Update for 2.5.11

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
MAINTAINERS: remove unmaintained releases
2016-02-01 20:44:56 +01:00 · 2016-01-31 00:45:17 +01:00 · 2016-01-31 00:24:52 +01:00 · 2016-01-31 00:24:52 +01:00 · 2016-01-31 00:24:52 +01:00 · 2016-01-31 00:24:52 +01:00
4952 changed files with 116933 additions and 428970 deletions
--- a/.gitignore
+++ b/.gitignore
@ -1,6 +1,5 @@
 *.a
 *.o
 *.o.*
 *.d
 *.def
 *.dll
@ -18,9 +17,9 @@
 *.so.*
 *.swp
 *.ver
 *-example
 *-test
 *_g
 \#*
 .\#*
 /.config
 /.version
 /ffmpeg
@ -29,7 +28,62 @@
 /ffserver
 /config.*
 /coverage.info
-/avversion.h
+/doc/*.1
 /doc/*.3
 /doc/*.html
 /doc/*.pod
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
 /doc/examples/extract_mvs
 /doc/examples/filter_audio
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
 /doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
 /doc/examples/transcode_aac
 /doc/examples/transcoding
 /doc/fate.txt
 /doc/print_options
 /lcov/
-/src
+/libavcodec/*_tablegen
-/mapfile
+/libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
 /tests/pixfmts.mak
 /tests/rotozoom
 /tests/test_copy.ffmeta
 /tests/tiny_psnr
 /tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
 /tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape
 /tools/ffeval
 /tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
 /tools/trasher
 /tools/seek_print
 /tools/uncoded_frame
 /tools/zmqsend
--- a/.travis.yml
+++ b/.travis.yml
@ -1,26 +0,0 @@
 language: c
 sudo: false
 os:
  - linux
  - osx
 addons:
  apt:
    packages:
      - yasm
      - diffutils
 compiler:
  - clang
  - gcc
 cache:
  directories:
    - ffmpeg-samples
 before_install:
  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
 install:
  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install yasm; fi
 script:
  - mkdir -p ffmpeg-samples
  - ./configure --samples=ffmpeg-samples --cc=$CC
  - make -j 8
  - make fate-rsync
  - make check -j 8
--- a/753
+++ b/753
@ -1,241 +1,536 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
-version <next>:
+version 2.5.11
- libopenmpt demuxer
+- MAINTAINERS: remove unmaintained releases
- tee protocol
+- avcodec/jpeg2000dec: More completely check cdef
- Changed metadata print option to accept general urls
+- avutil/opt: check for and handle errors in av_opt_set_dict2()
- Alias muxer for Ogg Video (.ogv)
+- avcodec/flacenc: fix calculation of bits required in case of custom sample rate
- VP8 in Ogg muxing
+- avformat: Document urls a bit
- curves filter doesn't automatically insert points at x=0 and x=1 anymore
+- avformat/libquvi: Set default demuxer and protocol limitations
- 16-bit support in curves filter
+- avformat/concat: Check protocol prefix
- 16-bit support in selectivecolor filter
+- doc/demuxers: Document enable_drefs and use_absolute_path
- OpenH264 decoder wrapper
+- avcodec/mjpegdec: Check for end for both bytes in unescaping
- MediaCodec hwaccel
+- avcodec/mpegvideo_enc: Check for integer overflow in ff_mpv_reallocate_putbitbuffer()
 - avformat/avformat: Replace some references to filenames by urls
 - avcodec/wmaenc: Check ff_wma_init() for failure
 - avcodec/mpeg12enc: Move high resolution thread check to before initializing threads
 - avformat/avio: Limit url option parsing to the documented cases
 - avcodec/ass_split: Fix null pointer dereference in ff_ass_style_get()
 - mov: Add an option to toggle dref opening
 - avcodec/gif: Fix lzw buffer size
 - avcodec/put_bits: Assert buf_ptr in flush_put_bits()
 - avcodec/tiff: Check subsample & rps values more completely
 - swscale/swscale: Add some sanity checks for srcSlice* parameters
 - swscale/x86/rgb2rgb_template: Fix planar2x() for short width
 - swscale/swscale_unscaled: Fix odd height inputs for bayer_to_yv12_wrapper()
 - swscale/swscale_unscaled: Fix odd height inputs for bayer_to_rgb24_wrapper()
 - avcodec/aacenc: Check both channels for finiteness
 - swscale/swscale-test: Fix slice height in random reference data creation.
 - dca: fix misaligned access in avpriv_dca_convert_bitstream
 - brstm: fix missing closing brace
 - brstm: also allocate b->table in read_packet
 - brstm: make sure an ADPC chunk was read for adpcm_thp
 - vorbisdec: reject rangebits 0 with non-0 partitions
 - vorbisdec: reject channel mapping with less than two channels
 - ffmdec: reset packet_end in case of failure
 - avformat/ipmovie: put video decoding_map_size into packet and use it in decoder
 - avcodec/samidec: make sure to properly restore parsing context after a tag
 version 2.5.10
 - configure: bump copyright year to 2016
 - avformat/hls: Even stricter URL checks
 - avformat/hls: More strict url checks
 - swscale/utils: Detect and skip unneeded sws_setColorspaceDetails() calls
 - swscale/yuv2rgb: Increase YUV2RGB table headroom
 - swscale/yuv2rgb: Factor YUVRGB_TABLE_LUMA_HEADROOM out
 - avformat/hls: forbid all protocols except http(s) & file
 - avformat/aviobuf: Fix end check in put_str16()
 - avformat/asfenc: Check pts
 - avcodec/mpeg4video: Check time_incr
 - avcodec/wavpackenc: Check the number of channels
 - avcodec/wavpackenc: Headers are per channel
 - avcodec/dvdec: Fix "left shift of negative value -254"
 - avcodec/mjpegdec: Fix negative shift
 - avcodec/mss2: Check for repeat overflow
 - avformat: Add integer fps from 31 to 60 to get_std_framerate()
 - avcodec/mpegvideo_enc: Clip bits_per_raw_sample within valid range
 - avfilter/vf_scale: set proper out frame color range
 - avcodec/motion_est: Fix mv_penalty table size
 - avcodec/h264_slice: Fix integer overflow in implicit weight computation
 - swscale/utils: Use normal bilinear scaler if fast cannot be used due to tiny dimensions
 - avcodec/put_bits: Always check buffer end before writing
 - mjpegdec: extend check for incompatible values of s->rgb and s->ls
 - swscale/utils: Fix intermediate format for cascaded alpha downscaling
 - avcodec/h264_refs: Fix long_idx check
 - avfilter/vf_mpdecimate: Add missing emms_c()
 - avformat/mxfenc: Do not crash if there is no packet in the first stream
 - avformat/utils: estimate_timings_from_pts - increase retry counter, fixes invalid duration for ts files with hevc codec
 - avformat/matroskaenc: Check codecdelay before use
 - avutil/mathematics: Fix division by 0
 - x86/float_dsp: zero extend offset from ff_scalarproduct_float_sse
 - avcodec/mpeg4videodec: also for empty partitioned slices
 - nuv: sanitize negative fps rate
 - rawdec: only exempt BIT0 with need_copy from buffer sanity check
 - mlvdec: check that index_entries exist
 - nutdec: reject negative value_len in read_sm_data
 - xwddec: prevent overflow of lsize * avctx->height
 - nutdec: only copy the header if it exists
 - exr: fix out of bounds read in get_code
 - on2avc: limit number of bits to 30 in get_egolomb
 - sonic: make sure num_taps * channels is not larger than frame_size
 - opus_silk: fix typo causing overflow in silk_stabilize_lsf
 - ffm: reject invalid codec_id and codec_type
 - aaccoder: prevent crash of anmr coder
 - ffmdec: reject zero-sized chunks
 - swscale/x86/rgb2rgb_template: Fallback to mmx in interleaveBytes() if the alignment is insufficient for SSE*
 - swscale/x86/rgb2rgb_template: Do not crash on misaligend stride
-version 3.1:
+version 2.5.9
- DXVA2-accelerated HEVC Main10 decoding
+- avcodec/hevc: Check max ctb addresses for WPP
- fieldhint filter
+- avcodec/vp3: ensure header is parsed successfully before tables
- loop video filter and aloop audio filter
+- avcodec/jpeg2000dec: Check bpno in decode_cblk()
- Bob Weaver deinterlacing filter
+- avcodec/pgssubdec: Fix left shift of 255 by 24 places cannot be represented in type int
- firequalizer filter
+- swscale/utils: Fix for runtime error: left shift of negative value -1
- datascope filter
+- avcodec/hevc: Fix integer overflow of entry_point_offset
- bench and abench filters
+- avcodec/dirac_parser: Check that there is a previous PU before accessing it
- ciescope filter
+- avcodec/dirac_parser: Add basic validity checks for next_pu_offset and prev_pu_offset
- protocol blacklisting API
+- avcodec/dirac_parser: Fix potential overflows in pointer checks
- MediaCodec H264 decoding
+- avcodec/wmaprodec: Check bits per sample to be within the range not causing integer overflows
- VC-2 HQ RTP payload format (draft v1) depacketizer and packetizer
+- avcodec/wmaprodec: Fix overflow of cutoff
- VP9 RTP payload format (draft v2) packetizer
+- avformat/smacker: fix integer overflow with pts_inc
- AudioToolbox audio decoders
+- avcodec/vp3: Fix "runtime error: left shift of negative value"
- AudioToolbox audio encoders
+- mpegencts: Fix overflow in cbr mode period calculations
- coreimage filter (GPU based image filtering on OSX)
+- avutil/timecode: Fix fps check
- libdcadec removed
+- avutil/mathematics: return INT64_MIN (=AV_NOPTS_VALUE) from av_rescale_rnd() for overflows
- bitstream filter for extracting DTS core
+- avcodec/apedec: Check length in long_filter_high_3800()
- ADPCM IMA DAT4 decoder
+- avcodec/vp3: always set pix_fmt in theora_decode_header()
- musx demuxer
+- avcodec/mpeg4videodec: Check available data before reading custom matrix
- aix demuxer
+- avutil/mathematics: Do not treat INT64_MIN as positive in av_rescale_rnd
- remap filter
+- avutil/integer: Fix av_mod_i() with negative dividend
- hash and framehash muxers
+- avformat/dump: Fix integer overflow in av_dump_format()
- colorspace filter
+- avcodec/utils: Clear dimensions in ff_get_buffer() on failure
- hdcd filter
+- avcodec/utils: Use 64bit for aspect ratio calculation in avcodec_string()
- readvitc filter
+- avcodec/vp3: Clear context on reinitialization failure
- VAAPI-accelerated format conversion and scaling
+- avcodec/hevc: allocate entries unconditionally
- libnpp/CUDA-accelerated format conversion and scaling
+- avcodec/hevc_cabac: Fix multiple integer overflows
- Duck TrueMotion 2.0 Real Time decoder
+- avcodec/jpeg2000dwt: Check ndeclevels before calling dwt_encode*()
- Wideband Single-bit Data (WSD) demuxer
+- avcodec/jpeg2000dwt: Check ndeclevels before calling dwt_decode*()
- VAAPI-accelerated H.264/HEVC/MJPEG encoding
+- avcodec/hevc: Check entry_point_offsets
- DTS Express (LBR) decoder
+- avcodec/cabac: Check initial cabac decoder state
- Generic OpenMAX IL encoder with support for Raspberry Pi
+- avcodec/cabac_functions: Fix "left shift of negative value -31767"
- IFF ANIM demuxer & decoder
+- avcodec/ffv1dec: Clear quant_table_count if its invalid
- Direct Stream Transfer (DST) decoder
+- avcodec/ffv1dec: Print an error if the quant table count is invalid
- loudnorm filter
+- doc/filters/drawtext: fix centering example
- MTAF demuxer and decoder
+- avcodec/h264_slice: Limit max_contexts when slice_context_count is initialized
- MagicYUV decoder
+- avcodec/vp8: Do not use num_coeff_partitions in thread/buffer setup
- OpenExr improvements (tile data and B44/B44A support)
+- rtmpcrypt: Do the xtea decryption in little endian mode
- BitJazz SheerVideo decoder
+- avformat/matroskadec: Check subtitle stream before dereferencing
- CUDA CUVID H264/HEVC decoder
+- avformat/utils: Do not init parser if probing is unfinished
- 10-bit depth support in native utvideo decoder
+- avcodec/jpeg2000dec: Fix potential integer overflow with tile dimensions
- libutvideo wrapper removed
+- avcodec/jpeg2000dec: Check SIZ dimensions to be within the supported range
- YUY2 Lossless Codec decoder
+- avcodec/jpeg2000: Check comp coords to be within the supported size
- VideoToolbox H.264 encoder
+- avcodec/jpeg2000: Use av_image_check_size() in ff_jpeg2000_init_component()
 - avcodec/wmaprodec: Check for overread in decode_packet()
 - avcodec/smacker: Check that the data size is a multiple of a sample vector
 - avcodec/takdec: Skip last p2 sample (which is unused)
 - avcodec/dxtory: Fix input size check in dxtory_decode_v1_410()
 - avcodec/dxtory: Fix input size check in dxtory_decode_v1_420()
 - avcodec/error_resilience: avoid accessing previous or next frames tables beyond height
 - avcodec/dpx: Move need_align to act per line
 - avcodec/flashsv: Check size before updating it
 - avcodec/ivi: Check image dimensions
 - avcodec/utils: Better check for channels in av_get_audio_frame_duration()
 - avcodec/jpeg2000dec: Check for duplicate SIZ marker
 - avcodec/jpeg2000dec: Clip all tile coordinates
 - avcodec/microdvddec: Check for string end in 'P' case
 - avcodec/dirac_parser: Fix undefined memcpy() use
 - avformat/xmv: Discard remainder of packet on error
 - avformat/xmv: factor return check out of if/else
 - libavutil/channel_layout: Check strtol*() for failure
 - avcodec/ffv1dec: Check for 0 quant tables
 - avcodec/mjpegdec: Reinitialize IDCT on BPP changes
 - avcodec/mjpegdec: Check index in ljpeg_decode_yuv_scan() before using it
 - avutil/file_open: avoid file handle inheritance on Windows
 - opusdec: Don't run vector_fmul_scalar on zero length arrays
 - avcodec/ffv1: Initialize vlc_state on allocation
 - avcodec/ffv1dec: update progress in case of broken pointer chains
 - avcodec/ffv1dec: Clear slice coordinates if they are invalid or slice header decoding fails for other reasons
 - avformat/httpauth: Add space after commas in HTTP/RTSP auth header
 - avcodec/x86/sbrdsp: Fix using uninitialized upper 32bit of noise
 - avcodec/ffv1dec: Fix off by 1 error in quant_table_count check
 - avcodec/ffv1dec: Explicitly check read_quant_table() return value
 - avcodec/rangecoder: Check e
 - lavf/webvttenc: Require webvtt file to contain exactly one WebVTT stream.
 - avcodec/mjpegdec: Fix decoding RGBA RCT LJPEG
 - avfilter/af_asyncts: use llabs for int64_t
 - avcodec/g2meet: Also clear tile dimensions on header_fail
 - avcodec/g2meet: Fix potential overflow in tile dimensions check
 - avcodec/svq1dec: Check init_get_bits8() for failure
 - avcodec/tta: Check init_get_bits8() for failure
 - swresample/swresample: Fix integer overflow in seed calculation
 - avformat/mov: Fix integer overflow in FFABS
 - avutil/common: Add FFNABS()
 - avutil/common: Document FFABS() corner case
 - avformat/dump: Fix integer overflow in aspect ratio calculation
 - avcodec/truemotion1: Check for even width
 - avcodec/mpeg12dec: Set dimensions in mpeg1_decode_sequence() only in absence of errors
 - avcodec/libopusenc: Fix infinite loop on flushing after 0 input
 - avformat/hevc: Check num_long_term_ref_pics_sps to avoid potentially long loops
 - avformat/hevc: Fix parsing errors
 - ffmpeg: Use correct codec_id for av_parser_change() check
 - ffmpeg: Check av_parser_change() for failure
 - ffmpeg: Check for RAWVIDEO and do not relay only on AVFMT_RAWPICTURE
 - ffmpeg: check avpicture_fill() return value
 - avformat/mux: Update sidedata in ff_write_chained()
 - avcodec/flashsvenc: Correct max dimension in error message
 - avcodec/svq1enc: Check dimensions
 - avcodec/dcaenc: clear bitstream end
 - libavcodec/aacdec_template: Use init_get_bits8() in aac_decode_frame()
 - mxfdec: check edit_rate also for physical_track
 - mpegvideo: clear overread in clear_context
 - dvdsubdec: validate offset2 similar to offset1
 - avcodec/takdec: Use memove, avoid undefined memcpy() use
 - jvdec: avoid unsigned overflow in comparison
 - avcodec/mpeg12dec: Do not call show_bits() with invalid bits
 - riffdec: prevent negative bit rate
 - Merge commit 'd80811c94e068085aab797f9ba35790529126f85'
 - imc: use correct position for flcoeffs2 calculation
 - wavpack: limit extra_bits to 32 and use get_bits_long
 - wavpack: use get_bits_long to read up to 32 bits
 - nutdec: check maxpos in read_sm_data before returning success
 - s302m: fix arithmetic exception
 - avcodec/s302m: Only set the sample rate when some data is output
 - vp9: add support for resolution changes in inter frames.
 - alsdec: limit avctx->bits_per_raw_sample to 32
 - vp9: avoid infinite loop with broken files
 - videodsp: don't overread edges in vfix3 emu_edge.
 - avcodec/h264_mp4toannexb_bsf: Reorder operations in nal_size check
 - avformat/oggenc: Check segments_count for headers too
 - avformat/avidec: Workaround broken initial frame
 - hevc: properly handle no_rasl_output_flag when removing pictures from the DPB
 - hevc: fix wpp threading deadlock.
 - avcodec/ffv1: separate slice_count from max_slice_count
 - lavf/img2dec: Fix memory leak
 - avcodec/mp3: fix skipping zeros
 - avformat/srtdec: make sure we probe a number
 - avformat/srtdec: more lenient first line probing
 - doc: mention libavcodec can decode Opus natively
 - MAINTAINERS: Remove myself as leader
-version 3.0:
+version 2.5.8
- Common Encryption (CENC) MP4 encoding and decoding support
+- snow: remove an obsolete av_assert2
- DXV decoding
+- huffyuvdec: validate image size
- extrastereo filter
+- vc1dec: use get_bits_long and limit the read bits to 32
- ocr filter
+- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
- alimiter filter
+- libshine: fix support for shine 3.0
- stereowiden filter
+- avidec: check for valid bit_rate range
- stereotools filter
+- avformat/nut: support WavPack
- rubberband filter
+- avcodec/diracdec: Check slices malloc and propagate error code
- tremolo filter
+- avcodec/vp8: Check buffer size in vp8_decode_frame_header()
- agate filter
+- avcodec/vp8: Fix null pointer dereference in ff_vp8_decode_free()
- chromakey filter
+- avcodec/diracdec: Check for hpel_base allocation failure
- maskedmerge filter
+- avcodec/rv34: Clear pointers in ff_rv34_decode_init_thread_copy()
- Screenpresso SPV1 decoding
+- avfilter/af_aresample: Check ff_all_* for allocation failures
- chromaprint fingerprinting muxer
+- avcodec/pthread_frame: clear priv_data, avoid stale pointer in error case
- ffplay dynamic volume control
+- swscale/utils: Clear pix buffers
- displace filter
+- avutil/fifo: Fix the case where func() returns less bytes than requested in av_fifo_generic_write()
- selectivecolor filter
+- avformat/mov: Fix deallocation when MOVStreamContext failed to allocate
- extensive native AAC encoder improvements and removal of experimental flag
+- ffmpeg: Fix crash with ost->last_frame allocation failure
- ADPCM PSX decoder
+- ffmpeg: Fix cleanup with ost = NULL
- 3dostr, dcstr, fsb, genh, vag, xvag, ads, msf, svag & vpk demuxer
+- avcodec/pthread_frame: check avctx on deallocation
- zscale filter
+- avcodec/sanm: Reset sizes in destroy_buffers()
- wve demuxer
+- avcodec/alac: Clear pointers in allocate_buffers()
- zero-copy Intel QSV transcoding in ffmpeg
+- bytestream2: set the reader to the end when reading more than available
- shuffleframes filter
+- avcodec/utils: use a minimum 32pixel width in  avcodec_align_dimensions2() for H.264
- SDX2 DPCM decoder
+- avcodec/mpegvideo: Clear pointers in ff_mpv_common_init()
- vibrato filter
+- oggparsedirac: check return value of init_get_bits
- innoHeim/Rsupport Screen Capture Codec decoder
+- wmalosslessdec: reset frame->nb_samples on packet loss
- ADPCM AICA decoder
+- wmalosslessdec: avoid reading 0 bits with get_bits
- Interplay ACM demuxer and audio decoder
+- avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2()
- XMA1 & XMA2 decoder
+- avcodec/aacsbr: Assert that bs_num_env is positive
- realtime filter
+- avcodec/aacsbr: check that the element type matches before applying SBR
- anoisesrc audio filter source
+- avcodec/h264_slice: Use w/h from the AVFrame instead of mb_w/h
- IVR demuxer
+- vp9/update_prob: prevent out of bounds table read
- compensationdelay filter
+- avfilter/vf_transpose: Fix rounding error
- acompressor filter
+- avcodec/pngdec: Check values before updating context in decode_fctl_chunk()
- support encoding 16-bit RLE SGI images
+- avcodec/pngdec: Require a IHDR chunk before fctl
- apulsator filter
+- avcodec/pngdec: Only allow one IHDR chunk
- sidechaingate audio filter
+- wmavoice: limit wmavoice_decode_packet return value to packet size
- mipsdspr1 option has been renamed to mipsdsp
+- swscale/swscale_unscaled: Fix rounding difference with RGBA output between little and big endian
- aemphasis filter
+- ffmpeg: Do not use the data/size of a bitstream filter after failure
- mips32r5 option has been removed
+- swscale/x86/rgb2rgb_template: fix signedness of v in shuffle_bytes_2103_{mmx,mmxext}
- mips64r6 option has been removed
+- swscale/x86/rgb2rgb_template: add missing xmm clobbers
- DXVA2-accelerated VP9 decoding
+- vda: unlock the pixel buffer base address.
- SOFAlizer: virtual binaural acoustics filter
+- swscale/rgb2rgb_template: Fix signedness of v in shuffle_bytes_2103_c()
- VAAPI VP9 hwaccel
+- swscale/rgb2rgb_template: Implement shuffle_bytes_0321_c and fix shuffle_bytes_2103_c on BE
- audio high-order multiband parametric equalizer
+- swscale/rgb2rgb_template: Disable shuffle_bytes_2103_c on big endian
- automatic bitstream filtering
+- swr: Remember previously set int_sample_format from user
- showspectrumpic filter
+- matroskadec: check audio sample rate
- libstagefright support removed
+- matroskadec: validate audio channels and bitdepth
- spectrumsynth filter
+- avcodec/dpxenc: implement write16/32 as functions
- ahistogram filter
+- postproc: fix unaligned access
- only seek with the right mouse button in ffplay
+- ffmpeg: Free last_frame instead of just unref
- toggle full screen when double-clicking with the left mouse button in ffplay
+- avio: fix potential crashes when combining ffio_ensure_seekback + crc
- afftfilt filter
+- h264: er: Copy from the previous reference only if compatible
- convolution filter
+- sonic: set avctx->channels in sonic_decode_init
- libquvi support removed
+- vp8: change mv_{min,max}.{x,y} type to int
- support for dvaudio in wav and avi
+- vp9: change type of tile_size from unsigned to int64_t
- libaacplus and libvo-aacenc support removed
+- arm: only enable setend on ARMv6
- Cineform HD decoder
+- libopenjpegdec: check existence of image component data
- new DCA decoder with full support for DTS-HD extensions
+- mov: abort on EOF in ff_mov_read_chan
- significant performance improvements in Windows Television (WTV) demuxer
+- ffmpeg_opt: Check for localtime() failure
- nnedi deinterlacer
+- avformat: Fix bug in parse_rps for HEVC.
- streamselect video and astreamselect audio filter
+- takdec: ensure chan2 is a valid channel index
- swaprect filter
+- avcodec/h264_slice: Use AVFrame diemensions for grayscale handling
- metadata video and ametadata audio filter
+- avdevice/lavfi: do not rescale AV_NOPTS_VALUE in lavfi_read_packet()
- SMPTE VC-2 HQ profile support for the Dirac decoder
+- libavutil/channel_layout: Correctly return layout when channel specification ends with a trailing 'c'.
- SMPTE VC-2 native encoder supporting the HQ profile
+- avcodec/jpeg2000dec: Check that coords match before applying ICT
 - avformat/ffmdec: Check ffio_set_buf_size() return value
 - avcodec/adpcm: Check for overreads
 - avcodec/alsdec: Check for overread
 - avcodec/atrac3plusdec: consume only as many bytes as available
 - libavutil/softfloat: Fix av_normalize1_sf bias.
 - swresample/swresample: Cleanup on init failure.
 - Revert "avformat/rtpenc: check av_packet_get_side_data() return, fix null ptr dereference"
 - avformat/mxfenc: Accept MXF D-10 with 49.999840 Mbit/sec
 - swresample/dither: check memory allocation
 - libopenjpegenc: add NULL check for img before accessing it
 - swresample: Check the return value of resampler->init()
 - h264: Make sure reinit failures mark the context as not initialized
 - ffmpeg_opt: Set the video VBV parameters only for the video stream from -target
 - avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits()
 - avcodec/put_bits: Assert that there is enough space left in skip_put_bytes()
 - avcodec/mpegvideo_enc: Update the buffer size as more slices are merged
 - avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size()
 - avformat/wavdec: Increase dts packet threshold to fix more misdetections
 - avformat/wavdec: Increase probe_packets limit
 - avformat/swfdec: Do not error out on pixel format changes
 - avfilter/x86/vf_hqdn3d: Fix register types
 - avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer()
 - avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out
 - avformat/mov: Mark avio context of decompressed atoms as seekable
 - avcodec/hevc_ps: Only discard overread VPS if a previous is available
 - avcodec/x86/h264_weight: handle weight1=128
 - avcodec/exr: fix crash caused by merge
-version 2.8:
+version 2.5.7
- colorkey video filter
+- avformat/nutdec: Fix recovery when immedeately after seeking a failure happens
- BFSTM/BCSTM demuxer
+- nutdec: fix memleaks on error in nut_read_header
- little-endian ADPCM_THP decoder
+- rtpenc_jpeg: handle case of picture dimensions not dividing by 8
- Hap decoder and encoder
+- avformat/mov: Fix parsing short loci
- DirectDraw Surface image/texture decoder
+- avcodec/shorten: Fix code depending on signed overflow behavior
- ssim filter
+- avcodec/proresdec2: Reset slice_count on deallocation
- optional new ASF demuxer
+- ffmpeg_opt: Fix -timestamp parsing
- showvolume filter
+- hevc: make avcodec_decode_video2() fail if get_format() fails
- Many improvements to the JPEG 2000 decoder
+- avcodec/mpeg4audio: add some padding/alignment to MAX_PCE_SIZE
- Go2Meeting decoding support
+- swr: fix alignment issue caused by 8ch sse functions
- adrawgraph audio and drawgraph video filter
+- libswscale/x86/hscale_fast_bilinear_simd.c: Include BX in the clobber list on x86_64, because it isn't implicitly included when PIC is on.
- removegrain video filter
+- aacdec: don't return frames without data
- Intel QSV-accelerated MPEG-2 video and HEVC encoding
+- avformat/matroskadec: Cleanup error handling for bz2 & zlib
- Intel QSV-accelerated MPEG-2 video and HEVC decoding
+- avformat/nutdec: Fix use of uinitialized value
- Intel QSV-accelerated VC-1 video decoding
+- tools/graph2dot: use larger data types than int for array/string sizes
- libkvazaar HEVC encoder
+- id3v2: catch avio_read errors in check_tag
- erosion, dilation, deflate and inflate video filters
+- aacsbr: break infinite loop in sbr_hf_calc_npatches
- Dynamic Audio Normalizer as dynaudnorm filter
+- diracdec: avoid overflow of bytes*8 in decode_lowdelay
- Reverse video and areverse audio filter
+- diracdec: prevent overflow in data_unit_size check
- Random filter
+- avidec: avoid infinite loop due to negative ast->sample_size
- deband filter
+- pngdec: don't use AV_PIX_FMT_MONOBLACK for apng
- AAC fixed-point decoding
+- avcodec/wavpack: Check L/R values before use to avoid harmless integer overflow and undefined behavior in fate
- sidechaincompress audio filter
+- xcbgrab: Validate the capture area
- bitstream filter for converting HEVC from MP4 to Annex B
+- xcbgrab: Do not assume the non shm image data is always available
- acrossfade audio filter
+- avfilter/lavfutils: disable frame threads when decoding a single image
- allyuv and allrgb video sources
+- nutdec: fix illegal count check in decode_main_header
- atadenoise video filter
+- ffmpeg: remove incorrect network deinit
- OS X VideoToolbox support
+- OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c
- aphasemeter filter
+- apedec: set s->samples only when init_frame_decoder succeeded
- showfreqs filter
+- swscale/ppc/swscale_altivec.c: POWER LE support in yuv2planeX_8() delete macro GET_VF()
- vectorscope filter
+- libvpxenc: only set noise reduction w/vp8
- waveform filter
+- tests/fate-run: do not attempt to parse tiny_psnrs output if it failed
- hstack and vstack filter
+- alac: reject rice_limit 0 if compression is used
- Support DNx100 (1440x1080@8)
+- alsdec: only adapt order for positive max_order
- VAAPI hevc hwaccel
+- alsdec: check sample pointer range in revert_channel_correlation
- VDPAU hevc hwaccel
+- tests: drop bc dependency
- framerate filter
+- fate: Include branch information in the payload header
 - Switched default encoders for webm to VP9 and Opus
 - Removed experimental flag from the JPEG 2000 encoder
-version 2.7:
+version 2.5.6
- FFT video filter
+- avcodec/atrac3plusdsp: fix on stack alignment
- TDSC decoder
+- ac3: validate end in ff_ac3_bit_alloc_calc_mask
- DTS lossless extension (XLL) decoding (not lossless, disabled by default)
+- aacpsy: avoid psy_band->threshold becoming NaN
- showwavespic filter
+- aasc: return correct buffer size from aasc_decode_frame
- DTS decoding through libdcadec
+- msrledec: use signed pixel_ptr in msrle_decode_pal4
- Drop support for nvenc API before 5.0
+- swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07)
- nvenc HEVC encoder
+- swresample: Allow reinitialization without ever setting channel counts
- Detelecine filter
+- avcodec/h264: Do not fail with randomly truncated VUIs
- Intel QSV-accelerated H.264 encoding
+- avcodec/h264_ps: Move truncation check from VUI to SPS
- MMAL-accelerated H.264 decoding
+- avcodec/h264: Be more tolerant to changing pps id between slices
- basic APNG encoder and muxer with default extension "apng"
+- avcodec/aacdec: Fix storing state before PCE decode
- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
+- avcodec/h264: reset the counts in the correct context
- WebM Live Chunk Muxer
+- avcodec/h264_slice: Do not reset mb_aff_frame per slice
- nvenc level and tier options
+- avcodec/h264: finish previous slices before switching to single thread mode
- chorus filter
+- avcodec/h264: Fix race between slices where one overwrites data from the next
- Canopus HQ/HQA decoder
+- avcodec/h264_refs: Do not set reference to things which do not exist
- Automatically rotate videos based on metadata in ffmpeg
+- avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode
- improved Quickdraw compatibility
+- h264: avoid unnecessary calls to get_format
- VP9 high bit-depth and extended colorspaces decoding support
+- avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
 - WebPAnimEncoder API when available for encoding and muxing WebP
 - Direct3D11-accelerated decoding
 - Support Secure Transport
 - Multipart JPEG demuxer
-version 2.6:
+version 2.5.5:
- nvenc encoder
+- vp9: make above buffer pointer 32-byte aligned.
- 10bit spp filter
+- avcodec/dnxhddec: Check that the frame is interlaced before using cur_field
- colorlevels filter
+- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
- RIFX format for *.wav files
+- avformat/mov: Check for string truncation in mov_open_dref()
- RTP/mpegts muxer
+- avformat/mov: Use sizeof(filename) instead of a literal number
- non continuous cache protocol support
+- eac3dec: fix scaling
- tblend filter
+- ac3_fixed: fix computation of spx_noise_blend
- cropdetect support for non 8bpp, absolute (if limit >= 1) and relative (if limit < 1.0) threshold
+- ac3_fixed: fix out-of-bound read
- Camellia symmetric block cipher
+- ac3dec_fixed: always use the USE_FIXED=1 variant of the AC3DecodeContext
- OpenH264 encoder wrapper
+- avcodec/012v: redesign main loop
- VOC seeking support
+- avcodec/012v: Check dimensions more completely
- Closed caption Decoder
+- asfenc: fix leaking asf->index_ptr on error
- fspp, uspp, pp7 MPlayer postprocessing filters ported to native filters
+- avcodec/options_table: remove extradata_size from the AVOptions table
- showpalette filter
+- ffmdec: limit the backward seek to the last resync position
- Twofish symmetric block cipher
+- ffmdec: make sure the time base is valid
- Support DNx100 (960x720@8)
+- ffmdec: fix infinite loop at EOF
- eq2 filter ported from libmpcodecs as eq filter
+- ffmdec: initialize f_cprv, f_stvi and f_stau
- removed libmpcodecs
+- avformat/rm: limit packet size
- Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
+- avcodec/webp: validate the distance prefix code
- ported softpulldown filter from libmpcodecs as repeatfields filter
+- avcodec/rv10: check size of s->mb_width * s->mb_height
- dcshift filter
+- eamad: check for out of bounds read
- RTP depacketizer for loss tolerant payload format for MP3 audio (RFC 5219)
+- mdec: check for out of bounds read
- RTP depacketizer for AC3 payload format (RFC 4184)
+- arm: Suppress tags about used cpu arch and extensions
- palettegen and paletteuse filters
+- aic: Fix decoding files with odd dimensions
- VP9 RTP payload format (draft 0) experimental depacketizer
+- avcodec/tiff: move bpp check to after "end:"
- RTP depacketizer for DV (RFC 6469)
+- mxfdec: Fix the error handling for when strftime fails
- DXVA2-accelerated HEVC decoding
+- avcodec/opusdec: Fix delayed sample value
- AAC ELD 480 decoding
+- avcodec/opusdec: Clear out pointers per packet
- Intel QSV-accelerated H.264 decoding
+- avcodec/utils: Align YUV411 by as much as the other YUV variants
- DSS SP decoder and DSS demuxer
+- vp9: fix segmentation map retention with threading enabled.
