Update for 0.10.15

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

.gitignore

@@ -1,60 +1,56 @@
-*.a
+.config
+.version
 *.o
 *.d
-*.def
-*.dll
 *.exe
-*.h.c
-*.lib
-*.pc
-*.so
-*.so.*
-*.ver
+*.ho
 *-example
 *-test
 *_g
-/.config
-/.version
-/ffmpeg
-/ffplay
-/ffprobe
-/ffserver
-/config.*
-/version.h
-/doc/*.1
-/doc/*.html
-/doc/*.pod
-/doc/avoptions_codec.texi
-/doc/avoptions_format.texi
-/doc/examples/decoding_encoding
-/doc/examples/demuxing
-/doc/examples/filtering_audio
-/doc/examples/filtering_video
-/doc/examples/metadata
-/doc/examples/muxing
-/doc/examples/scaling_video
-/doc/fate.txt
-/doc/print_options
-/doxy/
-/libavcodec/*_tablegen
-/libavcodec/*_tables.c
-/libavcodec/*_tables.h
-/libavutil/avconfig.h
-/tests/audiogen
-/tests/base64
-/tests/data/
-/tests/rotozoom
-/tests/tiny_psnr
-/tests/videogen
-/tests/vsynth1/
-/tools/aviocat
-/tools/ffbisect
-/tools/bisect.need
-/tools/cws2fws
-/tools/ffeval
-/tools/graph2dot
-/tools/ismindex
-/tools/pktdumper
-/tools/probetest
-/tools/qt-faststart
-/tools/trasher
+*.def
+*.dll
+*.lib
+*.exp
+config.*
+doc/*.1
+doc/*.html
+doc/*.pod
+doc/fate.txt
+doxy
+ffmpeg
+ffplay
+ffprobe
+ffserver
+avconv
+libavcodec/*_tablegen
+libavcodec/*_tables.c
+libavcodec/*_tables.h
+libavcodec/codec_names.h
+libavcodec/libavcodec*
+libavcore/libavcore*
+libavdevice/libavdevice*
+libavfilter/libavfilter*
+libavformat/libavformat*
+libavutil/avconfig.h
+libavutil/libavutil*
+libpostproc/libpostproc*
+libswresample/libswresample*
+libswscale/libswscale*
+tests/audiogen
+tests/base64
+tests/data
+tests/rotozoom
+tests/tiny_psnr
+tests/videogen
+tests/vsynth1
+tests/vsynth2
+tools/aviocat
+tools/cws2fws
+tools/graph2dot
+tools/ismindex
+tools/lavfi-showfiltfmts
+tools/pktdumper
+tools/probetest
+tools/qt-faststart
+tools/trasher
+version.h

COPYING.LGPLv2.1

@@ -500,3 +500,5 @@ necessary.  Here is a sample; alter the names:
   Ty Coon, President of Vice
 
 That's all there is to it!
+
+

Changelog

@@ -4,118 +4,299 @@ releases are sorted from youngest to oldest.
 version next:
 
 
-version 1.0:
-- INI and flat output in ffprobe
-- Scene detection in libavfilter
-- Indeo Audio decoder
-- channelsplit audio filter
-- setnsamples audio filter
-- atempo filter
-- ffprobe -show_data option
-- RTMPT protocol support
-- iLBC encoding/decoding via libilbc
-- Microsoft Screen 1 decoder
-- join audio filter
-- audio channel mapping filter
-- Microsoft ATC Screen decoder
-- RTSP listen mode
-- TechSmith Screen Codec 2 decoder
-- AAC encoding via libfdk-aac
-- Microsoft Expression Encoder Screen decoder
-- RTMPS protocol support
-- RTMPTS protocol support
-- RTMPE protocol support
-- RTMPTE protocol support
-- showwaves and showspectrum filter
-- LucasArts SMUSH playback support
-- SAMI, RealText and SubViewer demuxers and decoders
-- Heart Of Darkness PAF playback support
-- iec61883 device
-- asettb filter
-- new option: -progress
-- 3GPP Timed Text encoder/decoder
-- GeoTIFF decoder support
-- ffmpeg -(no)stdin option
-- Opus decoder using libopus
-- caca output device using libcaca
-- alphaextract and alphamerge filters
-- concat filter
-- flite filter
-- Canopus Lossless Codec decoder
-- bitmap subtitles in filters (experimental and temporary)
-- MP2 encoding via TwoLAME
-- bmp parser
-- smptebars source
-- asetpts filter
-- hue filter
-- ICO muxer
-- SubRip encoder and decoder without embedded timing
-- edge detection filter
-- framestep filter
-- ffmpeg -shortest option is now per-output file
-  -pass and -passlogfile are now per-output stream
-- volume measurement filter
-- Ut Video encoder
-- Microsoft Screen 2 decoder
-- Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP
-  instead of AV_CODEC_ID_TEXT
-- smartblur filter ported from MPlayer
-- CPiA decoder
-- decimate filter ported from MPlayer
-- RTP depacketization of JPEG
-- Smooth Streaming live segmenter muxer
-- F4V muxer
-- sendcmd and asendcmd filters
-- WebVTT demuxer and decoder (simple tags supported)
-- RTP packetization of JPEG
-- faststart option in the MOV/MP4 muxer
+version 0.10.11
+
+- pthread: Avoid spurious wakeups
+- pthread: Fix deadlock during thread initialization
+- mpegvideo: Initialize chroma_*_shift and codec_tag even if the size is 0
+- vc1dec: Don't decode slices when the latest slice header failed to decode
+- vc1dec: Make sure last_picture is initialized in vc1_decode_skip_blocks
+- r3d: Add more input value validation
+- fraps: Make the input buffer size checks more strict
+- svq3: Avoid a division by zero
+- rmdec: Validate the fps value
+- twinvqdec: Check the ibps parameter separately
+- asfdec: Check the return value of asf_read_stream_properties
+- mxfdec: set audio timebase to 1/samplerate
+- pcx: Check the packet size before assuming it fits a palette
+- rpza: Fix a buffer size check
+- xxan: Disallow odd width
+- xan: Only read within the data that actually was initialized
+- xan: Use bytestream2 to limit reading to within the buffer
+- pcx: Consume the whole packet if giving up due to missing palette
+- pngdec: Stop trying to decode once inflate returns Z_STREAM_END
+- mov: Make sure the read sample count is nonnegative
+- bfi: Add some very basic sanity checks for input packet sizes
+- bfi: Avoid divisions by zero
+- electronicarts: Add more sanity checking for the number of channels
+- riffdec: Add sanity checks for the sample rate
+- mvi: Add sanity checking for the audio frame size
+- xwma: Avoid division by zero
+- avidec: Make sure a packet is large enough before reading its data
+- vqf: Make sure the bitrate is in the valid range
+- vqf: Make sure sample_rate is set to a valid value
+- vc1dec: Undo mpegvideo initialization if unable to allocate tables
+- vc1dec: Fix leaks in ff_vc1_decode_init_alloc_tables on errors
+- wnv1: Make sure the input packet is large enough
+- dca: Validate the lfe parameter
+- rl2: Avoid a division by zero
+- wtv: Add more sanity checks for a length read from the file
+- segafilm: Validate the number of audio channels
+- qpeg: Add checks for running out of rows in qpeg_decode_inter
+- mpegaudiodec: Validate that the number of channels fits at the given offset
+- asv1: Verify the amount of extradata
+- idroqdec: Make sure a video stream has been allocated before returning packets
+- rv10: Validate the dimensions set from the container
+- xmv: Add more sanity checks for parameters read from the bitstream
+- ffv1: Make sure at least one slice context is initialized
+- truemotion2: Use av_freep properly in an error path
+- eacmv: Make sure a reference frame exists before referencing it
+- mpeg4videodec: Check the width/height in mpeg4_decode_sprite_trajectory
+- ivi_common: Make sure color planes have been initialized
+- oggparseogm: Convert to use bytestream2
+- rv34: Check the return value from ff_rv34_decode_init
+- matroskadec: Verify realaudio codec parameters
+- mace: Make sure that the channel count is set to a valid value
+- svq3: Check for any negative return value from ff_h264_check_intra_pred_mode
+- vp3: Check the framerate for validity
+- cavsdec: Make sure a sequence header has been decoded before decoding pictures
+- sierravmd: Do sanity checking of frame sizes
+- omadec: Properly check lengths before incrementing the position
+- mpc8: Make sure the first stream exists before parsing the seek table
+- mpc8: Check the seek table size parsed from the bitstream
+- zmbvdec: Check the buffer size for uncompressed data
+- ape: Don't allow the seektable to be omitted
+- shorten: Break out of loop looking for fmt chunk if none is found
+- shorten: Use a checked bytestream reader for the wave header
+- smacker: Make sure we don't fill in huffman codes out of range
+- smacker: Avoid integer overflow when allocating packets
+- smacker: Don't return packets in unallocated streams
+- dsicin: Add some basic sanity checks for fields read from the file
+- roqvideodec: check dimensions validity
+- qdm2: check array index before use, fix out of array accesses
+- alsdec: check block length
 
