(1, sce0->ics.swb_sizes[g], I34);
-        is_band_type = find_min_book(maxval, is_sf_idx);
-        dist1 += quantize_band_cost(s, &L[start + (w+w2)*128], L34,
-                                    sce0->ics.swb_sizes[g],
-                                    sce0->sf_idx[(w+w2)*16+g],
-                                    sce0->band_type[(w+w2)*16+g],
-                                    s->lambda / band0->threshold, INFINITY, NULL, 0);
-        dist1 += quantize_band_cost(s, &R[start + (w+w2)*128], R34,
-                                    sce1->ics.swb_sizes[g],
-                                    sce1->sf_idx[(w+w2)*16+g],
-                                    sce1->band_type[(w+w2)*16+g],
-                                    s->lambda / band1->threshold, INFINITY, NULL, 0);
-        dist2 += quantize_band_cost(s, IS, I34, sce0->ics.swb_sizes[g],
-                                    is_sf_idx, is_band_type,
-                                    s->lambda / minthr, INFINITY, NULL, 0);
-        for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-            dist_spec_err += (L34[i] - I34[i])*(L34[i] - I34[i]);
-            dist_spec_err += (R34[i] - I34[i]*e01_34)*(R34[i] - I34[i]*e01_34);
-        }
-        dist_spec_err *= s->lambda / minthr;
-        dist2 += dist_spec_err;
-    }
-
-    is_error.pass = dist2 <= dist1;
-    is_error.phase = phase;
-    is_error.error = fabsf(dist1 - dist2);
-    is_error.dist1 = dist1;
-    is_error.dist2 = dist2;
-
-    return is_error;
-}
-
-void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
-{
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    int start = 0, count = 0, w, w2, g, i;
-    const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
-
-    if (!cpe->common_window)
-        return;
-
-    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
-        start = 0;
-        for (g = 0;  g < sce0->ics.num_swb; g++) {
-            if (start*freq_mult > INT_STEREO_LOW_LIMIT*(s->lambda/170.0f) &&
-                cpe->ch[0].band_type[w*16+g] != NOISE_BT && !cpe->ch[0].zeroes[w*16+g] &&
-                cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g]) {
-                float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
-                struct AACISError ph_err1, ph_err2, *erf;
-                for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
-                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                        float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
-                        float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
-                        ener0  += coef0*coef0;
-                        ener1  += coef1*coef1;
-                        ener01 += (coef0 + coef1)*(coef0 + coef1);
-                    }
-                }
-                ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                                 ener0, ener1, ener01, 0, -1);
-                ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                                 ener0, ener1, ener01, 0, +1);
-                erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
-                if (erf->pass) {
-                    cpe->is_mask[w*16+g] = 1;
-                    cpe->ch[0].is_ener[w*16+g] = sqrt(ener0/ener01);
-                    cpe->ch[1].is_ener[w*16+g] = ener0/ener1;
-                    cpe->ch[1].band_type[w*16+g] = erf->phase ? INTENSITY_BT : INTENSITY_BT2;
-                    count++;
-                }
-            }
-            start += sce0->ics.swb_sizes[g];
-        }
-    }
-    cpe->is_mode = !!count;
-}

libavcodec/aacenc_is.h

@@ -1,50 +0,0 @@
-/*
- * AAC encoder intensity stereo
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * 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
- * AAC encoder Intensity Stereo
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_IS_H
-#define AVCODEC_AACENC_IS_H
-
-#include "aacenc.h"
-
-/** Frequency in Hz for lower limit of intensity stereo **/
-#define INT_STEREO_LOW_LIMIT 6100
-
-struct AACISError {
-    int pass;    /* 1 if dist2 <= dist1  */
-    int phase;   /* -1 or +1             */
-    float error; /* fabs(dist1 - dist2)  */
-    float dist1; /* From original coeffs */
-    float dist2; /* From IS'd coeffs     */
-};
-
-struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