- Fix stsd atom corruption in DNxHD QuickTimes
+- webp: ensure that each transform is only used once
- Canopus HQX decoder
+- doc/protocols/tcp: fix units of listen_timeout option value, from microseconds to milliseconds
- RTP depacketization of T.140 text (RFC 4103)
+- fix VP9 packet decoder returning 0 instead of the used data size
- Port MIPS optimizations to 64-bit
+- avformat/flvenc: check that the codec_tag fits in the available bits
 - avcodec/utils: use correct printf specifier in ff_set_sar
 - avutil/imgutils: correctly check for negative SAR components
 - swscale/utils: clear formatConvBuffer on allocation
 - avformat/bit: only accept the g729 codec and 1 channel
 - avformat/bit: check that pkt->size is 10 in write_packet
 - avformat/adxdec: check avctx->channels for invalid values
 - avformat/adxdec: set avctx->channels in adx_read_header
 - Fix buffer_size argument to init_put_bits() in multiple encoders.
 - mips/acelp_filters: fix incorrect register constraint
 - avcodec/hevc_ps: Sanity checks for some log2_* values
 - avcodec/zmbv: Check len before reading in decode_frame()
 - avcodec/h264: Only reinit quant tables if a new PPS is allowed
 - avcodec/snowdec: Fix ref value check
 - swscale/utils: More carefully merge and clear coefficients outside the input
 - avcodec/a64multienc: Assert that the Packet size does not grow
 - avcodec/a64multienc: simplify frame handling code
 - avcodec/a64multienc: fix use of uninitialized values in to_meta_with_crop
 - avcodec/a64multienc: initialize mc_meta_charset to zero
 - avcodec/a64multienc: don't set incorrect packet size
 - avcodec/a64multienc: use av_frame_ref instead of copying the frame
 - avcodec/x86/mlpdsp_init: Simplify mlp_filter_channel_x86()
 - h264: initialize H264Context.avctx in init_thread_copy
 - wtvdec: fix integer overflow resulting in errors with large files
 - avcodec/gif: fix off by one in column offsetting finding
 version 2.5.4:
 - avcodec/arm/videodsp_armv5te: Fix linking failure with shared libs
 - avcodec/mjpegdec: Skip blocks which are outside the visible area
 - avcodec/h264_slice: ignore SAR changes in slices after the first
 - avcodec/h264_slice: Check picture structure before setting the related fields
 - avcodec/h264_slice: Do not change frame_num after the first slice
 - avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT
 - avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT
 - avcodec/h264: Be more strict on rejecting pps/sps changes
 - avcodec/h264: Be more strict on rejecting pps_id changes
 - avcodec/h264_ps: More completely check the bit depths
 - avformat/thp: Check av_get_packet() for failure not only for partial output
 - swscale/utils: Limit filter shifting so as not to read from prior the array
 - avcodec/mpegvideo_motion: Fix gmc chroma dimensions
 - avcodec/mjpegdec: Check number of components for JPEG-LS
 - avcodec/mjpegdec: Check escape sequence validity
 - avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior
 - avformat/mpc8: fix broken pointer math
 - avformat/mpc8: fix hang with fuzzed file
 - avformat/tta: fix crash with corrupted files
 - avcodec/ppc/idctdsp.c: POWER LE support in idct_add_altivec()
 - swscale/input: fix rgba64 alpha non native
 - swscale/input: Fix alpha of YA16 input
 - libavcodec/ppc/mpegvideoencdsp.c: fix stack smashing in pix_norm1_altivec() and pix_sum_altivec()
 - avformat/rmdec: Check for overflow in ff_rm_read_mdpr_codecdata()
 - avformat/mpeg: do not count PES packets inside PES packets during probing
 - hevc: always clip luma_log2_weight_denom
 - rtpdec_h263_rfc2190: Clear the stored bits if discarding buffered data
 - aacenc: correctly check returned value
 - swscale: check memory allocations
 - opt: check memory allocation
 - avformat/utils: check for malloc failure
 - avcodec/flac_parser: fix handling EOF if no headers are found
 - avfilter/vf_framepack: Check and update frame_rate
 - vp8: improve memory allocation checks
 - configure: enable vsx together with altivec for ppc64el
 - avcodec/hevc: Fix handling of skipped_bytes() reallocation failures
 - qpeg: avoid pointless invalid memcpy()
 version 2.5.3:
 - vp9: fix parser return values in error case
 - ffmpeg: Clear error message array at init.
 - avcodec/dvdsubdec: fix accessing dangling pointers
 - avcodec/dvdsubdec: error on bitmaps with size 0
 - cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file()
 - mov: Fix negative size calculation in mov_read_default().
 - avformat/mov: fix integer overflow in mov_read_udta_string()
 - mov: Fix overflow and error handling in read_tfra().
 - mov: Avoid overflow with mov_metadata_raw()
 - avcodec/dvdsubdec: fix out of bounds accesses
 - avfilter/vf_sab: fix filtering tiny images
 - avformat/flvdec: Increase string array size
 - avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream
 - avformat/cdxl: Fix integer overflow of image_size
 - libavformat: Build hevc.o when building the RTP muxer
 version 2.5.2:
 - avcodec/indeo3: ensure offsets are non negative
 - avcodec/h264: Check *log2_weight_denom
 - avcodec/hevc_ps: Check diff_cu_qp_delta_depth
 - avcodec/h264: Clear delayed_pic on deallocation
 - avcodec/hevc: clear filter_slice_edges() on allocation
 - avcodec/dcadec: Check that the added xch channel isnt already there
 - avcodec/indeo3: use signed variables to avoid underflow
 - swscale: increase yuv2rgb table headroom
 - avformat/mov: fix integer overflow of size
 - avformat/mov: check atom nesting depth
 - avcodec/utvideodec: Fix handling of slice_height=0
 - avcodec/xface: correct the XFACE_MAX_* values
 - avcodec/vmdvideo: Check len before using it in method 3
 - configure: create the tests directory like the doc directory
 - mmvideo: check frame dimensions
 - jvdec: check frame dimensions
 version 2.5.1:
 - lavu/frame: fix malloc error path in av_frame_copy_props()
 - avformat/aviobuf: Check that avio_seek() target is non negative
 - swresample/soxr_resample: fix error handling
 - avformat/flvdec: fix potential use of uninitialized variables
 - avformat/crypto: fix key vs iv typo
 - configure: use use_pkg_config() instead of check_pkg_config() for libsmbclient
 - avcodec/ppc/vp3dsp_altivec: POWER LE support to vp3_idct_add_altivec()
 - avformat/matroskadec: fix handling of recursive SeekHead elements
 - doc/examples/filtering_video: fix frame rate
 - avcodec/mpegaudiodec_template: only allocate fdsp when its used
 - doc/examples/transcoding: check encoder before using it
 - update MAINTAINERS file
 - POWER LE support in put_vp8_epel_h_altivec_core() put_vp8_epel_v_altivec_core() put_vp8_pixels16_altivec()
 - POWER LE support in vc1_inv_trans_8x4_altivec()
 version 2.5:
 - HEVC/H.265 RTP payload format (draft v6) packetizer
 - SUP/PGS subtitle demuxer
@ -251,7 +546,7 @@ version 2.5:
 - creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
 - WebP muxer with animated WebP support
 - zygoaudio decoding support
- APNG demuxer
+- APNG decoder and demuxer
 - postproc visualization support
@ -262,7 +557,7 @@ version 2.4:
 - ICY metadata are now requested by default with the HTTP protocol
 - support for using metadata in stream specifiers in fftools
 - LZMA compression support in TIFF decoder
- H.261 RTP payload format (RFC 4587) depacketizer and experimental packetizer
+- support for H.261 RTP payload format (RFC 4587)
 - HEVC/H.265 RTP payload format (draft v6) depacketizer
 - added codecview filter to visualize information exported by some codecs
 - Matroska 3D support thorugh side data
@ -742,7 +1037,7 @@ 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
@ -931,8 +1226,8 @@ version 0.8:
 - showinfo filter added
 - SMPTE 302M AES3 audio decoder
 - Apple Core Audio Format muxer
- 9 bits and 10 bits per sample support in the H.264 decoder
+- 9bit and 10bit per sample support in the H.264 decoder
- 9 bits and 10 bits FFV1 encoding / decoding
+- 9bit and 10bit FFV1 encoding / decoding
 - split filter added
 - select filter added
 - sdl output device added
@ -1225,7 +1520,7 @@ version 0.4.9-pre1:
 - rate distorted optimal lambda->qp support
 - AAC encoding with libfaac
 - Sunplus JPEG codec (SP5X) support
- use Lagrange multiplier instead of QP for ratecontrol
+- use Lagrange multipler instead of QP for ratecontrol
 - Theora/VP3 decoding support
 - XA and ADX ADPCM codecs
 - export MPEG-2 active display area / pan scan
--- a/LICENSE.md
+++ b/LICENSE.md
@ -1,124 +1,106 @@
-# License
+#FFmpeg:
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
-or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
+or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
 files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
 FFmpeg.
 Some optional parts of FFmpeg are licensed under the GNU General Public License
-version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
+version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
-these parts are used by default, you have to explicitly pass `--enable-gpl` to
+these parts are used by default, you have to explicitly pass --enable-gpl to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are:
 - libpostproc
- optional x86 optimization in the files
+- libmpcodecs
-    - `libavcodec/x86/flac_dsp_gpl.asm`
+- optional x86 optimizations in the files
-    - `libavcodec/x86/idct_mmx.c`
+  libavcodec/x86/flac_dsp_gpl.asm
-    - `libavfilter/x86/vf_removegrain.asm`
+  libavcodec/x86/idct_mmx.c
- the X11 grabber in `libavdevice/x11grab.c`
+- libutvideo encoding/decoding wrappers in
- the following building and testing tools
+  libavcodec/libutvideo*.cpp
-    - `compat/solaris/make_sunver.pl`
+- the X11 grabber in libavdevice/x11grab.c
-    - `doc/t2h.pm`
+- the swresample test app in
-    - `doc/texi2pod.pl`
+  libswresample/swresample-test.c
-    - `libswresample/swresample-test.c`
+- the texi2pod.pl tool
    - `tests/checkasm/*`
    - `tests/tiny_ssim.c`
 - the following filters in libavfilter:
-    - `f_ebur128.c`
+    - f_ebur128.c
-    - `vf_blackframe.c`
+    - vf_blackframe.c
-    - `vf_boxblur.c`
+    - vf_boxblur.c
-    - `vf_colormatrix.c`
+    - vf_colormatrix.c
-    - `vf_cover_rect.c`
+    - vf_cropdetect.c
-    - `vf_cropdetect.c`
+    - vf_decimate.c
-    - `vf_delogo.c`
+    - vf_delogo.c
-    - `vf_eq.c`
+    - vf_geq.c
-    - `vf_find_rect.c`
+    - vf_histeq.c
-    - `vf_fspp.c`
+    - vf_hqdn3d.c
-    - `vf_geq.c`
+    - vf_interlace.c
-    - `vf_histeq.c`
+    - vf_kerndeint.c
-    - `vf_hqdn3d.c`
+    - vf_mcdeint.c
-    - `vf_interlace.c`
+    - vf_mp.c
-    - `vf_kerndeint.c`
+    - vf_owdenoise.c
-    - `vf_mcdeint.c`
+    - vf_perspective.c
-    - `vf_mpdecimate.c`
+    - vf_phase.c
-    - `vf_owdenoise.c`
+    - vf_pp.c
-    - `vf_perspective.c`
+    - vf_pullup.c
-    - `vf_phase.c`
+    - vf_sab.c
-    - `vf_pp.c`
+    - vf_smartblur.c
-    - `vf_pp7.c`
+    - vf_spp.c
-    - `vf_pullup.c`
+    - vf_stereo3d.c
-    - `vf_repeatfields.c`
+    - vf_super2xsai.c
-    - `vf_sab.c`
+    - vf_tinterlace.c
-    - `vf_smartblur.c`
+    - vsrc_mptestsrc.c
    - `vf_spp.c`
    - `vf_stereo3d.c`
    - `vf_super2xsai.c`
    - `vf_tinterlace.c`
    - `vf_uspp.c`
    - `vsrc_mptestsrc.c`
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter `--enable-version3` will activate this licensing option
+the configure parameter --enable-version3 will activate this licensing option
-for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
+for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
+COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 There are a handful of files under other licensing terms, namely:
-* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  `libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
+  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
  licensing details. Specifically note that you must credit the IJG in the
  documentation accompanying your program if you only distribute executables.
  You must also indicate any changes including additions and deletions to
  those three files in the documentation.
-* `tests/reference.pnm` is under the expat license.
+  tests/reference.pnm is under the expat license
-## External libraries
+external libraries
 ==================
 FFmpeg can be combined with a number of external libraries, which sometimes
 affect the licensing of binaries resulting from the combination.
-### Compatible libraries
+compatible libraries
 --------------------
 The following libraries are under GPL:
- frei0r
+    - frei0r
- libcdio
+    - libcdio
- librubberband
+    - libutvideo
- libvidstab
+    - libvidstab
- libx264
+    - libx264
- libx265
+    - libx265
- libxavs
+    - libxavs
- libxvid
+    - libxvid
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing `--enable-gpl` to configure.
+passing --enable-gpl to configure.
 The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
 license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing `--enable-version3` to configure.
+license version needs to be upgraded by passing --enable-version3 to configure.
-### Incompatible libraries
+incompatible libraries
 ----------------------
-There are certain libraries you can combine with FFmpeg whose licenses are not
+The Fraunhofer AAC library, FAAC and aacplus are under licenses which
-compatible with the GPL and/or the LGPL. If you wish to enable these
+are incompatible with the GPLv2 and v3. We do not know for certain if their
-libraries, even in circumstances that their license may be incompatible, pass
+licenses are compatible with the LGPL.
-`--enable-nonfree` to configure. But note that if you enable any of these
+If you wish to enable these libraries, pass --enable-nonfree to configure.
-libraries the resulting binary will be under a complex license mix that is
+But note that if you enable any of these libraries the resulting binary will
-more restrictive than the LGPL and that may result in additional obligations.
+be under a complex license mix that is more restrictive than the LGPL and that
-It is possible that these restrictions cause the resulting binary to be
+may result in additional obligations. It is possible that these
-unredistributable.
+restrictions cause the resulting binary to be unredistributeable.
 The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
 incompatible with the GPLv2 and v3. To the best of our knowledge, they are
 compatible with the LGPL.
 The FAAC library is incompatible with all versions of GPL and LGPL.
 The NVENC library, while its header file is licensed under the compatible MIT
 license, requires a proprietary binary blob at run time, and is deemed to be
 incompatible with the GPL. We are not certain if it is compatible with the
 LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
 it is not.
--- a/160
+++ b/160
@ -42,8 +42,9 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
+build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
 project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@ -55,9 +56,9 @@ Communication
 website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
 Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Baptiste Coudurier, 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
@ -70,7 +71,6 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer
 Other:
  aes_ctr.c, aes_ctr.h                  Eran Kornblau
  bprint                                Nicolas George
  bswap.h
  des                                   Reimar Doeffinger
@ -88,6 +88,7 @@ Other:
  rational.c, rational.h                Michael Niedermayer
  rc4                                   Reimar Doeffinger
  ripemd.c, ripemd.h                    James Almer
  timecode                              Clément Bœsch
 libavcodec
@ -115,6 +116,8 @@ Generic Parts:
    faandct.c, faandct.h                Michael Niedermayer
  Golomb coding:
    golomb.c, golomb.h                  Michael Niedermayer
  LPC:
    lpc.c, lpc.h                        Justin Ruggles
  motion estimation:
    motion*                             Michael Niedermayer
  rate control:
@ -134,36 +137,43 @@ 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
  alsdec.c                              Thilo Borgmann
  apedec.c                              Kostya Shishkov
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
  atrac3*                               Benjamin Larsson
  atrac3plus*                           Maxim Poliakovski
  bgmc.c, bgmc.h                        Thilo Borgmann
  bink.c                                Kostya Shishkov
  binkaudio.c                           Peter Ross
  bmp.c                                 Mans Rullgard, Kostya Shishkov
  cavs*                                 Stefan Gehrer
  cdxl.c                                Paul B Mahol
  celp_filters.*                        Vitor Sessak
  cinepak.c                             Roberto Togni
  cinepakenc.c                          Rl / Aetey G.T. AB
  ccaption_dec.c                        Anshul Maheshwari, Aman Gupta
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
  cpia.c                                Stephan Hilb
  crystalhd.c                           Philip Langdale
  cscd.c                                Reimar Doeffinger
-  cuvid.c                               Timo Rothenpieler
+  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dirac*                                Rostislav Pehlivanov
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
  dss_sp.c                              Oleksij Rempel
  dv.c                                  Roman Shaposhnik
  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
  fic.c                                 Derek Buitenhuis
  flac*                                 Justin Ruggles
  flashsv*                              Benjamin Larsson
  flicvideo.c                           Mike Melanson
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
@ -171,47 +181,58 @@ Codecs:
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
  h264*                                 Loren Merritt, Michael Niedermayer
  hap*                                  Tom Butterworth
  huffyuv*                              Michael Niedermayer, Christophe Gisquet
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
  indeo2*                               Kostya Shishkov
  indeo5*                               Kostya Shishkov
  interplayvideo.c                      Mike Melanson
-  jni*, ffjni*                          Matthieu Bouron
+  ivi*                                  Kostya Shishkov
  jacosub*                              Clément Bœsch
  jpeg2000*                             Nicolas Bertrand
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
  libkvazaar.c                          Arttu Ylä-Outinen
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
  libspeexdec.c                         Justin Ruggles
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
  libvpx*                               James Zern
  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
  libx265.c                             Derek Buitenhuis
  libxavs.c                             Stefan Gehrer
  libzvbi-teletextdec.c                 Marton Balint
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
  mimic.c                               Ramiro Polla
  mjpeg*.c                              Michael Niedermayer
  mlp*                                  Ramiro Polla
  mmvideo.c                             Peter Ross
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  msvideo1.c                            Mike Melanson
  nellymoserdec.c                       Benjamin Larsson
  nuv.c                                 Reimar Doeffinger
  nvenc*                                Timo Rothenpieler
  paf.*                                 Paul B Mahol
  pcx.c                                 Ivo van Poorten
  pgssubdec.c                           Reimar Doeffinger
  ptx.c                                 Ivo van Poorten
  qcelp*                                Reynaldo H. Verdejo Pinochet
-  qdm2.c, qdm2data.h                    Roberto Togni
+  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
-  qsv*                                  Ivan Uskov
+  qdrw.c                                Kostya Shishkov
  qpeg.c                                Kostya Shishkov
  qtrle.c                               Mike Melanson
  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
  resample2.c                           Michael Niedermayer
@ -219,8 +240,10 @@ Codecs:
  rpza.c                                Roberto Togni
  rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
  rv10.c                                Michael Niedermayer
-  rv4*                                  Christophe Gisquet
+  rv3*                                  Kostya Shishkov
  rv4*                                  Kostya Shishkov, Christophe Gisquet
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
  smvjpegdec.c                          Ash Hughes
  snow*                                 Michael Niedermayer, Loren Merritt
@ -229,41 +252,53 @@ Codecs:
  sunrast.c                             Ivo van Poorten
  svq3.c                                Michael Niedermayer
  tak*                                  Paul B Mahol
  targa.c                               Kostya Shishkov
  tiff.c                                Kostya Shishkov
  truemotion1*                          Mike Melanson
  truemotion2*                          Kostya Shishkov
  truespeech.c                          Kostya Shishkov
  tscc.c                                Kostya Shishkov
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
  ttaenc.c                              Paul B Mahol
  txd.c                                 Ivo van Poorten
-  vc1*                                  Christophe Gisquet
+  ulti*                                 Kostya Shishkov
-  vc2*                                  Rostislav Pehlivanov
+  v410*.c                               Derek Buitenhuis
  vb.c                                  Kostya Shishkov
  vble.c                                Derek Buitenhuis
  vc1*                                  Kostya Shishkov, Christophe Gisquet
  vcr1.c                                Michael Niedermayer
  vda_h264_dec.c                        Xidorn Quan
  videotoolboxenc.c                     Rick Kern
  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
  vorbisdec.c                           Denes Balatoni, David Conrad
  vorbisenc.c                           Oded Shimon
  vp3*                                  Mike Melanson
  vp5                                   Aurelien Jacobs
  vp6                                   Aurelien Jacobs
-  vp8                                   David Conrad, Ronald Bultje
+  vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
-  vp9                                   Ronald Bultje
+  vp9                                   Ronald Bultje, Clément Bœsch
  vqavideo.c                            Mike Melanson
  wavpack.c                             Kostya Shishkov
  wmaprodec.c                           Sascha Sommer
  wmavoice.c                            Ronald S. Bultje
  wmv2.c                                Michael Niedermayer
  wnv1.c                                Kostya Shishkov
  xan.c                                 Mike Melanson
  xbm*                                  Paul B Mahol
  xface                                 Stefano Sabatini
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
  xwd*                                  Paul B Mahol
  zerocodec.c                           Derek Buitenhuis
  zmbv*                                 Kostya Shishkov
 Hardware acceleration:
  crystalhd.c                           Philip Langdale
-  dxva2*                                Hendrik Leppkes, Laurent Aimar
+  dxva2*                                Laurent Aimar
-  mediacodec*                           Matthieu Bouron
+  libstagefright.cpp                    Mohamed Naufal
  vaapi*                                Gwenole Beauchesne
-  vaapi_encode*                         Mark Thompson
+  vda*                                  Sebastien Zwickert
-  vdpau*                                Philip Langdale, Carl Eugen Hoyos
+  vdpau*                                Carl Eugen Hoyos
  videotoolbox*                         Rick Kern
 libavdevice
@ -295,7 +330,6 @@ 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
@ -303,30 +337,19 @@ Filters:
  af_aphaser.c                          Paul B Mahol
  af_aresample.c                        Michael Niedermayer
  af_astats.c                           Paul B Mahol
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
  af_biquads.c                          Paul B Mahol
  af_chorus.c                           Paul B Mahol
  af_compand.c                          Paul B Mahol
  af_firequalizer.c                     Muhammad Faiz
  af_hdcd.c                             Burt P.
  af_ladspa.c                           Paul B Mahol
  af_loudnorm.c                         Kyle Swanson
  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_chromakey.c                        Timo Rothenpieler
  vf_colorchannelmixer.c                Paul B Mahol
  vf_colorbalance.c                     Paul B Mahol
  vf_colorkey.c                         Timo Rothenpieler
  vf_colorlevels.c                      Paul B Mahol
  vf_coreimage.m                        Thilo Borgmann
  vf_deband.c                           Paul B Mahol
  vf_dejudder.c                         Nicholas Robbins
-  vf_delogo.c                           Jean Delvare (CC <jdelvare@suse.com>)
+  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
  vf_drawbox.c/drawgrid                 Andrey Utkin
  vf_extractplanes.c                    Paul B Mahol
  vf_histogram.c                        Paul B Mahol
@ -335,17 +358,12 @@ 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_readvitc.c                         Tobias Rapp (CC t.rapp at noa-archive dot com)
  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
@ -358,16 +376,15 @@ Generic parts:
    libavformat/avformat.h              Michael Niedermayer
  Utility Code:
    libavformat/utils.c                 Michael Niedermayer
  Text Subtitles                        Clément Bœsch
 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
  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
  ape.c                                 Kostya Shishkov
  apngdec.c                             Benoit Fouet
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
@ -381,18 +398,18 @@ Muxers/Demuxers:
  cdxl.c                                Paul B Mahol
  crc.c                                 Michael Niedermayer
  daud.c                                Reimar Doeffinger
  dss.c                                 Oleksij Rempel
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
  electronicarts.c                      Peter Ross
  epafdec.c                             Paul B Mahol
  ffm*                                  Baptiste Coudurier
  flac*                                 Justin Ruggles
  flic.c                                Mike Melanson
  flvdec.c, flvenc.c                    Michael Niedermayer
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
  hls.c                                 Anssi Hannula
  hls encryption (hlsenc.c)             Christian Suloway
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
@ -400,10 +417,10 @@ Muxers/Demuxers:
  ipmovie.c                             Mike Melanson
  ircam*                                Paul B Mahol
  iss.c                                 Stefan Gehrer
  jacosub*                              Clément Bœsch
  jvdec.c                               Peter Ross
  libmodplug.c                          Clément Bœsch
  libnut.c                              Oded Shimon
  libopenmpt.c                          Josh de Kock
  lmlm4.c                               Ivo van Poorten
  lvfdec.c                              Paul B Mahol
  lxfdec.c                              Tomas Härdin
@ -415,9 +432,9 @@ Muxers/Demuxers:
  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
-  mov.c                                 Baptiste Coudurier
+  mov.c                                 Michael Niedermayer, Baptiste Coudurier
  movenc.c                              Baptiste Coudurier, Matthieu Bouron
-  movenccenc.c                          Eran Kornblau
+  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
  mpegts.c                              Marton Balint
@ -433,7 +450,6 @@ Muxers/Demuxers:
  oggdec.c, oggdec.h                    David Conrad
  oggenc.c                              Baptiste Coudurier
  oggparse*.c                           David Conrad
  oggparsedaala*                        Rostislav Pehlivanov
  oma.c                                 Maxim Poliakovski
  paf.c                                 Paul B Mahol
  psxstr.c                              Mike Melanson
@ -443,21 +459,20 @@ Muxers/Demuxers:
  raw.c                                 Michael Niedermayer
  rdt.c                                 Ronald S. Bultje
  rl2.c                                 Sascha Sommer
-  rmdec.c, rmenc.c                      Ronald S. Bultje
+  rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
  rtmp*                                 Kostya Shishkov
  rtp.c, rtpenc.c                       Martin Storsjo
  rtpdec_ac3.*                          Gilles Chanteperdrix
  rtpdec_dv.*                           Thomas Volkert
  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
  rtpdec_mpa_robust.*                   Gilles Chanteperdrix
  rtpdec_asf.*                          Ronald S. Bultje
  rtpdec_vc2hq.*, rtpenc_vc2hq.*        Thomas Volkert
  rtpdec_vp9.c                          Thomas Volkert
  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
  rtsp.c                                Luca Barbato
  sbgdec.c                              Nicolas George
  sdp.c                                 Martin Storsjo
  segafilm.c                            Mike Melanson
  segment.c                             Stefano Sabatini
  siff.c                                Kostya Shishkov
  smacker.c                             Kostya Shishkov
  smjpeg*                               Paul B Mahol
  spdif*                                Anssi Hannula
  srtdec.c                              Aurelien Jacobs
@ -472,10 +487,10 @@ Muxers/Demuxers:
  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
  wvenc.c                               Paul B Mahol
 Protocols:
  async.c                               Zhang Rui
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
@ -502,25 +517,25 @@ Resamplers:
 Operating systems / CPU architectures
 =====================================
-Alpha                                   Falk Hueffner
+Alpha                                   Mans Rullgard, Falk Hueffner
-MIPS                                    Nedeljko Babic
+ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
 MIPS                                    Mans Rullgard, Nedeljko Babic
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver, Hendrik Leppkes
+Windows MSVC                            Matthew Oliver
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
-OS/2                                    KO Myung-Hun
+x86                                     Michael Niedermayer
 Releases
 ========
 2.8                                     Michael Niedermayer
 2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 If you want to maintain an older release, please contact us
@ -531,6 +546,7 @@ GnuPG Fingerprints of maintainers and contributors
 Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
 Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
 Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
@ -538,19 +554,19 @@ Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
 Clément Bœsch                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
 FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
 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
--- a/29
+++ b/29
@ -4,7 +4,6 @@ include config.mak
 vpath %.c    $(SRC_PATH)
 vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
 vpath %.inc  $(SRC_PATH)
 vpath %.m    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
@ -30,18 +29,12 @@ $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
 $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
 OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
 OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
 OBJS-ffmpeg-$(CONFIG_LIBMFX)  += ffmpeg_qsv.o
 OBJS-ffmpeg-$(CONFIG_VAAPI)   += ffmpeg_vaapi.o
 ifndef CONFIG_VIDEOTOOLBOX
 OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
 endif
 OBJS-ffmpeg-$(CONFIG_CUVID)   += ffmpeg_cuvid.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
 OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
 OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
 OBJS-ffserver                 += ffserver_config.o
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher uncoded_frame
 TOOLS-$(CONFIG_ZLIB) += cws2fws
@ -61,14 +54,12 @@ FFLIBS := avutil
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
 EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
-SKIPHEADERS = cmdutils_common_opts.h                                    \
+SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
              compat/w32pthreads.h
 include $(SRC_PATH)/common.mak
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
 all: $(AVPROGS)
@ -89,8 +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 MIPSDSP-OBJS MSA-OBJS         \
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               OBJS SLIBOBJS HOSTOBJS TESTOBJS
 define RESET
 $(1) :=
@ -180,15 +171,11 @@ clean::
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
-	$(RM) -rf coverage.info coverage.info.in lcov
+	$(RM) -rf coverage.info lcov
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version mapfile avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h libavcodec/bsf_list.c libavformat/protocol_list.c
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
 ifeq ($(SRC_LINK),src)
 	$(RM) src
 endif
 	$(RM) -rf doc/examples/pc-uninstalled
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
--- a/README.md
+++ b/README.md
@ -16,21 +16,19 @@ such as audio, video, subtitles and related metadata.
 ## Tools
-* [ffmpeg](https://ffmpeg.org/ffmpeg.html) is a command line toolbox to
+* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
  manipulate, convert and stream multimedia content.
-* [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
+* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
-* [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
+* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analisys tool to inspect
  multimedia content.
 * [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
  for live broadcasts.
 * Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
 ## Documentation
 The offline documentation is available in the **doc/** directory.
-The online documentation is available in the main [website](https://ffmpeg.org)
+The online documentation is available in the main [website](http://ffmpeg.org)
-and in the [wiki](https://trac.ffmpeg.org).
+and in the [wiki](http://trac.ffmpeg.org).
 ### Examples
@ -40,10 +38,3 @@ Coding examples are available in the **doc/examples** directory.
 FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
 GPL. Please refer to the LICENSE file for detailed information.
 ## Contributing
 Patches should be submitted to the ffmpeg-devel mailing list using
 `git format-patch` or `git send-email`. Github pull requests should be
 avoided because they are not part of our review process. Few developers
 follow pull requests so they will likely be ignored.
--- a/2
+++ b/2
@ -1 +1 @@
-3.0.git
+2.5.11
--- a/101
+++ b/101
@ -0,0 +1,101 @@
 ┌────────────────────────────────────────┐
 │ RELEASE NOTES for FFmpeg 2.5 "Bohr"    │
 └────────────────────────────────────────┘
   The FFmpeg Project proudly presents FFmpeg 2.5 "Bohr", 2.5 months after the
   release of 2.4.
   The most important new features are AVFoundation screen-grabbing support,
   animated WebP decoding support, and Animated PNG support. In addition, many
   exciting features for video streaming are also implemented, including MPEG-
   DASH fragmenting muxer, HEVC RTP payload muxer, and UDP Lite support.
   As usual, if you have any question on this release or any FFmpeg related
   topic, feel free to join us on the #ffmpeg IRC channel (on
   irc.freenode.net).
   ┌────────────────────────────┐
   │ 🔨  API Information         │
   └────────────────────────────┘
     FFmpeg 2.5 includes the following library versions:
       • libavutil      54.15.100
       • libavcodec     56.13.100
       • libavformat    56.15.102
       • libavdevice    56. 3.100
       • libavfilter     5. 2.103
       • libswscale      3. 1.101
       • libswresample   1. 1.100
       • libpostproc    53. 3.100
     Important API changes since 2.4:
       • avpriv_dv_frame_profile2() has been deprecated
     Please refer to the doc/APIchanges file for more information.