 
-version 0.11:
+version 0.10.10
+
+- x86: fft: Remove 3DNow! optimizations, they break FATE
+- x86: ac3dsp: Drop mmx variant of ac3_max_msb_abs_int16
+- aac: Check init_get_bits return value
+- aac: return meaningful errors
+- dsicinav: K&R formatting cosmetics
+- mov: Seek back if overreading an individual atom
+- vcr1: add sanity checks
+- pictordec: pass correct context to avpriv_request_sample
+- dsicinav: Clip the source size to the expected maximum
+- alsdec: Clean up error paths
+- ogg: Fix potential infinite discard loop
+- nuv: check rtjpeg_decode_frame_yuv420 return value
+- nuv: Reset the frame on resize
+- nuv: Use av_fast_realloc
+- nuv: return meaningful error codes.
+- nuv: Pad the lzo outbuf
+- nuv: Do not ignore lzo decompression failures
+- oma: correctly mark and decrypt partial packets
+- oma: check geob tag boundary
+- oma: refactor seek function
+- 8bps: Bound-check the input buffer
+- rtmp: Do not misuse memcmp
+- rtmp: rename data_size to size
+- lavc: set the default rc_initial_buffer_occupancy
+- 4xm: Reject not a multiple of 16 dimension
+- 4xm: do not overread the prestream buffer
+- 4xm: validate the buffer size before parsing it
+- indeo: Do not reference mismatched tiles
+- indeo: Sanitize ff_ivi_init_planes fail paths
+- indeo: Bound-check before applying motion compensation
+- indeo: Bound-check before applying transform
+- indeo: reject negative array indexes
+- indeo: Cosmetic formatting
+- indeo: Refactor ff_ivi_init_tiles and ivi_decode_blocks
+- indeo: Refactor ff_ivi_dec_huff_desc
+- lavf: fix the comparison in an overflow check
+- dv: Add a guard to not overread the ppcm array
+- mpegvideo: Avoid 32-bit wrapping of linesize multiplications
+- mjpegb: Detect changing number of planes in interlaced video
+- matroskadec: Check that .lang was allocated and set before reading it
+- ape demuxer: check for EOF in potentially long loops
+- lavf: avoid integer overflow when estimating bitrate
+- pictordec: break out of both decoding loops when y drops below 0
+- ac3: Return proper error codes
+- ac3: Clean up the error paths
+- ac3: Do not clash with normal AVERROR
+- dxa: Make sure the reference frame exists
+- h261: check the mtype index
+- segafilm: Error out on impossible packet size
+- ogg: Always alloc the private context in vorbis_header
+- vc1: check mb_height validity.
+- vc1: check the source buffer in vc1_mc functions
+- bink: Bound check the quantization matrix.
+- xl: Make sure the width is valid
+- alsdec: Fix the clipping range
+- dsicinav: Bound-check the source buffer when needed
+- mov: Do not allow updating the time scale after it has been set
+- ac3dec: Don't consume more data than the actual input packet size
+- indeo: Reject impossible FRAMETYPE_NULL
+- indeo5: return proper error codes
+- indeo4: Validate scantable dimension
+- indeo4: Check the quantization matrix index
+- indeo4: Do not access missing reference MV
+- adpcm: Unbreak ima-dk4
+- ac3dec: validate channel output mode against channel count
+- dca: Respect the current limits in the downmixing capabilities
+- dca: Error out on missing DSYNC
+- pcm: always use codec->id instead of codec_id
+- mlpdec: Do not set invalid context in read_restart_header
+- pcx: Do not overread source buffer in pcx_rle_decode
+- wmavoice: conceal clearly corrupted blocks
+- iff: Do not read over the source buffer
+- qdm2: Conceal broken samples
+- qdm2: refactor joined stereo support
+- adpcm: Write the correct number of samples for ima-dk4
+- imc: Catch a division by zero
+- atrac3: Error on impossible encoding/channel combinations
+- atrac3: set the getbits context the right buffer_end
+- atrac3: fix error handling
+- qdm2: check and reset dithering index per channel
+- westwood_vqa: do not free extradata on error in read_header
+- vqavideo: check the version
+- rmdec: Use the AVIOContext given as parameter in rm_read_metadata()
+- avio: Handle AVERROR_EOF in the same way as the return value 0
+- wtv: Mark attachment with a negative stream id
+- avidec: Let the inner dv demuxer take care of discarding
+- swfdec: do better validation of tag length
+
+
+version 0.10.8
+- kmvc: Clip pixel position to valid range
+- kmvc: use fixed sized arrays in the context
+- indeo: use a typedef for the mc function pointer
+- lavc: check for overflow in init_get_bits
+- mjpegdec: properly report unsupported disabled features
+- jpegls: return meaningful errors
+- jpegls: factorize return paths
+- jpegls: check the scan offset
+- wavpack: validate samples size parsed in wavpack_decode_block
+- ljpeg: use the correct number of components in yuv
+- mjpeg: Validate sampling factors
+- mjpegdec: validate parameters in mjpeg_decode_scan_progressive_ac
+- wavpack: check packet size early
+- wavpack: return meaningful errors
+- apetag: use int64_t for filesize
+- tiff: do not overread the source buffer
+- Prepare for 0.8.8 Release
+- smacker: fix an off by one in huff.length computation
+- smacker: check the return value of smacker_decode_tree
+- smacker: pad the extradata allocation
+- smacker: check frame size validity
+- vmdav: convert to bytestream2
+- 4xm: don't rely on get_buffer() initializing the frame.
+- 4xm: check the return value of read_huffman_tables().
+- 4xm: use the correct logging context
+- 4xm: reject frames not compatible with the declared version
+- 4xm: check bitstream_size boundary before using it
+- 4xm: do not overread the source buffer in decode_p_block
+- avfiltergraph: check for sws opts being non-NULL before using them
+- bmv: check for len being valid in bmv_decode_frame()
+- dfa: check for invalid access in decode_wdlt()
+- indeo3: check motion vectors
+- indeo3: fix data size check
+- indeo3: switch parsing the header to bytestream2
+- lavf: make sure stream probe data gets freed.
+- oggdec: fix faulty cleanup prototype
+- oma: Validate sample rates
+- qdm2: check that the FFT size is a power of 2
+- rv10: check that extradata is large enough
+- xmv: check audio track parameters validity
+- xmv: do not leak memory in the error paths in xmv_read_header()
+- aac: check the maximum number of channels
+- indeo3: fix off by one in MV validity check, Bug #503
+- id3v2: check for end of file while unescaping tags
+- wav: Always seek to an even offset, Bug #500, LP: #1174737
+- proresdec: support mixed interlaced/non-interlaced content
+
+
+version 0.10.6:
+
+- many bug fixes that where found with Coverity
+
+- The following CVE fixes where backported:
+  CVE-2012-2796, CVE-2012-2775, CVE-2012-2772, CVE-2012-2776,
+  CVE-2012-2779, CVE-2012-2787, CVE-2012-2794, CVE-2012-2800,
+  CVE-2012-2802, CVE-2012-2801, CVE-2012-2786, CVE-2012-2798,
+  CVE-2012-2793, CVE-2012-2789, CVE-2012-2788, CVE-2012-2790,
+  CVE-2012-2777, CVE-2012-2784
+
+- hundreads of other bug fixes, some possibly security relevant,
+  see the git log for details.
+
+
+version 0.10.5:
+
+- Several bugs and crashes have been fixed as well as build problems
+  with recent mingw64
+
+
+version 0.10.4:
+
+- Several bugs and crashes have been fixed
+  Note, CVE-2012-0851 and CVE-2011-3937 have been fixed in previous releases
+
+version 0.10.3:
+
+- Security fixes in the 4xm demuxer, avi demuxer, cook decoder,
+  mm demuxer, mpegvideo decoder, vqavideo decoder (CVE-2012-0947) and
+  xmv demuxer.
+
+- Several bugs and crashes have been fixed in the following codecs: AAC,
+  APE, H.263, H.264, Indeo 4, Mimic, MJPEG, Motion Pixels Video, RAW,
+  TTA, VC1, VQA, WMA Voice, vqavideo.
+
+- Several bugs and crashes have been fixed in the following formats:
+  ASF, ID3v2, MOV, xWMA
+
+- This release additionally updates the following codecs to the
+  bytestream2 API, and therefore benefit from additional overflow
+  checks: truemotion2, utvideo, vqavideo
+
+
+version 0.10.1
+- Several security fixes, many bugfixes affecting many formats and
+  codecs, the list below is not complete.
 