-                                         int start, int w, int g, float ener0,
-                                         float ener1, float ener01,
-                                         int use_pcoeffs, int phase);
-void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
-
-#endif /* AVCODEC_AACENC_IS_H */

libavcodec/aacenc_pred.c

@@ -1,342 +0,0 @@
-/*
- * AAC encoder main-type prediction
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * 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
- * AAC encoder Intensity Stereo
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#include "aactab.h"
-#include "aacenc_pred.h"
-#include "aacenc_utils.h"
-#include "aacenc_is.h"            /* <- Needed for common window distortions */
-#include "aacenc_quantization.h"
-
-#define RESTORE_PRED(sce, sfb) \
-        if (sce->ics.prediction_used[sfb]) {\
-            sce->ics.prediction_used[sfb] = 0;\
-            sce->band_type[sfb] = sce->band_alt[sfb];\
-        }
-
-static inline float flt16_round(float pf)
-{
-    union av_intfloat32 tmp;
-    tmp.f = pf;
-    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
-    return tmp.f;
-}
-
-static inline float flt16_even(float pf)
-{
-    union av_intfloat32 tmp;
-    tmp.f = pf;
-    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
-    return tmp.f;
-}
-
-static inline float flt16_trunc(float pf)
-{
-    union av_intfloat32 pun;
-    pun.f = pf;
-    pun.i &= 0xFFFF0000U;
-    return pun.f;
-}
-
-static inline void predict(PredictorState *ps, float *coef, float *rcoef, int set)
-{
-    float k2;
-    const float a     = 0.953125; // 61.0 / 64
-    const float alpha = 0.90625;  // 29.0 / 32
-    const float   k1 = ps->k1;
-    const float   r0 = ps->r0,     r1 = ps->r1;
-    const float cor0 = ps->cor0, cor1 = ps->cor1;
-    const float var0 = ps->var0, var1 = ps->var1;
-    const float e0 = *coef - ps->x_est;
-    const float e1 = e0 - k1 * r0;
-
-    if (set)
-        *coef = e0;
-
-    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
-    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
-    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
-    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
-    ps->r1   = flt16_trunc(a * (r0 - k1 * e0));
-    ps->r0   = flt16_trunc(a * e0);
-
-    /* Prediction for next frame */
-    ps->k1   = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
-    k2       = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
-    *rcoef   = ps->x_est = flt16_round(ps->k1*ps->r0 + k2*ps->r1);
-}
-
-static inline void reset_predict_state(PredictorState *ps)
-{
-    ps->r0    = 0.0f;
-    ps->r1    = 0.0f;
-    ps->k1    = 0.0f;
-    ps->cor0  = 0.0f;
-    ps->cor1  = 0.0f;
-    ps->var0  = 1.0f;
-    ps->var1  = 1.0f;
-    ps->x_est = 0.0f;
-}
-
-static inline void reset_all_predictors(PredictorState *ps)
-{
-    int i;
-    for (i = 0; i < MAX_PREDICTORS; i++)
-        reset_predict_state(&ps[i]);
-}
-
-static inline void reset_predictor_group(SingleChannelElement *sce, int group_num)
-{
-    int i;
-    PredictorState *ps = sce->predictor_state;
-    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
-        reset_predict_state(&ps[i]);
-}
-
-void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce)
-{
-    int sfb, k;
-    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-
-    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
-        for (sfb = 0; sfb < pmax; sfb++) {
-            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
-                predict(&sce->predictor_state[k], &sce->coeffs[k], &sce->prcoeffs[k],
-                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
-            }
-        }
-        if (sce->ics.predictor_reset_group) {
-            reset_predictor_group(sce, sce->ics.predictor_reset_group);
-        }
-    } else {
-        reset_all_predictors(sce->predictor_state);
-    }
-}
-
-/* If inc = 0 you can check if this returns 0 to see if you can reset freely */
-static inline int update_counters(IndividualChannelStream *ics, int inc)
-{
-    int i;
-    for (i = 1; i < 31; i++) {
-        ics->predictor_reset_count[i] += inc;
-        if (ics->predictor_reset_count[i] > PRED_RESET_FRAME_MIN)
-            return i; /* Reset this immediately */
-    }
-    return 0;
-}
-