 ┌────────────────────────────┐
 │ ★  List of New Features    │
 └────────────────────────────┘
   ┌────────────────────────────┐
   │ ffprobe                    │
   └────────────────────────────┘
    • -show_pixel_formats option
   ┌────────────────────────────┐
   │ ffserver                   │
   └────────────────────────────┘
    • codec private options support
   ┌────────────────────────────┐
   │ libavcodec                 │
   └────────────────────────────┘
    • STL subtitle decoder
    • libutvideo YUV 4:2:2 10bit support
    • animated WebP decoding support
    • zygoaudio decoding support
    • APNG decoder
   ┌────────────────────────────┐
   │ libavdevice                │
   └────────────────────────────┘
    • XCB-based screen-grabber
    • AVFoundation screen capturing support
   ┌────────────────────────────┐
   │ libavformat                │
   └────────────────────────────┘
    • HEVC/H.265 RTP payload format (draft v6) packetizer
    • SUP/PGS subtitle demuxer
    • STL subtitle demuxer
    • UDP-Lite support (RFC 3828)
    • MPEG-DASH segmenting muxer, which allows creating DASH compatible
      fragmented MP4
    • WebP muxer
    • APNG demuxer
   ┌────────────────────────────┐
   │ libavfilter                │
   └────────────────────────────┘
    • xBR scaling filter
   ┌────────────────────────────┐
   │ libavutil                  │
   └────────────────────────────┘
    • CAST128 symmetric block cipher, ECB mode
   ┌────────────────────────────┐
   │ libpostproc                │
   └────────────────────────────┘
    • visualization support
--- a/arch.mak
+++ b/arch.mak
@ -5,13 +5,11 @@ OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
-OBJS-$(HAVE_MIPSDSP)   += $(MIPSDSP-OBJS)    $(MIPSDSP-OBJS-yes)
+OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
 OBJS-$(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)
--- a/cmdutils.c
+++ b/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"
@ -52,7 +50,6 @@
 #include "libavutil/opt.h"
 #include "libavutil/cpu.h"
 #include "libavutil/ffversion.h"
 #include "libavutil/version.h"
 #include "cmdutils.h"
 #if CONFIG_NETWORK
 #include "libavformat/network.h"
@ -64,7 +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;
@ -74,13 +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);
@ -286,14 +290,10 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
    if (po->flags & OPT_SPEC) {
        SpecifierOpt **so = dst;
        char *p = strchr(opt, ':');
        char *str;
        dstcount = (int *)(so + 1);
        *so = grow_array(*so, sizeof(**so), dstcount, *dstcount + 1);
-        str = av_strdup(p ? p + 1 : "");
+        (*so)[*dstcount - 1].specifier = av_strdup(p ? p + 1 : "");
        if (!str)
            return AVERROR(ENOMEM);
        (*so)[*dstcount - 1].specifier = str;
        dst = &(*so)[*dstcount - 1].u;
    }
@ -301,8 +301,6 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
        char *str;
        str = av_strdup(arg);
        av_freep(dst);
        if (!str)
            return AVERROR(ENOMEM);
        *(char **)dst = str;
    } else if (po->flags & OPT_BOOL || po->flags & OPT_INT) {
        *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
@ -476,22 +474,10 @@ 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])
@ -523,7 +509,7 @@ 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;
@ -534,12 +520,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
 #if CONFIG_AVRESAMPLE
    const AVClass *rc = avresample_get_class();
 #endif
-#if CONFIG_SWSCALE
+    const AVClass *sc, *swr_class;
    const AVClass *sc = sws_get_class();
 #endif
 #if CONFIG_SWRESAMPLE
    const AVClass *swr_class = swr_get_class();
 #endif
    if (!strcmp(opt, "debug") || !strcmp(opt, "fdebug"))
        av_log_set_level(AV_LOG_DEBUG);
@ -563,24 +544,15 @@ int opt_default(void *optctx, const char *opt, const char *arg)
        consumed = 1;
    }
 #if CONFIG_SWSCALE
-    if (!consumed && (o = opt_find(&sc, opt, NULL, 0,
+    sc = sws_get_class();
-                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+    if (!consumed && opt_find(&sc, opt, NULL, 0,
-        struct SwsContext *sws = sws_alloc_context();
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
-        int ret = av_opt_set(sws, opt, arg, 0);
+        // XXX we only support sws_flags, not arbitrary sws options
-        sws_freeContext(sws);
+        int ret = av_opt_set(sws_opts, opt, arg, 0);
        if (!strcmp(opt, "srcw") || !strcmp(opt, "srch") ||
            !strcmp(opt, "dstw") || !strcmp(opt, "dsth") ||
            !strcmp(opt, "src_format") || !strcmp(opt, "dst_format")) {
            av_log(NULL, AV_LOG_ERROR, "Directly using swscale dimensions/format options is not supported, please use the -s or -pix_fmt options\n");
            return AVERROR(EINVAL);
        }
        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
@ -590,6 +562,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    }
 #endif
 #if CONFIG_SWRESAMPLE
    swr_class = swr_get_class();
    if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0,
                                    AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        struct SwrContext *swr = swr_alloc();
@ -653,7 +626,9 @@ 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;
@ -662,7 +637,9 @@ 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();
@ -718,8 +695,9 @@ 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);
-
+#if CONFIG_SWSCALE
-            av_dict_free(&l->groups[j].sws_dict);
+            sws_freeContext(l->groups[j].sws_opts);
 #endif
            av_dict_free(&l->groups[j].swr_opts);
        }
        av_freep(&l->groups);
@ -861,7 +839,6 @@ 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;
@ -1059,8 +1036,7 @@ static int warned_cfg = 0;
                   LIB##LIBNAME##_VERSION_MAJOR,                        \
                   LIB##LIBNAME##_VERSION_MINOR,                        \
                   LIB##LIBNAME##_VERSION_MICRO,                        \
-                   AV_VERSION_MAJOR(version), AV_VERSION_MINOR(version),\
+                   version >> 16, version >> 8 & 0xff, version & 0xff); \
                   AV_VERSION_MICRO(version));                          \
        }                                                               \
        if (flags & SHOW_CONFIG) {                                      \
            const char *cfg = libname##_configuration();                \
@ -1079,15 +1055,15 @@ static int warned_cfg = 0;
 static void print_all_libs_info(int flags, int level)
 {
-    PRINT_LIB_INFO(avutil,     AVUTIL,     flags, level);
+    PRINT_LIB_INFO(avutil,   AVUTIL,   flags, level);
-    PRINT_LIB_INFO(avcodec,    AVCODEC,    flags, level);
+    PRINT_LIB_INFO(avcodec,  AVCODEC,  flags, level);
-    PRINT_LIB_INFO(avformat,   AVFORMAT,   flags, level);
+    PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
-    PRINT_LIB_INFO(avdevice,   AVDEVICE,   flags, level);
+    PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
-    PRINT_LIB_INFO(avfilter,   AVFILTER,   flags, level);
+    PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
-    PRINT_LIB_INFO(swscale,    SWSCALE,    flags, level);
+    PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
-    PRINT_LIB_INFO(swresample, SWRESAMPLE, flags, level);
+    PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
-    PRINT_LIB_INFO(postproc,   POSTPROC,   flags, level);
+    PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
 }
 static void print_program_info(int flags, int level)
@ -1099,7 +1075,8 @@ static void print_program_info(int flags, int level)
        av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
               program_birth_year, CONFIG_THIS_YEAR);
    av_log(NULL, level, "\n");
-    av_log(NULL, level, "%sbuilt with %s\n", indent, CC_IDENT);
+    av_log(NULL, level, "%sbuilt on %s %s with %s\n",
           indent, __DATE__, __TIME__, CC_IDENT);
    av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }
@ -1236,7 +1213,12 @@ static int is_device(const AVClass *avclass)
 {
    if (!avclass)
        return 0;
-    return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
+    return avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_OUTPUT ||
           avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_INPUT ||
           avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT ||
           avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT ||
           avclass->category == AV_CLASS_CATEGORY_DEVICE_OUTPUT ||
           avclass->category == AV_CLASS_CATEGORY_DEVICE_INPUT;
 }
 static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
@ -1324,47 +1306,16 @@ static void print_codec(const AVCodec *c)
    printf("%s %s [%s]:\n", encoder ? "Encoder" : "Decoder", c->name,
           c->long_name ? c->long_name : "");
    printf("    General capabilities: ");
    if (c->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)
        printf("horizband ");
    if (c->capabilities & AV_CODEC_CAP_DR1)
        printf("dr1 ");
    if (c->capabilities & AV_CODEC_CAP_TRUNCATED)
        printf("trunc ");
    if (c->capabilities & AV_CODEC_CAP_DELAY)
        printf("delay ");
    if (c->capabilities & AV_CODEC_CAP_SMALL_LAST_FRAME)
        printf("small ");
    if (c->capabilities & AV_CODEC_CAP_SUBFRAMES)
        printf("subframes ");
    if (c->capabilities & AV_CODEC_CAP_EXPERIMENTAL)
        printf("exp ");
    if (c->capabilities & AV_CODEC_CAP_CHANNEL_CONF)
        printf("chconf ");
    if (c->capabilities & AV_CODEC_CAP_PARAM_CHANGE)
        printf("paramchange ");
    if (c->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
        printf("variable ");
    if (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
                           AV_CODEC_CAP_SLICE_THREADS |
                           AV_CODEC_CAP_AUTO_THREADS))
        printf("threads ");
    if (!c->capabilities)
        printf("none");
    printf("\n");
    if (c->type == AVMEDIA_TYPE_VIDEO ||
        c->type == AVMEDIA_TYPE_AUDIO) {
        printf("    Threading capabilities: ");
-        switch (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
+        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
-                                   AV_CODEC_CAP_SLICE_THREADS |
+                                   CODEC_CAP_SLICE_THREADS)) {
-                                   AV_CODEC_CAP_AUTO_THREADS)) {
+        case CODEC_CAP_FRAME_THREADS |
-        case AV_CODEC_CAP_FRAME_THREADS |
+             CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
-             AV_CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
+        case CODEC_CAP_FRAME_THREADS: printf("frame");           break;
-        case AV_CODEC_CAP_FRAME_THREADS: printf("frame");           break;
+        case CODEC_CAP_SLICE_THREADS: printf("slice");           break;
-        case AV_CODEC_CAP_SLICE_THREADS: printf("slice");           break;
+        default:                      printf("no");              break;
        case AV_CODEC_CAP_AUTO_THREADS : printf("auto");            break;
        default:                         printf("none");            break;
        }
        printf("\n");
    }
@ -1423,7 +1374,7 @@ static int compare_codec_desc(const void *a, const void *b)
    const AVCodecDescriptor * const *da = a;
    const AVCodecDescriptor * const *db = b;
-    return (*da)->type != (*db)->type ? FFDIFFSIGN((*da)->type, (*db)->type) :
+    return (*da)->type != (*db)->type ? (*da)->type - (*db)->type :
           strcmp((*da)->name, (*db)->name);
 }
@ -1537,11 +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 & CODEC_CAP_FRAME_THREADS) ? "F" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
+            printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
+            printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
+            printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_DR1)           ? "D" : ".");
+            printf((codec->capabilities & CODEC_CAP_DR1)           ? "D" : ".");
            printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : "");
            if (strcmp(codec->name, desc->name))
@ -1584,10 +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;
 }
@ -1602,7 +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"
@ -1615,17 +1566,17 @@ 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)) ||
                                  ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
        }
        *descr_cur = 0;
-        printf(" %c%c%c %-17s %-10s %s\n",
+        printf(" %c%c%c %-16s %-10s %s\n",
               filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
               filter->flags & AVFILTER_FLAG_SLICE_THREADS    ? 'S' : '.',
               filter->process_command                        ? 'C' : '.',
@ -1870,8 +1821,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
    av_log_set_callback(log_callback_help);
    topic = av_strdup(arg ? arg : "");
    if (!topic)
        return AVERROR(ENOMEM);
    par = strchr(topic, '=');
    if (par)
        *par++ = 0;
@ -1909,6 +1858,50 @@ 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;
    }
    fseek(f, 0, SEEK_END);
    *size = ftell(f);
    fseek(f, 0, SEEK_SET);
    if (*size == (size_t)-1) {
        ret = AVERROR(errno);
        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno));
        fclose(f);
        return ret;
    }
    *bufptr = av_malloc(*size + 1);
    if (!*bufptr) {
        av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
        fclose(f);
        return AVERROR(ENOMEM);
    }
    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';
    }
    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)
@ -2051,7 +2044,7 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
        exit_program(1);
    }
    if (*size < new_size) {
-        uint8_t *tmp = av_realloc_array(array, new_size, elem_size);
+        uint8_t *tmp = av_realloc(array, new_size*elem_size);
        if (!tmp) {
            av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
            exit_program(1);
@ -2063,38 +2056,13 @@ 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)
 {
    int ret, i;
    AVFormatContext *dev = NULL;
    AVDeviceInfoList *device_list = NULL;
    AVDictionary *tmp_opts = NULL;
    if (!fmt || !fmt->priv_class  || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
        return AVERROR(EINVAL);
@ -2106,7 +2074,15 @@ static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
        goto fail;
    }
-    if ((ret = avdevice_list_input_sources(fmt, NULL, opts, &device_list)) < 0) {
+    /* TODO: avformat_open_input calls read_header callback which is not necessary.
             Function like avformat_alloc_output_context2 for input could be helpful here. */
    av_dict_copy(&tmp_opts, opts, 0);
    if ((ret = avformat_open_input(&dev, NULL, fmt, &tmp_opts)) < 0) {
        printf("Cannot open device: %s.\n", fmt->name);
        goto fail;
    }
    if ((ret = avdevice_list_devices(dev, &device_list)) < 0) {
        printf("Cannot list sources.\n");
        goto fail;
    }
@ -2117,14 +2093,18 @@ static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
    }
  fail:
    av_dict_free(&tmp_opts);
    avdevice_free_list_devices(&device_list);
    avformat_close_input(&dev);
    return ret;
 }
 static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
 {
    int ret, i;
    AVFormatContext *dev = NULL;
    AVDeviceInfoList *device_list = NULL;
    AVDictionary *tmp_opts = NULL;
    if (!fmt || !fmt->priv_class  || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
        return AVERROR(EINVAL);
@ -2136,7 +2116,14 @@ static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
        goto fail;
    }
-    if ((ret = avdevice_list_output_sinks(fmt, NULL, opts, &device_list)) < 0) {
+    if ((ret = avformat_alloc_output_context2(&dev, fmt, NULL, NULL)) < 0) {
        printf("Cannot open device: %s.\n", fmt->name);
        goto fail;
    }
    av_dict_copy(&tmp_opts, opts, 0);
    av_opt_set_dict2(dev, &tmp_opts, AV_OPT_SEARCH_CHILDREN);
    if ((ret = avdevice_list_devices(dev, &device_list)) < 0) {
        printf("Cannot list sinks.\n");
        goto fail;
    }
@ -2147,7 +2134,9 @@ static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
    }
  fail:
    av_dict_free(&tmp_opts);
    avdevice_free_list_devices(&device_list);
    avformat_free_context(dev);
    return ret;
 }
@ -2191,7 +2180,7 @@ int show_sources(void *optctx, const char *opt, const char *arg)
        if (fmt) {
            if (!strcmp(fmt->name, "lavfi"))
                continue; //it's pointless to probe lavfi
-            if (dev && !av_match_name(dev, fmt->name))
+            if (dev && strcmp(fmt->name, dev))
                continue;
            print_device_sources(fmt, opts);
        }
@ -2199,7 +2188,7 @@ int show_sources(void *optctx, const char *opt, const char *arg)
    do {
        fmt = av_input_video_device_next(fmt);
        if (fmt) {
-            if (dev && !av_match_name(dev, fmt->name))
+            if (dev && strcmp(fmt->name, dev))
                continue;
            print_device_sources(fmt, opts);
        }
@ -2227,7 +2216,7 @@ int show_sinks(void *optctx, const char *opt, const char *arg)
    do {
        fmt = av_output_audio_device_next(fmt);
        if (fmt) {
-            if (dev && !av_match_name(dev, fmt->name))
+            if (dev && strcmp(fmt->name, dev))
                continue;
            print_device_sinks(fmt, opts);
        }
@ -2235,7 +2224,7 @@ int show_sinks(void *optctx, const char *opt, const char *arg)
    do {
        fmt = av_output_video_device_next(fmt);
        if (fmt) {
-            if (dev && !av_match_name(dev, fmt->name))
+            if (dev && strcmp(fmt->name, dev))
                continue;
            print_device_sinks(fmt, opts);
        }
@ -2246,5 +2235,4 @@ int show_sinks(void *optctx, const char *opt, const char *arg)
    av_log_set_level(error_level);
    return ret;
 }
 #endif
--- a/cmdutils.h
+++ b/cmdutils.h
@ -19,8 +19,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
-#ifndef CMDUTILS_H
+#ifndef FFMPEG_CMDUTILS_H
-#define CMDUTILS_H
+#define FFMPEG_CMDUTILS_H
 #include <stdint.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 */
--- a/cmdutils_opencl.c
+++ b/cmdutils_opencl.c
@ -206,9 +206,7 @@ end:
 static int compare_ocl_device_desc(const void *a, const void *b)
 {
-    const OpenCLDeviceBenchmark* va = (const OpenCLDeviceBenchmark*)a;
+    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
    const OpenCLDeviceBenchmark* vb = (const OpenCLDeviceBenchmark*)b;
    return FFDIFFSIGN(va->runtime , vb->runtime);
 }
 int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
--- a/common.mak
+++ b/common.mak
@ -5,20 +5,12 @@
 # first so "all" becomes default target
 all: all-yes
 DEFAULT_YASMD=.dbg
 ifeq ($(DBG),1)
 YASMD=$(DEFAULT_YASMD)
 else
 YASMD=
 endif
 ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 MSG    = $@
@ -32,14 +24,12 @@ endif
 ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
 # NASM requires -I path terminated with /
-IFLAGS     := -I. -I$(SRC_LINK)/
+IFLAGS     := -I. -I$(SRC_PATH)/
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
 CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 OBJCFLAGS  += $(EOBJCFLAGS)
 OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
+CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
 YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@ -47,13 +37,12 @@ LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 define COMPILE
       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
+       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
 endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
 COMPILE_M = $(call COMPILE,OBJCC)
 COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.c
@ -63,10 +52,10 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 	$(COMPILE_CXX)
 %.o: %.m
-	$(COMPILE_M)
+	$(COMPILE_C)
 %.s: %.c
-	$(CC) $(CCFLAGS) -S -o $@ $<
+	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
 %.o: %.S
 	$(COMPILE_S)
@ -83,7 +72,10 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
-%.c %.h %.ver: TAG = GEN
+%.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
 %.c %.h: TAG = GEN
 # Dummy rule to stop make trying to rebuild removed or renamed headers
 %.h:
@ -109,8 +101,8 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
-TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
+TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
-TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
+TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
 HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
 TOOLS     += $(TOOLS-yes)
@ -118,9 +110,8 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)
-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
 STATIC_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))
@ -147,15 +138,17 @@ $(TOOLOBJS): | tools
 OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.gcda *.gcno *.map *.ver *.ho *$(DEFAULT_YASMD).asm
+CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 define RULES
 clean::
-	$(RM) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
+	$(RM) $(OBJS) $(OBJS:.o=.d)
 	$(RM) $(HOSTPROGS)
 	$(RM) $(TOOLS)
 endef
 $(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@ -19,8 +19,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
-#ifndef COMPAT_AIX_MATH_H
+#ifndef FFMPEG_COMPAT_AIX_MATH_H
-#define COMPAT_AIX_MATH_H
+#define FFMPEG_COMPAT_AIX_MATH_H
 #define class class_in_math_h_causes_problems
@ -28,4 +28,4 @@
 #undef class
-#endif /* COMPAT_AIX_MATH_H */
+#endif /* FFMPEG_COMPAT_AIX_MATH_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@ -38,9 +38,40 @@
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__
-#include "avs/config.h"
+#ifdef __cplusplus
-#include "avs/capi.h"
+#  define EXTERN_C extern "C"
-#include "avs/types.h"
+#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__
+#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 6 };
+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_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 16,
+      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=0,
-  AVS_CACHE_NOTHING=10, // Do not cache video.
+  AVS_CACHE_RANGE=1,
-  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
+  AVS_CACHE_ALL=2,
-  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
+  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
+  AVS_CACHE_AUDIO_NONE=4,
-
+  AVS_CACHE_AUDIO_AUTO=5
  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)
  };
-#ifdef BUILDING_AVSCORE
+#define AVS_FRAME_ALIGN 16
 struct AVS_ScriptEnvironment {
        IScriptEnvironment * env;
        const char * error;
        AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
                : env(e), error(0) {}
 };
 #endif
 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_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
+        { 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_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_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);}
+        return p->pitch;}
 #endif
-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);}
+        return p->vfb->data + p->offset;}
 #endif
-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     = 0x800,   //  Nehalem
  AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
 //AVS_CPUF_AVX2      = 0x2000,   //  Haswell
 //AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
 };
 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);
--- a/compat/avisynth/avisynth_c_25.h
+++ b/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); }
--- a/compat/avisynth/avs/capi.h
+++ b/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
--- a/compat/avisynth/avs/config.h
+++ b/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
--- a/compat/avisynth/avs/types.h
+++ b/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
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@ -513,21 +513,21 @@ AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
 // only use these functions on am AVS_Value that does not already have
 // an active value.  Remember, treat AVS_Value as a fat pointer.
 AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
 AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
+        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
+        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
+        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
 AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
+        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
+        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
 #endif
 AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
 /////////////////////////////////////////////////////////////////////
 //
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@ -52,8 +52,8 @@ namespace avxsynth {
 //
 // Functions
 //
-#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
+#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
-#define MAKEWORD(a,b) (((a) << 8) | (b))
+#define MAKEWORD(a,b) ((a << 8) | (b))
 #define lstrlen                             strlen
 #define lstrcpy                             strcpy
--- a/compat/dispatch_semaphore/semaphore.h
+++ b/compat/dispatch_semaphore/semaphore.h
@ -1,42 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
 #define COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
 #include <dispatch/dispatch.h>
 #include <errno.h>
 #define sem_t dispatch_semaphore_t
 #define sem_post(psem)              dispatch_semaphore_signal(*psem)
 #define sem_wait(psem)              dispatch_semaphore_wait(*psem, DISPATCH_TIME_FOREVER)
 #define sem_timedwait(psem, val)    dispatch_semaphore_wait(*psem, dispatch_walltime(val, 0))
 #define sem_destroy(psem)           dispatch_release(*psem)
 static inline int compat_sem_init(dispatch_semaphore_t *psem,
                                  int unused, int val)
 {
    int ret = !!(*psem = dispatch_semaphore_create(val)) - 1;
    if (ret < 0)
        errno = ENOMEM;
    return ret;
 }
 #define sem_init compat_sem_init
 #endif /* COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H */
--- a/compat/msvcrt/snprintf.h
+++ b/compat/msvcrt/snprintf.h
@ -19,8 +19,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
-#ifndef COMPAT_MSVCRT_SNPRINTF_H
+#ifndef COMPAT_SNPRINTF_H
-#define COMPAT_MSVCRT_SNPRINTF_H
+#define COMPAT_SNPRINTF_H
 #include <stdarg.h>
 #include <stdio.h>
@ -35,4 +35,4 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
 #define _snprintf avpriv_snprintf
 #define vsnprintf avpriv_vsnprintf
-#endif /* COMPAT_MSVCRT_SNPRINTF_H */
+#endif /* COMPAT_SNPRINTF_H */
--- a/compat/os2threads.h
+++ b/compat/os2threads.h
@ -23,8 +23,8 @@
 * os2threads to pthreads wrapper
 */
-#ifndef COMPAT_OS2THREADS_H
+#ifndef AVCODEC_OS2PTHREADS_H
-#define COMPAT_OS2THREADS_H
+#define AVCODEC_OS2PTHREADS_H
 #define INCL_DOS
 #include <os2.h>
@ -32,71 +32,57 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
-#include <sys/builtin.h>
+#include "libavutil/mem.h"
 #include <sys/fmutex.h>
 #include "libavutil/attributes.h"
 typedef struct {
    TID tid;
    void *(*start_routine)(void *);
    void *arg;
    void *result;
 } pthread_t;
 typedef TID  pthread_t;
 typedef void pthread_attr_t;
 typedef HMTX pthread_mutex_t;
 typedef void pthread_mutexattr_t;
 typedef struct {
-    HEV event_sem;
+    HEV  event_sem;
-    HEV ack_sem;
+    int  wait_count;
    volatile unsigned  wait_count;
 } pthread_cond_t;
 typedef void pthread_condattr_t;
-typedef struct {
+struct thread_arg {
-    volatile int done;
+    void *(*start_routine)(void *);
-    _fmutex mtx;
+    void *arg;
-} pthread_once_t;
+};
 #define PTHREAD_ONCE_INIT {0, _FMUTEX_INITIALIZER}
 static void thread_entry(void *arg)
 {
-    pthread_t *thread = arg;
+    struct thread_arg *thread_arg = arg;
-    thread->result = thread->start_routine(thread->arg);
+    thread_arg->start_routine(thread_arg->arg);
    av_free(thread_arg);
 }
-static av_always_inline int pthread_create(pthread_t *thread,
+static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
                                           const pthread_attr_t *attr,
                                           void *(*start_routine)(void*),
                                           void *arg)
 {
-    thread->start_routine = start_routine;
+    struct thread_arg *thread_arg;
    thread->arg = arg;
    thread->result = NULL;
-    thread->tid = _beginthread(thread_entry, NULL, 1024 * 1024, thread);
+    thread_arg = av_mallocz(sizeof(struct thread_arg));
    thread_arg->start_routine = start_routine;
    thread_arg->arg = arg;
    *thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg);
    return 0;
 }
 static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
 {
-    DosWaitThread(&thread.tid, DCWW_WAIT);
+    DosWaitThread((PTID)&thread, DCWW_WAIT);
    if (value_ptr)
        *value_ptr = thread.result;
    return 0;
 }
-static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
+static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
                                               const pthread_mutexattr_t *attr)
 {
    DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
@ -124,11 +110,9 @@ static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
    return 0;
 }
-static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
+static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr)
                                              const pthread_condattr_t *attr)
 {
    DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE);
    DosCreateEventSem(NULL, &cond->ack_sem, DCE_POSTONE, FALSE);
    cond->wait_count = 0;
@ -138,16 +122,16 @@ static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
 static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond)
 {
    DosCloseEventSem(cond->event_sem);
    DosCloseEventSem(cond->ack_sem);
    return 0;
 }
 static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
 {
-    if (!__atomic_cmpxchg32(&cond->wait_count, 0, 0)) {
+    if (cond->wait_count > 0) {
        DosPostEventSem(cond->event_sem);
-        DosWaitEventSem(cond->ack_sem, SEM_INDEFINITE_WAIT);
+
        cond->wait_count--;
    }
    return 0;
@ -155,47 +139,26 @@ static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
 static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
 {
-    while (!__atomic_cmpxchg32(&cond->wait_count, 0, 0))
+    while (cond->wait_count > 0) {
-        pthread_cond_signal(cond);
+        DosPostEventSem(cond->event_sem);
        cond->wait_count--;
    }
    return 0;
 }
-static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
+static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
                                              pthread_mutex_t *mutex)
 {
-    __atomic_increment(&cond->wait_count);
+    cond->wait_count++;
    pthread_mutex_unlock(mutex);
    DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
    __atomic_decrement(&cond->wait_count);
    DosPostEventSem(cond->ack_sem);
    pthread_mutex_lock(mutex);
    return 0;
 }
-static av_always_inline int pthread_once(pthread_once_t *once_control,
+#endif /* AVCODEC_OS2PTHREADS_H */
                                         void (*init_routine)(void))
 {
    if (!once_control->done)
    {
        _fmutex_request(&once_control->mtx, 0);
        if (!once_control->done)
        {
            init_routine();
            once_control->done = 1;
        }
        _fmutex_release(&once_control->mtx);
    }
    return 0;
 }
 #endif /* COMPAT_OS2THREADS_H */
--- a/compat/solaris/make_sunver.pl
+++ b/compat/solaris/make_sunver.pl
@ -1,352 +0,0 @@
 #!/usr/bin/env perl
 # make_sunver.pl
 #
 #   Copyright (C) 2010, 2011, 2012, 2013
 #   Free Software Foundation, Inc.
 #
 # This file 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 3 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; see the file COPYING.GPLv3.  If not see
 # <http://www.gnu.org/licenses/>.
 # This script takes at least two arguments, a GNU style version script and
 # a list of object and archive files, and generates a corresponding Sun
 # style version script as follows:
 #
 # Each glob pattern, C++ mangled pattern or literal in the input script is
 # matched against all global symbols in the input objects, emitting those
 # that matched (or nothing if no match was found).
 # A comment with the original pattern and its type is left in the output
 # file to make it easy to understand the matches.
 #
 # It uses elfdump when present (native), GNU readelf otherwise.
 # It depends on the GNU version of c++filt, since it must understand the
 # GNU mangling style.
 use FileHandle;
 use IPC::Open2;
 # Enforce C locale.
 $ENV{'LC_ALL'} = "C";
 $ENV{'LANG'} = "C";
 # Input version script, GNU style.
 my $symvers = shift;
 ##########
 # Get all the symbols from the library, match them, and add them to a hash.
 my %sym_hash = ();
 # List of objects and archives to process.
 my @OBJECTS = ();
 # List of shared objects to omit from processing.
 my @SHAREDOBJS = ();
 # Filter out those input archives that have corresponding shared objects to
 # avoid adding all symbols matched in the archive to the output map.
 foreach $file (@ARGV) {
    if (($so = $file) =~ s/\.a$/.so/ && -e $so) {
    printf STDERR "omitted $file -> $so\n";
    push (@SHAREDOBJS, $so);
    } else {
    push (@OBJECTS, $file);
    }
 }
 # We need to detect and ignore hidden symbols.  Solaris nm can only detect
 # this in the harder to parse default output format, and GNU nm not at all,
 # so use elfdump -s in the native case and GNU readelf -s otherwise.
 # GNU objdump -t cannot be used since it produces a variable number of
 # columns.
 # The path to elfdump.
 my $elfdump = "/usr/ccs/bin/elfdump";
 if (-f $elfdump) {
    open ELFDUMP,$elfdump.' -s '.(join ' ',@OBJECTS).'|' or die $!;
    my $skip_arsym = 0;
    while (<ELFDUMP>) {
    chomp;
    # Ignore empty lines.
    if (/^$/) {
        # End of archive symbol table, stop skipping.
        $skip_arsym = 0 if $skip_arsym;
        next;
    }
    # Keep skipping until end of archive symbol table.
    next if ($skip_arsym);
    # Ignore object name header for individual objects and archives.
    next if (/:$/);
    # Ignore table header lines.
    next if (/^Symbol Table Section:/);
    next if (/index.*value.*size/);
    # Start of archive symbol table: start skipping.
    if (/^Symbol Table: \(archive/) {
        $skip_arsym = 1;
        next;
    }
    # Split table.
    (undef, undef, undef, undef, $bind, $oth, undef, $shndx, $name) = split;
    # Error out for unknown input.
    die "unknown input line:\n$_" unless defined($bind);
    # Ignore local symbols.
    next if ($bind eq "LOCL");
    # Ignore hidden symbols.
    next if ($oth eq "H");
    # Ignore undefined symbols.
    next if ($shndx eq "UNDEF");
    # Error out for unhandled cases.
    if ($bind !~ /^(GLOB|WEAK)/ or $oth ne "D") {
        die "unhandled symbol:\n$_";
    }
    # Remember symbol.
    $sym_hash{$name}++;
    }
    close ELFDUMP or die "$elfdump error";
 } else {
    open READELF, 'readelf -s -W '.(join ' ',@OBJECTS).'|' or die $!;
    # Process each symbol.
    while (<READELF>) {
    chomp;
    # Ignore empty lines.
    next if (/^$/);
    # Ignore object name header.
    next if (/^File: .*$/);
    # Ignore table header lines.
    next if (/^Symbol table.*contains.*:/);
    next if (/Num:.*Value.*Size/);
    # Split table.
    (undef, undef, undef, undef, $bind, $vis, $ndx, $name) = split;
    # Error out for unknown input.
    die "unknown input line:\n$_" unless defined($bind);
    # Ignore local symbols.
    next if ($bind eq "LOCAL");
    # Ignore hidden symbols.
    next if ($vis eq "HIDDEN");
    # Ignore undefined symbols.
    next if ($ndx eq "UND");
    # Error out for unhandled cases.
    if ($bind !~ /^(GLOBAL|WEAK)/ or $vis ne "DEFAULT") {
        die "unhandled symbol:\n$_";
    }
    # Remember symbol.
    $sym_hash{$name}++;
    }
    close READELF or die "readelf error";
 }
 ##########
 # The various types of glob patterns.
 #
 # A glob pattern that is to be applied to the demangled name: 'cxx'.
 # A glob patterns that applies directly to the name in the .o files: 'glob'.
 # This pattern is ignored; used for local variables (usually just '*'): 'ign'.
 # The type of the current pattern.
 my $glob = 'glob';
 # We're currently inside `extern "C++"', which Sun ld doesn't understand.
 my $in_extern = 0;
 # The c++filt command to use.  This *must* be GNU c++filt; the Sun Studio
 # c++filt doesn't handle the GNU mangling style.
 my $cxxfilt = $ENV{'CXXFILT'} || "c++filt";
 # The current version name.
 my $current_version = "";
 # Was there any attempt to match a symbol to this version?
 my $matches_attempted;
 # The number of versions which matched this symbol.
 my $matched_symbols;
 open F,$symvers or die $!;
 # Print information about generating this file
 print "# This file was generated by make_sunver.pl.  DO NOT EDIT!\n";
 print "# It was generated by:\n";
 printf "# %s %s %s\n", $0, $symvers, (join ' ',@ARGV);
 printf "# Omitted archives with corresponding shared libraries: %s\n",
    (join ' ', @SHAREDOBJS) if $#SHAREDOBJS >= 0;
 print "#\n\n";
 print "\$mapfile_version 2\n";
 while (<F>) {
    # Lines of the form '};'
    if (/^([ \t]*)(\}[ \t]*;[ \t]*)$/) {
    $glob = 'glob';
    if ($in_extern) {
        $in_extern--;
        print "$1##$2\n";
    } else {
        print;
    }
    next;
    }
    # Lines of the form '} SOME_VERSION_NAME_1.0;'
    if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
    $glob = 'glob';
    # We tried to match symbols agains this version, but none matched.
    # Emit dummy hidden symbol to avoid marking this version WEAK.
    if ($matches_attempted && $matched_symbols == 0) {
        print "  hidden:\n";
        print "    .force_WEAK_off_$current_version = DATA S0x0 V0x0;\n";
    }
    print; next;
    }
    # Comment and blank lines
    if (/^[ \t]*\#/) { print; next; }
    if (/^[ \t]*$/) { print; next; }
    # Lines of the form '{'
    if (/^([ \t]*){$/) {
    if ($in_extern) {
        print "$1##{\n";
    } else {
        print;
    }
    next;
    }
    # Lines of the form 'SOME_VERSION_NAME_1.1 {'
    if (/^([A-Z0-9_.]+)[ \t]+{$/) {
    # Record version name.
    $current_version = $1;
    # Reset match attempts, #matched symbols for this version.
    $matches_attempted = 0;
    $matched_symbols = 0;
    print "SYMBOL_VERSION $1 {\n";
    next;
    }
    # Ignore 'global:'
    if (/^[ \t]*global:$/) { print; next; }
    # After 'local:', globs should be ignored, they won't be exported.
    if (/^[ \t]*local:$/) {
    $glob = 'ign';
    print;
    next;
    }
    # After 'extern "C++"', globs are C++ patterns
    if (/^([ \t]*)(extern \"C\+\+\"[ \t]*)$/) {
    $in_extern++;
    $glob = 'cxx';
    # Need to comment, Sun ld cannot handle this.
    print "$1##$2\n"; next;
    }
    # Chomp newline now we're done with passing through the input file.
    chomp;
    # Catch globs.  Note that '{}' is not allowed in globs by this script,
    # so only '*' and '[]' are available.
    if (/^([ \t]*)([^ \t;{}#]+);?[ \t]*$/) {
    my $ws = $1;
    my $ptn = $2;
    # Turn the glob into a regex by replacing '*' with '.*', '?' with '.'.
    # Keep $ptn so we can still print the original form.
    ($pattern = $ptn) =~ s/\*/\.\*/g;
    $pattern =~ s/\?/\./g;
    if ($glob eq 'ign') {
        # We're in a local: * section; just continue.
        print "$_\n";
        next;
    }
    # Print the glob commented for human readers.
    print "$ws##$ptn ($glob)\n";
    # We tried to match a symbol to this version.
    $matches_attempted++;
    if ($glob eq 'glob') {
        my %ptn_syms = ();
        # Match ptn against symbols in %sym_hash.
        foreach my $sym (keys %sym_hash) {
        # Maybe it matches one of the patterns based on the symbol in
        # the .o file.
        $ptn_syms{$sym}++ if ($sym =~ /^$pattern$/);
        }
        foreach my $sym (sort keys(%ptn_syms)) {
        $matched_symbols++;
        print "$ws$sym;\n";
        }
    } elsif ($glob eq 'cxx') {
        my %dem_syms = ();
        # Verify that we're actually using GNU c++filt.  Other versions
        # most likely cannot handle GNU style symbol mangling.
        my $cxxout = `$cxxfilt --version 2>&1`;
        $cxxout =~ m/GNU/ or die "$0 requires GNU c++filt to function";
        # Talk to c++filt through a pair of file descriptors.
        # Need to start a fresh instance per pattern, otherwise the
        # process grows to 500+ MB.
        my $pid = open2(*FILTIN, *FILTOUT, $cxxfilt) or die $!;
        # Match ptn against symbols in %sym_hash.
        foreach my $sym (keys %sym_hash) {
        # No?  Well, maybe its demangled form matches one of those
        # patterns.
        printf FILTOUT "%s\n",$sym;
        my $dem = <FILTIN>;
        chomp $dem;
        $dem_syms{$sym}++ if ($dem =~ /^$pattern$/);
        }
        close FILTOUT or die "c++filt error";
        close FILTIN or die "c++filt error";
        # Need to wait for the c++filt process to avoid lots of zombies.
        waitpid $pid, 0;
        foreach my $sym (sort keys(%dem_syms)) {
        $matched_symbols++;
        print "$ws$sym;\n";