-- Fixes: CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
-         CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
-         CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
-         CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
-         CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
-         CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
-- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
-- setfield filter
-- CDXL demuxer and decoder
-- Apple ProRes encoder
-- ffprobe -count_packets and -count_frames options
-- Sun Rasterfile Encoder
-- ID3v2 attached pictures reading and writing
-- WMA Lossless decoder
-- bluray protocol
-- blackdetect filter
-- libutvideo encoder wrapper (--enable-libutvideo)
 - swapuv filter
-- bbox filter
-- XBM encoder and decoder
-- RealAudio Lossless decoder
-- ZeroCodec decoder
-- tile video filter
-- Metal Gear Solid: The Twin Snakes demuxer
-- OpenEXR image decoder
-- removelogo filter
-- drop support for ffmpeg without libavfilter
-- drawtext video filter: fontconfig support
-- ffmpeg -benchmark_all option
-- super2xsai filter ported from libmpcodecs
-- add libavresample audio conversion library for compatibility
-- MicroDVD decoder
-- Avid Meridien (AVUI) encoder and decoder
-- accept + prefix to -pix_fmt option to disable automatic conversions.
-- complete audio filtering in libavfilter and ffmpeg
-- add fps filter
-- vorbis parser
-- png parser
-- audio mix filter
+
+- Several bugs and crashes have been fixed in the following codecs: AAC,
+  AC-3, ADPCM, AMR (both NB and WB), ATRAC3, CAVC, Cook, camstudio, DCA,
+  DPCM, DSI CIN, DV, EA TGQ, FLAC, fraps, G.722 (both encoder and
+  decoder), H.264, huvffyuv, BB JV decoder, Indeo 3, KGV1, LCL, the
+  libx264 wrapper, MJPEG, mp3on4, Musepack, MPEG1/2, PNG, QDM2, Qt RLE,
+  ROQ, RV10, RV30/RV34/RV40, shorten, smacker, subrip, SVQ3, TIFF,
+  Truemotion2, TTA, VC1, VMware Screen codec, Vorbis, VP5, VP6, WMA,
+  Westwood SNDx, XXAN.
+
+- This release additionally updates the following codecs to the
+  bytestream2 API, and therefore benefit from additional overflow
+  checks: XXAN, ALG MM, TQG, SMC, Qt SMC, ROQ, PNG
+
+- Several bugs and crashes have been fixed in the following formats:
+  AIFF, ASF, DV, Matroska, NSV, MOV, MPEG-TS, Smacker, Sony OpenMG, RM,
+  SWF.
+
+- Libswscale has an potential overflow for large image size fixed.
+
+- The following APIs have been added:
+
+  avcodec_is_open()
+  avformat_get_riff_video_tags()
+  avformat_get_riff_audio_tags()
+
+  Please see the file doc/APIchanges and the Doxygen documentation for
+  further information.
 
 
 version 0.10:
-
 - Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
          CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
          CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
@@ -856,7 +1037,7 @@ version 0.4.5:
 - MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
 - ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
 - Windows porting of file converter
-- added MJPEG raw format (input/output)
+- added MJPEG raw format (input/ouput)
 - added JPEG image format support (input/output)
 
 

Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 1.0.6
+PROJECT_NUMBER         = 0.10.15
 
 # With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
@@ -1378,7 +1378,7 @@ PREDEFINED             = "__attribute__(x)=" \
                          "DEF(x)=x ## _TMPL" \
                          HAVE_AV_CONFIG_H \
                          HAVE_MMX \
-                         HAVE_MMXEXT \
+                         HAVE_MMX2 \
                          HAVE_AMD3DNOW \
                          "DECLARE_ALIGNED(a,t,n)=t n" \
                          "offsetof(x,y)=0x42"

LICENSE

@@ -1,4 +1,5 @@
 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
@@ -13,36 +14,14 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are
 
 - libpostproc
-- libmpcodecs
 - optional x86 optimizations in the files
   libavcodec/x86/idct_mmx.c
-- libutvideo encoding/decoding wrappers in
-  libavcodec/libutvideo*.cpp
 - the X11 grabber in libavdevice/x11grab.c
-- the swresample test app in
-  libswresample/swresample-test.c
-- the texi2pod.pl tool
-- the following filters in libavfilter:
-    - vf_blackframe.c
-    - vf_boxblur.c
-    - vf_colormatrix.c
-    - vf_cropdetect.c
-    - vf_delogo.c
-    - vf_hqdn3d.c
-    - vf_mp.c
-    - vf_super2xsai.c
-    - vf_tinterlace.c
-    - vf_yadif.c
-    - vsrc_mptestsrc.c
 
 There are a handful of files under other licensing terms, namely:
 
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
-  licensing details. Specifically note that you must credit the IJG in the
-  documentation accompanying your program if you only distribute executables.
-  You must also indicate any changes including additions and deletions to
-  those three files in the documentation.
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
+  are taken from libjpeg, see the top of the files for licensing details.
 