-void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe)
-{
-    int start, w, w2, g, i, count = 0;
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    const int pmax0 = FFMIN(sce0->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-    const int pmax1 = FFMIN(sce1->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-    const int pmax  = FFMIN(pmax0, pmax1);
-
-    if (!cpe->common_window ||
-        sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
-        sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
-        return;
-
-    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
-        start = 0;
-        for (g = 0; g < sce0->ics.num_swb; g++) {
-            int sfb = w*16+g;
-            int sum = sce0->ics.prediction_used[sfb] + sce1->ics.prediction_used[sfb];
-            float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
-            struct AACISError ph_err1, ph_err2, *erf;
-            if (sfb < PRED_SFB_START || sfb > pmax || sum != 2) {
-                RESTORE_PRED(sce0, sfb);
-                RESTORE_PRED(sce1, sfb);
-                start += sce0->ics.swb_sizes[g];
-                continue;
-            }
-            for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
-                for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                    float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
-                    float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
-                    ener0  += coef0*coef0;
-                    ener1  += coef1*coef1;
-                    ener01 += (coef0 + coef1)*(coef0 + coef1);
-                }
-            }
-            ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                             ener0, ener1, ener01, 1, -1);
-            ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                             ener0, ener1, ener01, 1, +1);
-            erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
-            if (erf->pass) {
-                sce0->ics.prediction_used[sfb] = 1;
-                sce1->ics.prediction_used[sfb] = 1;
-                count++;
-            } else {
-                RESTORE_PRED(sce0, sfb);
-                RESTORE_PRED(sce1, sfb);
-            }
-            start += sce0->ics.swb_sizes[g];
-        }
-    }
-
-    sce1->ics.predictor_present = sce0->ics.predictor_present = !!count;
-}
-
-static void update_pred_resets(SingleChannelElement *sce)
-{
-    int i, max_group_id_c, max_frame = 0;
-    float avg_frame = 0.0f;
-    IndividualChannelStream *ics = &sce->ics;
-
-    /* Update the counters and immediately update any frame behind schedule */
-    if ((ics->predictor_reset_group = update_counters(&sce->ics, 1)))
-        return;
-
-    for (i = 1; i < 31; i++) {
-        /* Count-based */
-        if (ics->predictor_reset_count[i] > max_frame) {
-            max_group_id_c = i;
-            max_frame = ics->predictor_reset_count[i];
-        }
-        avg_frame = (ics->predictor_reset_count[i] + avg_frame)/2;
-    }
-
-    if (max_frame > PRED_RESET_MIN) {
-        ics->predictor_reset_group = max_group_id_c;
-    } else {
-        ics->predictor_reset_group = 0;
-    }
-}
-
-void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce)
-{
-    int sfb, i, count = 0, cost_coeffs = 0, cost_pred = 0;
-    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-    float *O34  = &s->scoefs[128*0], *P34 = &s->scoefs[128*1];
-    float *SENT = &s->scoefs[128*2], *S34 = &s->scoefs[128*3];
-    float *QERR = &s->scoefs[128*4];
-
-    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-        sce->ics.predictor_present = 0;
-        return;
-    }
-
-    if (!sce->ics.predictor_initialized) {
-        reset_all_predictors(sce->predictor_state);
-        sce->ics.predictor_initialized = 1;
-        memcpy(sce->prcoeffs, sce->coeffs, 1024*sizeof(float));
-        for (i = 1; i < 31; i++)
-            sce->ics.predictor_reset_count[i] = i;
-    }
-
-    update_pred_resets(sce);
-    memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
-
-    for (sfb = PRED_SFB_START; sfb < pmax; sfb++) {
-        int cost1, cost2, cb_p;
-        float dist1, dist2, dist_spec_err = 0.0f;
-        const int cb_n = sce->band_type[sfb];
-        const int start_coef = sce->ics.swb_offset[sfb];