        }
    } else {
        # No?  Well, then ignore it.
    }
    next;
    }
    # Important sanity check.  This script can't handle lots of formats
    # that GNU ld can, so be sure to error out if one is seen!
    die "strange line `$_'";
 }
 close F;
--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@ -16,8 +16,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
-#ifndef COMPAT_TMS470_MATH_H
+#ifndef FFMPEG_COMPAT_TMS470_MATH_H
-#define COMPAT_TMS470_MATH_H
+#define FFMPEG_COMPAT_TMS470_MATH_H
 #include_next <math.h>
@ -27,4 +27,4 @@
 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
-#endif /* COMPAT_TMS470_MATH_H */
+#endif /* FFMPEG_COMPAT_TMS470_MATH_H */
--- a/compat/va_copy.h
+++ b/compat/va_copy.h
@ -19,9 +19,6 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_VA_COPY_H
 #define COMPAT_VA_COPY_H
 #include <stdarg.h>
 #if !defined(va_copy) && defined(_MSC_VER)
@ -30,5 +27,3 @@
 #if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
 #define va_copy(dst, src) __va_copy(dst, src)
 #endif
 #endif /* COMPAT_VA_COPY_H */
--- a/compat/w32pthreads.h
+++ b/compat/w32pthreads.h
@ -26,8 +26,8 @@
 * w32threads to pthreads wrapper
 */
-#ifndef COMPAT_W32PTHREADS_H
+#ifndef FFMPEG_COMPAT_W32PTHREADS_H
-#define COMPAT_W32PTHREADS_H
+#define FFMPEG_COMPAT_W32PTHREADS_H
 /* Build up a pthread-like API using underlying Windows API. Have only static
 * methods so as to not conflict with a potentially linked in pthread-win32
@ -39,11 +39,6 @@
 #include <windows.h>
 #include <process.h>
 #if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
 #undef MemoryBarrier
 #define MemoryBarrier __sync_synchronize
 #endif
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
@ -87,29 +82,19 @@ static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
 {
    thread->func   = start_routine;
    thread->arg    = arg;
 #if HAVE_WINRT
    thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
                                           0, NULL);
 #else
    thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
                                           0, NULL);
 #endif
    return !thread->handle;
 }
-static av_unused int pthread_join(pthread_t thread, void **value_ptr)
+static av_unused void pthread_join(pthread_t thread, void **value_ptr)
 {
    DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
-    if (ret != WAIT_OBJECT_0) {
+    if (ret != WAIT_OBJECT_0)
-        if (ret == WAIT_ABANDONED)
+        return;
            return EINVAL;
        else
            return EDEADLK;
    }
    if (value_ptr)
        *value_ptr = thread.ret;
    CloseHandle(thread.handle);
    return 0;
 }
 static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
@ -134,19 +119,6 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
 }
 #if _WIN32_WINNT >= 0x0600
 typedef INIT_ONCE pthread_once_t;
 #define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
 static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
 {
    BOOL pending = FALSE;
    InitOnceBeginInitialize(once_control, 0, &pending, NULL);
    if (pending)
        init_routine();
    InitOnceComplete(once_control, 0, NULL);
    return 0;
 }
 static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
    InitializeConditionVariable(cond);
@ -154,15 +126,14 @@ static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_att
 }
 /* native condition variables do not destroy */
-static inline int pthread_cond_destroy(pthread_cond_t *cond)
+static inline void pthread_cond_destroy(pthread_cond_t *cond)
 {
-    return 0;
+    return;
 }
-static inline int pthread_cond_broadcast(pthread_cond_t *cond)
+static inline void pthread_cond_broadcast(pthread_cond_t *cond)
 {
    WakeAllConditionVariable(cond);
    return 0;
 }
 static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
@ -171,77 +142,14 @@ static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex
    return 0;
 }
-static inline int pthread_cond_signal(pthread_cond_t *cond)
+static inline void pthread_cond_signal(pthread_cond_t *cond)
 {
    WakeConditionVariable(cond);
    return 0;
 }
 #else // _WIN32_WINNT < 0x0600
 /* atomic init state of dynamically loaded functions */
 static LONG w32thread_init_state = 0;
 static av_unused void w32thread_init(void);
 /* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
 * compatible to the one used in the native API */
 typedef union pthread_once_t  {
    void * Ptr;    ///< For the Windows 6.0+ native functions
    LONG state;    ///< For the pre-Windows 6.0 compat code
 } pthread_once_t;
 #define PTHREAD_ONCE_INIT {0}
 /* function pointers to init once API on windows 6.0+ kernels */
 static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
 static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
 /* pre-Windows 6.0 compat using a spin-lock */
 static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
 {
    switch (InterlockedCompareExchange(state, 1, 0)) {
    /* Initial run */
    case 0:
        init_routine();
        InterlockedExchange(state, 2);
        break;
    /* Another thread is running init */
    case 1:
        while (1) {
            MemoryBarrier();
            if (*state == 2)
                break;
            Sleep(0);
        }
        break;
    /* Initialization complete */
    case 2:
        break;
    }
 }
 static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
 {
    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
    /* Use native functions on Windows 6.0+ */
    if (initonce_begin && initonce_complete) {
        BOOL pending = FALSE;
        initonce_begin(once_control, 0, &pending, NULL);
        if (pending)
            init_routine();
        initonce_complete(once_control, 0, NULL);
        return 0;
    }
    w32thread_once_fallback(&once_control->state, init_routine);
    return 0;
 }
 /* for pre-Windows 6.0 platforms we need to define and use our own condition
 * variable and api */
 typedef struct  win32_cond_t {
    pthread_mutex_t mtx_broadcast;
    pthread_mutex_t mtx_waiter_count;
@ -261,9 +169,6 @@ static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
 static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
    win32_cond_t *win32_cond = NULL;
    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
    if (cond_init) {
        cond_init(cond);
        return 0;
@ -286,12 +191,12 @@ static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_
    return 0;
 }
-static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
+static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    /* native condition variables do not destroy */
    if (cond_init)
-        return 0;
+        return;
    /* non native condition variables */
    CloseHandle(win32_cond->semaphore);
@ -300,17 +205,16 @@ static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
    av_freep(&win32_cond);
    cond->Ptr = NULL;
    return 0;
 }
-static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
+static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    int have_waiter;
    if (cond_broadcast) {
        cond_broadcast(cond);
-        return 0;
+        return;
    }
    /* non native condition variables */
@ -332,7 +236,6 @@ static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
    } else
        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
    return 0;
 }
 static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
@ -367,13 +270,13 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
    return pthread_mutex_lock(mutex);
 }
-static av_unused int pthread_cond_signal(pthread_cond_t *cond)
+static av_unused void pthread_cond_signal(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->Ptr;
    int have_waiter;
    if (cond_signal) {
        cond_signal(cond);
-        return 0;
+        return;
    }
    pthread_mutex_lock(&win32_cond->mtx_broadcast);
@ -390,7 +293,6 @@ static av_unused int pthread_cond_signal(pthread_cond_t *cond)
    }
    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
    return 0;
 }
 #endif
@ -407,12 +309,8 @@ static av_unused void w32thread_init(void)
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
    initonce_begin =
        (void*)GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
    initonce_complete =
        (void*)GetProcAddress(kernel_dll, "InitOnceComplete");
 #endif
 }
-#endif /* COMPAT_W32PTHREADS_H */
+#endif /* FFMPEG_COMPAT_W32PTHREADS_H */
--- a/compat/windows/mslink
+++ b/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 $?
--- a/1817
+++ b/1817
--- a/doc/.gitignore
+++ b/doc/.gitignore
@ -1,9 +0,0 @@
 /*.1
 /*.3
 /*.html
 /*.pod
 /config.texi
 /avoptions_codec.texi
 /avoptions_format.texi
 /fate.txt
 /print_options
--- a/doc/APIchanges
+++ b/doc/APIchanges
@ -2,383 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 The last version increases were:
-libavcodec:    2015-08-28
+libavcodec:    2014-08-09
-libavdevice:   2015-08-28
+libavdevice:   2014-08-09
-libavfilter:   2015-08-28
+libavfilter:   2014-08-09
-libavformat:   2015-08-28
+libavformat:   2014-08-09
-libavresample: 2015-08-28
+libavresample: 2014-08-09
-libpostproc:   2015-08-28
+libpostproc:   2014-08-09
-libswresample: 2015-08-28
+libswresample: 2014-08-09
-libswscale:    2015-08-28
+libswscale:    2014-08-09
-libavutil:     2015-08-28
+libavutil:     2014-08-09
 API changes, most recent first:
 2016-07-09 - xxxxxxx / 90f469a - lavc 57.50.100 / 57.20.0 - avcodec.h
  Add FF_PROFILE_H264_MULTIVIEW_HIGH and FF_PROFILE_H264_STEREO_HIGH.
 2016-06-30 - c1c7e0ab - lavf 57.41.100 - avformat.h
  Moved codecpar field from AVStream to the end of the struct, so that
  the following private fields are in the same location as in FFmpeg 3.0 (lavf 57.25.100).
 2016-06-30 - 042fb69d - lavu 55.28.100 - frame.h
  Moved hw_frames_ctx field from AVFrame to the end of the struct, so that
  the following private fields are in the same location as in FFmpeg 3.0 (lavu 55.17.103).
 2016-06-29 - 1a751455 - lavfi 6.47.100 - avfilter.h
  Fix accidental ABI breakage in AVFilterContext.
  ABI was broken in 8688d3a, lavfi 6.42.100 and released as ffmpeg 3.1.
  Because of this, ffmpeg and ffplay built against lavfi>=6.42.100 will not be
  compatible with lavfi>=6.47.100. Potentially also affects other users of
  libavfilter if they are using one of the affected fields.
 -------- 8< --------- FFmpeg 3.1 was cut here -------- 8< ---------
 2016-06-26 - 481f320 / 1c9e861 - lavu 55.27.100 / 55.13.0 - hwcontext.h
  Add av_hwdevice_ctx_create().
 2016-06-26 - b95534b / e47b8bb - lavc 57.48.101 / 57.19.1 - avcodec.h
  Adjust values for JPEG 2000 profiles.
 2016-06-23 - 5d75e46 / db7968b - lavf 57.40.100 / 57.7.0 - avio.h
  Add AVIODataMarkerType, write_data_type, ignore_boundary_point and
  avio_write_marker.
 2016-06-23 - abb3cc4 / 0c4468d - lavu 55.26.100 / 55.12.0 - opt.h
  Add av_stereo3d_type_name() and av_stereo3d_from_name().
 2016-06-22 - 3689efe / c46db38 - lavu 55.25.100 / 55.11.0 - hwcontext_dxva2.h
  Add new installed header with DXVA2-specific hwcontext definitions.
 2016-04-27 - fb91871 - lavu 55.23.100 - log.h
  Add a new function av_log_format_line2() which returns number of bytes
  written to the target buffer.
 2016-04-21 - 7fc329e - lavc 57.37.100 - avcodec.h
  Add a new audio/video encoding and decoding API with decoupled input
  and output -- avcodec_send_packet(), avcodec_receive_frame(),
  avcodec_send_frame() and avcodec_receive_packet().
 2016-04-17 - af9cac1 / 33d1898 - lavc 57.35.100 / 57.15.0 - avcodec.h
  Add a new bitstream filtering API working with AVPackets.
  Deprecate the old bitstream filtering API.
 2016-04-14 - 8688d3a / 07a844f - lavfi 6.42.100 / 6.3.0 - avfilter.h
  Add AVFilterContext.hw_device_ctx.
 2016-04-14 - 28abb21 / 551c677 - lavu 55.22.100 / 55.9.0 - hwcontext_vaapi.h
  Add new installed header with VAAPI-specific hwcontext definitions.
 2016-04-14 - afccfaf / b1f01e8 - lavu 55.21.100 / 55.7.0 - hwcontext.h
  Add AVHWFramesConstraints and associated API.
 2016-04-11 - 6f69f7a / 9200514 - lavf 57.33.100 / 57.5.0 - avformat.h
  Add AVStream.codecpar, deprecate AVStream.codec.
 2016-04-02 - e8a9b64 - lavu 55.20.100 - base64.h
  Add AV_BASE64_DECODE_SIZE(x) macro.
 2016-xx-xx - lavc 57.33.100 / 57.14.0 - avcodec.h
  f9b1cf1 / 998e1b8 - Add AVCodecParameters and its related API.
  e6053b3 / a806834 - Add av_get_audio_frame_duration2().
 2016-03-11 - 6d8ab35 - lavf/lavc 57.28.101
  Add requirement to bitstream filtering API that returned packets with
  size == 0 and side_data_elems == 0 are to be skipped by the caller.
 2016-03-04 - 9362973 - lavf 57.28.100
  Add protocol blacklisting API
 2016-02-28 - 4dd4d53 - lavc 57.27.101
  Validate AVFrame returned by get_buffer2 to have required
  planes not NULL and unused planes set to NULL as crashes
  and buffer overflow are possible with certain streams if
  that is not the case.
 2016-02-26 - 30e7685 - lavc 57.27.100 - avcodec.h
  "flags2" decoding option now allows the flag "ass_ro_flush_noop" preventing
  the reset of the ASS ReadOrder field on flush. This affects the content of
  AVSubtitles.rects[N]->ass when "sub_text_format" is set to "ass" (see
  previous entry).
 2016-02-26 - 2941282 - lavc 57.26.100 - avcodec.h
  Add a "sub_text_format" subtitles decoding option allowing the values "ass"
  (recommended) and "ass_with_timings" (not recommended, deprecated, default).
  The default value for this option will change to "ass" at the next major
  libavcodec version bump.
  The current default is "ass_with_timings" for compatibility. This means that
  all subtitles text decoders currently still output ASS with timings printed
  as strings in the AVSubtitles.rects[N]->ass fields.
  Setting "sub_text_format" to "ass" allows a better timing accuracy (ASS
  timing is limited to a 1/100 time base, so this is relevant for any subtitles
  format needing a bigger one), ease timing adjustments, and prevents the need
  of removing the timing from the decoded string yourself. This form is also
  known as "the Matroska form". The timing information (start time, duration)
  can be found in the AVSubtitles fields.
 2016-02-24 - 7e49cdd / 7b3214d0 - lavc 57.25.100 / 57.13.0 - avcodec.h
  Add AVCodecContext.hw_frames_ctx.
 2016-02-24 - 1042402 / b3dd30d - lavfi 6.36.100 / 6.2.0 - avfilter.h
  avfilter.h - Add AVFilterLink.hw_frames_ctx.
  buffersrc.h - Add AVBufferSrcParameters and functions for handling it.
 2016-02-23 - 14f7a3d - lavc 57.25.100
  Add AV_PKT_DATA_MPEGTS_STREAM_ID for exporting the MPEGTS stream ID.
 2016-02-18 - 08acab8 - lavu 55.18.100 - audio_fifo.h
  Add av_audio_fifo_peek_at().
 2016-xx-xx - lavu 55.18.100 / 55.6.0
  26abd51 / 721a4ef buffer.h - Add av_buffer_pool_init2().
  1a70878 / 89923e4 hwcontext.h - Add a new installed header hwcontext.h with a new API
                        for handling hwaccel frames.
  6992276 / ad884d1 hwcontext_cuda.h - Add a new installed header hwcontext_cuda.h with
                             CUDA-specific hwcontext definitions.
  d779d8d / a001ce3 hwcontext_vdpau.h - Add a new installed header hwcontext_vdpau.h with
                              VDPAU-specific hwcontext definitions.
  63c3e35 / 7bc780c pixfmt.h - Add AV_PIX_FMT_CUDA.
 -------- 8< --------- FFmpeg 3.0 was cut here -------- 8< ---------
 2016-02-10 - bc9a596 / 9f61abc - lavf 57.25.100 / 57.3.0 - avformat.h
  Add AVFormatContext.opaque, io_open and io_close, allowing custom IO
 2016-02-01 - 1dba837 - lavf 57.24.100 - avformat.h, avio.h
  Add protocol_whitelist to AVFormatContext, AVIOContext
 2016-01-31 - 66e9d2f - lavu 55.17.100 - frame.h
  Add AV_FRAME_DATA_GOP_TIMECODE for exporting MPEG1/2 GOP timecodes.
 2016-01-01 - 5e8b053 / 2c68113 - lavc 57.21.100 / 57.12.0 - avcodec.h
  Add AVCodecDescriptor.profiles and avcodec_profile_name().
 2015-12-28 - 1f9139b - lavf 57.21.100 - avformat.h
  Add automatic bitstream filtering; add av_apply_bitstream_filters()
 2015-12-22 - 39a09e9 - lavfi 6.21.101 - avfilter.h
  Deprecate avfilter_link_set_closed().
  Applications are not supposed to mess with links,
  they should close the sinks.
 2015-12-17 - lavc 57.18.100 / 57.11.0 - avcodec.h dirac.h
  xxxxxxx - Add av_packet_add_side_data().
  xxxxxxx - Add AVCodecContext.coded_side_data.
  xxxxxxx - Add AVCPBProperties API.
  xxxxxxx - Add a new public header dirac.h containing
            av_dirac_parse_sequence_header()
 2015-12-11 - 676a93f - lavf 57.20.100 - avformat.h
  Add av_program_add_stream_index()
 2015-11-29 - 93fb4a4 - lavc 57.16.101 - avcodec.h
  Deprecate rtp_callback without replacement, i.e. it won't be possible to
  get image slices before the full frame is encoded any more. The libavformat
  rtpenc muxer can still be used for RFC-2190 packetization.
 2015-11-22 - fe20e34 - lavc 57.16.100 - avcodec.h
  Add AV_PKT_DATA_FALLBACK_TRACK for making fallback associations between
  streams.
 2015-11-22 - ad317c9 - lavf 57.19.100 - avformat.h
  Add av_stream_new_side_data().
 2015-11-22 - e12f403 - lavu 55.8.100 - xtea.h
    Add av_xtea_le_init and av_xtea_le_crypt
 2015-11-18 - lavu 55.7.100 - mem.h
  Add av_fast_mallocz()
 2015-10-29 - lavc 57.12.100 / 57.8.0 - avcodec.h
  xxxxxx - Deprecate av_free_packet(). Use av_packet_unref() as replacement,
           it resets the packet in a more consistent way.
  xxxxxx - Deprecate av_dup_packet(), it is a no-op for most cases.
           Use av_packet_ref() to make a non-refcounted AVPacket refcounted.
  xxxxxx - Add av_packet_alloc(), av_packet_clone(), av_packet_free().
           They match the AVFrame functions with the same name.
 2015-10-27 - 1e477a9 - lavu 55.5.100 - cpu.h
  Add AV_CPU_FLAG_AESNI.
 2015-10-22 - ee573b4 / a17a766 - lavc 57.9.100 / 57.5.0 - avcodec.h
  Add data and linesize array to AVSubtitleRect, to be used instead of
  the ones from the embedded AVPicture.
 2015-10-22 - 866a417 / dc923bc - lavc 57.8.100 / 57.0.0 - qsv.h
  Add an API for allocating opaque surfaces.
 2015-10-15 - 2c2d162 - lavf 57.4.100
  Remove the latm demuxer that was a duplicate of the loas demuxer.
 2015-10-14 - b994788 / 11c5f43 - lavu 55.4.100 / 55.2.0 - dict.h
  Change return type of av_dict_copy() from void to int, so that a proper
  error code can be reported.
 2015-09-29 - b01891a / 948f3c1 - lavc 57.3.100 / 57.2.0 - avcodec.h
  Change type of AVPacket.duration from int to int64_t.
 2015-09-17 - 7c46f24 / e3d4784 - lavc 57.3.100 / 57.2.0 - d3d11va.h
  Add av_d3d11va_alloc_context(). This function must from now on be used for
  allocating AVD3D11VAContext.
 2015-09-15 - lavf 57.2.100 - avformat.h
  probesize and max_analyze_duration switched to 64bit, both
  are only accessible through AVOptions
 2015-09-15 - lavf 57.1.100 - avformat.h
  bit_rate was changed to 64bit, make sure you update any
  printf() or other type sensitive code
 2015-09-15 - lavc 57.2.100 - avcodec.h
  bit_rate/rc_max_rate/rc_min_rate were changed to 64bit, make sure you update
  any printf() or other type sensitive code
 2015-09-07 - lavu 55.0.100 / 55.0.0
  c734b34 / b8b5d82 - Change type of AVPixFmtDescriptor.flags from uint8_t to uint64_t.
  f53569a / 6b3ef7f - Change type of AVComponentDescriptor fields from uint16_t to int
            and drop bit packing.
  151aa2e / 2268db2 - Add step, offset, and depth to AVComponentDescriptor to replace
            the deprecated step_minus1, offset_plus1, and depth_minus1.
 -------- 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
  Add avformat_flush()
 2015-03-03 - 81a9126 - lavf 56.24.100
  Add avio_put_str16be()
 2015-02-19 - 560eb71 / 31d2039 - lavc 56.23.100 / 56.13.0
  Add width, height, coded_width, coded_height and format to
  AVCodecParserContext.
 2015-02-19 - e375511 / 5b1d9ce - lavu 54.19.100 / 54.9.0
  Add AV_PIX_FMT_QSV for QSV hardware acceleration.
 2015-02-14 - ba22295 - lavc 56.21.102
  Deprecate VIMA decoder.
 2015-01-27 - 62a82c6 / 728685f - lavc 56.21.100 / 56.12.0, lavu 54.18.100 / 54.8.0 - avcodec.h, frame.h
  Add AV_PKT_DATA_AUDIO_SERVICE_TYPE and AV_FRAME_DATA_AUDIO_SERVICE_TYPE for
  storing the audio service type as side data.
 2015-01-16 - a47c933 - lavf 56.19.100 - avformat.h
  Add data_codec and data_codec_id for storing codec of data stream
 2015-01-11 - 007c33d - lavd 56.4.100 - avdevice.h
  Add avdevice_list_input_sources().
  Add avdevice_list_output_sinks().
 2014-12-25 - d7aaeea / c220a60 - lavc 56.19.100 / 56.10.0 - vdpau.h
  Add av_vdpau_get_surface_parameters().
 2014-12-25 - ddb9a24 / 6c99c92 - lavc 56.18.100 / 56.9.0 - avcodec.h
  Add AV_HWACCEL_FLAG_ALLOW_HIGH_DEPTH flag to av_vdpau_bind_context().
 2014-12-25 - d16079a / 57b6704 - lavc 56.17.100 / 56.8.0 - avcodec.h
  Add AVCodecContext.sw_pix_fmt.
 2014-12-04 - 6e9ac02 - lavc 56.14.100 - dv_profile.h
  Add av_dv_codec_profile2().
 -------- 8< --------- FFmpeg 2.5 was cut here -------- 8< ---------
 2014-11-21 - ab922f9 - lavu 54.15.100 - dict.h
   Add av_dict_get_string().
@ -434,12 +70,12 @@ API changes, most recent first:
 -------- 8< --------- FFmpeg 2.4 was cut here -------- 8< ---------
 2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
  Add AVFormatContext.max_ts_probe.
 2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h
  Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information.
 2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
  Add AVFormatContext.max_ts_probe.
 2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h
  AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now
  freed even on error. This is consistent with the behaviour all users
@ -617,7 +253,7 @@ API changes, most recent first:
  Add avcodec_free_context(). From now on it should be used for freeing
  AVCodecContext.
-2014-05-17 - 0eec06e / 1bd0bdc - lavu 52.84.100 / 54.5.0 - time.h
+2014-05-17 - 0eec06e - lavu 52.84.100 - time.h
  Add av_gettime_relative() av_gettime_relative_is_monotonic()
 2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
@ -957,9 +593,6 @@ 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
@ -1212,7 +845,7 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
                * base -- is now stored in AVBufferRef
                * reference, type, buffer_hints -- are unnecessary in the new API
                * hwaccel_picture_private, owner, thread_opaque -- should not
-                  have been accessed from outside of lavc
+                  have been acessed from outside of lavc
                * qscale_table, qstride, qscale_type, mbskip_table, motion_val,
                  mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
                  which are not exported anymore.
@ -1249,14 +882,15 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
  Add avresample_set_channel_mapping() for input channel reordering,
  duplication, and silencing.
-2012-12-29 - lavu 52.13.100 / 52.3.0 - avstring.h
+2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h
-  2ce43b3 / d8fd06c - Add av_basename() and av_dirname().
+  Add av_basename() and av_dirname().
  e13d5e9 / c1a02e8 - Add av_pix_fmt_get_chroma_sub_sample and deprecate
                      avcodec_get_chroma_sub_sample.
 2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h
  Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated.
 2012-11-05 - 7d26be6 / dfde8a3 - lavu 52.5.100 / 52.1.0 - intmath.h
  Add av_ctz() for trailing zero bit count
 2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h
  Add AVERROR_EXPERIMENTAL
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
--- a/doc/Makefile
+++ b/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-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
@ -47,7 +46,6 @@ DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
 DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
 DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
 DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
 DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
 DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
 DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
 DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
@ -116,20 +114,19 @@ doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 doc/%.1 doc/%.3: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
+	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
 doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
+	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
 $(DOCS) doc/doxy/html: | doc/
 $(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
 OBJDIRS += doc/examples
-DOXY_INPUT      = $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)
+DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
 DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) config.mak
 doc/doxy/html: TAG = DOXY
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
-	$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
 install-doc: install-html install-man
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@ -67,10 +67,6 @@ the header stored in extradata to the key packets:
 ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dca_core
 Extract DCA core from DTS-HD streams.
@section h264_mp4toannexb
 Convert an H.264 bitstream from length prefixed mode to start code
@ -143,26 +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
--- a/doc/build_system.txt
+++ b/doc/build_system.txt
@ -7,28 +7,20 @@ V
    Disable the default terse mode, the full command issued by make and its
    output will be shown on the screen.
 DBG
    Preprocess x86 external assembler files to a .dbg.asm file in the object
    directory, which then gets compiled. Helps in developing those assembler
    files.
 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
    Default target, builds all the libraries and the executables.
 fate
-    Run the fate test suite, note that you must have installed it.
+    Run the fate test suite, note you must have installed it
 fate-list
-    List all fate/regression test targets.
+    Will list all fate/regression test targets
 install
    Install headers, libraries and programs.
@ -39,23 +31,26 @@ examples
 libavformat/output-example
    Build the libavformat basic example.
 libavcodec/api-example
    Build the libavcodec basic example.
 libswscale/swscale-test
-    Build the swscale self-test (useful also as an example).
+    Build the swscale self-test (useful also as example).
 config
-    Reconfigure the project with the current configuration.
+    Reconfigure the project with current configuration.
 Useful standard make commands:
 make -t <target>
-    Touch all files that otherwise would be built, this is useful to reduce
+    Touch all files that otherwise would be build, this is useful to reduce
-    unneeded rebuilding when changing headers, but note that you must force rebuilds
+    unneeded rebuilding when changing headers, but note you must force rebuilds
    of files that actually need it by hand then.
 make -j<num>
-    Rebuild with multiple jobs at the same time. Faster on multi processor systems.
+    rebuild with multiple jobs at the same time. Faster on multi processor systems
 make -k
-    Continue build in case of errors, this is useful for the regression tests
+    continue build in case of errors, this is useful for the regression tests
-    sometimes but note that it will still not run all reg tests.
+    sometimes but note it will still not run all reg tests.
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
@ -7,7 +7,7 @@ all the encoders and decoders. In addition each codec may support
 so-called private options, which are specific for a given codec.
 Sometimes, a global option may only affect a specific kind of codec,
-and may be nonsensical or ignored by another, so you need to be aware
+and may be unsensical or ignored by another, so you need to be aware
 of the meaning of the specified options. Also some options are
 meant only for decoding or encoding.
@ -129,7 +129,7 @@ should be @code{1 / frame_rate} and timestamp increments should be
 identically 1.
@item g @var{integer} (@emph{encoding,video})
-Set the group of picture (GOP) size. Default value is 12.
+Set the group of picture size. Default value is 12.
@item ar @var{integer} (@emph{decoding/encoding,audio})
 Set audio sampling rate (in Hz).
@ -257,7 +257,7 @@ Specify how strictly to follow the standards.
 Possible values:
@table @samp
@item very
-strictly conform to an older more strict version of the spec or reference software
+strictly conform to a older more strict version of the spec or reference software
@item strict
 strictly conform to all the things in the spec no matter what consequences
@item normal
@ -456,9 +456,6 @@ Possible values:
@item aspect @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio.
@item sar @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio. Alias to @var{aspect}.
@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
 Print specific debug info.
@ -478,9 +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
@ -501,8 +495,6 @@ visualize block types
 picture buffer allocations
@item thread_ops
 threading operations
@item nomc
 skip motion compensation
@end table
@item vismv @var{integer} (@emph{decoding,video})
@ -820,17 +812,13 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
 Deprecated, use mpegvideo private options instead.
@item threads @var{integer} (@emph{decoding/encoding,video})
 Set the number of threads to be used, in case the selected codec
 implementation supports multi-threading.
 Possible values:
@table @samp
-@item auto, 0
+@item auto
-automatically select the number of threads to set
+detect a good number of threads
@end table
 Default value is @samp{auto}.
@item me_threshold @var{integer} (@emph{encoding,video})
 Set motion estimation threshold.
@ -875,14 +863,6 @@ Possible values:
@item mpeg2_aac_he
@item mpeg4_sp
@item mpeg4_core
@item mpeg4_main
@item mpeg4_asp
@item dts
@item dts_es
@ -1050,44 +1030,9 @@ Possible values:
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
@item color_trc @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@table @samp
@item bt709
 BT.709
@item gamma22
 BT.470 M
@item gamma28
 BT.470 BG
@item linear
 SMPTE 170 M
@item log
 SMPTE 240 M
@item log_sqrt
 Linear
@item iec61966_2_4
 Log
@item bt1361
 Log square root
@item iec61966_2_1
 IEC 61966-2-4
@item bt2020_10bit
 BT.1361
@item bt2020_12bit
 IEC 61966-2-1
@item smpte2084
 BT.2020 - 10 bit
@item smpte428_1
 BT.2020 - 12 bit
@end table
@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}
--- a/doc/decoders.texi
+++ b/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
@ -279,16 +236,10 @@ present between the subtitle lines because of double-sized teletext charactes.
 Default value is 1.
@item txt_duration
 Sets the display duration of the decoded teletext pages or subtitles in
-milliseconds. Default value is 30000 which is 30 seconds.
+miliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
 Force transparent background of the generated teletext bitmaps. Default value
-is 0 which means an opaque background.
+is 0 which means an opaque (black) background.
@item txt_opacity
 Sets the opacity (0-255) of the teletext background. If
@option{txt_transparent} is not set, it only affects characters between a start
 box and an end box, typically subtitles. Default value is 0 if
@option{txt_transparent} is set, 255 otherwise.
@end table
@c man end SUBTILES DECODERS
--- a/doc/demuxers.texi
+++ b/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.
@ -104,7 +98,7 @@ All subsequent file-related directives apply to that file.
@item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
-to 1 if it was -1.
+to 1 if it was to its default -1.
 To make FFmpeg recognize the format automatically, this directive must
 appears exactly as is (no extra space or byte-order-mark) on the very first
@ -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
@ -192,57 +145,19 @@ component.
 If set to 0, any file name is accepted.
-The default is 1.
+The default is -1, it is equivalent to 1 if the format was automatically
 -1 is equivalent to 1 if the format was automatically
 probed and 0 otherwise.
@item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
 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
 there are resolution changes.
@item segment_time_metadata
 If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
@var{lavf.concat.duration} packet metadata values which are the start_time and
 the duration of the respective file segments in the concatenated output
 expressed in microseconds. The duration metadata is only set if it is known
 based on the concat file.
 The default is 0.
@end table
@subsection Examples
@itemize
@item
 Use absolute filenames and include some comments:
@example
 # my first filename
 file /mnt/share/file-1.wav
 # my second filename including whitespace
 file '/mnt/share/file 2.wav'
 # my third filename including whitespace plus single quote
 file '/mnt/share/file 3'\''.wav'
@end example
@item
 Allow for input format auto-probing, use safe filenames and set the duration of
 the first file:
@example
 ffconcat version 1.0
 file file-1.wav
 duration 20.0
 file subdir/file-2.wav
@end example
@end itemize
@section flv
 Adobe Flash Video Format demuxer.
@ -267,32 +182,17 @@ track. Track indexes start at 0. The demuxer exports the number of tracks as
 For very large files, the @option{max_size} option may have to be adjusted.
-@section libopenmpt
+@section libquvi
-libopenmpt based module demuxer
+Play media from Internet services using the quvi project.
-See @url{https://lib.openmpt.org/libopenmpt/} for more information.
+The demuxer accepts a @option{format} option to request a specific quality. It
 is by default set to @var{best}.
-Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
+See @url{http://quvi.sourceforge.net/} for more information.
 option.
-It accepts the following options:
+FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
-
+enabled.
@table @option
@item subsong
 Set the subsong index. This can be either  'all', 'auto', or the index of the
 subsong. Subsong indexes start at 0. The default is 'auto'.
 The default value is to let libopenmpt choose.
@item layout
 Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
 The default value is STEREO.
@item sample_rate
 Set the sample rate for libopenmpt to output.
 Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section gif
@ -305,11 +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.
@ -358,10 +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.
@ -468,9 +359,9 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
-@section mov/mp4/3gp/QuickTime
+@section mov/mp4/3gp/Quicktme
-QuickTime / MP4 demuxer.
+Quicktime / MP4 demuxer.
 This demuxer accepts the following options:
@table @option
@ -489,41 +380,13 @@ is known to be non malicious.
 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 mpjpeg
 MJPEG encapsulated in multi-part MIME demuxer.
 This demuxer allows reading of MJPEG, where each frame is represented as a part of
 multipart/x-mixed-replace stream.
@table @option
@item strict_mime_boundary
 Default implementation applies a relaxed standard to multi-part MIME boundary detection,
 to prevent regression with numerous existing endpoints not generating a proper MIME
 MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
 of the boundary value.
@end table
@section rawvideo
--- a/doc/developer.texi
+++ b/doc/developer.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Developer Documentation
@titlepage
@ -28,14 +27,14 @@ this document.
 For more detailed legal information about the use of FFmpeg in
 external programs read the @file{LICENSE} file in the source tree and
-consult @url{https://ffmpeg.org/legal.html}.
+consult @url{http://ffmpeg.org/legal.html}.
@section Contributing
-There are 3 ways by which code gets into FFmpeg.
+There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
-@item Submitting patches to the main developer mailing list.
+@item Submitting Patches to the main developer mailing list
-      See @ref{Submitting patches} for details.
+      see @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
      gitorious.org. And asking us to merge these changes.
@ -65,9 +64,6 @@ rejected by the git repository.
@item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@item
 K&R coding style is used.
@end itemize
 The presentation is one inspired by 'indent -i4 -kr -nut'.
@ -127,10 +123,10 @@ the @samp{inline} keyword;
@samp{//} comments;
@item
-designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
+designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
-compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
+compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
 These features are supported by all compilers we care about, so we will not
@ -159,7 +155,7 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
-for example structs and enums; they should always be in CamelCase.
+for example structs and enums; they should always be in the CamelCase
 There are the following conventions for naming variables and functions:
@ -231,7 +227,7 @@ 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)
@ -242,7 +238,7 @@ 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
@ -397,41 +393,12 @@ or obfuscates the code.
 Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
 your name after it.
-If at some point you no longer want to maintain some code, then please help in
+If at some point you no longer want to maintain some code, then please help
-finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
+finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
@end enumerate
 We think our rules are not too hard. If you have comments, contact us.
@section Code of conduct
 Be friendly and respectful towards others and third parties.
 Treat others the way you yourself want to be treated.
 Be considerate. Not everyone shares the same viewpoint and priorities as you do.
 Different opinions and interpretations help the project.
 Looking at issues from a different perspective assists development.
 Do not assume malice for things that can be attributed to incompetence. Even if
 it is malice, it's rarely good to start with that as initial assumption.
 Stay friendly even if someone acts contrarily. Everyone has a bad day
 once in a while.
 If you yourself have a bad day or are angry then try to take a break and reply
 once you are calm and without anger if you have to.
 Try to help other team members and cooperate if you can.
 The goal of software development is to create technical excellence, not for any
 individual to be better and "win" against the others. Large software projects
 are only possible and successful through teamwork.
 If someone struggles do not put them down. Give them a helping hand
 instead and point them in the right direction.
 Finally, keep in mind the immortal words of Bill and Ted,
 "Be excellent to each other."
@anchor{Submitting patches}
@section Submitting patches
@ -439,7 +406,7 @@ First, read the @ref{Coding Rules} above if you did not yet, in particular
 the rules regarding patch submission.
 When you submit your patch, please use @code{git format-patch} or
-@code{git send-email}. We cannot read other diffs :-).
+@code{git send-email}. We cannot read other diffs :-)
 Also please do not submit a patch which contains several unrelated changes.
 Split it into separate, self-contained pieces. This does not mean splitting
@ -462,7 +429,7 @@ Also please if you send several patches, send each patch as a separate mail,
 do not attach several unrelated patches to the same mail.
 Patches should be posted to the
-@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
+@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
 mailing list. Use @code{git send-email} when possible since it will properly
 send patches without requiring extra care. If you cannot, then send patches
 as base64-encoded attachments, so your patch is not trashed during
@ -575,10 +542,6 @@ 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{https://samples.ffmpeg.org}.
@item
 Does the patch not mix functional and cosmetic changes?
@ -599,7 +562,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
 If the patch fixes a bug, did you provide enough information, including
 a sample, so the bug can be reproduced and the fix can be verified?
 Note please do not attach samples >100k to mails but rather provide a
-URL, you can upload to ftp://upload.ffmpeg.org.
+URL, you can upload to ftp://upload.ffmpeg.org
@item
 Did you provide a verbose summary about what the patch does change?
@ -628,10 +591,10 @@ Lines with similar content should be aligned vertically when doing so
 improves readability.
@item
-Consider adding a regression test for your code.
+Consider to add a regression test for your code.
@item
-If you added YASM code please check that things still work with --disable-yasm.
+If you added YASM code please check that things still work with --disable-yasm
@item
 Make sure you check the return values of function and return appropriate
@ -669,10 +632,6 @@ 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
@ -688,14 +647,15 @@ accordingly].
@subsection Adding files to the fate-suite dataset
 When there is no muxer or encoder available to generate test media for a
-specific test then the media has to be included in the fate-suite.
+specific test then the media has to be inlcuded in the fate-suite.
 First please make sure that the sample file is as small as possible to test the
 respective decoder or demuxer sufficiently. Large files increase network
 bandwidth and disk space requirements.
 Once you have a working fate test and fate sample, provide in the commit
-message or introductory message for the patch series that you post to
+message or introductionary message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.
@subsection Visualizing Test Coverage
 The FFmpeg build system allows visualizing the test coverage in an easy
@ -743,7 +703,7 @@ FFmpeg maintains a set of @strong{release branches}, which are the
 recommended deliverable for system integrators and distributors (such as
 Linux distributions, etc.). At regular times, a @strong{release
 manager} prepares, tests and publishes tarballs on the
-@url{https://ffmpeg.org} website.
+@url{http://ffmpeg.org} website.
 There are two kinds of releases:
@ -822,7 +782,7 @@ Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
 supplementing files that contain @code{gpg} signatures
@item
-Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and
+Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
 push an annotated tag in the form @code{nX}, with @code{X}
 containing the version number.
@ -834,7 +794,7 @@ with a news entry for the website.
 Publish the news entry.
@item
-Send an announcement to the mailing list.
+Send announcement to the mailing list.
@end enumerate
@bye
--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@ -1,21 +1,13 @@
 #!/bin/sh
-OUT_DIR="${1}"
+SRC_PATH="${1}"
 DOXYFILE="${2}"
 DOXYGEN="${3}"
 shift 3
 if [ -e "VERSION" ]; then
    VERSION=`cat "VERSION"`
 else
    VERSION=`git describe`
 fi
 $DOXYGEN - <<EOF
@INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-HTML_TIMESTAMP  = NO
+EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
 PROJECT_NUMBER  = $VERSION
 OUTPUT_DIRECTORY = $OUT_DIR
 EOF
--- a/doc/doxy/.gitignore
+++ b/doc/doxy/.gitignore
@ -1 +0,0 @@
 /html/
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@ -30,119 +30,81 @@ follows.
 Advanced Audio Coding (AAC) encoder.
-This encoder is the default AAC encoder, natively implemented into FFmpeg. Its
+This encoder is an experimental FFmpeg-native AAC encoder. Currently only the
-quality is on par or better than libfdk_aac at the default bitrate of 128kbps.
+low complexity (AAC-LC) profile is supported. To use this encoder, you must set
-This encoder also implements more options, profiles and samplerates than
+@option{strict} option to @samp{experimental} or lower.
-other encoders (with only the AAC-HE profile pending to be implemented) so this
+
-encoder has become the default and is the recommended choice.
+As this encoder is experimental, unexpected behavior may exist from time to
 time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
 that it has a worse quality reported by some users.
@c todo @ref{libaacplus}
 See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
@subsection Options
@table @option
@item b
 Set bit rate in bits/s. Setting this automatically activates constant bit rate
-(CBR) mode. If this option is unspecified it is set to 128kbps.
+(CBR) mode.
@item q
 Set quality for variable bit rate (VBR) mode. This option is valid only using
 the @command{ffmpeg} command-line tool. For library interface users, use
@option{global_quality}.
-@item cutoff
+@item stereo_mode
-Set cutoff frequency. If unspecified will allow the encoder to dynamically
+Set stereo encoding mode. Possible values:
-adjust the cutoff to improve clarity on low bitrates.
+
@table @samp
@item auto
 Automatically selected by the encoder.
@item ms_off
 Disable middle/side encoding. This is the default.
@item ms_force
 Force middle/side encoding.
@end table
@item aac_coder
 Set AAC encoder coding method. Possible values:
@table @samp
@item faac
 FAAC-inspired method.
 This method is a simplified reimplementation of the method used in FAAC, which
 sets thresholds proportional to the band energies, and then decreases all the
 thresholds with quantizer steps to find the appropriate quantization with
 distortion below threshold band by band.
 The quality of this method is comparable to the two loop searching method
 described below, but somewhat a little better and slower.
@item anmr
 Average noise to mask ratio (ANMR) trellis-based solution.
 This has a theoretic best quality out of all the coding methods, but at the
 cost of the slowest speed.
@item twoloop
 Two loop searching (TLS) method.
 This method first sets quantizers depending on band thresholds and then tries
 to find an optimal combination by adding or subtracting a specific value from
 all quantizers and adjusting some individual quantizer a little.
 Will tune itself based on whether aac_is/aac_ms/aac_pns are enabled.
 This is the default choice for a coder.
-@item anmr
+This method produces similar quality with the FAAC method and is the default.
 Average noise to mask ratio (ANMR) trellis-based solution.
 This is an experimental coder which currently produces a lower quality, is more
 unstable and is slower than the default twoloop coder but has potential.
 Currently has no support for the @option{aac_is} or @option{aac_pns} options.
 Not currently recommended.
@item fast
 Constant quantizer method.
 This method sets a constant quantizer for all bands. This is the fastest of all
-the methods and has no rate control or support for @option{aac_is} or
+the methods, yet produces the worst quality.
@option{aac_pns}.
 Not recommended.
@end table
@item aac_ms
 Sets mid/side coding mode. The default value of auto will automatically use
 M/S with bands which will benefit from such coding. Can be forced for all bands
 using the value "enable", which is mainly useful for debugging or disabled using
 "disable".
@item aac_is
 Sets intensity stereo coding tool usage. By default, it's enabled and will
 automatically toggle IS for similar pairs of stereo bands if it's benefitial.
 Can be disabled for debugging by setting the value to "disable".
@item aac_pns
 Uses perceptual noise substitution to replace low entropy high frequency bands
 with imperceivable white noise during the decoding process. By default, it's
 enabled, but can be disabled for debugging purposes by using "disable".
@item aac_tns
 Enables the use of a multitap FIR filter which spans through the high frequency
 bands to hide quantization noise during the encoding process and is reverted
 by the decoder. As well as decreasing unpleasant artifacts in the high range
 this also reduces the entropy in the high bands and allows for more bits to
 be used by the mid-low bands. By default it's enabled but can be disabled for
 debugging by setting the option to "disable".
@item aac_ltp
 Enables the use of the long term prediction extension which increases coding
 efficiency in very low bandwidth situations such as encoding of voice or
 solo piano music by extending constant harmonic peaks in bands throughout
 frames. This option is implied by profile:a aac_low and is incompatible with
 aac_pred. Use in conjunction with @option{-ar} to decrease the samplerate.
@item aac_pred
 Enables the use of a more traditional style of prediction where the spectral
 coefficients transmitted are replaced by the difference of the current
 coefficients minus the previous "predicted" coefficients. In theory and sometimes
 in practice this can improve quality for low to mid bitrate audio.
 This option implies the aac_main profile and is incompatible with aac_ltp.
@item profile
 Sets the encoding profile, possible values:
@table @samp
@item aac_low
 The default, AAC "Low-complexity" profile. Is the most compatible and produces
 decent quality.
@item mpeg2_aac_low
 Equivalent to -profile:a aac_low -aac_pns 0. PNS was introduced with the MPEG4
 specifications.
@item aac_ltp
 Long term prediction profile, is enabled by and will enable the aac_ltp option.
 Introduced in MPEG4.
@item aac_main
 Main-type prediction profile, is enabled by and will enable the aac_pred option.
 Introduced in MPEG2.
 If this option is unspecified it is set to @samp{aac_low}.
@end table
@end table
@section ac3 and ac3_fixed
@ -532,98 +494,21 @@ 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
 libfaac AAC (Advanced Audio Coding) encoder wrapper.
-This encoder is of much lower quality and is more unstable than any other AAC
+Requires the presence of the libfaac headers and library during
-encoders, so it's highly recommended to instead use other encoders, like
+configuration. You need to explicitly configure the build with
@ref{aacenc,,the native FFmpeg AAC encoder}.
 This encoder also requires the presence of the libfaac headers and library
 during configuration. You need to explicitly configure the build with
@code{--enable-libfaac --enable-nonfree}.
 This encoder is considered to be of higher quality with respect to the
@ref{aacenc,,the native experimental FFmpeg AAC encoder}.
 For more information see the libfaac project at
@url{http://www.audiocoding.com/faac.html/}.
@subsection Options
 The following shared FFmpeg codec options are recognized.
@ -730,10 +615,9 @@ configuration. You need to explicitly configure the build with
 so if you allow the use of GPL, you should configure with
@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
-This encoder is considered to produce output on par or worse at 128kbps to the
+This encoder is considered to be of higher quality with respect to
-@ref{aacenc,,the native FFmpeg AAC encoder} but can often produce better
+both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
-sounding audio at identical or lower bitrates and has support for the
+@ref{libfaac}.
 AAC-HE profiles.
 VBR encoding, enabled through the @option{vbr} or @option{flags
 +qscale} options, is experimental and only works with some
@ -1075,6 +959,31 @@ Set MPEG audio original flag when set to 1. The default value is 0
@end table
@anchor{libvo-aacenc}
@section libvo-aacenc
 VisualOn AAC encoder.
 Requires the presence of the libvo-aacenc headers and library during
 configuration. You need to explicitly configure the build with
@code{--enable-libvo-aacenc --enable-version3}.
 This encoder is considered to be worse than the
@ref{aacenc,,native experimental FFmpeg AAC encoder}, according to
 multiple sources.
@subsection Options
 The VisualOn AAC encoder only support encoding AAC-LC and up to 2
 channels. It is also CBR-only.
@table @option
@item b
 Set bit rate in bits/s.
@end table
@section libvo-amrwbenc
 VisualOn Adaptive Multi-Rate Wideband encoder.
@ -1137,7 +1046,7 @@ kilobits/s.
@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
 Set VBR mode. The FFmpeg @option{vbr} option has the following
-valid arguments, with the @command{opusenc} equivalent options
+valid arguments, with the their @command{opusenc} equivalent options
 in parentheses:
@table @samp
@ -1184,17 +1093,6 @@ following: 4000, 6000, 8000, 12000, or 20000, corresponding to
 narrowband, mediumband, wideband, super wideband, and fullband
 respectively. The default is 0 (cutoff disabled).
@item mapping_family (@emph{mapping_family})
 Set channel mapping family to be used by the encoder. The default value of -1
 uses mapping family 0 for mono and stereo inputs, and mapping family 1
 otherwise. The default also disables the surround masking and LFE bandwidth
 optimzations in libopus, and requires that the input contains 8 channels or
 fewer.
 Other values include 0 for mono and stereo, 1 for surround sound with masking
 and LFE bandwidth optimizations, and 255 for independent streams with an
 unspecified channel layout.
@end table
@section libvorbis
@ -1365,96 +1263,6 @@ disabled
 A description of some of the currently available video encoders
 follows.
@section libopenh264
 Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.
 This encoder requires the presence of the libopenh264 headers and
 library during configuration. You need to explicitly configure the
 build with @code{--enable-libopenh264}. The library is detected using
@command{pkg-config}.
 For more information about the library see
@url{http://www.openh264.org}.
@subsection Options
 The following FFmpeg global options affect the configurations of the
 libopenh264 encoder.
@table @option
@item b
 Set the bitrate (as a number of bits per second).
@item g
 Set the GOP size.
@item maxrate
 Set the max bitrate (as a number of bits per second).
@item flags +global_header
 Set global header in the bitstream.
@item slices
 Set the number of slices, used in parallelized encoding. Default value
 is 0. This is only used when @option{slice_mode} is set to
@samp{fixed}.
@item slice_mode
 Set slice mode. Can assume one of the following possible values:
@table @samp
@item fixed
 a fixed number of slices
@item rowmb
 one slice per row of macroblocks
@item auto
 automatic number of slices according to number of threads
@item dyn
 dynamic slicing
@end table
 Default value is @samp{auto}.
@item loopfilter
 Enable loop filter, if set to 1 (automatically enabled). To disable
 set a value of 0.
@item profile
 Set profile restrictions. If set to the value of @samp{main} enable
 CABAC (set the @code{SEncParamExt.iEntropyCodingModeFlag} flag to 1).
@item max_nal_size
 Set maximum NAL size in bytes.
@item allow_skip_frames
 Allow skipping frames to hit the target bitrate if set to 1.
@end table
@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
 makes it possible 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.
@ -1529,159 +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
+Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
@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.
@table @option
-@item b (@emph{target-bitrate})
+@item threads
-Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
+g_threads
 expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in
 kilobits/s.
-@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})
+@item qmin
-Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
+rc_min_quantizer
 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 rc_init_occupancy (@emph{buf-initial-sz})
+@item qmax
-Set number of bits which should be loaded into the rc buffer before decoding
+rc_max_quantizer
 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 undershoot-pct
+@item bufsize, vb
-Set datarate undershoot (min) percentage of the target bitrate.
+rc_buf_sz
@code{(bufsize * 1000 / vb)}
-@item overshoot-pct
+rc_buf_optimal_sz
-Set datarate overshoot (max) percentage of the target bitrate.
+@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})
+@item skip_threshold
-Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
+rc_dropframe_thresh
 percentage of the target bitrate, the libvpx wrapper converts this value as
 follows: @code{(maxrate * 100 / bitrate)}.
-@item minrate (@emph{minsection-pct})
+@item qcomp
-Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
+rc_2pass_vbr_bias_pct
 percentage of the target bitrate, the libvpx wrapper converts this value as
 follows: @code{(minrate * 100 / bitrate)}.
-@item minrate, maxrate, b @emph{end-usage=cbr}
+@item maxrate, vb
-@code{(minrate == maxrate == bitrate)}.
+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 tune (@emph{tune})
+@item minrate, maxrate, vb
-@table @samp
+@code{VPX_CBR}
-@item psnr (@emph{psnr})
+@code{(minrate == maxrate == vb)}
-@item ssim (@emph{ssim})
+
@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 quality, deadline (@emph{deadline})
+@item speed
-@table @samp
+@code{VP8E_SET_CPUUSED}
@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.
@end table
-@item speed, cpu-used (@emph{cpu-used})
+@item nr
-Set quality/speed ratio modifier. Higher values speed up the encode at the cost
+@code{VP8E_SET_NOISE_SENSITIVITY}
 of quality.
-@item nr (@emph{noise-sensitivity})
+@item mb_threshold
@code{VP8E_SET_STATIC_THRESHOLD}
-@item static-thresh
+@item slices
-Set a change threshold on blocks below which they will be skipped by the
+@code{VP8E_SET_TOKEN_PARTITIONS}
 encoder.
@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 max-intra-rate
-Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0
+@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
 means unlimited.
@item force_key_frames
@code{VPX_EFLAG_FORCE_KF}
@item Alternate reference frame related
@table @option
-@item auto-alt-ref
+@item vp8flags altref
-Enable use of alternate reference frames (2-pass only).
+@code{VP8E_SET_ENABLEAUTOALTREF}
-@item arnr-max-frames
+@item @var{arnr_max_frames}
-Set altref noise reduction max frame count.
+@code{VP8E_SET_ARNR_MAXFRAMES}
-@item arnr-type
+@item @var{arnr_type}
-Set altref noise reduction filter type: backward, forward, centered.
+@code{VP8E_SET_ARNR_TYPE}
-@item arnr-strength
+@item @var{arnr_strength}
-Set altref noise reduction filter strength.
+@code{VP8E_SET_ARNR_STRENGTH}
-@item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
+@item @var{rc_lookahead}
-Set number of frames to look ahead for frametype and ratecontrol.
+g_lag_in_frames
@end table
-@item error-resilient
+@item vp8flags error_resilient
-Enable error resiliency features.
+g_error_resilient
-@item VP9-specific options
+@item aq_mode
-@table @option
+@code{VP9E_SET_AQ_MODE}
@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
@end table
@ -2106,10 +1868,6 @@ For example to specify libx264 encoding options with @command{ffmpeg}:
 ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
@item a53cc @var{boolean}
 Import closed captions (which must be ATSC compatible format) into output.
 Only the mpeg2 and h264 decoders provide these. Default is 0 (off).
@item x264-params (N.A.)
 Override the x264 configuration using a :-separated list of key=value
 parameters.
@ -2413,7 +2171,7 @@ Use @var{0} to disable alpha plane coding.
@subsection Speed considerations
 In the default mode of operation the encoder has to honor frame constraints
-(i.e. not produce frames with size bigger than requested) while still making
+(i.e. not produc frames with size bigger than requested) while still making
 output picture as good as possible.
 A frame containing a lot of small details is harder to compress and the encoder
 would spend more time searching for appropriate quantizers for each slice.
@ -2423,180 +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 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
@section QSV encoders
 The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
 The ratecontrol method is selected as follows:
@itemize @bullet
@item
 When @option{global_quality} is specified, a quality-based mode is used.
 Specifically this means either
@itemize @minus
@item
@var{CQP} - constant quantizer scale, when the @option{qscale} codec flag is
 also set (the @option{-qscale} ffmpeg option).
@item
@var{LA_ICQ} - intelligent constant quality with lookahead, when the
@option{look_ahead} option is also set.
@item
@var{ICQ} -- intelligent constant quality otherwise.
@end itemize
@item
 Otherwise, a bitrate-based mode is used. For all of those, you should specify at
 least the desired average bitrate with the @option{b} option.
@itemize @minus
@item
@var{LA} - VBR with lookahead, when the @option{look_ahead} option is specified.
@item
@var{VCM} - video conferencing mode, when the @option{vcm} option is set.
@item
@var{CBR} - constant bitrate, when @option{maxrate} is specified and equal to
 the average bitrate.
@item
@var{VBR} - variable bitrate, when @option{maxrate} is specified, but is higher
 than the average bitrate.
@item
@var{AVBR} - average VBR mode, when @option{maxrate} is not specified. This mode
 is further configured by the @option{avbr_accuracy} and
@option{avbr_convergence} options.
@end itemize
@end itemize
 Note that depending on your system, a different mode than the one you specified
 may be selected by the encoder. Set the verbosity level to @var{verbose} or
 higher to see the actual settings used by the QSV runtime.
 Additional libavcodec global options are mapped to MSDK options as follows:
@itemize
@item
@option{g/gop_size} -> @option{GopPicSize}
@item
@option{bf/max_b_frames}+1 -> @option{GopRefDist}
@item
@option{rc_init_occupancy/rc_initial_buffer_occupancy} ->
@option{InitialDelayInKB}
@item
@option{slices} -> @option{NumSlice}
@item
@option{refs} -> @option{NumRefFrame}
@item
@option{b_strategy/b_frame_strategy} -> @option{BRefType}
@item
@option{cgop/CLOSED_GOP} codec flag -> @option{GopOptFlag}
@item
 For the @var{CQP} mode, the @option{i_qfactor/i_qoffset} and
@option{b_qfactor/b_qoffset} set the difference between @var{QPP} and @var{QPI},
 and @var{QPP} and @var{QPB} respectively.
@item
 Setting the @option{coder} option to the value @var{vlc} will make the H.264
 encoder use CAVLC instead of CABAC.
@end itemize
@section vc2
 SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at
 professional broadcasting but since it supports yuv420, yuv422 and yuv444 at
 8 (limited range or full range), 10 or 12 bits, this makes it suitable for
 other tasks which require low overhead and low compression (like screen
 recording).
@subsection Options
@table @option
@item b
 Sets target video bitrate. Usually that's around 1:6 of the uncompressed
 video bitrate (e.g. for 1920x1080 50fps yuv422p10 that's around 400Mbps). Higher
 values (close to the uncompressed bitrate) turn on lossless compression mode.
@item field_order
 Enables field coding when set (e.g. to tt - top field first) for interlaced
 inputs. Should increase compression with interlaced content as it splits the
 fields and encodes each separately.
@item wavelet_depth
 Sets the total amount of wavelet transforms to apply, between 1 and 5 (default).
 Lower values reduce compression and quality. Less capable decoders may not be
 able to handle values of @option{wavelet_depth} over 3.
@item wavelet_type
 Sets the transform type. Currently only @var{5_3} (LeGall) and @var{9_7}
 (Deslauriers-Dubuc)
 are implemented, with 9_7 being the one with better compression and thus
 is the default.
@item slice_width
@item slice_height
 Sets the slice size for each slice. Larger values result in better compression.
 For compatibility with other more limited decoders use @option{slice_width} of
 32 and @option{slice_height} of 8.
@item tolerance
 Sets the undershoot tolerance of the rate control system in percent. This is
 to prevent an expensive search from being run.
@item qm
 Sets the quantization matrix preset to use by default or when @option{wavelet_depth}
 is set to 5
@itemize @minus
@item
@var{default}
 Uses the default quantization matrix from the specifications, extended with
 values for the fifth level. This provides a good balance between keeping detail
 and omitting artifacts.
@item
@var{flat}
 Use a completely zeroed out quantization matrix. This increases PSNR but might
 reduce perception. Use in bogus benchmarks.
@item
@var{color}
 Reduces detail but attempts to preserve color at extremely low bitrates.
@end itemize
@end table
@c man end VIDEO ENCODERS
@chapter Subtitles Encoders
--- a/doc/errno.txt
+++ b/doc/errno.txt
@ -76,7 +76,7 @@ EMFILE           POSIX     ++++++  Too many open files
 EMLINK           POSIX     ++++++  Too many links
 EMSGSIZE         POSIX     +++..+  Message too long
 EMULTIHOP        POSIX     ++4...  Multihop attempted
-ENAMETOOLONG     POSIX  -  ++++++  File name too long
+ENAMETOOLONG     POSIX  -  ++++++  Filen ame too long
 ENAVAIL                    +.....  No XENIX semaphores available
 ENEEDAUTH                  .++...  Need authenticator
 ENETDOWN         POSIX     +++..+  Network is down
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@ -1,16 +0,0 @@
 /avio_dir_cmd
 /avio_reading
 /decoding_encoding
 /demuxing_decoding
 /extract_mvs
 /filter_audio
 /filtering_audio
 /filtering_video
 /metadata
 /muxing
 /pc-uninstalled
 /remuxing
 /resampling_audio
 /scaling_video
 /transcode_aac
 /transcoding
--- a/doc/examples/Makefile
+++ b/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                       \
+EXAMPLES=       avio_reading                       \
                avio_reading                       \
                decoding_encoding                  \
                demuxing_decoding                  \
                extract_mvs                        \
                filtering_video                    \
                filtering_audio                    \
                http_multiclient                   \
                metadata                           \
                muxing                             \
                remuxing                           \
--- a/doc/examples/avio_dir_cmd.c
+++ b/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;
 }
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@ -25,9 +25,9 @@
 * libavcodec API use example.
 *
 * @example decoding_encoding.c
- * Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...),
+ * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
- * not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...).
+ * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
- * See library 'libavformat' for the format handling
+ * format handling
 */
 #include <math.h>
@ -211,7 +211,7 @@ static void audio_encode_example(const char *filename)
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }
@ -225,7 +225,7 @@ static void audio_encode_example(const char *filename)
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }
    fclose(f);
@ -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;
@ -253,7 +253,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    printf("Decode audio file %s to %s\n", filename, outfilename);
-    /* find the MPEG audio decoder */
+    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
@ -356,7 +356,7 @@ static void video_encode_example(const char *filename, int codec_id)
    printf("Encode video file %s\n", filename);
-    /* find the video encoder */
+    /* find the mpeg1 video encoder */
    codec = avcodec_find_encoder(codec_id);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
@ -454,7 +454,7 @@ static void video_encode_example(const char *filename, int codec_id)
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }
@ -471,11 +471,11 @@ static void video_encode_example(const char *filename, int codec_id)
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }
-    /* add sequence end code to have a real MPEG file */
+    /* add sequence end code to have a real mpeg file */
    fwrite(endcode, 1, sizeof(endcode), f);
    fclose(f);
@ -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) {
@ -538,17 +538,17 @@ 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) */
+    /* 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);
-    /* find the MPEG-1 video decoder */
+    /* find the mpeg1 video decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
@ -561,8 +561,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }
-    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+    if(codec->capabilities&CODEC_CAP_TRUNCATED)
-        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+        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
@ -613,9 +613,9 @@ static void video_decode_example(const char *outfilename, const char *filename)
                exit(1);
    }
-    /* Some codecs, such as MPEG, transmit the I- and P-frame with a
+    /* some codecs, such as MPEG, transmit the I and P frame with a
       latency of one frame. You must do the following to have a
-       chance to get the last frame of the video. */
+       chance to get the last frame of the video */
    avpkt.data = NULL;
    avpkt.size = 0;
    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@ -36,8 +36,6 @@
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
 static int width, height;
 static enum AVPixelFormat pix_fmt;
 static AVStream *video_stream = NULL, *audio_stream = NULL;
 static const char *src_filename = NULL;
 static const char *video_dst_filename = NULL;
@ -55,11 +53,17 @@ static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;
-/* Enable or disable frame reference counting. You are not supposed to support
+/* The different ways of decoding and managing data memory. You are not
- * both paths in your application but pick the one most appropriate to your
+ * supposed to support all the modes in your application but pick the one most
- * needs. Look for the use of refcount in this example to see what are the
+ * appropriate to your needs. Look for the use of api_mode in this example to
- * differences of API usage between them. */
+ * see what are the differences of API usage between them */
-static int refcount = 0;
+enum {
    API_MODE_OLD                  = 0, /* old method, deprecated */
    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
 };
 static int api_mode = API_MODE_OLD;
 static int decode_packet(int *got_frame, int cached)
 {
@ -77,22 +81,6 @@ static int decode_packet(int *got_frame, int cached)
        }
        if (*got_frame) {
            if (frame->width != width || frame->height != height ||
                frame->format != pix_fmt) {
                /* To handle this change, one could call av_image_alloc again and
                 * decode the following frames into another rawvideo file. */
                fprintf(stderr, "Error: Width, height and pixel format have to be "
                        "constant in a rawvideo file, but the width, height or "
                        "pixel format of the input video changed:\n"
                        "old: width = %d, height = %d, format = %s\n"
                        "new: width = %d, height = %d, format = %s\n",
                        width, height, av_get_pix_fmt_name(pix_fmt),
                        frame->width, frame->height,
                        av_get_pix_fmt_name(frame->format));
                return -1;
            }
            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   video_frame_count++, frame->coded_picture_number,
@ -102,7 +90,7 @@ static int decode_packet(int *got_frame, int cached)
             * this is required since rawvideo expects non aligned data */
            av_image_copy(video_dst_data, video_dst_linesize,
                          (const uint8_t **)(frame->data), frame->linesize,
-                          pix_fmt, width, height);
+                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
            /* write to rawvideo file */
            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
@ -139,9 +127,9 @@ static int decode_packet(int *got_frame, int cached)
        }
    }
-    /* If we use frame reference counting, we own the data and need
+    /* If we use the new API with reference counting, we own the data and need
     * to de-reference it when we don't use it anymore */
-    if (*got_frame && refcount)
+    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
        av_frame_unref(frame);
    return decoded;
@ -150,7 +138,7 @@ static int decode_packet(int *got_frame, int cached)
 static int open_codec_context(int *stream_idx,
                              AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
-    int ret, stream_index;
+    int ret;
    AVStream *st;
    AVCodecContext *dec_ctx = NULL;
    AVCodec *dec = NULL;
@ -162,8 +150,8 @@ static int open_codec_context(int *stream_idx,
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
-        stream_index = ret;
+        *stream_idx = ret;
-        st = fmt_ctx->streams[stream_index];
+        st = fmt_ctx->streams[*stream_idx];
        /* find decoder for the stream */
        dec_ctx = st->codec;
@ -175,13 +163,13 @@ static int open_codec_context(int *stream_idx,
        }
        /* Init the decoders, with or without reference counting */
-        av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
+        if (api_mode == API_MODE_NEW_API_REF_COUNT)
            av_dict_set(&opts, "refcounted_frames", "1", 0);
        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
        }
        *stream_idx = stream_index;
    }
    return 0;
@ -221,19 +209,28 @@ int main (int argc, char **argv)
    int ret = 0, got_frame;
    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
+        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
                "input_file video_output_file audio_output_file\n"
                "API example program to show how to read frames from an input file.\n"
                "This program reads frames from a file, decodes them, and writes decoded\n"
                "video frames to a rawvideo file named video_output_file, and decoded\n"
                "audio frames to a rawaudio file named audio_output_file.\n\n"
                "If the -refcount option is specified, the program use the\n"
                "reference counting frame system which allows keeping a copy of\n"
-                "the data for longer than one decode call.\n"
+                "the data for longer than one decode call. If unset, it's using\n"
                "the classic old method.\n"
                "\n", argv[0]);
        exit(1);
    }
-    if (argc == 5 && !strcmp(argv[1], "-refcount")) {
+    if (argc == 5) {
-        refcount = 1;
+        const char *mode = argv[1] + strlen("-refcount=");
        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
        else {
            fprintf(stderr, "unknow mode '%s'\n", mode);
            exit(1);
        }
        argv++;
    }
    src_filename = argv[1];
@ -267,11 +264,9 @@ int main (int argc, char **argv)
        }
        /* allocate image where the decoded image will be put */
        width = video_dec_ctx->width;
        height = video_dec_ctx->height;
        pix_fmt = video_dec_ctx->pix_fmt;
        ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             width, height, pix_fmt, 1);
+                             video_dec_ctx->width, video_dec_ctx->height,
                             video_dec_ctx->pix_fmt, 1);
        if (ret < 0) {
            fprintf(stderr, "Could not allocate raw video buffer\n");
            goto end;
@ -299,7 +294,12 @@ int main (int argc, char **argv)
        goto end;
    }
-    frame = av_frame_alloc();
+    /* When using the new API, you need to use the libavutil/frame.h API, while
     * the classic frame management is available in libavcodec */
    if (api_mode == API_MODE_OLD)
        frame = avcodec_alloc_frame();
    else
        frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
@ -326,7 +326,7 @@ int main (int argc, char **argv)
            pkt.data += ret;
            pkt.size -= ret;
        } while (pkt.size > 0);
-        av_packet_unref(&orig_pkt);
+        av_free_packet(&orig_pkt);
    }
    /* flush cached frames */
@ -341,7 +341,7 @@ int main (int argc, char **argv)
    if (video_stream) {
        printf("Play the output video file with the command:\n"
               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(pix_fmt), width, height,
+               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
               video_dst_filename);
    }
@ -376,7 +376,10 @@ end:
        fclose(video_dst_file);
    if (audio_dst_file)
        fclose(audio_dst_file);
-    av_frame_free(&frame);
+    if (api_mode == API_MODE_OLD)
        avcodec_free_frame(&frame);
    else
        av_frame_free(&frame);
    av_free(video_dst_data[0]);
    return ret < 0;
--- a/doc/examples/extract_mvs.c
+++ b/doc/examples/extract_mvs.c
@ -167,7 +167,7 @@ int main(int argc, char **argv)
            pkt.data += ret;
            pkt.size -= ret;
        } while (pkt.size > 0);
-        av_packet_unref(&orig_pkt);
+        av_free_packet(&orig_pkt);
    }
    /* flush cached frames */
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@ -33,6 +33,7 @@
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@ -65,7 +66,7 @@ static int open_input_file(const char *filename)
    /* select the audio stream */
    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
+        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
        return ret;
    }
    audio_stream_index = ret;
@ -144,28 +145,12 @@ static int init_filters(const char *filters_descr)
        goto end;
    }
-    /*
+    /* Endpoints for the filter graph. */
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */
    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
@ -273,10 +258,10 @@ int main(int argc, char **argv)
            }
            if (packet.size <= 0)
-                av_packet_unref(&packet0);
+                av_free_packet(&packet0);
        } else {
            /* discard non-wanted packets */
-            av_packet_unref(&packet0);
+            av_free_packet(&packet0);
        }
    }
 end:
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@ -33,14 +33,12 @@
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
-const char *filter_descr = "scale=78:24,transpose=cclock";
+const char *filter_descr = "scale=78:24";
 /* other way:
   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
 */
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@ -130,28 +128,12 @@ static int init_filters(const char *filters_descr)
        goto end;
    }
-    /*
+    /* Endpoints for the filter graph. */
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */
    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
@ -262,7 +244,7 @@ int main(int argc, char **argv)
                av_frame_unref(frame);
            }
        }
-        av_packet_unref(&packet);
+        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
--- a/doc/examples/http_multiclient.c
+++ b/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;
 }
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@ -52,7 +52,6 @@
 // a wrapper around a single output AVStream
 typedef struct OutputStream {
    AVStream *st;
    AVCodecContext *enc;
    /* pts of the next frame that will be generated */
    int64_t next_pts;
@ -105,18 +104,13 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
        exit(1);
    }
-    ost->st = avformat_new_stream(oc, NULL);
+    ost->st = avformat_new_stream(oc, *codec);
    if (!ost->st) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
    ost->st->id = oc->nb_streams-1;
-    c = avcodec_alloc_context3(*codec);
+    c = ost->st->codec;
    if (!c) {
        fprintf(stderr, "Could not alloc an encoding context\n");
        exit(1);
    }
    ost->enc = c;
    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
@ -161,7 +155,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
        c->pix_fmt       = STREAM_PIX_FMT;
        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
-            /* just for testing, we also add B-frames */
+            /* just for testing, we also add B frames */
            c->max_b_frames = 2;
        }
        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
@ -178,7 +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;
 }
 /**************************************************************/
@ -219,7 +213,7 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
    int ret;
    AVDictionary *opt = NULL;
-    c = ost->enc;
+    c = ost->st->codec;
    /* open it */
    av_dict_copy(&opt, opt_arg, 0);
@ -236,7 +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;
@ -246,13 +240,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
                                       c->sample_rate, nb_samples);
    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
    if (ret < 0) {
        fprintf(stderr, "Could not copy the stream parameters\n");
        exit(1);
    }
    /* create resampler context */
        ost->swr_ctx = swr_alloc();
        if (!ost->swr_ctx) {
@ -284,13 +271,13 @@ static AVFrame *get_audio_frame(OutputStream *ost)
    int16_t *q = (int16_t*)frame->data[0];
    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->enc->time_base,
+    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
        return NULL;
    for (j = 0; j <frame->nb_samples; j++) {
        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->enc->channels; i++)
+        for (i = 0; i < ost->st->codec->channels; i++)
            *q++ = v;
        ost->t     += ost->tincr;
        ost->tincr += ost->tincr2;
@ -316,7 +303,7 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
    int dst_nb_samples;
    av_init_packet(&pkt);
-    c = ost->enc;
+    c = ost->st->codec;
    frame = get_audio_frame(ost);
@ -396,7 +383,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
 static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
    int ret;
-    AVCodecContext *c = ost->enc;
+    AVCodecContext *c = ost->st->codec;
    AVDictionary *opt = NULL;
    av_dict_copy(&opt, opt_arg, 0);
@ -427,13 +414,6 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
            exit(1);
        }
    }
    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
    if (ret < 0) {
        fprintf(stderr, "Could not copy the stream parameters\n");
        exit(1);
    }
 }
 /* Prepare a dummy image. */
@ -468,10 +448,10 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
 static AVFrame *get_video_frame(OutputStream *ost)
 {
-    AVCodecContext *c = ost->enc;
+    AVCodecContext *c = ost->st->codec;
    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, c->time_base,
+    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
        return NULL;
@ -513,25 +493,44 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
    AVCodecContext *c;
    AVFrame *frame;
    int got_packet = 0;
    AVPacket pkt = { 0 };
-    c = ost->enc;
+    c = ost->st->codec;
    frame = get_video_frame(ost);
-    av_init_packet(&pkt);
+    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
        /* a hack to avoid data copy with some raw video muxers */
        AVPacket pkt;
        av_init_packet(&pkt);
-    /* encode the image */
+        if (!frame)
-    ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+            return 1;
    if (ret < 0) {
        fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
        exit(1);
    }
-    if (got_packet) {
+        pkt.flags        |= AV_PKT_FLAG_KEY;
-        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        pkt.stream_index  = ost->st->index;
        pkt.data          = (uint8_t *)frame;
        pkt.size          = sizeof(AVPicture);
        pkt.pts = pkt.dts = frame->pts;
        av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
-        ret = 0;
+        AVPacket pkt = { 0 };
        av_init_packet(&pkt);
        /* encode the image */
        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
        if (ret < 0) {
            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
            exit(1);
        }
        if (got_packet) {
            ret = write_frame(oc, &c->time_base, ost->st, &pkt);
        } else {
            ret = 0;
        }
    }
    if (ret < 0) {
@ -544,7 +543,7 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
 static void close_stream(AVFormatContext *oc, OutputStream *ost)
 {
-    avcodec_free_context(&ost->enc);
+    avcodec_close(ost->st->codec);
    av_frame_free(&ost->frame);
    av_frame_free(&ost->tmp_frame);
    sws_freeContext(ost->sws_ctx);
@ -565,7 +564,6 @@ int main(int argc, char **argv)
    int have_video = 0, have_audio = 0;
    int encode_video = 0, encode_audio = 0;
    AVDictionary *opt = NULL;
    int i;
    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
@ -582,9 +580,8 @@ int main(int argc, char **argv)
    }
    filename = argv[1];
-    for (i = 2; i+1 < argc; i+=2) {
+    if (argc > 3 && !strcmp(argv[2], "-flags")) {
-        if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
+        av_dict_set(&opt, argv[2]+1, argv[3], 0);
            av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
    }
    /* allocate the output media context */
@ -642,8 +639,8 @@ int main(int argc, char **argv)
    while (encode_video || encode_audio) {
        /* select the stream to encode */
        if (encode_video &&
-            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
+            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
-                                            audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
+                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
            encode_video = !write_video_frame(oc, &video_st);
        } else {
            encode_audio = !write_audio_frame(oc, &audio_st);
@ -664,7 +661,7 @@ int main(int argc, char **argv)
    if (!(fmt->flags & AVFMT_NOFILE))
        /* Close the output file. */
-        avio_closep(&oc->pb);
+        avio_close(oc->pb);
    /* free the stream */
    avformat_free_context(oc);
--- a/doc/examples/qsvdec.c
+++ b/doc/examples/qsvdec.c
@ -1,487 +0,0 @@
 /*
 * Copyright (c) 2015 Anton Khirnov
 *
 * 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
 * Intel QSV-accelerated H.264 decoding example.