 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
@@ -50,32 +29,18 @@ for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
 COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 
 
-external libraries
-==================
+external libraries:
+-------------------
 
-FFmpeg can be combined with a number of external libraries, which sometimes
-affect the licensing of binaries resulting from the combination.
+Some external libraries, e.g. libx264, are under GPL and can be used in
+conjunction with FFmpeg. They require --enable-gpl to be passed to configure
+as well.
 
-compatible libraries
---------------------
+The OpenCORE external 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 the OpenCORE libraries with FFmpeg, the license
+version needs to be upgraded by passing --enable-version3 to configure.
 
-The libcdio, libx264, libxavs and libxvid libraries are under GPL. When
-combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing --enable-gpl to configure.
-
-The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
-license is incompatible with the LGPL v2.1 and the GPL v2, but not with
-version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing --enable-version3 to configure.
-
-incompatible libraries
-----------------------
-
-The Fraunhofer AAC library, FAAC and aacplus are under licenses which
-are incompatible with the GPLv2 and v3. We do not know for certain if their
-licenses are compatible with the LGPL.
-If you wish to enable these libraries, pass --enable-nonfree to configure.
-But note that if you enable any of these libraries the resulting binary will
-be under a complex license mix that is more restrictive than the LGPL and that
-may result in additional obligations. It is possible that these
-restrictions cause the resulting binary to be unredistributeable.
+The nonfree external libraries libfaac and libaacplus can be hooked up in FFmpeg.
+You need to pass --enable-nonfree to configure to enable it. Employ this option
+with care as FFmpeg then becomes nonfree and unredistributable.

MAINTAINERS

@@ -4,7 +4,7 @@ FFmpeg maintainers
 Below is a list of the people maintaining different parts of the
 FFmpeg code.
 
-Please try to keep entries where you are the maintainer up to date!
+Please try to keep entries where you are the maintainer upto date!
 
 Names in () mean that the maintainer currently has no time to maintain the code.
 A CC after the name means that the maintainer prefers to be CC-ed on patches
@@ -132,9 +132,7 @@ Codecs:
   celp_filters.*                        Vitor Sessak
   cinepak.c                             Roberto Togni
   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
   dca.c                                 Kostya Shishkov, Benjamin Larsson
@@ -161,7 +159,6 @@ Codecs:
   indeo5*                               Kostya Shishkov
   interplayvideo.c                      Mike Melanson
   ivi*                                  Kostya Shishkov
-  jacosub*                              Clément Bœsch
   jpeg_ls.c                             Kostya Shishkov
   jvdec.c                               Peter Ross
   kmvc.c                                Kostya Shishkov
@@ -174,7 +171,6 @@ Codecs:
   libschroedinger*                      David Conrad
   libspeexdec.c                         Justin Ruggles
   libtheoraenc.c                        David Conrad
-  libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
   libxavs.c                             Stefan Gehrer
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
@@ -231,7 +227,6 @@ Codecs:
   vble.c                                Derek Buitenhuis
   vc1*                                  Kostya Shishkov
   vcr1.c                                Michael Niedermayer
-  vda_h264_dec.c                        Xidorn Quan
   vmnc.c                                Kostya Shishkov
   vorbis_enc.c                          Oded Shimon
   vorbis_dec.c                          Denes Balatoni, David Conrad
@@ -248,7 +243,6 @@ Codecs:
   xan.c                                 Mike Melanson
   xl.c                                  Kostya Shishkov
   xvmc.c                                Ivan Kalvachev
-  zerocodec.c                           Derek Buitenhuis
   zmbv*                                 Kostya Shishkov
 
 Hardware acceleration:
@@ -266,7 +260,6 @@ libavdevice
     libavdevice/avdevice.h
 
 
-  iec61883.c                            Georg Lippitsch
   libdc1394.c                           Roman Shaposhnik
   v4l2.c                                Luca Abeni
   vfwcap.c                              Ramiro Polla
@@ -275,18 +268,14 @@ libavdevice
 libavfilter
 ===========
 
-Generic parts:
+Video filters:
   graphdump.c                           Nicolas George
-
-Filters:
   af_amerge.c                           Nicolas George
   af_astreamsync.c                      Nicolas George
-  af_atempo.c                           Pavel Koshevoy
   af_pan.c                              Nicolas George
+  vsrc_mandelbrot.c                     Michael Niedermayer
   vf_yadif.c                            Michael Niedermayer
 
-Sources:
-  vsrc_mandelbrot.c                     Michael Niedermayer
 
 libavformat
 ===========
@@ -324,7 +313,6 @@ Muxers/Demuxers:
   ipmovie.c                             Mike Melanson
   img2.c                                Michael Niedermayer
   iss.c                                 Stefan Gehrer
-  jacosub*                              Clément Bœsch
   jvdec.c                               Peter Ross
   libmodplug.c                          Clément Bœsch
   libnut.c                              Oded Shimon
@@ -382,7 +370,6 @@ Muxers/Demuxers:
   wv.c                                  Kostya Shishkov
 
 Protocols:
-  bluray.c                              Petri Hintukainen
   http.c                                Ronald S. Bultje
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
@@ -408,11 +395,8 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-1.2                                     Michael Niedermayer
-1.1                                     Michael Niedermayer
-1.0                                     Michael Niedermayer
+0.9                                     Michael Niedermayer
 
-If you want to maintain an older release, please contact us
 
 
 GnuPG Fingerprints of maintainers and contributors
@@ -424,7 +408,6 @@ Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
-Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
@@ -443,5 +426,4 @@ Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
-Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tomas Härdin                  D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD

Makefile

@@ -15,12 +15,11 @@ PROGS-$(CONFIG_FFPLAY)   += ffplay
 PROGS-$(CONFIG_FFPROBE)  += ffprobe
 PROGS-$(CONFIG_FFSERVER) += ffserver
 
-PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS      := $(PROGS-yes:%=%$(EXESUF))
 INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-OBJS        = cmdutils.o
-OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
+OBJS        = $(PROGS-yes:%=%.o) cmdutils.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+HOSTPROGS  := $(TESTTOOLS:%=tests/%)
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
@@ -32,7 +31,6 @@ ALLMANPAGES = $(BASENAMES:%=%.1)
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
-FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
@@ -41,7 +39,6 @@ FFLIBS-$(CONFIG_SWSCALE)  += swscale
 FFLIBS := avutil
 
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile
 
 SKIPHEADERS = cmdutils_common_opts.h
 
@@ -52,14 +49,14 @@ FF_DEP_LIBS  := $(DEP_LIBS)
 
 all: $(PROGS)
 
-$(PROGS): %$(EXESUF): %_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+	$(CP) $< $@$(PROGSSUF)
+	$(STRIP) $@$(PROGSSUF)
 
 $(TOOLS): %$(EXESUF): %.o
-	$(LD) $(LDFLAGS) $(LD_O) $< $(ELIBS)
+	$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
 
-tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
+tools/cws2fws$(EXESUF): ELIBS = -lz
 