-        const int num_coeffs = sce->ics.swb_offset[sfb + 1] - start_coef;
-        const FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[sfb];
-
-        if (start_coef + num_coeffs > MAX_PREDICTORS)
-            continue;
-
-        /* Normal coefficients */
-        abs_pow34_v(O34, &sce->coeffs[start_coef], num_coeffs);
-        dist1 = quantize_and_encode_band_cost(s, NULL, &sce->coeffs[start_coef], NULL,
-                                              O34, num_coeffs, sce->sf_idx[sfb],
-                                              cb_n, s->lambda / band->threshold, INFINITY, &cost1, 0);
-        cost_coeffs += cost1;
-
-        /* Encoded coefficients - needed for #bits, band type and quant. error */
-        for (i = 0; i < num_coeffs; i++)
-            SENT[i] = sce->coeffs[start_coef + i] - sce->prcoeffs[start_coef + i];
-        abs_pow34_v(S34, SENT, num_coeffs);
-        if (cb_n < RESERVED_BT)
-            cb_p = find_min_book(find_max_val(1, num_coeffs, S34), sce->sf_idx[sfb]);
-        else
-            cb_p = cb_n;
-        quantize_and_encode_band_cost(s, NULL, SENT, QERR, S34, num_coeffs,
-                                      sce->sf_idx[sfb], cb_p, s->lambda / band->threshold, INFINITY,
-                                      &cost2, 0);
-
-        /* Reconstructed coefficients - needed for distortion measurements */
-        for (i = 0; i < num_coeffs; i++)
-            sce->prcoeffs[start_coef + i] += QERR[i] != 0.0f ? (sce->prcoeffs[start_coef + i] - QERR[i]) : 0.0f;
-        abs_pow34_v(P34, &sce->prcoeffs[start_coef], num_coeffs);
-        if (cb_n < RESERVED_BT)
-            cb_p = find_min_book(find_max_val(1, num_coeffs, P34), sce->sf_idx[sfb]);
-        else
-            cb_p = cb_n;
-        dist2 = quantize_and_encode_band_cost(s, NULL, &sce->prcoeffs[start_coef], NULL,
-                                              P34, num_coeffs, sce->sf_idx[sfb],
-                                              cb_p, s->lambda / band->threshold, INFINITY, NULL, 0);
-        for (i = 0; i < num_coeffs; i++)
-            dist_spec_err += (O34[i] - P34[i])*(O34[i] - P34[i]);
-        dist_spec_err *= s->lambda / band->threshold;
-        dist2 += dist_spec_err;
-
-        if (dist2 <= dist1 && cb_p <= cb_n) {
-            cost_pred += cost2;
-            sce->ics.prediction_used[sfb] = 1;
-            sce->band_alt[sfb]  = cb_n;
-            sce->band_type[sfb] = cb_p;
-            count++;
-        } else {
-            cost_pred += cost1;
-            sce->band_alt[sfb] = cb_p;
-        }
-    }
-
-    if (count && cost_coeffs < cost_pred) {
-        count = 0;
-        for (sfb = PRED_SFB_START; sfb < pmax; sfb++)
-            RESTORE_PRED(sce, sfb);
-        memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
-    }
-
-    sce->ics.predictor_present = !!count;
-}
-
-/**
- * Encoder predictors data.
- */
-void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce)
-{
-    int sfb;
-    IndividualChannelStream *ics = &sce->ics;
-    const int pmax = FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-
-    if (!ics->predictor_present)
-        return;
-
-    put_bits(&s->pb, 1, !!ics->predictor_reset_group);
-    if (ics->predictor_reset_group)
-        put_bits(&s->pb, 5, ics->predictor_reset_group);
-    for (sfb = 0; sfb < pmax; sfb++)
-        put_bits(&s->pb, 1, ics->prediction_used[sfb]);
-}

libavcodec/aacenc_pred.h

@@ -1,47 +0,0 @@
-/*
- * AAC encoder main-type prediction
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * 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
- * AAC encoder main prediction
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_PRED_H
-#define AVCODEC_AACENC_PRED_H
-
-#include "aacenc.h"
-
-/* Every predictor group needs to get reset at least once in this many frames */
-#define PRED_RESET_FRAME_MIN 240
-
-/* Any frame with less than this amount of frames since last reset is ok */
-#define PRED_RESET_MIN 64
-
-/* Raise to filter any low frequency artifacts due to prediction */
-#define PRED_SFB_START 10
-
-void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce);
-void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe);
-void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce);
-void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce);
-
-#endif /* AVCODEC_AACENC_PRED_H */