 *
 * @example qsvdec.c
 * This example shows how to do QSV-accelerated H.264 decoding with output
 * frames in the VA-API video surfaces.
 */
 #include "config.h"
 #include <stdio.h>
 #include <mfx/mfxvideo.h>
 #include <va/va.h>
 #include <va/va_x11.h>
 #include <X11/Xlib.h>
 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 #include "libavcodec/avcodec.h"
 #include "libavcodec/qsv.h"
 #include "libavutil/error.h"
 #include "libavutil/mem.h"
 typedef struct DecodeContext {
    mfxSession mfx_session;
    VADisplay va_dpy;
    VASurfaceID *surfaces;
    mfxMemId    *surface_ids;
    int         *surface_used;
    int       nb_surfaces;
    mfxFrameInfo frame_info;
 } DecodeContext;
 static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
                             mfxFrameAllocResponse *resp)
 {
    DecodeContext *decode = pthis;
    int err, i;
    if (decode->surfaces) {
        fprintf(stderr, "Multiple allocation requests.\n");
        return MFX_ERR_MEMORY_ALLOC;
    }
    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
        return MFX_ERR_UNSUPPORTED;
    }
    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
        fprintf(stderr, "Unsupported surface properties.\n");
        return MFX_ERR_UNSUPPORTED;
    }
    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
        goto fail;
    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
                           req->Info.Width, req->Info.Height,
                           decode->surfaces, req->NumFrameSuggested,
                           NULL, 0);
    if (err != VA_STATUS_SUCCESS) {
        fprintf(stderr, "Error allocating VA surfaces\n");
        goto fail;
    }
    decode->nb_surfaces = req->NumFrameSuggested;
    for (i = 0; i < decode->nb_surfaces; i++)
        decode->surface_ids[i] = &decode->surfaces[i];
    resp->mids           = decode->surface_ids;
    resp->NumFrameActual = decode->nb_surfaces;
    decode->frame_info = req->Info;
    return MFX_ERR_NONE;
 fail:
    av_freep(&decode->surfaces);
    av_freep(&decode->surface_ids);
    av_freep(&decode->surface_used);
    return MFX_ERR_MEMORY_ALLOC;
 }
 static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
 {
    return MFX_ERR_NONE;
 }
 static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
 {
    return MFX_ERR_UNSUPPORTED;
 }
 static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
 {
    return MFX_ERR_UNSUPPORTED;
 }
 static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
 {
    *hdl = mid;
    return MFX_ERR_NONE;
 }
 static void free_surfaces(DecodeContext *decode)
 {
    if (decode->surfaces)
        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
    av_freep(&decode->surfaces);
    av_freep(&decode->surface_ids);
    av_freep(&decode->surface_used);
    decode->nb_surfaces = 0;
 }
 static void free_buffer(void *opaque, uint8_t *data)
 {
    int *used = opaque;
    *used = 0;
    av_freep(&data);
 }
 static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
 {
    DecodeContext *decode = avctx->opaque;
    mfxFrameSurface1 *surf;
    AVBufferRef *surf_buf;
    int idx;
    for (idx = 0; idx < decode->nb_surfaces; idx++) {
        if (!decode->surface_used[idx])
            break;
    }
    if (idx == decode->nb_surfaces) {
        fprintf(stderr, "No free surfaces\n");
        return AVERROR(ENOMEM);
    }
    surf = av_mallocz(sizeof(*surf));
    if (!surf)
        return AVERROR(ENOMEM);
    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
    if (!surf_buf) {
        av_freep(&surf);
        return AVERROR(ENOMEM);
    }
    surf->Info       = decode->frame_info;
    surf->Data.MemId = &decode->surfaces[idx];
    frame->buf[0]  = surf_buf;
    frame->data[3] = (uint8_t*)surf;
    decode->surface_used[idx] = 1;
    return 0;
 }
 static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
 {
    while (*pix_fmts != AV_PIX_FMT_NONE) {
        if (*pix_fmts == AV_PIX_FMT_QSV) {
            if (!avctx->hwaccel_context) {
                DecodeContext *decode = avctx->opaque;
                AVQSVContext *qsv = av_qsv_alloc_context();
                if (!qsv)
                    return AV_PIX_FMT_NONE;
                qsv->session   = decode->mfx_session;
                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
                avctx->hwaccel_context = qsv;
            }
            return AV_PIX_FMT_QSV;
        }
        pix_fmts++;
    }
    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
    return AV_PIX_FMT_NONE;
 }
 static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
                         AVFrame *frame, AVPacket *pkt,
                         AVIOContext *output_ctx)
 {
    int ret = 0;
    int got_frame = 1;
    while (pkt->size > 0 || (!pkt->data && got_frame)) {
        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
        if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            return ret;
        }
        pkt->data += ret;
        pkt->size -= ret;
        /* A real program would do something useful with the decoded frame here.
         * We just retrieve the raw data and write it to a file, which is rather
         * useless but pedagogic. */
        if (got_frame) {
            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
            VAImageFormat img_fmt = {
                .fourcc         = VA_FOURCC_NV12,
                .byte_order     = VA_LSB_FIRST,
                .bits_per_pixel = 8,
                .depth          = 8,
            };
            VAImage img;
            VAStatus err;
            uint8_t *data;
            int i, j;
            img.buf      = VA_INVALID_ID;
            img.image_id = VA_INVALID_ID;
            err = vaCreateImage(decode->va_dpy, &img_fmt,
                                frame->width, frame->height, &img);
            if (err != VA_STATUS_SUCCESS) {
                fprintf(stderr, "Error creating an image: %s\n",
                        vaErrorStr(err));
                ret = AVERROR_UNKNOWN;
                goto fail;
            }
            err = vaGetImage(decode->va_dpy, surface, 0, 0,
                             frame->width, frame->height,
                             img.image_id);
            if (err != VA_STATUS_SUCCESS) {
                fprintf(stderr, "Error getting an image: %s\n",
                        vaErrorStr(err));
                ret = AVERROR_UNKNOWN;
                goto fail;
            }
            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
            if (err != VA_STATUS_SUCCESS) {
                fprintf(stderr, "Error mapping the image buffer: %s\n",
                        vaErrorStr(err));
                ret = AVERROR_UNKNOWN;
                goto fail;
            }
            for (i = 0; i < img.num_planes; i++)
                for (j = 0; j < (img.height >> (i > 0)); j++)
                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
 fail:
            if (img.buf != VA_INVALID_ID)
                vaUnmapBuffer(decode->va_dpy, img.buf);
            if (img.image_id != VA_INVALID_ID)
                vaDestroyImage(decode->va_dpy, img.image_id);
            av_frame_unref(frame);
            if (ret < 0)
                return ret;
        }
    }
    return 0;
 }
 int main(int argc, char **argv)
 {
    AVFormatContext *input_ctx = NULL;
    AVStream *video_st = NULL;
    AVCodecContext *decoder_ctx = NULL;
    const AVCodec *decoder;
    AVPacket pkt = { 0 };
    AVFrame *frame = NULL;
    DecodeContext decode = { NULL };
    Display *dpy = NULL;
    int va_ver_major, va_ver_minor;
    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
    mfxVersion mfx_ver = { { 1, 1 } };
    mfxFrameAllocator frame_allocator = {
        .pthis = &decode,
        .Alloc = frame_alloc,
        .Lock  = frame_lock,
        .Unlock = frame_unlock,
        .GetHDL = frame_get_hdl,
        .Free   = frame_free,
    };
    AVIOContext *output_ctx = NULL;
    int ret, i, err;
    av_register_all();
    if (argc < 3) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        return 1;
    }
    /* open the input file */
    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
    if (ret < 0) {
        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
        goto finish;
    }
    /* find the first H.264 video stream */
    for (i = 0; i < input_ctx->nb_streams; i++) {
        AVStream *st = input_ctx->streams[i];
        if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
            video_st = st;
        else
            st->discard = AVDISCARD_ALL;
    }
    if (!video_st) {
        fprintf(stderr, "No H.264 video stream in the input file\n");
        goto finish;
    }
    /* initialize VA-API */
    dpy = XOpenDisplay(NULL);
    if (!dpy) {
        fprintf(stderr, "Cannot open the X display\n");
        goto finish;
    }
    decode.va_dpy = vaGetDisplay(dpy);
    if (!decode.va_dpy) {
        fprintf(stderr, "Cannot open the VA display\n");
        goto finish;
    }
    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
    if (err != VA_STATUS_SUCCESS) {
        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
        goto finish;
    }
    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
    /* initialize an MFX session */
    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
    if (err != MFX_ERR_NONE) {
        fprintf(stderr, "Error initializing an MFX session\n");
        goto finish;
    }
    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
    /* initialize the decoder */
    decoder = avcodec_find_decoder_by_name("h264_qsv");
    if (!decoder) {
        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
        goto finish;
    }
    decoder_ctx = avcodec_alloc_context3(decoder);
    if (!decoder_ctx) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }
    decoder_ctx->codec_id = AV_CODEC_ID_H264;
    if (video_st->codecpar->extradata_size) {
        decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
                                            AV_INPUT_BUFFER_PADDING_SIZE);
        if (!decoder_ctx->extradata) {
            ret = AVERROR(ENOMEM);
            goto finish;
        }
        memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
               video_st->codecpar->extradata_size);
        decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
    }
    decoder_ctx->refcounted_frames = 1;
    decoder_ctx->opaque      = &decode;
    decoder_ctx->get_buffer2 = get_buffer;
    decoder_ctx->get_format  = get_format;
    ret = avcodec_open2(decoder_ctx, NULL, NULL);
    if (ret < 0) {
        fprintf(stderr, "Error opening the decoder: ");
        goto finish;
    }
    /* open the output stream */
    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
    if (ret < 0) {
        fprintf(stderr, "Error opening the output context: ");
        goto finish;
    }
    frame = av_frame_alloc();
    if (!frame) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }
    /* actual decoding */
    while (ret >= 0) {
        ret = av_read_frame(input_ctx, &pkt);
        if (ret < 0)
            break;
        if (pkt.stream_index == video_st->index)
            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
        av_packet_unref(&pkt);
    }
    /* flush the decoder */
    pkt.data = NULL;
    pkt.size = 0;
    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
 finish:
    if (ret < 0) {
        char buf[1024];
        av_strerror(ret, buf, sizeof(buf));
        fprintf(stderr, "%s\n", buf);
    }
    avformat_close_input(&input_ctx);
    av_frame_free(&frame);
    if (decoder_ctx)
        av_freep(&decoder_ctx->hwaccel_context);
    avcodec_free_context(&decoder_ctx);
    free_surfaces(&decode);
    if (decode.mfx_session)
        MFXClose(decode.mfx_session);
    if (decode.va_dpy)
        vaTerminate(decode.va_dpy);
    if (dpy)
        XCloseDisplay(dpy);
    avio_close(output_ctx);
    return ret;
 }
--- a/doc/examples/remuxing.c
+++ b/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);
@ -143,7 +143,7 @@ int main(int argc, char **argv)
            fprintf(stderr, "Error muxing packet\n");
            break;
        }
-        av_packet_unref(&pkt);
+        av_free_packet(&pkt);
    }
    av_write_trailer(ofmt_ctx);
@ -153,7 +153,7 @@ end:
    /* close output */
    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
+        avio_close(ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);
    if (ret < 0 && ret != AVERROR_EOF) {
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@ -41,9 +41,11 @@
 #include "libswresample/swresample.h"
 /** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 96000
+#define OUTPUT_BIT_RATE 48000
 /** The number of output channels */
 #define OUTPUT_CHANNELS 2
 /** The audio sample output format */
 #define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
 /**
 * Convert an error code into a text message.
@ -62,7 +64,6 @@ static int open_input_file(const char *filename,
                           AVFormatContext **input_format_context,
                           AVCodecContext **input_codec_context)
 {
    AVCodecContext *avctx;
    AVCodec *input_codec;
    int error;
@ -92,39 +93,23 @@ static int open_input_file(const char *filename,
    }
    /** Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
+    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
        fprintf(stderr, "Could not find input codec\n");
        avformat_close_input(input_format_context);
        return AVERROR_EXIT;
    }
    /** allocate a new decoding context */
    avctx = avcodec_alloc_context3(input_codec);
    if (!avctx) {
        fprintf(stderr, "Could not allocate a decoding context\n");
        avformat_close_input(input_format_context);
        return AVERROR(ENOMEM);
    }
    /** initialize the stream parameters with demuxer information */
    error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
    if (error < 0) {
        avformat_close_input(input_format_context);
        avcodec_free_context(&avctx);
        return error;
    }
    /** Open the decoder for the audio stream to use it later. */
-    if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
+    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
                               input_codec, NULL)) < 0) {
        fprintf(stderr, "Could not open input codec (error '%s')\n",
                get_error_text(error));
        avcodec_free_context(&avctx);
        avformat_close_input(input_format_context);
        return error;
    }
    /** Save the decoder context for easier access later. */
-    *input_codec_context = avctx;
+    *input_codec_context = (*input_format_context)->streams[0]->codec;
    return 0;
 }
@ -139,7 +124,6 @@ static int open_output_file(const char *filename,
                            AVFormatContext **output_format_context,
                            AVCodecContext **output_codec_context)
 {
    AVCodecContext *avctx          = NULL;
    AVIOContext *output_io_context = NULL;
    AVStream *stream               = NULL;
    AVCodec *output_codec          = NULL;
@ -179,64 +163,43 @@ static int open_output_file(const char *filename,
    }
    /** Create a new audio stream in the output file container. */
-    if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
+    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
        fprintf(stderr, "Could not create new stream\n");
        error = AVERROR(ENOMEM);
        goto cleanup;
    }
-    avctx = avcodec_alloc_context3(output_codec);
+    /** Save the encoder context for easiert access later. */
-    if (!avctx) {
+    *output_codec_context = stream->codec;
        fprintf(stderr, "Could not allocate an encoding context\n");
        error = AVERROR(ENOMEM);
        goto cleanup;
    }
    /**
     * Set the basic encoder parameters.
     * The input file's sample rate is used to avoid a sample rate conversion.
     */
-    avctx->channels       = OUTPUT_CHANNELS;
+    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
-    avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
-    avctx->sample_rate    = input_codec_context->sample_rate;
+    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
-    avctx->sample_fmt     = output_codec->sample_fmts[0];
+    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
-    avctx->bit_rate       = OUTPUT_BIT_RATE;
+    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
    /** Allow the use of the experimental AAC encoder */
    avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
    /** Set the sample rate for the container. */
    stream->time_base.den = input_codec_context->sample_rate;
    stream->time_base.num = 1;
    /**
     * Some container formats (like MP4) require global headers to be present
     * Mark the encoder so that it behaves accordingly.
     */
    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        avctx->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(avctx, output_codec, NULL)) < 0) {
+    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
        fprintf(stderr, "Could not open output codec (error '%s')\n",
                get_error_text(error));
        goto cleanup;
    }
    error = avcodec_parameters_from_context(stream->codecpar, avctx);
    if (error < 0) {
        fprintf(stderr, "Could not initialize stream parameters\n");
        goto cleanup;
    }
    /** Save the encoder context for easier access later. */
    *output_codec_context = avctx;
    return 0;
 cleanup:
-    avcodec_free_context(&avctx);
+    avio_close((*output_format_context)->pb);
    avio_closep(&(*output_format_context)->pb);
    avformat_free_context(*output_format_context);
    *output_format_context = NULL;
    return error < 0 ? error : AVERROR_EXIT;
@ -308,11 +271,10 @@ static int init_resampler(AVCodecContext *input_codec_context,
 }
 /** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
+static int init_fifo(AVAudioFifo **fifo)
 {
    /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
+    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
                                      output_codec_context->channels, 1))) {
        fprintf(stderr, "Could not allocate FIFO\n");
        return AVERROR(ENOMEM);
    }
@ -364,7 +326,7 @@ static int decode_audio_frame(AVFrame *frame,
                                       data_present, &input_packet)) < 0) {
        fprintf(stderr, "Could not decode frame (error '%s')\n",
                get_error_text(error));
-        av_packet_unref(&input_packet);
+        av_free_packet(&input_packet);
        return error;
    }
@ -374,7 +336,7 @@ static int decode_audio_frame(AVFrame *frame,
     */
    if (*finished && *data_present)
        *finished = 0;
-    av_packet_unref(&input_packet);
+    av_free_packet(&input_packet);
    return 0;
 }
@ -575,9 +537,6 @@ static int init_output_frame(AVFrame **frame,
    return 0;
 }
 /** Global timestamp for the audio frames */
 static int64_t pts = 0;
 /** Encode one frame worth of audio to the output file. */
 static int encode_audio_frame(AVFrame *frame,
                              AVFormatContext *output_format_context,
@ -589,12 +548,6 @@ static int encode_audio_frame(AVFrame *frame,
    int error;
    init_packet(&output_packet);
    /** Set a timestamp based on the sample rate for the container. */
    if (frame) {
        frame->pts = pts;
        pts += frame->nb_samples;
    }
    /**
     * Encode the audio frame and store it in the temporary packet.
     * The output audio stream encoder is used to do this.
@ -603,7 +556,7 @@ static int encode_audio_frame(AVFrame *frame,
                                       frame, data_present)) < 0) {
        fprintf(stderr, "Could not encode frame (error '%s')\n",
                get_error_text(error));
-        av_packet_unref(&output_packet);
+        av_free_packet(&output_packet);
        return error;
    }
@ -612,11 +565,11 @@ static int encode_audio_frame(AVFrame *frame,
        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
            fprintf(stderr, "Could not write frame (error '%s')\n",
                    get_error_text(error));
-            av_packet_unref(&output_packet);
+            av_free_packet(&output_packet);
            return error;
        }
-        av_packet_unref(&output_packet);
+        av_free_packet(&output_packet);
    }
    return 0;
@ -706,7 +659,7 @@ int main(int argc, char **argv)
                       &resample_context))
        goto cleanup;
    /** Initialize the FIFO buffer to store audio samples to be encoded. */
-    if (init_fifo(&fifo, output_codec_context))
+    if (init_fifo(&fifo))
        goto cleanup;
    /** Write the header of the output file container. */
    if (write_output_file_header(output_format_context))
@ -788,13 +741,13 @@ cleanup:
        av_audio_fifo_free(fifo);
    swr_free(&resample_context);
    if (output_codec_context)
-        avcodec_free_context(&output_codec_context);
+        avcodec_close(output_codec_context);
    if (output_format_context) {
-        avio_closep(&output_format_context->pb);
+        avio_close(output_format_context->pb);
        avformat_free_context(output_format_context);
    }
    if (input_codec_context)
-        avcodec_free_context(&input_codec_context);
+        avcodec_close(input_codec_context);
    if (input_format_context)
        avformat_close_input(&input_format_context);
--- a/doc/examples/transcoding.c
+++ b/doc/examples/transcoding.c
@ -31,6 +31,7 @@
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@ -116,7 +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;
            }
@ -128,10 +129,7 @@ static int open_output_file(const char *filename)
                enc_ctx->width = dec_ctx->width;
                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
                /* take first format from list of supported formats */
-                if (encoder->pix_fmts)
+                enc_ctx->pix_fmt = encoder->pix_fmts[0];
                    enc_ctx->pix_fmt = encoder->pix_fmts[0];
                else
                    enc_ctx->pix_fmt = dec_ctx->pix_fmt;
                /* video time_base can be set to whatever is handy and supported by encoder */
                enc_ctx->time_base = dec_ctx->time_base;
            } else {
@ -163,7 +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);
@ -451,7 +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) {
@ -539,7 +537,7 @@ int main(int argc, char **argv)
            if (ret < 0)
                goto end;
        }
-        av_packet_unref(&packet);
+        av_free_packet(&packet);
    }
    /* flush filters and encoders */
@ -563,7 +561,7 @@ int main(int argc, char **argv)
    av_write_trailer(ofmt_ctx);
 end:
-    av_packet_unref(&packet);
+    av_free_packet(&packet);
    av_frame_free(&frame);
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        avcodec_close(ifmt_ctx->streams[i]->codec);
@ -575,7 +573,7 @@ end:
    av_free(filter_ctx);
    avformat_close_input(&ifmt_ctx);
    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
+        avio_close(ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);
    if (ret < 0)
--- a/doc/faq.texi
+++ b/doc/faq.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg FAQ
@titlepage
@ -91,63 +90,13 @@ To build FFmpeg, you need to install the development package. It is usually
 called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
 build is finished, but be sure to keep the main package.
@section How do I make @command{pkg-config} find my libraries?
 Somewhere along with your libraries, there is a @file{.pc} file (or several)
 in a @file{pkgconfig} directory. You need to set environment variables to
 point @command{pkg-config} to these files.
 If you need to @emph{add} directories to @command{pkg-config}'s search list
 (typical use case: library installed separately), add it to
@code{$PKG_CONFIG_PATH}:
@example
 export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
@end example
 If you need to @emph{replace} @command{pkg-config}'s search list
 (typical use case: cross-compiling), set it in
@code{$PKG_CONFIG_LIBDIR}:
@example
 export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
@end example
 If you need to know the library's internal dependencies (typical use: static
 linking), add the @code{--static} option to @command{pkg-config}:
@example
 ./configure --pkg-config-flags=--static
@end example
@section How do I use @command{pkg-config} when cross-compiling?
 The best way is to install @command{pkg-config} in your cross-compilation
 environment. It will automatically use the cross-compilation libraries.
 You can also use @command{pkg-config} from the host environment by
 specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
 In that case, you must point @command{pkg-config} to the correct directories
 using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
 As an intermediate solution, you can place in your cross-compilation
 environment a script that calls the host @command{pkg-config} with
@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
@example
 #!/bin/sh
 PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
 export PKG_CONFIG_LIBDIR
 exec /usr/bin/pkg-config "$@@"
@end example
@chapter Usage
@section ffmpeg does not work; what is wrong?
 Try a @code{make distclean} in the ffmpeg source directory before the build.
 If this does not help see
-(@url{https://ffmpeg.org/bugreports.html}).
+(@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
@ -311,18 +260,18 @@ invoking ffmpeg with several @option{-i} options.
 For audio, to put all channels together in a single stream (example: two
 mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
-@url{https://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
+@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
 For audio, to play one on top of the other: this is called to @emph{mix}
 them, and can be done by first merging them into a single stream and then
-using the @url{https://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
+using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
 the channels at will.
@item
 For video, to display both together, side by side or one on top of a part of
 the other; it can be done using the
-@url{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
+@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@ -333,19 +282,19 @@ There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
-FFmpeg has a @url{https://ffmpeg.org/ffmpeg-filters.html#concat,
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
 documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
-FFmpeg has a @url{https://www.ffmpeg.org/ffmpeg-formats.html#concat,
+FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
 your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
-FFmpeg has a @url{https://ffmpeg.org/ffmpeg-protocols.html#concat,
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
 documentation.
@ -467,40 +416,6 @@ point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
 of the encoder you chose.
@section I have a stretched video, why does scaling does not fix it?
 A lot of video codecs and formats can store the @emph{aspect ratio} of the
 video: this is the ratio between the width and the height of either the full
 image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
 ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
 SAR.
 Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
 of video standards, especially from the analogic-numeric transition era, use
 non-square pixels.
 Most processing filters in FFmpeg handle the aspect ratio to avoid
 stretching the image: cropping adjusts the DAR to keep the SAR constant,
 scaling adjusts the SAR to keep the DAR constant.
 If you want to stretch, or “unstretch”, the image, you need to override the
 information with the
@url{https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
 Do not forget to examine carefully the original video to check whether the
 stretching comes from the image or from the aspect ratio information.
 For example, to fix a badly encoded EGA capture, use the following commands,
 either the first one to upscale to square pixels or the second one to set
 the correct aspect ratio or the third one to avoid transcoding (may not work
 depending on the format / codec / player / phase of the moon):
@example
 ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
 ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
 ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@ -589,7 +504,7 @@ see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
-see @url{https://www.ffmpeg.org/~michael/}
+see @url{http://www.ffmpeg.org/~michael/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
--- a/doc/fate.texi
+++ b/doc/fate.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Automated Testing Environment
@titlepage
@ -13,36 +12,36 @@
@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 +50,11 @@ 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
+`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the @var{SAMPLES} makefile
+invoke the makefile targets without setting the SAMPLES makefile
 variable. This is illustrated by the following commands:
@example
@ -64,7 +63,7 @@ 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 +84,7 @@ 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 +92,23 @@ 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
+  Create a configuration that suits your needs, based on the configuration
-template. The @env{slot} configuration variable can be any string that is not
+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
+pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-configuration file itself will be sourced in a shell script, therefore all
+itself will be sourced in a shell script, therefore all shell features may
-shell features may be used. This enables you to setup the environment as you
+be used. This enables you to setup the environment as you need it for your
-need it for your build.
+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 +121,29 @@ 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 +164,7 @@ 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 +182,20 @@ 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},
+Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @samp{slice+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
--- a/doc/fate_config.sh.template
+++ b/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
--- a/doc/ffmpeg-bitstream-filters.texi
+++ b/doc/ffmpeg-bitstream-filters.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Bitstream Filters Documentation
@titlepage
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Codecs Documentation
@titlepage
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Devices Documentation
@titlepage
--- a/doc/ffmpeg-filters.texi
+++ b/doc/ffmpeg-filters.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Filters Documentation
@titlepage
--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Formats Documentation
@titlepage
--- a/doc/ffmpeg-protocols.texi
+++ b/doc/ffmpeg-protocols.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Protocols Documentation
@titlepage
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Resampler Documentation
@titlepage
--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Scaler Documentation
@titlepage
--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Utilities Documentation
@titlepage
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffmpeg Documentation
@titlepage
@ -80,7 +79,7 @@ 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 +98,7 @@ 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 +123,7 @@ 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 +135,19 @@ 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 +160,7 @@ 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 +178,7 @@ 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 +197,14 @@ 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
@ -253,10 +252,6 @@ Overwrite output files without asking.
 Do not overwrite output files, and exit immediately if a specified
 output file already exists.
@item -stream_loop @var{number} (@emph{input})
 Set number of times input stream shall be looped. Loop 0 means no loop,
 loop -1 means infinite loop.
@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 Select an encoder (when used before an output file) or a decoder (when used
@ -284,27 +279,23 @@ 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,
+@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 -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,
+@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 -to and -t are mutually exclusive and -t has priority.
@item -fs @var{limit_size} (@emph{output})
-Set the file size limit, expressed in bytes. No further chunk of bytes is written
+Set the file size limit, expressed in bytes.
 after the limit is exceeded. The size of the output file is slightly more than the
 requested file size.
@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,
+@var{position}. Note the in most formats it is not possible to seek exactly, so
-so @command{ffmpeg} will seek to the closest seek point before @var{position}.
+@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
@ -313,13 +304,7 @@ 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,
+@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
 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.
@item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
@ -334,15 +319,15 @@ 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})
 Set a metadata key/value pair.
 An optional @var{metadata_specifier} may be given to set metadata
-on streams, chapters or programs. See @code{-map_metadata}
+on streams or chapters. See @code{-map_metadata} documentation for
-documentation for details.
+details.
 This option overrides metadata set with @code{-map_metadata}. It is
 also possible to delete metadata by using an empty value.
@ -357,11 +342,6 @@ To set the language of the first audio stream:
 ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
@end example
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
 Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@var{stream}(s) to it.
@item -target @var{type} (@emph{output})
 Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv},
@code{dv50}). @var{type} may be prefixed with @code{pal-}, @code{ntsc-} or
@ -481,9 +461,6 @@ 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
@ -682,16 +659,6 @@ Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@item dxva2
 Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
@item qsv
 Use the Intel QuickSync Video acceleration for video transcoding.
 Unlike most other values, this option does not enable accelerated decoding (that
 is used automatically whenever a qsv decoder is selected), but accelerated
 transcoding, without copying the frames into the system memory.
 For it to work, both the decoder and the encoder must support QSV acceleration
 and no filters must be used.
@end table
 This option has no effect if the selected hwaccel is not available or not
@ -718,27 +685,9 @@ is not specified, the value of the @var{DISPLAY} environment variable is used
@item dxva2
 For DXVA2, this option should contain the number of the display adapter to use.
 If this option is not specified, the default adapter is used.
@item qsv
 For QSV, this option corresponds to the values of MFX_IMPL_* . Allowed values
 are:
@table @option
@item auto
@item sw
@item hw
@item auto_any
@item hw_any
@item hw2
@item hw3
@item hw4
@end table
@end table
@item -hwaccels
 List all hardware acceleration methods supported in this build of ffmpeg.
@end table
@section Audio Options
@table @option
@ -882,14 +831,6 @@ 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
@ -1053,13 +994,6 @@ With -map you can select from which stream the timestamps should be
 taken. You can leave either video or audio unchanged and sync the
 remaining stream(s) to the unchanged one.
@item -frame_drop_threshold @var{parameter}
 Frame drop threshold, which specifies how much behind video frames can
 be before they are dropped. In frame rate units, so 1.0 is one frame.
 The default is -1.1. One possible usecase is to avoid framedrops in case
 of noisy timestamps or to increase frame drop precision in case of exact
 timestamps.
@item -async @var{samples_per_second}
 Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
 the parameter is the maximum samples per second by which the audio is changed.