 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -67,13 +64,9 @@ config.h: .config
 	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
-SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
-               ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS NEON-OBJS             \
-               MMI-OBJS ALTIVEC-OBJS VIS-OBJS                            \
-               MMX-OBJS YASM-OBJS                                        \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS HOSTOBJS TESTOBJS
+SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
+               ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
+               HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS ARMV6-OBJS TOOLS
 
 define RESET
 $(1) :=
@@ -90,19 +83,12 @@ endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-define DOPROG
-OBJS-$(1) += $(1).o
-$(1)$(PROGSSUF)_g$(EXESUF): $(OBJS-$(1))
-$$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
-$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
-$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
--include $$(OBJS-$(1):.o=.d)
-endef
-
-$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
+ffplay.o: CFLAGS += $(SDL_CFLAGS)
+ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
+ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)
 
 %$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) cmdutils.o $(FF_EXTRALIBS)
+	$(LD) $(LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
 
 OBJDIRS += tools
 
@@ -136,10 +122,9 @@ install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
-install-data: $(DATA_FILES) $(EXAMPLES_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
-	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
 
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
@@ -152,14 +137,13 @@ uninstall-data:
 clean::
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(TOOLS)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) coverage.info
 	$(RM) -r coverage-html
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h
+	$(RM) config.* .version version.h libavutil/avconfig.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
@@ -173,7 +157,7 @@ coverage-html: coverage.info
 	$(Q)genhtml -o $@ $<
 	$(Q)touch $@
 
-check: all alltools examples testprogs fate
+check: all alltools checkheaders examples testprogs fate
 
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile
@@ -189,5 +173,5 @@ $(sort $(OBJDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:
 
-.PHONY: all all-yes alltools check *clean config install*
+.PHONY: all all-yes alltools check *clean config examples install*
 .PHONY: testprogs uninstall*

README

@@ -4,15 +4,9 @@ FFmpeg README
 1) Documentation
 ----------------
 
-* Read the documentation in the doc/ directory in git.
-  You can also view it online at http://ffmpeg.org/documentation.html
+* Read the documentation in the doc/ directory.
 
 2) Licensing
 ------------
 
 * See the LICENSE file.
-
-3) Build and Install
---------------------
-
-* See the INSTALL file.

RELEASE

@@ -1 +1 @@
-1.0.6
+0.10.15

VERSION

@@ -1 +1 @@
-1.0.6
+0.10.15

arch.mak

@@ -1,17 +0,0 @@
-OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
-OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMVFP)  += $(ARMVFP-OBJS)  $(ARMVFP-OBJS-yes)
-OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
-
-OBJS-$(HAVE_MMI)     += $(MMI-OBJS)     $(MMI-OBJS-yes)
-OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
-OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
-
-OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-
-OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
-
-OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
-OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.h

@@ -51,7 +51,6 @@ extern const int this_year;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern struct SwrContext *swr_opts;
 extern AVDictionary *format_opts, *codec_opts;
 
 /**
@@ -75,25 +74,23 @@ void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
  * Fallback for options that are not explicitly handled, these will be
  * parsed through AVOptions.
  */
-int opt_default(void *optctx, const char *opt, const char *arg);
+int opt_default(const char *opt, const char *arg);
 
 /**
  * Set the libav* libraries log level.
  */
-int opt_loglevel(void *optctx, const char *opt, const char *arg);
+int opt_loglevel(const char *opt, const char *arg);
 
 int opt_report(const char *opt);
 
-int opt_max_alloc(void *optctx, const char *opt, const char *arg);
+int opt_max_alloc(const char *opt, const char *arg);
 
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
-int opt_codec_debug(void *optctx, const char *opt, const char *arg);
+int opt_codec_debug(const char *opt, const char *arg);
 
 /**
  * Limit the execution time.
  */
-int opt_timelimit(void *optctx, const char *opt, const char *arg);
+int opt_timelimit(const char *opt, const char *arg);
 
 /**
  * Parse a string and return its corresponding value as a double.
@@ -148,14 +145,14 @@ typedef struct {
 #define OPT_STRING 0x0008
 #define OPT_VIDEO  0x0010
 #define OPT_AUDIO  0x0020
+#define OPT_GRAB   0x0040
 #define OPT_INT    0x0080
 #define OPT_FLOAT  0x0100
 #define OPT_SUBTITLE 0x0200
 #define OPT_INT64  0x0400
 #define OPT_EXIT   0x0800
 #define OPT_DATA   0x1000
-#define OPT_PERFILE  0x2000     /* the option is per-file (currently ffmpeg-only).
-                                   implied by OPT_OFFSET or OPT_SPEC */
+#define OPT_FUNC2  0x2000
 #define OPT_OFFSET 0x4000       /* option is specified as an offset in a passed optctx */
 #define OPT_SPEC   0x8000       /* option is to be stored in an array of SpecifierOpt.
                                    Implies OPT_OFFSET. Next element after the offset is
@@ -164,24 +161,16 @@ typedef struct {
 #define OPT_DOUBLE 0x20000
      union {
         void *dst_ptr;
-        int (*func_arg)(void *, const char *, const char *);
+        int (*func_arg)(const char *, const char *);
+        int (*func2_arg)(void *, const char *, const char *);
         size_t off;
     } u;
     const char *help;
     const char *argname;
 } OptionDef;
 
-/**
- * Print help for all options matching specified flags.
- *
- * @param options a list of options
- * @param msg title of this group. Only printed if at least one option matches.
- * @param req_flags print only options which have all those flags set.
- * @param rej_flags don't print options which have any of those flags set.
- * @param alt_flags print only options that have at least one of those flags set
- */
-void show_help_options(const OptionDef *options, const char *msg, int req_flags,
-                       int rej_flags, int alt_flags);
+void show_help_options(const OptionDef *options, const char *msg, int mask,
+                       int value);
 
 /**
  * Show help for all options with given flags in class and all its
@@ -189,17 +178,6 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
  */
 void show_help_children(const AVClass *class, int flags);
 
-/**
- * Per-avtool specific help handler. Implemented in each
- * avtool, called by show_help().
- */
-void show_help_default(const char *opt, const char *arg);
-
-/**
- * Generic -h handler common to all avtools.
- */
-int show_help(void *optctx, const char *opt, const char *arg);
-
 /**
  * Parse the command line arguments.
  *
@@ -226,12 +204,6 @@ int parse_option(void *optctx, const char *opt, const char *arg,
  */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 
-/**
- * Return index of option opt in argv or 0 if not found.
- */
-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname);
-
 /**
  * Check if the given stream matches a stream specifier.
  *
@@ -251,12 +223,10 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
  *
  * @param s Corresponding format context.
  * @param st A stream from s for which the options should be filtered.
- * @param codec The particular codec for which the options should be filtered.
- *              If null, the default one is looked up according to the codec id.
  * @return a pointer to the created dictionary
  */
-AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
-                                AVFormatContext *s, AVStream *st, AVCodec *codec);
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
+                                AVFormatContext *s, AVStream *st);
 
 /**
  * Setup AVCodecContext options for avformat_find_stream_info().
@@ -296,81 +266,62 @@ void show_banner(int argc, char **argv, const OptionDef *options);
  * libraries.
  * This option processing function does not utilize the arguments.
  */
-int show_version(void *optctx, const char *opt, const char *arg);
+int opt_version(const char *opt, const char *arg);
 