@ -1216,19 +1150,6 @@ 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})
 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
 avoid it.
@item -override_ffserver (@emph{global})
 Overrides the input specifications from @command{ffserver}. Using this
 option you can map any input stream to @command{ffserver} and control
@ -1239,11 +1160,6 @@ requested by @command{ffserver}.
 The option is intended for cases where features are needed that cannot be
 specified to @command{ffserver} but can be to @command{ffmpeg}.
@item -sdp_file @var{file} (@emph{global})
 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. (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.
 Not all demuxers support this.
@ -1268,17 +1184,6 @@ Discard all frames excepts keyframes.
 Discard all frames.
@end table
@item -abort_on @var{flags} (@emph{global})
 Stop and abort on various conditions. The following flags are available:
@table @option
@item empty_output
 No packets were passed to the muxer, the output is empty.
@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
@ -1304,10 +1209,7 @@ awkward to specify on the command line. Lines starting with the hash
 ('#') character are ignored and are used to provide comments. Check
 the @file{presets} directory in the FFmpeg source tree for examples.
-There are two types of preset files: ffpreset and avpreset files.
+Preset files are specified with the @code{vpre}, @code{apre},
@subsection ffpreset files
 ffpreset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
 filename of the preset instead of a preset name as input and can be
 used for any kind of codec. For the @code{vpre}, @code{apre}, and
@ -1332,31 +1234,67 @@ directories, where @var{codec_name} is the name of the codec to which
 the preset file options will be applied. For example, if you select
 the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p},
 then it will search for the file @file{libvpx-1080p.ffpreset}.
@subsection avpreset files
 avpreset files are specified with the @code{pre} option. They work similar to
 ffpreset files, but they only allow encoder- specific options. Therefore, an
@var{option}=@var{value} pair specifying an encoder cannot be used.
 When the @code{pre} option is specified, ffmpeg will look for files with the
 suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and
@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually
@file{PREFIX/share/ffmpeg}), in that order.
 First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in
 the above-mentioned directories, where @var{codec_name} is the name of the codec
 to which the preset file options will be applied. For example, if you select the
 video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will
 search for the file @file{libvpx-1080p.avpreset}.
 If no such file is found, then ffmpeg will search for a file named
@var{arg}.avpreset in the same directories.
@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
@section Preset files
 A preset file contains a sequence of @var{option=value} pairs, one for
 each line, specifying a sequence of options which can be specified also on
 the command line. Lines starting with the hash ('#') character are ignored and
 are used to provide comments. Empty lines are also ignored. Check the
@file{presets} directory in the FFmpeg source tree for examples.
 Preset files are specified with the @code{pre} option, this option takes a
 preset name as input.  FFmpeg searches for a file named @var{preset_name}.avpreset in
 the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
 the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg})
 in that order.  For example, if the argument is @code{libx264-max}, it will
 search for the file @file{libx264-max.avpreset}.
@section Video and Audio grabbing
 If you specify the input format and device then ffmpeg can grab video
@ -1504,7 +1442,7 @@ 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
@ -1519,7 +1457,7 @@ 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
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffplay Documentation
@titlepage
@ -47,17 +46,9 @@ 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
+Seek to a given position in seconds.
 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}.
@item -t @var{duration}
-Play @var{duration} seconds of audio/video.
+play <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}.
@item -bytes
 Seek by bytes.
@item -nodisp
@ -133,20 +124,24 @@ master clock is used to control audio-video synchronization. Most media
 players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
-@item -ast @var{audio_stream_specifier}
+@item -threads @var{count}
-Select the desired audio stream using the given stream specifier. The stream
+Set the thread count. By default, @command{ffplay} automatically detects a
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
+suitable number of threads to use.
-is not specified, the "best" audio stream is selected in the program of the
+@item -ast @var{audio_stream_number}
-already selected video stream.
+Select the desired audio stream number, counting from 0. The number
-@item -vst @var{video_stream_specifier}
+refers to the list of all the input audio streams. If it is greater
-Select the desired video stream using the given stream specifier. The stream
+than the number of audio streams minus one, then the last one is
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
+selected, if it is negative the audio playback is disabled.
-is not specified, the "best" video stream is selected.
+@item -vst @var{video_stream_number}
-@item -sst @var{subtitle_stream_specifier}
+Select the desired video stream number, counting from 0. The number
-Select the desired subtitle stream using the given stream specifier. The stream
+refers to the list of all the input video streams. If it is greater
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
+than the number of video streams minus one, then the last one is
-is not specified, the "best" subtitle stream is selected in the program of the
+selected, if it is negative the video playback is disabled.
-already selected video or audio stream.
+@item -sst @var{subtitle_stream_number}
 Select the desired subtitle stream number, counting from 0. The number
 refers to the list of all the input subtitle streams. If it is greater
 than the number of subtitle streams minus one, then the last one is
 selected, if it is negative the subtitle rendering is disabled.
@item -autoexit
 Exit when video is done playing.
@item -exitonkeydown
@ -169,7 +164,7 @@ 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
@ -197,15 +192,6 @@ Toggle full screen.
@item p, SPC
 Pause.
@item m
 Toggle mute.
@item 9, 0
 Decrease and increase volume respectively.
@item /, *
 Decrease and increase volume respectively.
@item a
 Cycle audio channel in the current program.
@ -238,12 +224,9 @@ Seek to the previous/next chapter.
 or if there are no chapters
 Seek backward/forward 10 minutes.
-@item right mouse click
+@item mouse click
 Seek to percentage in file corresponding to fraction of width.
@item left mouse double-click
 Toggle full screen.
@end table
@c man end
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffprobe Documentation
@titlepage
@ -447,17 +446,17 @@ writer).
 It can assume one of the following values:
@table @option
@item c
-Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
+Perform C-like escaping. Strings containing a newline ('\n'), carriage
-return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
+return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
-character (@samp{\}) or the item separator character @var{SEP} are escaped
+character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
-using C-like fashioned escaping, so that a newline is converted to the
+escaping, so that a newline is converted to the sequence "\n", a
-sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
+carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
-the separator @var{SEP} is converted to @samp{\@var{SEP}}.
+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
+containing a newline ('\n'), a carriage return ('\r'), a double quote
-(@samp{"}), or @var{SEP} are enclosed in double-quotes.
+('"'), or @var{SEP} are enclosed in double-quotes.
@item none
 Perform no escaping.
@ -485,7 +484,7 @@ 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 +506,21 @@ 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
+newline, '\t', '\f', '\b' and the following characters are escaped
 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.
--- a/doc/ffprobe.xsd
+++ b/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"/>
@ -129,7 +113,6 @@
    <xsd:complexType name="frameSideDataType">
        <xsd:attribute name="side_data_type"              type="xsd:string"/>
        <xsd:attribute name="side_data_size"              type="xsd:int"   />
        <xsd:attribute name="timecode"                    type="xsd:string"/>
    </xsd:complexType>
    <xsd:complexType name="subtitleType">
@ -172,7 +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"/>
@ -189,8 +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"/>
@ -202,7 +182,6 @@
      <xsd:attribute name="color_primaries"      type="xsd:string"/>
      <xsd:attribute name="chroma_location"      type="xsd:string"/>
      <xsd:attribute name="timecode"             type="xsd:string"/>
      <xsd:attribute name="refs"                 type="xsd:int"/>
      <!-- audio attributes -->
      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
@ -274,8 +253,8 @@
    <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_date"       type="xsd:string" use="required"/>
-      <xsd:attribute name="build_time"       type="xsd:string"/>
+      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
      <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>
--- a/doc/ffserver.conf
+++ b/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)
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffserver Documentation
@titlepage
@ -118,8 +117,7 @@ Multiple streams can be connected to the same feed.
 For example, you can have a situation described by the following
 graph:
-
+@example
@verbatim
               _________       __________
              |         |     |          |
 ffmpeg 1 -----| feed 1  |-----| stream 1 |
@ -144,8 +142,7 @@ ffmpeg 2 -----| feed 3  |-----| stream 4 |
              |         |     |          |
              | file 1  |-----| stream 5 |
              |_________|     |__________|
-
+@end example
@end verbatim
@anchor{FFM}
@section FFM, FFM2 formats
--- a/doc/fftools-common-opts.texi
+++ b/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'
+@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
-for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
+'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
 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
 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.
@ -170,29 +165,28 @@ omitted. "repeat" can also be used alone.
 If "repeat" is used alone, and with no prior loglevel set, the default
 loglevel will be used. If multiple loglevel parameters are given, using
 'repeat' will not change the loglevel.
-@var{loglevel} is a string or a number containing one of the following values:
+@var{loglevel} is a number or a string containing one of the following values:
@table @samp
-@item quiet, -8
+@item quiet
 Show nothing at all; be silent.
-@item panic, 0
+@item panic
 Only show fatal errors which could lead the process to crash, such as
 and assert failure. This is not currently used for anything.
-@item fatal, 8
+@item fatal
 Only show fatal errors. These are errors after which the process absolutely
 cannot continue after.
-@item error, 16
+@item error
 Show all errors, including ones which can be recovered from.
-@item warning, 24
+@item warning
 Show all warnings and errors. Any message related to possibly
 incorrect or unexpected events will be shown.
-@item info, 32
+@item info
 Show informative messages during processing. This is in addition to
 warnings and errors. This is the default value.
-@item verbose, 40
+@item verbose
 Same as @code{info}, except more verbose.
-@item debug, 48
+@item debug
 Show everything, including debugging information.
@item trace, 56
@end table
 By default the program logs to stderr, if coloring is supported by the
@ -210,29 +204,21 @@ directory.
 This file can be useful for bug reports.
 It also implies @code{-loglevel verbose}.
-Setting the environment variable @env{FFREPORT} to any value has the
+Setting the environment variable @code{FFREPORT} to any value has the
 same effect. If the value is a ':'-separated key=value sequence, these
-options will affect the report; option values must be escaped if they
+options will affect the report; options values must be escaped if they
 contain special characters or the options delimiter ':' (see the
-``Quoting and escaping'' section in the ffmpeg-utils manual).
+``Quoting and escaping'' section in the ffmpeg-utils manual). The
-
+following option is recognized:
 The following options are recognized:
@table @option
@item file
 set the file name to use for the report; @code{%p} is expanded to the name
 of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
 to a plain @code{%}
@item level
-set the log verbosity level using a numerical value (see @code{-loglevel}).
+set the log level
@end table
 For example, to output a report to a file named @file{ffreport.log}
 using a log level of @code{32} (alias for log level @code{info}):
@example
 FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
@end example
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.
@ -311,41 +297,8 @@ Possible flags for this option are:
@end table
@item -opencl_bench
-This option is used to benchmark all available OpenCL devices and print the
+Benchmark all available OpenCL devices and show the results. This option
-results. This option is only available when FFmpeg has been compiled with
+is only available when FFmpeg has been compiled with @code{--enable-opencl}.
@code{--enable-opencl}.
 When FFmpeg is configured with @code{--enable-opencl}, the options for the
 global OpenCL context are set via @option{-opencl_options}. See the
 "OpenCL Options" section in the ffmpeg-utils manual for the complete list of
 supported options. Amongst others, these options include the ability to select
 a specific platform and device to run the OpenCL code on. By default, FFmpeg
 will run on the first device of the first platform. While the options for the
 global OpenCL context provide flexibility to the user in selecting the OpenCL
 device of their choice, most users would probably want to select the fastest
 OpenCL device for their system.
 This option assists the selection of the most efficient configuration by
 identifying the appropriate device for the user's system. The built-in
 benchmark is run on all the OpenCL devices and the performance is measured for
 each device. The devices in the results list are sorted based on their
 performance with the fastest device listed first. The user can subsequently
 invoke @command{ffmpeg} using the device deemed most appropriate via
@option{-opencl_options} to obtain the best performance for the OpenCL
 accelerated code.
 Typical usage to use the fastest OpenCL device involve the following steps.
 Run the command:
@example
 ffmpeg -opencl_bench
@end example
 Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
 i.e. fastest device in the list.
 Select the platform and device using the command:
@example
 ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
@end example
@item -opencl_options options (@emph{global})
 Set OpenCL environment options. This option is only available when
--- a/doc/filter_design.txt
+++ b/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
@ -232,8 +232,7 @@ Frame scheduling
    one of its inputs, repeatedly until at least one frame has been pushed.
    Return values:
-    if request_frame could produce a frame, or at least make progress
+    if request_frame could produce a frame, it should return 0;
    towards producing a frame, it should return 0;
    if it could not for temporary reasons, it should return AVERROR(EAGAIN);
    if it could not because there are no more frames, it should return
    AVERROR_EOF.
@ -245,18 +244,20 @@ Frame scheduling
            push_one_frame();
            return 0;
        }
-        input = input_where_a_frame_is_most_needed();
+        while (!frame_pushed) {
-        ret = ff_request_frame(input);
+            input = input_where_a_frame_is_most_needed();
-        if (ret == AVERROR_EOF) {
+            ret = ff_request_frame(input);
-            process_eof_on_input();
+            if (ret == AVERROR_EOF) {
-        } else if (ret < 0) {
+                process_eof_on_input();
-            return ret;
+            } else if (ret < 0) {
                return ret;
            }
        }
        return 0;
    Note that, except for filters that can have queued frames, request_frame
    does not push frames: it requests them to its input, and as a reaction,
-    the filter_frame method possibly will be called and do the work.
+    the filter_frame method will be called and do the work.
 Legacy API
 ==========
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/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
@ -127,27 +125,8 @@ 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 if set to 1. Default is 0.
+Use wallclock as timestamps.
@item avoid_negative_ts @var{integer} (@emph{output})
--- a/doc/general.texi
+++ b/doc/general.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle General Documentation
@titlepage
@ -53,6 +52,14 @@ instructions for installing the libraries.
 Then pass @code{--enable-libopencore-amrnb} and/or
@code{--enable-libopencore-amrwb} to configure to enable them.
@subsection VisualOn AAC encoder library
 FFmpeg can make use of the VisualOn AACenc library for AAC encoding.
 Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libvo-aacenc} to configure to enable it.
@subsection VisualOn AMR-WB encoder library
 FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding.
@ -101,21 +108,6 @@ Go to @url{http://www.wavpack.com/} and follow the instructions for
 installing the library. Then pass @code{--enable-libwavpack} to configure to
 enable it.
@section OpenH264
 FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.
 Go to @url{http://www.openh264.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libopenh264} to configure to
 enable it.
 For decoding, this library is much more limited than the built-in decoder
 in libavcodec; currently, this library lacks support for decoding B-frames
 and some other main/high profile features. (It currently only supports
 constrained baseline profile and CABAC.) Using it is mostly useful for
 testing and for taking advantage of Cisco's patent portfolio license
 (@url{http://www.openh264.org/BINARY_LICENSE.txt}).
@section x264
 FFmpeg can make use of the x264 library for H.264 encoding.
@ -144,14 +136,6 @@ 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
@ -159,7 +143,7 @@ by Google as part of the WebRTC project. libilbc is a packaging friendly
 copy of the iLBC codec. FFmpeg can make use of the libilbc library for
 iLBC encoding and decoding.
-Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
+Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.
@ -172,6 +156,12 @@ Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions
 installing the library. Then pass @code{--enable-libzvbi} to configure to
 enable it.
@float NOTE
 libzvbi is licensed under the GNU General Public License Version 2 or later
 (see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
 you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section AviSynth
 FFmpeg can read AviSynth scripts as input. To enable support, pass
@ -180,8 +170,8 @@ 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://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
+@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}.
@ -193,17 +183,6 @@ 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
@ -216,14 +195,9 @@ library:
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 3dostr                    @tab   @tab X
@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
@ -235,15 +209,10 @@ library:
    @tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab   @tab X
@item Artworx Data Format       @tab   @tab X
@item Interplay ACM             @tab   @tab X
    @tab Audio only format used in some Interplay games.
@item ADP                       @tab   @tab X
    @tab Audio format used on the Nintendo Gamecube.
@item AFC                       @tab   @tab X
    @tab Audio format used on the Nintendo Gamecube.
@item ADS/SS2                   @tab   @tab X
    @tab Audio format used on the PS2.
@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.
@ -264,8 +233,6 @@ 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
@ -276,14 +243,9 @@ library:
    @tab Used in the game Cyberia from Interplay.
@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
@item Phantom Cine              @tab   @tab X
@item Cineform HD               @tab   @tab X
@item Commodore CDXL            @tab   @tab X
    @tab Amiga CD video format
@item Core Audio Format         @tab X @tab X
@ -295,10 +257,8 @@ library:
    @tab Audio format used in some games by CRYO Interactive Entertainment.
@item D-Cinema audio            @tab X @tab X
@item Deluxe Paint Animation    @tab   @tab X
@item DCSTR                     @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
@ -322,8 +282,6 @@ library:
@item G.723.1                   @tab X @tab X
@item G.729 BIT                 @tab X @tab X
@item G.729 raw                 @tab   @tab X
@item GENH                      @tab   @tab X
    @tab Audio format for various games.
@item GIF Animation             @tab X @tab X
@item GXF                       @tab X @tab X
    @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
@ -346,7 +304,6 @@ library:
    @tab A format generated by IndigoVision 8000 video server.
@item IVF (On2)                 @tab X @tab X
    @tab A format used by libvpx
@item Internet Video Recording  @tab   @tab X
@item IRCAM                     @tab X @tab X
@item LATM                      @tab X @tab X
@item LMLM4                     @tab   @tab X
@ -383,8 +340,6 @@ library:
    @tab also known as DVB Transport Stream
@item MPEG-4                    @tab X @tab X
    @tab MPEG-4 is a variant of QuickTime.
@item MSF                       @tab   @tab X
    @tab Audio format used on the PS3.
@item Mirillis FIC video        @tab   @tab X
    @tab No cursor rendering.
@item MIME multipart JPEG       @tab X @tab
@ -468,7 +423,6 @@ library:
@item Redirector                @tab   @tab X
@item RedSpark                  @tab   @tab X
@item Renderware TeXture Dictionary @tab   @tab X
@item Resolume DXV              @tab   @tab X
@item RL2                       @tab   @tab X
    @tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie               @tab   @tab X
@ -501,22 +455,14 @@ 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 SVAG                      @tab   @tab X
    @tab Audio format used in Konami PS2 games.
@item TDSC                      @tab   @tab X
@item Text files                @tab   @tab X
@item THP                       @tab   @tab X
    @tab Used on the Nintendo GameCube.
@item Tiertex Limited SEQ       @tab   @tab X
    @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio                @tab   @tab X
@item VAG                       @tab   @tab X
    @tab Audio format used in many Sony PS2 games.
@item VC-1 test bitstream       @tab X @tab X
@item Vidvox Hap                @tab X @tab X
@item Vivo                      @tab   @tab X
@item VPK                       @tab   @tab X
    @tab Audio format used in Sony PS games.
@item WAV                       @tab X @tab X
@item WavPack                   @tab X @tab X
@item WebM                      @tab X @tab X
@ -527,12 +473,8 @@ library:
    @tab Multimedia format used in Westwood Studios games.
@item Westwood Studios VQA      @tab   @tab X
    @tab Multimedia format used in Westwood Studios games.
@item Wideband Single-bit Data (WSD) @tab   @tab X
@item WVE                       @tab   @tab X
@item XMV                       @tab   @tab X
    @tab Microsoft video container used in Xbox games.
@item XVAG                      @tab   @tab X
    @tab Audio format used on the PS3.
@item xWMA                      @tab   @tab X
    @tab Microsoft audio container used by XAudio 2.
@item eXtended BINary text (XBIN) @tab @tab X
@ -554,7 +496,6 @@ 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
@ -651,7 +592,6 @@ following image formats are supported:
@item Bethesda VID video     @tab     @tab  X
    @tab Used in some games from Bethesda Softworks.
@item Bink Video             @tab     @tab  X
@item BitJazz SheerVideo     @tab     @tab  X
@item Bitmap Brothers JV video  @tab   @tab X
@item y41p Brooktree uncompressed 4:1:1 12-bit     @tab  X  @tab  X
@item Brute Force & Ignorance   @tab   @tab X
@ -686,8 +626,6 @@ following image formats are supported:
    @tab fourcc: DUCK
@item Duck TrueMotion 2.0    @tab     @tab  X
    @tab fourcc: TM20
@item Duck TrueMotion 2.0 RT @tab     @tab  X
    @tab fourcc: TR20
@item DV (Digital Video)     @tab  X  @tab  X
@item Dxtory capture format  @tab     @tab  X
@item Feeble Files/ScummVM DXA  @tab     @tab  X
@ -709,17 +647,15 @@ 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
@item H.263 / H.263-1996     @tab  X  @tab  X
@item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
-    @tab encoding supported through external library libx264 and OpenH264
+    @tab encoding supported through external library libx264
@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
@ -754,7 +690,6 @@ following image formats are supported:
@item LucasArts SANM/Smush   @tab     @tab  X
    @tab Used in LucasArts games / SMUSH animations.
@item lossless MJPEG         @tab  X  @tab  X
@item MagicYUV Video         @tab     @tab  X
@item Microsoft ATC Screen   @tab     @tab  X
    @tab Also known as Microsoft Screen 3.
@item Microsoft Expression Encoder Screen  @tab     @tab  X
@ -819,7 +754,6 @@ following image formats are supported:
    @tab Texture dictionaries used by the Renderware Engine.
@item RL2 video              @tab     @tab  X
    @tab used in some games by Entertainment Software Partners
@item Screenpresso           @tab     @tab  X
@item Sierra VMD video       @tab     @tab  X
    @tab Used in Sierra VMD files.
@item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
@ -886,13 +820,12 @@ 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 EX  @tab  X
+@item AAC+                   @tab  E  @tab  X
-    @tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac
+    @tab encoding supported through external library libaacplus
-@item AAC+                   @tab  E  @tab  IX
+@item AAC                    @tab  E  @tab  X
-    @tab encoding supported through external library libfdk-aac
+    @tab encoding supported through external library libfaac and libvo-aacenc
@item AC-3                   @tab IX  @tab  IX
@item ADPCM 4X Movie         @tab     @tab  X
@item APDCM Yamaha AICA      @tab     @tab  X
@item ADPCM CDROM XA         @tab     @tab  X
@item ADPCM Creative Technology @tab     @tab  X
    @tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2
@ -927,8 +860,7 @@ 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 APDCM Playstation      @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.
@ -936,7 +868,7 @@ following image formats are supported:
@item ADPCM Sound Blaster Pro 2-bit  @tab     @tab  X
@item ADPCM Sound Blaster Pro 2.6-bit  @tab     @tab  X
@item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
-@item ADPCM VIMA             @tab     @tab  X
+@item ADPCM VIMA
    @tab Used in LucasArts SMUSH animations.
@item ADPCM Westwood Studios IMA @tab     @tab  X
    @tab Used in Westwood Studios games like Command and Conquer.
@ -957,18 +889,14 @@ following image formats are supported:
    @tab decoding supported through external library libcelt
@item Delphine Software International CIN audio  @tab     @tab  X
    @tab Codec used in Delphine Software International games.
@item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
@item Discworld II BMV Audio @tab     @tab  X
@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, XXCH, X96, XBR, XLL, LBR (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
    @tab Used in various Interplay computer games.
@item DPCM Squareroot-Delta-Exact  @tab  @tab  X
    @tab Used in various games.
@item DPCM Sierra Online     @tab     @tab  X
    @tab Used in Sierra Online game audio files.
@item DPCM Sol               @tab     @tab  X
@ -979,12 +907,11 @@ following image formats are supported:
@item DSD (Direct Stream Digitial), least significant bit first, planar  @tab  @tab  X
@item DSD (Direct Stream Digitial), most significant bit first, planar   @tab  @tab  X
@item DSP Group TrueSpeech   @tab     @tab  X
@item DST (Direct Stream Transfer) @tab  @tab  X
@item DV audio               @tab     @tab  X
@item Enhanced AC-3          @tab  X  @tab  X
@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
@item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
-@item G.723.1                @tab X   @tab  X
+@item G.723.1                @tab X @tab X
@item G.729                  @tab     @tab  X
@item GSM                    @tab  E  @tab  X
    @tab encoding supported through external library libgsm
@ -994,7 +921,6 @@ following image formats are supported:
@item iLBC (Internet Low Bitrate Codec) @tab  E  @tab  E
    @tab encoding and decoding supported through external library libilbc
@item IMC (Intel Music Coder)  @tab     @tab  X
@item Interplay ACM            @tab     @tab  X
@item MACE (Macintosh Audio Compression/Expansion) 3:1  @tab     @tab  X
@item MACE (Macintosh Audio Compression/Expansion) 6:1  @tab     @tab  X
@item MLP (Meridian Lossless Packing)  @tab     @tab  X
@ -1079,8 +1005,6 @@ following image formats are supported:
@item Windows Media Audio Lossless @tab  @tab  X
@item Windows Media Audio Pro @tab    @tab  X
@item Windows Media Audio Voice @tab  @tab  X
@item Xbox Media Audio 1     @tab     @tab  X
@item Xbox Media Audio 2     @tab     @tab  X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@ -1,10 +1,9 @@
 \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 +12,9 @@
@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 +31,22 @@ 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.
@ -65,21 +65,6 @@ git clone git@@source.ffmpeg.org:ffmpeg <target>
 This will put the FFmpeg sources into the directory @var{<target>} and let
 you push back your changes to the remote repository.
@example
 git clone gil@@ffmpeg.org:ffmpeg-web <target>
@end example
 This will put the source of the FFmpeg website into the directory
@var{<target>} and let you push back your changes to the remote repository.
 (Note that @var{gil} stands for GItoLite and is not a typo of @var{git}.)
 If you don't have write-access to the ffmpeg-web repository, you can
 create patches after making a read-only ffmpeg-web clone:
@example
 git clone git://ffmpeg.org/ffmpeg-web <target>
@end example
 Make sure that you do not have Windows line endings in your checkouts,
 otherwise you may experience spurious compilation failures. One way to
 achieve this is to run
@ -89,7 +74,6 @@ 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
@ -122,7 +106,7 @@ 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.
@ -142,8 +126,8 @@ 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}
+You may also use the graphical tools like gitview or gitk or the web
-or the web interface available at @url{http://source.ffmpeg.org/}.
+interface available at http://source.ffmpeg.org/
@section Checking source tree status
@ -164,7 +148,6 @@ 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.
@ -187,14 +170,14 @@ 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
@ -225,7 +208,7 @@ 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
@ -341,14 +324,12 @@ 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
+Will push the changes to the default remote (@var{origin}).
 (@var{origin}). And list which branches and ranges or commits would have been
 pushed.
 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>
@ -367,24 +348,23 @@ 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
@ -393,7 +373,7 @@ 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
@ -404,21 +384,21 @@ 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.
-Next let the code pass through a full run of our test suite.
+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
 Make sure all your changes have been checked before pushing them, the
-test suite only checks against regressions and that only to some extend. It does
+testsuite only checks against regressions and that only to some extend. It does
 obviously not check newly added features/code to be working unless you have
 added a test for that (which is recommended).
@ -431,5 +411,5 @@ recommended.
@chapter Server Issues
-Contact the project admins at @email{root@@ffmpeg.org} if you have technical
+Contact the project admins @email{root@@ffmpeg.org} if you have technical
-problems with the Git server.
+problems with the GIT server.
--- a/doc/indevs.texi
+++ b/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.
@ -121,24 +109,11 @@ Specify the audio device by its index. Overrides anything given in the input fil
@item -pixel_format <FORMAT>
 Request the video device to use a specific pixel format.
 If the specified format is not supported, a list of available formats is given
-and the first one in this list is used instead. Available pixel formats are:
+und the first one in this list is used instead. Available pixel formats are:
@code{monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
 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
@ -175,142 +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
 DeckLink devices.
 To enable this input device, you need the Blackmagic DeckLink SDK and you
 need to configure with the appropriate @code{--extra-cflags}
 and @code{--extra-ldflags}.
 On Windows, you need to run the IDL files through @command{widl}.
 DeckLink is very picky about the formats it supports. Pixel format is
 uyvy422 or v210, framerate and video size must be determined for your device with
@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
 of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single
 audio track.
@subsection Options
@table @option
@item list_devices
 If set to @option{true}, print a list of devices and exit.
 Defaults to @option{false}.
@item list_formats
 If set to @option{true}, print a list of supported formats and exit.
 Defaults to @option{false}.
@item bm_v210
 If set to @samp{1}, video is captured in 10 bit v210 instead
 of uyvy422. Not all Blackmagic devices support this option.
@item teletext_lines
 If set to nonzero, an additional teletext stream will be captured from the
 vertical ancillary data. This option is a bitmask of the VBI lines checked,
 specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask.
 Selected lines which do not contain teletext information will be ignored. You
 can use the special @option{all} constant to select all possible lines, or
@option{standard} to skip lines 6, 318 and 319, which are not compatible with all
 receivers. Capturing teletext only works for SD PAL sources in 8 bit mode.
 To use this option, ffmpeg needs to be compiled with @code{--enable-libzvbi}.
@item channels
 Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp{16}.
 Defaults to @samp{2}.
@item duplex_mode
 Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.
@item video_input
 Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
 Defaults to @samp{unset}.
@item audio_input
 Sets the audio input source. Must be @samp{unset}, @samp{embedded},
@samp{aes_ebu}, @samp{analog}, @samp{analog_xlr}, @samp{analog_rca} or
@samp{microphone}. Defaults to @samp{unset}.
@item video_pts
 Sets the video packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference} or @samp{wallclock}. Defaults to @samp{video}.
@item audio_pts
 Sets the audio packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference} or @samp{wallclock}. Defaults to @samp{audio}.
@end table
@subsection Examples
@itemize
@item
 List input devices:
@example
 ffmpeg -f decklink -list_devices 1 -i dummy
@end example
@item
 List supported formats:
@example
 ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
@end example
@item
 Capture video clip at 1080i50 (format 11):
@example
 ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
@end example
@item
 Capture video clip at 1080i50 10 bit:
@example
 ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@item
 Capture video clip at 1080i50 with 16 audio channels:
@example
 ffmpeg -channels 16 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@section dshow
 Windows DirectShow input device.
@ -328,7 +167,7 @@ The input name should be in the format:
@end example
 where @var{TYPE} can be either @var{audio} or @var{video},
-and @var{NAME} is the device's name or alternative name..
+and @var{NAME} is the device's name.
@subsection Options
@ -361,11 +200,11 @@ 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
@ -381,85 +220,6 @@ Setting this value too low can degrade performance.
 See also
@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
@item video_pin_name
 Select video capture pin to use by name or alternative name.
@item audio_pin_name
 Select audio capture pin to use by name or alternative name.
@item crossbar_video_input_pin_number
 Select video input pin number for crossbar device. This will be
 routed to the crossbar device's Video Decoder output pin.
 Note that changing this value can affect future invocations
 (sets a new default) until system reboot occurs.
@item crossbar_audio_input_pin_number
 Select audio input pin number for crossbar device. This will be
 routed to the crossbar device's Audio Decoder output pin.
 Note that changing this value can affect future invocations
 (sets a new default) until system reboot occurs.
@item show_video_device_dialog
 If set to @option{true}, before capture starts, popup a display dialog
 to the end user, allowing them to change video filter properties
 and configurations manually.
 Note that for crossbar devices, adjusting values in this dialog
 may be needed at times to toggle between PAL (25 fps) and NTSC (29.97)
 input frame rates, sizes, interlacing, etc.  Changing these values can
 enable different scan rates/frame rates and avoiding green bars at
 the bottom, flickering scan lines, etc.
 Note that with some devices, changing these properties can also affect future
 invocations (sets new defaults) until system reboot occurs.
@item show_audio_device_dialog
 If set to @option{true}, before capture starts, popup a display dialog
 to the end user, allowing them to change audio filter properties
 and configurations manually.
@item show_video_crossbar_connection_dialog
 If set to @option{true}, before capture starts, popup a display
 dialog to the end user, allowing them to manually
 modify crossbar pin routings, when it opens a video device.
@item show_audio_crossbar_connection_dialog
 If set to @option{true}, before capture starts, popup a display
 dialog to the end user, allowing them to manually
 modify crossbar pin routings, when it opens an audio device.
@item show_analog_tv_tuner_dialog
 If set to @option{true}, before capture starts, popup a display
 dialog to the end user, allowing them to manually
 modify TV channels and frequencies.
@item show_analog_tv_tuner_audio_dialog
 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
@ -496,46 +256,12 @@ Print the list of supported options in selected device and exit:
 $ ffmpeg -list_options true -f dshow -i video="Camera"
@end example
@item
 Specify pin names to capture by name or alternative name, specify alternative device name:
@example
 $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#@{65e8773d-8f56-11d0-a3b9-00a0c9223196@}\@{ca465100-deb0-4d59-818f-8c477184adf6@}":audio="Microphone"
@end example
@item
 Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup:
@example
 $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
     -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
@end example
@end itemize
@section dv1394
 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.
@ -548,27 +274,18 @@ 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
+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@table @option
@item framerate
 Set the frame rate. Default is 25.
@end table
@section gdigrab
@ -673,7 +390,7 @@ not work and result in undefined behavior.
 The values @option{auto}, @option{dv} and @option{hdv} are supported.
@item dvbuffer
-Set maximum size of buffer for incoming data, in frames. For DV, this
+Set maxiumum size of buffer for incoming data, in frames. For DV, this
 is an exact value. For HDV, it is not frame exact, since HDV does
 not have a fixed frame size.
@ -754,15 +471,6 @@ $ 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.
@ -787,14 +495,6 @@ generated by the device.
 The first unlabelled output is automatically assigned to the "out0"
 label, but all the others need to be specified explicitly.
 The suffix "+subcc" can be appended to the output label to create an extra
 stream with the closed captions packets attached to that output
 (experimental; only for EIA-608 / CEA-708 for now).
 The subcc streams are created after all the normal streams, in the order of
 the corresponding stream.
 For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the
 stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19.
 If not specified defaults to the filename specified for the input
 device.
@ -803,9 +503,6 @@ 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
@ -844,57 +541,24 @@ Read an audio stream and a video stream and play it back with
 ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end example
@item
 Dump decoded frames to images and closed captions to a file (experimental):
@example
 ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
@end example
@end itemize
@section libcdio
-Audio-CD input device based on libcdio.
+Audio-CD input device based on cdio.
 To enable this input device during configuration you need libcdio
-installed on your system. It requires the configure option
+installed on your system. Requires the configure option
@code{--enable-libcdio}.
 This device allows playing and grabbing from an Audio-CD.
-For example to copy with @command{ffmpeg} the entire Audio-CD in @file{/dev/sr0},
+For example to copy with @command{ffmpeg} the entire Audio-CD in /dev/sr0,
 you may run the command:
@example
 ffmpeg -f libcdio -i /dev/sr0 cd.wav
@end example
@subsection Options
@table @option
@item speed
 Set drive reading speed. Default value is 0.
 The speed is specified CD-ROM speed units. The speed is set through
 the libcdio @code{cdio_cddap_speed_set} function. On many CD-ROM
 drives, specifying a value too large will result in using the fastest
 speed.
@item paranoia_mode
 Set paranoia recovery mode flags. It accepts one of the following values:
@table @samp
@item disable
@item verify
@item overlap
@item neverskip
@item full
@end table
 Default value is @samp{disable}.