 /**
  * Print the license of the program to stdout. The license depends on
  * the license of the libraries compiled into the program.
  * This option processing function does not utilize the arguments.
  */
-int show_license(void *optctx, const char *opt, const char *arg);
+int opt_license(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the formats supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_formats(void *optctx, const char *opt, const char *arg);
+int opt_formats(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the codecs supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_codecs(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the decoders supported by the
- * program.
- */
-int show_decoders(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the encoders supported by the
- * program.
- */
-int show_encoders(void *optctx, const char *opt, const char *arg);
+int opt_codecs(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the filters supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_filters(void *optctx, const char *opt, const char *arg);
+int opt_filters(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the bit stream filters supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_bsfs(void *optctx, const char *opt, const char *arg);
+int opt_bsfs(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the protocols supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_protocols(void *optctx, const char *opt, const char *arg);
+int opt_protocols(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the pixel formats supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_pix_fmts(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the standard channel layouts supported by
- * the program.
- * This option processing function does not utilize the arguments.
- */
-int show_layouts(void *optctx, const char *opt, const char *arg);
+int opt_pix_fmts(const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the sample formats supported by the
  * program.
  */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+int show_sample_fmts(const char *opt, const char *arg);
 
 /**
  * Return a positive value if a line read from standard input
@@ -398,7 +349,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
  * at configuration time or in a "ffpresets" folder along the executable
  * on win32, in that order. If no such file is found and
  * codec_name is defined, then search for a file named
- * codec_name-preset_name.avpreset in the above-mentioned directories.
+ * codec_name-preset_name.ffpreset in the above-mentioned directories.
  *
  * @param filename buffer where the name of the found filename is written
  * @param filename_size size in bytes of the filename buffer
@@ -414,7 +365,7 @@ FILE *get_preset_file(char *filename, size_t filename_size,
  * Do all the necessary cleanup and abort.
  * This function is implemented in the avtools, not cmdutils.
  */
-av_noreturn void exit_program(int ret);
+void exit_program(int ret);
 
 /**
  * Realloc array to hold new_size elements of elem_size.
@@ -426,65 +377,4 @@ av_noreturn void exit_program(int ret);
  */
 void *grow_array(void *array, int elem_size, int *size, int new_size);
 
-typedef struct FrameBuffer {
-    uint8_t *base[4];
-    uint8_t *data[4];
-    int  linesize[4];
-
-    int h, w;
-    enum PixelFormat pix_fmt;
-
-    int refcount;
-    struct FrameBuffer **pool;  ///< head of the buffer pool
-    struct FrameBuffer *next;
-} FrameBuffer;
-
-/**
- * Get a frame from the pool. This is intended to be used as a callback for
- * AVCodecContext.get_buffer.
- *
- * @param s codec context. s->opaque must be a pointer to the head of the
- *          buffer pool.
- * @param frame frame->opaque will be set to point to the FrameBuffer
- *              containing the frame data.
- */
-int codec_get_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVCodecContext.release_buffer along with
- * codec_get_buffer().
- */
-void codec_release_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVFilterBuffer.free.
- * @param fb buffer to free. fb->priv must be a pointer to the FrameBuffer
- *           containing the buffer data.
- */
-void filter_release_buffer(AVFilterBuffer *fb);
-
-/**
- * Free all the buffers in the pool. This must be called after all the
- * buffers have been released.
- */
-void free_buffer_pool(FrameBuffer **pool);
-
-#define GET_PIX_FMT_NAME(pix_fmt)\
-    const char *name = av_get_pix_fmt_name(pix_fmt);
-
-#define GET_SAMPLE_FMT_NAME(sample_fmt)\
-    const char *name = av_get_sample_fmt_name(sample_fmt)
-
-#define GET_SAMPLE_RATE_NAME(rate)\
-    char name[16];\
-    snprintf(name, sizeof(name), "%d", rate);
-
-#define GET_CH_LAYOUT_NAME(ch_layout)\
-    char name[16];\
-    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
-
-#define GET_CH_LAYOUT_DESC(ch_layout)\
-    char name[128];\
-    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
-
 #endif /* CMDUTILS_H */

cmdutils_common_opts.h

@@ -1,23 +1,18 @@
-    { "L"          , OPT_EXIT, {.func_arg = show_license},      "show license" },
-    { "h"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "?"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
-    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
-    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
-    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
-    { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
-    { "bsfs"       , OPT_EXIT, {.func_arg = show_bsfs     },    "show available bit stream filters" },
-    { "protocols"  , OPT_EXIT, {.func_arg = show_protocols},    "show available protocols" },
-    { "filters"    , OPT_EXIT, {.func_arg = show_filters  },    "show available filters" },
-    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
-    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
+    { "L", OPT_EXIT, {(void*)opt_license}, "show license" },
+    { "h", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "?", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "help", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "-help", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "version", OPT_EXIT, {(void*)opt_version}, "show version" },
+    { "formats"  , OPT_EXIT, {(void*)opt_formats  }, "show available formats" },
+    { "codecs"   , OPT_EXIT, {(void*)opt_codecs   }, "show available codecs" },
+    { "bsfs"     , OPT_EXIT, {(void*)opt_bsfs     }, "show available bit stream filters" },
+    { "protocols", OPT_EXIT, {(void*)opt_protocols}, "show available protocols" },
+    { "filters",   OPT_EXIT, {(void*)opt_filters  }, "show available filters" },
+    { "pix_fmts" , OPT_EXIT, {(void*)opt_pix_fmts }, "show available pixel formats" },
     { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
-    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
-    { "debug"      , HAS_ARG,  {.func_arg = opt_codec_debug},   "set debug flags", "flags" },
-    { "fdebug"     , HAS_ARG,  {.func_arg = opt_codec_debug},   "set debug flags", "flags" },
-    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
-    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
+    { "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
+    { "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
+    { "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
+    { "report", 0, {(void*)opt_report}, "generate a report" },
+    { "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },

common.mak

@@ -10,9 +10,8 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
-SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
-
+BRIEF  = CC CXX AS YASM AR LD HOSTCC STRIP CP
+SILENT = DEPCC YASMDEP RM RANLIB
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
@@ -21,23 +20,21 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_PATH)/
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
-CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
-ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
-YASMFLAGS  += $(IFLAGS:%=%/) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
-
-HOSTCCFLAGS = $(IFLAGS) $(HOSTCFLAGS)
-LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
+CCFLAGS     = $(CFLAGS)
+CXXFLAGS   := $(CFLAGS) $(CXXFLAGS)
+YASMFLAGS  += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
+HOSTCFLAGS += $(IFLAGS)
+LDFLAGS    := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
 
 define COMPILE
-       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
+       $($(1)DEP)
+       $($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
 endef
 
 COMPILE_C = $(call COMPILE,CC)
@@ -56,8 +53,8 @@ COMPILE_S = $(call COMPILE,AS)
 %.o: %.S
 	$(COMPILE_S)
 
-%.h.c:
-	$(Q)echo '#include "$*.h"' >$@
+%.ho: %.h
+	$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
 
 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
@@ -76,14 +73,13 @@ COMPILE_S = $(call COMPILE,AS)
 $(OBJS):
 endif
 
-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)
 
 OBJS      += $(OBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
-LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
-FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
+FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(EXTRALIBS)
 
 EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
@@ -100,31 +96,34 @@ DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
-HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
-checkheaders: $(HOBJS)
-.SECONDARY:   $(HOBJS:.o=.c)
+checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
 
 alltools: $(TOOLS)
 
 $(HOSTOBJS): %.o: %.c
-	$(call COMPILE,HOSTCC)
+	$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
+	$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
-LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
 
+define RULES
 clean::
 	$(RM) $(OBJS) $(OBJS:.o=.d)
+	$(RM) $(HOSTPROGS)
+	$(RM) $(TOOLS)
+endef
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
+$(eval $(RULES))
+
+-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))

compat/getopt.c

@@ -1,85 +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
- */
-
-/*
- * This file was copied from the following newsgroup posting:
- *
- * Newsgroups: mod.std.unix
- * Subject: public domain AT&T getopt source
- * Date: 3 Nov 85 19:34:15 GMT
- *
- * Here's something you've all been waiting for:  the AT&T public domain
- * source for getopt(3).  It is the code which was given out at the 1985
- * UNIFORUM conference in Dallas.  I obtained it by electronic mail
- * directly from AT&T.  The people there assure me that it is indeed
- * in the public domain.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-static int opterr = 1;
-static int optind = 1;
-static int optopt;
-static char *optarg;
-
-#undef fprintf
-
-static int getopt(int argc, char *argv[], char *opts)
-{
-    static int sp = 1;
-    int c;
-    char *cp;
-
-    if (sp == 1)
-        if (optind >= argc ||
-            argv[optind][0] != '-' || argv[optind][1] == '\0')
-            return EOF;
-        else if (!strcmp(argv[optind], "--")) {
-            optind++;
-            return EOF;
-        }
-    optopt = c = argv[optind][sp];
-    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
-        fprintf(stderr, ": illegal option -- %c\n", c);
-        if (argv[optind][++sp] == '\0') {
-            optind++;
-            sp = 1;
-        }
-        return '?';
-    }
-    if (*++cp == ':') {
-        if (argv[optind][sp+1] != '\0')
-            optarg = &argv[optind++][sp+1];
-        else if(++optind >= argc) {
-            fprintf(stderr, ": option requires an argument -- %c\n", c);
-            sp = 1;
-            return '?';
-        } else
-            optarg = argv[optind++];
-        sp = 1;
-    } else {
-        if (argv[optind][++sp] == '\0') {
-            sp = 1;
-            optind++;
-        }
-        optarg = NULL;
-    }
-
-    return c;
-}

compat/msvcrt/snprintf.c

@@ -1,71 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <stdio.h>
-#include <stdarg.h>
-#include <limits.h>
-#include <string.h>
-
-#include "compat/va_copy.h"
-#include "libavutil/error.h"
-
-#if defined(__MINGW32__)
-#define EOVERFLOW EFBIG
-#endif
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...)
-{
-    va_list ap;
-    int ret;
-
-    va_start(ap, fmt);
-    ret = avpriv_vsnprintf(s, n, fmt, ap);
-    va_end(ap);
-
-    return ret;
-}
-
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
-                     va_list ap)
-{
-    int ret;
-    va_list ap_copy;
-
-    if (n == 0)
-        return _vscprintf(fmt, ap);
-    else if (n > INT_MAX)
-        return AVERROR(EOVERFLOW);
-
-    /* we use n - 1 here because if the buffer is not big enough, the MS
-     * runtime libraries don't add a terminating zero at the end. MSDN
-     * recommends to provide _snprintf/_vsnprintf() a buffer size that
-     * is one less than the actual buffer, and zero it before calling
-     * _snprintf/_vsnprintf() to workaround this problem.
-     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
-    memset(s, 0, n);
-    va_copy(ap_copy, ap);
-    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
-    va_end(ap_copy);
-    if (ret == -1)
-        ret = _vscprintf(fmt, ap);
-
-    return ret;
-}

compat/msvcrt/snprintf.h

@@ -1,38 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef COMPAT_SNPRINTF_H
-#define COMPAT_SNPRINTF_H
-
-#include <stdarg.h>
-#include <stdio.h>
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...);
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
-
-#undef snprintf
-#undef _snprintf
-#undef vsnprintf
-#define snprintf avpriv_snprintf
-#define _snprintf avpriv_snprintf
-#define vsnprintf avpriv_vsnprintf
-
-#endif /* COMPAT_SNPRINTF_H */

compat/strtod.c

@@ -1,94 +0,0 @@
-/*
- * C99-compatible strtod() implementation
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <ctype.h>
-#include <limits.h>
-#include <stdlib.h>
-
-#include "libavutil/avstring.h"
-#include "libavutil/mathematics.h"
-
-static char *check_nan_suffix(char *s)
-{
-    char *start = s;
-
-    if (*s++ != '(')
-        return start;
-
-    while ((*s >= 'a' && *s <= 'z') || (*s >= 'A' && *s <= 'Z') ||
-           (*s >= '0' && *s <= '9') ||  *s == '_')
-        s++;
-
-    return *s == ')' ? s + 1 : start;
-}
-
-#undef strtod
-double strtod(const char *, char **);
-
-double avpriv_strtod(const char *nptr, char **endptr)
-{
-    char *end;
-    double res;
-
-    /* Skip leading spaces */
-    while (isspace(*nptr))
-        nptr++;
-
-    if (!av_strncasecmp(nptr, "infinity", 8)) {
-        end = nptr + 8;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "inf", 3)) {
-        end = nptr + 3;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+infinity", 9)) {
-        end = nptr + 9;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+inf", 4)) {
-        end = nptr + 4;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "-infinity", 9)) {
-        end = nptr + 9;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "-inf", 4)) {
-        end = nptr + 4;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "nan", 3)) {
-        end = check_nan_suffix(nptr + 3);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "+nan", 4) ||
-               !av_strncasecmp(nptr, "-nan", 4)) {
-        end = check_nan_suffix(nptr + 4);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "0x", 2) ||
-               !av_strncasecmp(nptr, "-0x", 3) ||
-               !av_strncasecmp(nptr, "+0x", 3)) {
-        /* FIXME this doesn't handle exponents, non-integers (float/double)
-         * and numbers too large for long long */
-        res = strtoll(nptr, &end, 16);
-    } else {
-        res = strtod(nptr, &end);
-    }
-
-    if (endptr)
-        *endptr = end;
-
-    return res;
-}

compat/va_copy.h

@@ -1,26 +0,0 @@
-/*
- * MSVC Compatible va_copy macro
- * Copyright (c) 2012 Derek Buitenhuis
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <stdarg.h>
-
-#if !defined(va_copy) && defined(_MSC_VER)
-#define va_copy(dst, src) ((dst) = (src))
-#endif

doc/Makefile

@@ -6,21 +6,18 @@ HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
               doc/fate.html                                             \
               doc/general.html                                          \
               doc/git-howto.html                                        \
+              doc/libavfilter.html                                      \
               doc/platform.html                                         \
-              doc/syntax.html                                           \
 
 TXTPAGES    = doc/fate.txt                                              \
 
 
-DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
-DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
-DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
-DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
-DOCS = $(DOCS-yes)
+DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
+ifdef HAVE_MAKEINFO
+DOCS += $(TXTPAGES)
+endif
 
-all-$(CONFIG_DOC): doc
-
-doc: documentation
+all-$(CONFIG_DOC): documentation
 
 documentation: $(DOCS)
 
@@ -31,38 +28,28 @@ doc/%.txt: doc/%.texi
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
 
-GENTEXI  = format codec
-GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
-
-$(GENTEXI): TAG = GENTEXI
-$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
-	$(M)doc/print_options $* > $@
-
 doc/%.html: TAG = HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
+doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+	$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
+doc/%.pod: doc/%.texi
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
+	$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
 
 doc/%.1: TAG = MAN
-doc/%.1: doc/%.pod $(GENTEXI)
+doc/%.1: doc/%.pod
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
 
-$(DOCS): | doc/
+$(DOCS): | doc
+OBJDIRS += doc
 
-install-man:
-
-ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man
 
 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
 	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
-endif
 
 uninstall: uninstall-man
 
@@ -70,8 +57,8 @@ uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 
 clean::
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
 
 -include $(wildcard $(DOCS:%=%.d))
 
-.PHONY: doc documentation
+.PHONY: documentation

doc/RELEASE_NOTES

@@ -1,11 +1,13 @@
 Release Notes
 =============
 
-* 1.0 "Angel"  September, 2012
+* 0.10 "Freedom"  January, 2012
 
 
 General notes
 -------------
+This release is binary compatible with 0.8 and 0.9.
+
 See the Changelog file for a list of significant changes. Note, there
 are many more new features and bugfixes than whats listed there.
 