 For more information about the available recovery modes, consult the
 paranoia project documentation.
@end table
@section libdc1394
 IIDC1394 input device, based on libdc1394 and libraw1394.
@ -1007,19 +671,6 @@ 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.
@ -1060,10 +711,6 @@ 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
@ -1099,22 +746,6 @@ 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.
@ -1132,18 +763,6 @@ 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.
@ -1176,12 +795,6 @@ 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
@ -1266,10 +879,6 @@ 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
@ -1280,29 +889,12 @@ 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.
-To enable this input device during configuration you need libxcb
+Depends on X11, Xext, and Xfixes. Requires the configure option
-installed on your system. It will be automatically detected during
+@code{--enable-x11grab}.
 configuration.
 Alternatively, the configure option @option{--enable-x11grab} exists
 for legacy Xlib users.
 This device allows one to capture a region of an X11 display.
@ -1320,12 +912,10 @@ omitted, and defaults to "localhost". The environment variable
 area with respect to the top-left border of the X11 screen. They
 default to 0.
-Check the X11 documentation (e.g. @command{man X}) for more detailed
+Check the X11 documentation (e.g. man X) for more detailed information.
 information.
-Use the @command{xdpyinfo} program for getting basic information about
+Use the @command{dpyinfo} program for getting basic information about the
-the properties of your X11 display (e.g. grep for "name" or
+properties of your X11 display (e.g. grep for "name" or "dimensions").
 "dimensions").
 For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
@ -1374,10 +964,6 @@ If @var{show_region} is specified with @code{1}, then the grabbing
 region will be indicated on screen. With this option, it is easy to
 know what is being grabbed if only a portion of the screen is grabbed.
@item region_border
 Set the region border thickness if @option{-show_region 1} is used.
 Range is 1 to 128 and default is 3 (XCB-based x11grab only).
 For example:
@example
 ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@ -1393,15 +979,61 @@ Set the video frame size. Default value is @code{vga}.
@item use_shm
 Use the MIT-SHM extension for shared memory. Default value is @code{1}.
-It may be necessary to disable it for remote displays (legacy x11grab
+It may be necessary to disable it for remote displays.
 only).
@item grab_x
@item grab_y
 Set the grabbing region coordinates. They are expressed as offset from
 the top left corner of the X11 window and correspond to the
@var{x_offset} and @var{y_offset} parameters in the device name. The
 default value for both options is 0.
@end table
@section decklink
 The decklink input device provides capture capabilities for Blackmagic
 DeckLink devices.
 To enable this input device, you need the Blackmagic DeckLink SDK and you
 need to configure with the appropriate @code{--extra-cflags}
 and @code{--extra-ldflags}.
 On Windows, you need to run the IDL files through @command{widl}.
 DeckLink is very picky about the formats it supports. Pixel format is always
 uyvy422, framerate and video size must be determined for your device with
@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
 of channels currently is limited to 2 (stereo).
@subsection Options
@table @option
@item list_devices
 If set to @option{true}, print a list of devices and exit.
 Defaults to @option{false}.
@item list_formats
 If set to @option{true}, print a list of supported formats and exit.
 Defaults to @option{false}.
@end table
@subsection Examples
@itemize
@item
 List input devices:
@example
 ffmpeg -f decklink -list_devices 1 -i dummy
@end example
@item
 List supported formats:
@example
 ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
@end example
@item
 Capture video clip at 1080i50 (format 11):
@example
 ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@c man end INPUT DEVICES
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@ -1,6 +1,8 @@
 FFmpeg's bug/feature request tracker manual
 =================================================
 NOTE: This is a draft.
 Overview:
 ---------
@ -20,9 +22,9 @@ a mail for every change to every issue.
 (the above does all work already after light testing)
 The subscription URL for the ffmpeg-trac list is:
-https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
+http(s)://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-https://trac.ffmpeg.org
+http(s)://trac.ffmpeg.org
 Type:
 -----
@ -40,16 +42,12 @@ feature request / enhancement
    where the current implementation cannot be considered wrong.
 license violation
-    Ticket to keep track of (L)GPL violations of ffmpeg by others.
+    ticket to keep track of (L)GPL violations of ffmpeg by others
 sponsoring request
    Developer requests for hardware, software, specifications, money,
    refunds, etc.
 task
    A task/reminder such as setting up a FATE client, adding filters to
    Trac, etc.
 Priority:
 ---------
 critical
@ -68,8 +66,7 @@ important
    don't exist in a past revision or another branch.
 normal
-   Default setting. Use this if the bug does not match the other
+
   priorities or if you are unsure of what priority to choose.
 minor
    Bugs about things like spelling errors, "mp2" instead of
@ -166,23 +163,14 @@ Component:
 avcodec
    issues in libavcodec/*
 avdevice
    issues in libavdevice/*
 avfilter
    issues in libavfilter/*
 avformat
    issues in libavformat/*
 avutil
    issues in libavutil/*
-build system
+regression test
-    issues in or related to configure/Makefile
+    issues in tests/*
 documentation
    issues in or related to doc/*
 ffmpeg
    issues in or related to ffmpeg.c
@ -196,23 +184,11 @@ ffprobe
 ffserver
    issues in or related to ffserver.c
-postproc
+build system
-    issues in libpostproc/*
+    issues in or related to configure/Makefile
-swresample
+regression
-    issues in libswresample/*
+    bugs which were not present in a past revision
 swscale
    issues in libswscale/*
 trac
    issues related to our issue tracker
 undetermined
    default component; choose this if unsure
 website
    issues related to the website
 wiki
    issues related to the wiki
--- a/doc/libavcodec.texi
+++ b/doc/libavcodec.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavcodec Documentation
@titlepage
--- a/doc/libavdevice.texi
+++ b/doc/libavdevice.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavdevice Documentation
@titlepage
--- a/doc/libavfilter.texi
+++ b/doc/libavfilter.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavfilter Documentation
@titlepage
--- a/doc/libavformat.texi
+++ b/doc/libavformat.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavformat Documentation
@titlepage
--- a/doc/libavutil.texi
+++ b/doc/libavutil.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavutil Documentation
@titlepage
--- a/doc/libswresample.texi
+++ b/doc/libswresample.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libswresample Documentation
@titlepage
--- a/doc/libswscale.texi
+++ b/doc/libswscale.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libswscale Documentation
@titlepage
--- a/doc/metadata.texi
+++ b/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
+used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
-@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
+den are integers. If the timebase is missing then start/end times are assumed to
 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
+'START=num', 'END=num', where num is a positive integer.
 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{;},
+Metadata keys or values containing special characters ('=', ';', '#', '\' and a
-@samp{#}, @samp{\} and a newline) must be escaped with a backslash @samp{\}.
+newline) must be escaped with a backslash '\'.
@item
-Note that whitespace in metadata (e.g. @samp{foo = bar}) is considered to be
+Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
-a part of the tag (in the example above key is @samp{foo }, value is
+the tag (in the example above key is 'foo ', value is ' bar').
@samp{ bar}).
@end enumerate
 A ffmetadata file might look like this:
--- a/doc/mips.txt
+++ b/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
--- a/doc/multithreading.txt
+++ b/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
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@ -37,61 +37,6 @@ ID3v2.3 and ID3v2.4) are supported. The default is version 4.
@end table
@anchor{asf}
@section asf
 Advanced Systems Format muxer.
 Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this
 muxer too.
@subsection Options
 It accepts the following options:
@table @option
@item packet_size
 Set the muxer packet size. By tuning this setting you may reduce data
 fragmentation or muxer overhead depending on your source. Default value is
 3200, minimum is 100, maximum is 64k.
@end table
@anchor{chromaprint}
@section chromaprint
 Chromaprint fingerprinter
 This muxer feeds audio data to the Chromaprint library, which generates
 a fingerprint for the provided audio data. It takes a single signed
 native-endian 16-bit raw audio stream.
@subsection Options
@table @option
@item silence_threshold
 Threshold for detecting silence, ranges from 0 to 32767. -1 for default
 (required for use with the AcoustID service).
@item algorithm
 Algorithm index to fingerprint with.
@item fp_format
 Format to output the fingerprint as. Accepts the following options:
@table @samp
@item raw
 Binary raw fingerprint
@item compressed
 Binary compressed fingerprint
@item base64
 Base64 compressed fingerprint
@end table
@end table
@anchor{crc}
@section crc
@ -174,70 +119,30 @@ ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
 See also the @ref{crc} muxer.
@anchor{framehash}
@section framehash
 Per-packet hash testing format.
 This muxer computes and prints a cryptographic hash for each audio
 and video packet. This can be used for packet-by-packet equality
 checks without having to individually do a binary comparison on each.
 By default audio frames are converted to signed 16-bit raw audio and
 video frames to raw video before computing the hash, but the output
 of explicit conversions to other codecs can also be used. It uses the
 SHA-256 cryptographic hash function by default, but supports several
 other algorithms.
 The output of the muxer consists of a line for each audio and video
 packet of the form:
@example
@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{hash}
@end example
@var{hash} is a hexadecimal number representing the computed hash
 for the packet.
@table @option
@item hash @var{algorithm}
 Use the cryptographic hash function specified by the string @var{algorithm}.
 Supported values include @code{MD5}, @code{murmur3}, @code{RIPEMD128},
@code{RIPEMD160}, @code{RIPEMD256}, @code{RIPEMD320}, @code{SHA160},
@code{SHA224}, @code{SHA256} (default), @code{SHA512/224}, @code{SHA512/256},
@code{SHA384}, @code{SHA512}, @code{CRC32} and @code{adler32}.
@end table
@subsection Examples
 To compute the SHA-256 hash of the audio and video frames in @file{INPUT},
 converted to raw audio and video packets, and store it in the file
@file{out.sha256}:
@example
 ffmpeg -i INPUT -f framehash out.sha256
@end example
 To print the information to stdout, using the MD5 hash function, use
 the command:
@example
 ffmpeg -i INPUT -f framehash -hash md5 -
@end example
 See also the @ref{hash} muxer.
@anchor{framemd5}
@section framemd5
 Per-packet MD5 testing format.
-This is a variant of the @ref{framehash} muxer. Unlike that muxer,
+This muxer computes and prints the MD5 hash for each audio
-it defaults to using the MD5 hash function.
+and video packet. By default audio frames are converted to signed
 16-bit raw audio and video frames to raw video before computing the
 hash.
 The output of the muxer consists of a line for each audio and video
 packet of the form:
@example
@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
@end example
@var{MD5} is a hexadecimal number representing the computed MD5 hash
 for the packet.
@subsection Examples
-To compute the MD5 hash of the audio and video frames in @file{INPUT},
+For example to compute the MD5 of the audio and video frames in
-converted to raw audio and video packets, and store it in the file
+@file{INPUT}, converted to raw audio and video packets, and store it
-@file{out.md5}:
+in the file @file{out.md5}:
@example
 ffmpeg -i INPUT -f framemd5 out.md5
@end example
@ -247,7 +152,7 @@ To print the information to stdout, use the command:
 ffmpeg -i INPUT -f framemd5 -
@end example
-See also the @ref{framehash} and @ref{md5} muxers.
+See also the @ref{md5} muxer.
@anchor{gif}
@section gif
@ -283,51 +188,6 @@ ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
 Note 2: the GIF format has a very small time base: the delay between two frames
 can not be smaller than one centi second.
@anchor{hash}
@section hash
 Hash testing format.
 This muxer computes and prints a cryptographic hash of all the input
 audio and video frames. This can be used for equality checks without
 having to do a complete binary comparison.
 By default audio frames are converted to signed 16-bit raw audio and
 video frames to raw video before computing the hash, but the output
 of explicit conversions to other codecs can also be used. Timestamps
 are ignored. It uses the SHA-256 cryptographic hash function by default,
 but supports several other algorithms.
 The output of the muxer consists of a single line of the form:
@var{algo}=@var{hash}, where @var{algo} is a short string representing
 the hash function used, and @var{hash} is a hexadecimal number
 representing the computed hash.
@table @option
@item hash @var{algorithm}
 Use the cryptographic hash function specified by the string @var{algorithm}.
 Supported values include @code{MD5}, @code{murmur3}, @code{RIPEMD128},
@code{RIPEMD160}, @code{RIPEMD256}, @code{RIPEMD320}, @code{SHA160},
@code{SHA224}, @code{SHA256} (default), @code{SHA512/224}, @code{SHA512/256},
@code{SHA384}, @code{SHA512}, @code{CRC32} and @code{adler32}.
@end table
@subsection Examples
 To compute the SHA-256 hash of the input converted to raw audio and
 video, and store it in the file @file{out.sha256}:
@example
 ffmpeg -i INPUT -f hash out.sha256
@end example
 To print an MD5 hash to stdout use the command:
@example
 ffmpeg -i INPUT -f hash -hash md5 -
@end example
 See also the @ref{framehash} muxer.
@anchor{hls}
@section hls
@ -358,8 +218,7 @@ This muxer supports the following options:
@table @option
@item hls_time @var{seconds}
-Set the target segment length in seconds. Default value is 2.
+Set the segment length in seconds. Default value is 2.
 Segment will be cut on the next key frame after this time has passed.
@item hls_list_size @var{size}
 Set the maximum number of playlist entries. If set to 0 the list file
@ -395,91 +254,6 @@ and it is not to be confused with the segment filename sequence number
 which can be cyclic, for example if the @option{wrap} option is
 specified.
@item hls_segment_filename @var{filename}
 Set the segment filename. Unless @code{hls_flags single_file} is set,
@var{filename} is used as a string format with the segment number:
@example
 ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
@end example
 This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc.
@item use_localtime
 Use strftime on @var{filename} to expand the segment filename with localtime.
 The segment number (%d) is not available in this mode.
@example
 ffmpeg in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
@end example
 This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file-20160215-1455569023.ts}, @file{file-20160215-1455569024.ts}, etc.
@item use_localtime_mkdir
 Used together with -use_localtime, it will create up to one subdirectory which
 is expanded in @var{filename}.
@example
 ffmpeg in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
@end example
 This example will create a directory 201560215 (if it does not exist), and then
 produce the playlist, @file{out.m3u8}, and segment files:
@file{201560215/file-20160215-1455569023.ts}, @file{201560215/file-20160215-1455569024.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
@ -490,46 +264,6 @@ ffmpeg -i in.nut -hls_flags single_file out.m3u8
@end example
 Will produce the playlist, @file{out.m3u8}, and a single segment file,
@file{out.ts}.
@item hls_flags delete_segments
 Segment files removed from the playlist are deleted after a period of time
 equal to the duration of the segment plus the duration of the playlist.
@item hls_flags round_durations
 Round the duration info in the playlist file segment info to integer
 values, instead of using floating point.
@item hls_flags discont_starts
 Add the @code{#EXT-X-DISCONTINUITY} tag to the playlist, before the
 first segment's information.
@item hls_flags omit_endlist
 Do not append the @code{EXT-X-ENDLIST} tag at the end of the playlist.
@item hls_flags split_by_time
 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. This flag should be used with the @code{hls_time} option.
@item hls_playlist_type event
 Emit @code{#EXT-X-PLAYLIST-TYPE:EVENT} in the m3u8 header. Forces
@option{hls_list_size} to 0; the playlist can only be appended to.
@item hls_playlist_type vod
 Emit @code{#EXT-X-PLAYLIST-TYPE:VOD} in the m3u8 header. Forces
@option{hls_list_size} to 0; the playlist must not change.
@item method
 Use the given HTTP method to create the hls files.
@example
 ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8
@end example
 This example will upload all the mpegts segment files to the HTTP
 server using the HTTP PUT method, and update the m3u8 files every
@code{refresh} times using the same method.
 Note that the HTTP server must support the given method for uploading
 files.
@end table
@anchor{ico}
@ -634,7 +368,8 @@ ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
@table @option
@item start_number
-Start the sequence from the specified number. Default value is 0.
+Start the sequence from the specified number. Default value is 1. Must
 be a non-negative number.
@item update
 If set to 1, the filename will always be interpreted as just a
@ -743,12 +478,16 @@ have no effect if it is not.
 MD5 testing format.
-This is a variant of the @ref{hash} muxer. Unlike that muxer, it
+This muxer computes and prints the MD5 hash of all the input audio
-defaults to using the MD5 hash function.
+and video frames. By default audio frames are converted to signed
 16-bit raw audio and video frames to raw video before computing the
 hash.
-@subsection Examples
+The output of the muxer consists of a single line of the form:
 MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing
 the computed MD5 hash.
-To compute the MD5 hash of the input converted to raw
+For example to compute the MD5 hash of the input converted to raw
 audio and video, and store it in the file @file{out.md5}:
@example
 ffmpeg -i INPUT -f md5 out.md5
@ -759,7 +498,7 @@ You can print the MD5 to stdout with the command:
 ffmpeg -i INPUT -f md5 -
@end example
-See also the @ref{hash} and @ref{framemd5} muxers.
+See also the @ref{framemd5} muxer.
@section mov, mp4, ismv
@ -860,13 +599,6 @@ 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:
@ -936,41 +668,34 @@ and @code{service_name}. If they are not set the default for
 The muxer options are:
@table @option
-@item mpegts_original_network_id @var{number}
+@item -mpegts_original_network_id @var{number}
 Set the original_network_id (default 0x0001). This is unique identifier
 of a network in DVB. Its main use is in the unique identification of a
 service through the path Original_Network_ID, Transport_Stream_ID.
-@item mpegts_transport_stream_id @var{number}
+@item -mpegts_transport_stream_id @var{number}
 Set the transport_stream_id (default 0x0001). This identifies a
 transponder in DVB.
-@item mpegts_service_id @var{number}
+@item -mpegts_service_id @var{number}
 Set the service_id (default 0x0001) also known as program in DVB.
-@item mpegts_service_type @var{number}
+@item -mpegts_pmt_start_pid @var{number}
 Set the program service_type (default @var{digital_tv}), see below
 a list of pre defined values.
@item mpegts_pmt_start_pid @var{number}
 Set the first PID for PMT (default 0x1000, max 0x1f00).
-@item mpegts_start_pid @var{number}
+@item -mpegts_start_pid @var{number}
 Set the first PID for data packets (default 0x0100, max 0x0f00).
-@item mpegts_m2ts_mode @var{number}
+@item -mpegts_m2ts_mode @var{number}
 Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
-@item muxrate @var{number}
+@item -muxrate @var{number}
 Set a constant muxrate (default VBR).
-@item pcr_period @var{numer}
+@item -pcr_period @var{numer}
 Override the default PCR retransmission time (default 20ms), ignored
 if variable muxrate is selected.
-@item pat_period @var{number}
+@item -pes_payload_size @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}
+@item -mpegts_flags @var{flags}
 Set flags (see below).
-@item mpegts_copyts @var{number}
+@item -mpegts_copyts @var{number}
 Preserve original timestamps, if value is set to 1. Default value is -1, which
 results in shifting timestamps so that they start from 0.
-@item tables_version @var{number}
+@item -tables_version @var{number}
 Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
 This option allows updating stream structure so that standard consumer may
 detect the change. To do so, reopen output AVFormatContext (in case of API
@ -986,38 +711,13 @@ ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
@end example
@end table
-Option @option{mpegts_service_type} accepts the following values:
+Option mpegts_flags may take a set of such flags:
@table @option
@item hex_value
 Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468.
@item digital_tv
 Digital TV service.
@item digital_radio
 Digital Radio service.
@item teletext
 Teletext service.
@item advanced_codec_digital_radio
 Advanced Codec Digital Radio service.
@item mpeg2_digital_hdtv
 MPEG2 Digital HDTV service.
@item advanced_codec_digital_sdtv
 Advanced Codec Digital SDTV service.
@item advanced_codec_digital_hdtv
 Advanced Codec Digital HDTV service.
@end table
 Option @option{mpegts_flags} may take a set of such flags:
@table @option
@item resend_headers
 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.
@item system_b
 Conform to System B (DVB) instead of System A (ATSC).
@end table
@subsection Example
@ -1034,21 +734,6 @@ ffmpeg -i file.mpg -c copy \
     -y out.ts
@end example
@section mxf, mxf_d10
 MXF muxer.
@subsection Options
 The muxer options are:
@table @option
@item store_user_comments @var{bool}
 Set if user comments should be stored if available or never.
 IRT D-10 does not allow user comments. The default is thus to write them for
 mxf but not for mxf_d10
@end table
@section null
 Null muxer.
@ -1083,7 +768,7 @@ Change the syncpoint usage in nut:
    sensitive and seeking is not possible. Also in general the overhead from
    syncpoints is negligible. Note, -@code{write_index} 0 can be used to disable
    all growing data tables, allowing to mux endless streams with limited memory
-    and without these disadvantages.
+    and wihout these disadvantages.
@item @var{timestamped} extend the syncpoint with a wallclock field.
@end table
 The @var{none} and @var{timestamped} flags are experimental.
@ -1108,11 +793,6 @@ is 1 second. A value of 0 will fill all segments, making pages as large as
 possible. A value of 1 will effectively use 1 packet-per-page in most
 situations, giving a small seek granularity at the cost of additional container
 overhead.
@item -serial_offset @var{value}
 Serial value from which to set the streams serial number.
 Setting it to different and sufficiently large values ensures that the produced
 ogg files can be safely chained.
@end table
@anchor{segment}
@ -1121,9 +801,8 @@ ogg files can be safely chained.
 Basic stream segmenter.
 This muxer outputs streams to a number of separate files of nearly
-fixed duration. Output filename pattern can be set in a fashion
+fixed duration. Output filename pattern can be set in a fashion similar to
-similar to @ref{image2}, or by using a @code{strftime} template if
+@ref{image2}.
 the @option{strftime} option is enabled.
@code{stream_segment} is a variant of the muxer used to write to
 streaming output formats, i.e. which do not require global headers,
@ -1155,12 +834,6 @@ implementation for HLS segmentation.
 The segment muxer supports the following options:
@table @option
@item increment_tc @var{1|0}
 if set to @code{1}, increment timecode between each segment
 If this is selected, the input need to have
 a timecode in the first video stream. Default value is
@code{0}.
@item reference_stream @var{specifier}
 Set the reference stream, as specified by the string @var{specifier}.
 If @var{specifier} is set to @code{auto}, the reference is chosen
@ -1193,6 +866,13 @@ 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
@ -1202,9 +882,6 @@ 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
@ -1264,28 +941,6 @@ to create files at 12:00 o'clock, 12:15, 12:30, etc.
 Default value is "0".
@item segment_clocktime_offset @var{duration}
 Delay the segment splitting times with the specified duration when using
@option{segment_atclocktime}.
 For example with @option{segment_time} set to "900" and
@option{segment_clocktime_offset} set to "300" this makes it possible to
 create files at 12:05, 12:20, 12:35, etc.
 Default value is "0".
@item segment_clocktime_wrap_duration @var{duration}
 Force the segmenter to only start a new segment if a packet reaches the muxer
 within the specified duration after the segmenting clock time. This way you
 can make the segmenter more resilient to backward local time jumps, such as
 leap seconds or transition to standard time from daylight savings time.
 Assuming that the delay between the packets of your source is less than 0.5
 second you can detect a leap second by specifying 0.5 as the duration.
 Default is the maximum possible duration which means starting a new segment
 regardless of the elapsed time since the last clock time.
@item segment_time_delta @var{delta}
 Specify the accuracy time when selecting the start time for a
 segment, expressed as a duration specification. Default value is "0".
@ -1327,18 +982,6 @@ Wrap around segment index once it reaches @var{limit}.
@item segment_start_number @var{number}
 Set the sequence number of the first segment. Defaults to @code{0}.
@item strftime @var{1|0}
 Use the @code{strftime} function to define the name of the new
 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
@ -1348,11 +991,6 @@ muxers/codecs. It is set to @code{0} by default.
@item initial_offset @var{offset}
 Specify timestamp offset to apply to the output packet timestamps. The
 argument must be a time duration specification, and defaults to 0.
@item write_empty_segments @var{1|0}
 If enabled, write an empty segment if there are no packets during the period a
 segment would usually span. Otherwise, the segment will be filled with the next
 packet written. Defaults to @code{0}.
@end table
@subsection Examples
@ -1480,14 +1118,7 @@ Several bitstream filters can be specified, separated by ",".
@item select
 Select the streams that should be mapped to the slave output,
 specified by a stream specifier. If not specified, this defaults to
-all the input streams. You may use multiple stream specifiers
+all the input streams.
 separated by commas (@code{,}) e.g.: @code{a:0,v}
@item onfail
 Specify behaviour on output failure. This can be set to either @code{abort} (which is
 default) or @code{ignore}. @code{abort} will cause whole process to fail in case of failure
 on this slave output. @code{ignore} will ignore failure on this output, so other outputs
 will continue without being affected.
@end table
@subsection Examples
@ -1501,14 +1132,6 @@ ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
  "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@end example
@item
 As above, but continue streaming even if output to local file fails
 (for example local drive fills up):
@example
 ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
  "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@end example
@item
 Use @command{ffmpeg} to encode the input, and send the output
 to three different destinations. The @code{dump_extra} bitstream
@ -1539,17 +1162,7 @@ is the @option{global_header} flag.
 WebM DASH Manifest muxer.
-This muxer implements the WebM DASH Manifest specification to generate the DASH
+This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
 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
@subsection Options
@ -1560,32 +1173,6 @@ 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
@ -1601,47 +1188,4 @@ 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
--- a/doc/nut.texi
+++ b/doc/nut.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle NUT
@ -34,7 +33,7 @@ NUT has some variants signaled by using the flags field in its main header.
 The BROADCAST variant provides a secondary time reference to facilitate
 detecting endpoint latency and network delays.
-It assumes all the endpoint clocks are synchronized.
+It assumes all the endpoint clocks are syncronized.
 To be used in real-time scenarios.
@section PIPE
--- a/doc/optimization.txt
+++ b/doc/optimization.txt
@ -7,7 +7,7 @@ If you plan to do non-x86 architecture specific optimizations (SIMD normally),
 then take a look in the x86/ directory, as most important functions are
 already optimized for MMX.
-If you want to do x86 optimizations then you can either try to fine-tune the
+If you want to do x86 optimizations then you can either try to finetune the
 stuff in the x86 directory or find some other functions in the C source to
 optimize, but there aren't many left.
@ -163,7 +163,7 @@ general x86 registers (e.g. eax) as well as XMM registers. This last one is
 particularly important on Win64, where xmm6-15 are callee-save, and not
 restoring their contents leads to undefined results. In external asm (e.g.
 yasm), you do this by using:
-cglobal function_name, num_args, num_regs, num_xmm_regs
+cglobal functon_name, num_args, num_regs, num_xmm_regs
 In inline asm, you specify clobbered registers at the end of your asm:
 __asm__(".." ::: "%eax").
 If gcc is not set to support sse (-msse) it will not accept xmm registers
@ -191,11 +191,6 @@ __asm__() block.
 Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
 The latter requires a good optimizing compiler which gcc is not.
 When debugging a x86 external asm compilation issue, if lost in the macro
 expansions, add DBG=1 to your make command-line: the input file will be
 preprocessed, stripped of the debug/empty lines, then compiled, showing the
 actual lines causing issues.
 Inline asm vs. external asm
 ---------------------------
 Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)
--- a/doc/platform.texi
+++ b/doc/platform.texi
@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Platform Specific Information
@titlepage
@ -63,7 +62,7 @@ bash ./configure
@section Darwin (Mac OS X, iPhone)
 The toolchain provided with Xcode is sufficient to build the basic
-unaccelerated code.
+unacelerated code.
 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{https://github.com/FFmpeg/gas-preprocessor} or
@ -97,9 +96,9 @@ the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
@section Native Windows compilation using MinGW or MinGW-w64
-FFmpeg can be built to run natively on Windows using the MinGW-w64
+FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
-toolchain. Install the latest versions of MSYS2 and MinGW-w64 from
+toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
-@url{http://msys2.github.io/} and/or @url{http://mingw-w64.sourceforge.net/}.
+@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}.
 You can find detailed installation instructions in the download section and
 the FAQ.
@ -107,13 +106,8 @@ Notes:
@itemize
-@item Building for the MSYS environment is discouraged, MSYS2 provides a full
+@item Building natively using MSYS can be sped up by disabling implicit rules
-MinGW-w64 environment through @file{mingw64_shell.bat} or
+in the Makefile by calling @code{make -r} instead of plain @code{make}. This
@file{mingw32_shell.bat} that should be used instead of the environment
 provided by @file{msys2_shell.bat}.
@item Building using MSYS2 can be sped up by disabling implicit rules in the
 Makefile by calling @code{make -r} instead of plain @code{make}. This
 speed up is close to non-existent for normal one-off builds and is only
 noticeable when running make for a second time (for example during
@code{make install}).
@ -127,25 +121,6 @@ libavformat) as DLLs.
@end itemize
@subsection Native Windows compilation using MSYS2
 The MSYS2 MinGW-w64 environment provides ready to use toolchains and dependencies
 through @command{pacman}.
 Make sure to use @file{mingw64_shell.bat} or @file{mingw32_shell.bat} to have
 the correct MinGW-w64 environment. The default install provides shortcuts to
 them under @command{MinGW-w64 Win64 Shell} and @command{MinGW-w64 Win32 Shell}.
@example
 # normal msys2 packages
 pacman -S make pkgconf diffutils
 # mingw-w64 packages and toolchains
 pacman -S mingw-w64-x86_64-yasm mingw-w64-x86_64-gcc mingw-w64-x86_64-SDL
@end example
 To target 32 bits replace @code{x86_64} with @code{i686} in the command above.
@section Microsoft Visual C++ or Intel C++ Compiler for Windows
 FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
@ -158,12 +133,11 @@ You will need the following prerequisites:
 (if using MSVC 2012 or earlier)
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
 (if using MSVC 2012 or earlier)
-@item @uref{http://msys2.github.io/, MSYS2}
+@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
 (Also available via MSYS2's package manager.)
@end itemize
-To set up a proper environment in MSYS2, you need to run @code{msys_shell.bat} from
+To set up a proper environment in MSYS, you need to run @code{msys.bat} from
 the Visual Studio or Intel Compiler command prompt.
 Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
@ -199,6 +173,12 @@ 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
@ -314,7 +294,7 @@ These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
@example
-yasm, libSDL-devel, libfaac-devel, libgsm-devel, libmp3lame-devel,
+yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
 libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
@end example
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@ -1,22 +1,3 @@
@chapter Protocol Options
@c man begin PROTOCOL OPTIONS
 The libavformat library provides some generic global options, which
 can be set on all the protocols. In addition each protocol may support
 so-called private options, which are specific for that component.
 The list of supported options follows:
@table @option
@item protocol_whitelist @var{list} (@emph{input})
 Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
 prefixed by "-" are disabled.
 All protocols are allowed by default but protocols used by an another
 protocol (nested protocols) are restricted to a per protocol subset.
@end table
@c man end PROTOCOL OPTIONS
@chapter Protocols
@c man begin PROTOCOLS
@ -36,28 +17,8 @@ particular protocol using the option
 The option "-protocols" of the ff* tools will display the list of
 supported protocols.
 All protocols accept the following options:
@table @option
@item rw_timeout
 Maximum time to wait for (network) read/write operations to complete,
 in microseconds.
@end table
 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.
@ -102,7 +63,7 @@ cache:@var{URL}
 Physical concatenation protocol.
-Read and seek from many resources in sequence as if they were
+Allow to read and seek from many resource in sequence as if they were
 a unique resource.
 A URL accepted by this protocol has the syntax:
@ -156,7 +117,7 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////
 File access protocol.
-Read from or write to a file.
+Allow to read from or write to a file.
 A file URL can have the form:
@example
@ -194,7 +155,7 @@ time, which is valuable for files on slow medium.
 FTP (File Transfer Protocol).
-Read from or write to remote resources using FTP protocol.
+Allow to read from or write to remote resources using FTP protocol.
 Following syntax is required.
@example
@ -224,17 +185,6 @@ it, unless special care is taken (tests, customized server configuration
 etc.). Different FTP servers behave in different way during seek
 operation. ff* tools may produce incomplete content due to server limitations.
 This protocol accepts the following options:
@table @option
@item follow
 If set to 1, the protocol will retry reading at the end of the file, allowing
 reading files that still are being written. In order for this to terminate,
 you either need to use the rw_timeout option, or use the interrupt callback
 (for API users).
@end table
@section gopher
 Gopher protocol.
@ -278,9 +228,6 @@ If set to 1 use chunked Transfer-Encoding for posts, default is 1.
@item content_type
 Set a specific content type for the POST messages.
@item http_proxy
 set HTTP proxy to tunnel through e.g. http://example.com:1234
@item headers
 Set custom HTTP headers, can override built in default headers. The
 value must be a string encoding the headers.
@ -301,16 +248,6 @@ Set timeout in microseconds of socket I/O operations used by the underlying low
 operation. By default it is set to -1, which means that the timeout is
 not specified.
@item reconnect_at_eof
 If set then eof is treated like an error and causes reconnection, this is useful
 for live / endless streams.
@item reconnect_streamed
 If set then even streamed/non seekable streams will be reconnected on errors.
@item reconnect_delay_max
 Sets the maximum delay in seconds after which to give up reconnecting
@item mime_type
 Export the MIME type.
@ -340,43 +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
@ -474,7 +374,7 @@ be seekable, so they will fail with the MD5 output protocol.
 UNIX pipe access protocol.
-Read and write from UNIX pipes.
+Allow to read and write from UNIX pipes.
 The accepted syntax is:
@example
@ -695,7 +595,7 @@ This protocol accepts the following options.
@table @option
@item timeout
-Set timeout in milliseconds of socket I/O operations used by the underlying
+Set timeout in miliseconds of socket I/O operations used by the underlying
 low level operation. By default it is set to -1, which means that the timeout
 is not specified.
@ -714,7 +614,7 @@ For more information see: @url{http://www.samba.org/}.
 Secure File Transfer Protocol via libssh
-Read from or write to remote resources using SFTP protocol.
+Allow to read from or write to remote resources using SFTP protocol.
 Following syntax is required.
@ -1155,18 +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 tee
 Writes the output to multiple protocols. The individual outputs are separated
 by |
@example
 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
@end example
@section tcp
@ -1194,12 +1083,6 @@ than this time interval, raise error.
@item listen_timeout=@var{milliseconds}
 Set listen timeout, expressed in milliseconds.
@item recv_buffer_size=@var{bytes}
 Set receive buffer size, expressed bytes.
@item send_buffer_size=@var{bytes}
 Set send buffer size, expressed bytes.
@end table
 The following example shows how to setup a listening TCP connection
@ -1294,14 +1177,6 @@ Set the UDP maximum socket buffer size in bytes. This is used to set either
 the receive or send buffer size, depending on what the socket is used for.
 Default is 64KB.  See also @var{fifo_size}.
@item bitrate=@var{bitrate}
 If set to nonzero, the output will have the specified constant bitrate if the
 input has enough packets to sustain it.
@item burst_bits=@var{bits}
 When using @var{bitrate} this specifies the maximum number of bits in
 packet bursts.
@item localport=@var{port}
 Override the local UDP port to bind with.
--- a/Show More
+++ b/Show More