@@ -14,3 +16,34 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
+
+
+API changes
+-----------
+
+A number of additional APIs have been introduced and some existing
+functions have been deprecated and are scheduled for removal in the next
+release. Significant API changes include:
+
+* new audio decoding API which decodes from an AVPacket to an AVFrame and
+is able to use AVCodecContext.get_buffer() in the similar way as video decoding.
+
+* new audio encoding API which encodes from an AVFrame to an AVPacket, thus
+allowing it to properly output timing information and side data.
+
+Please see the git history and the file doc/APIchanges for details.
+
+
+Other notable changes
+---------------------
+
+Libavcodec and libavformat built as shared libraries now hide non-public
+symbols. This will break applications using those symbols. Possible solutions
+are, in order of preference:
+1) Try finding a way of accomplishing the same with public API.
+2) If there is no corresponding public API, but you think there should be,
+post a request on the developer mailing list or IRC channel.
+3) Finally if your program needs access to FFmpeg / libavcodec / libavformat
+internals for some special reason then the best solution is to link statically.
+
+Please see the Changelog file and git history for a more detailed list of changes.

doc/avtools-common-opts.texi

@@ -18,10 +18,10 @@ are used to precisely specify which stream(s) does a given option belong to.
 
 A stream specifier is a string generally appended to the option name and
 separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
-@code{a:1} stream specifier, which matches the second audio stream. Therefore it
+@code{a:1} stream specifer, which matches the second audio stream. Therefore it
 would select the ac3 codec for the second audio stream.
 
-A stream specifier can match several streams, the option is then applied to all
+A stream specifier can match several stream, the option is then applied to all
 of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
 streams.
 
@@ -41,10 +41,7 @@ streams of this type.
 @item p:@var{program_id}[:@var{stream_index}]
 If @var{stream_index} is given, then matches stream number @var{stream_index} in
 program with id @var{program_id}. Otherwise matches all streams in this program.
-@item #@var{stream_id}
-Matches the stream by format-specific ID.
 @end table
-
 @section Generic options
 
 These options are shared amongst the av* tools.
@@ -54,29 +51,8 @@ These options are shared amongst the av* tools.
 @item -L
 Show license.
 
-@item -h, -?, -help, --help [@var{arg}]
-Show help. An optional parameter may be specified to print help about a specific
-item.
-
-Possible values of @var{arg} are:
-@table @option
-@item decoder=@var{decoder_name}
-Print detailed information about the decoder named @var{decoder_name}. Use the
-@option{-decoders} option to get a list of all decoders.
-
-@item encoder=@var{encoder_name}
-Print detailed information about the encoder named @var{encoder_name}. Use the
-@option{-encoders} option to get a list of all encoders.
-
-@item demuxer=@var{demuxer_name}
-Print detailed information about the demuxer named @var{demuxer_name}. Use the
-@option{-formats} option to get a list of all demuxers and muxers.
-
-@item muxer=@var{muxer_name}
-Print detailed information about the muxer named @var{muxer_name}. Use the
-@option{-formats} option to get a list of all muxers and demuxers.
-
-@end table
+@item -h, -?, -help, --help
+Show help.
 
 @item -version
 Show version.
@@ -93,16 +69,23 @@ Encoding available
 @end table
 
 @item -codecs
-Show all codecs known to libavcodec.
+Show available codecs.
 
-Note that the term 'codec' is used throughout this documentation as a shortcut
-for what is more correctly called a media bitstream format.
-
-@item -decoders
-Show available decoders.
-
-@item -encoders
-Show all available encoders.
+The fields preceding the codec names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@item V/A/S
+Video/audio/subtitle codec
+@item S
+Codec supports slices
+@item D
+Codec supports direct rendering
+@item T
+Codec can handle input truncated at random locations instead of only at frame boundaries
+@end table
 
 @item -bsfs
 Show available bitstream filters.
@@ -119,9 +102,6 @@ Show available pixel formats.
 @item -sample_fmts
 Show available sample formats.
 
-@item -layouts
-Show channel names and standard channel layouts.
-
 @item -loglevel @var{loglevel} | -v @var{loglevel}
 Set the logging level used by the library.
 @var{loglevel} is a number or a string containing one of the following values:
@@ -154,15 +134,6 @@ It also implies @code{-loglevel verbose}.
 Note: setting the environment variable @code{FFREPORT} to any value has the
 same effect.
 
-@item -cpuflags flags (@emph{global})
-Allows setting and clearing cpu flags. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpuflags -sse+mmx ...
-ffmpeg -cpuflags mmx ...
-ffmpeg -cpuflags 0 ...
-@end example
-
 @end table
 
 @section AVOptions
@@ -195,6 +166,3 @@ use @option{-option 0}/@option{-option 1}.
 
 Note2 old undocumented way of specifying per-stream AVOptions by prepending
 v/a/s to the options name is now obsolete and will be removed soon.
-
-@include avoptions_codec.texi
-@include avoptions_format.texi

doc/bitstream_filters.texi

@@ -71,7 +71,7 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to
 produce fully qualified JPEG images.
 
 @example
-ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
+ffmpeg -i mjpeg-movie.avi -c:v copy -vbsf mjpeg2jpeg frame_%d.jpg
 exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 @end example

doc/demuxers.texi

@@ -23,31 +23,8 @@ The description of some of the currently available demuxers follows.
 Image file demuxer.
 
 This demuxer reads from a list of image files specified by a pattern.
-The syntax and meaning of the pattern is specified by the
-option @var{pattern_type}.
 
-The pattern may contain a suffix which is used to automatically
-determine the format of the images contained in the files.
-
-The size, the pixel format, and the format of each image must be the
-same for all the files in the sequence.
-
-This demuxer accepts the following options:
-@table @option
-@item framerate
-Set the framerate for the video stream. It defaults to 25.
-@item loop
-If set to 1, loop over the input. Default value is 0.
-@item pattern_type
-Select the pattern type used to interpret the provided filename.
-
-@var{pattern_type} accepts one of the following values.
-@table @option
-@item sequence
-Select a sequence pattern type, used to specify a sequence of files
-indexed by sequential numbers.
-
-A sequence pattern may contain the string "%d" or "%0@var{N}d", which
+The pattern may contain the string "%d" or "%0@var{N}d", which
 specifies the position of the characters representing a sequential
 number in each filename matched by the pattern. If the form
 "%d0@var{N}d" is used, the string representing the number in each
@@ -55,11 +32,13 @@ filename is 0-padded and @var{N} is the total number of 0-padded
 digits representing the number. The literal character '%' can be
 specified in the pattern with the string "%%".
 
-If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
+If the pattern contains "%d" or "%0@var{N}d", the first filename of
 the file list specified by the pattern must contain a number
-inclusively contained between @var{start_number} and
-@var{start_number}+@var{start_number_range}-1, and all the following
-numbers must be sequential.
+inclusively contained between 0 and 4, all the following numbers must
+be sequential. This limitation may be hopefully fixed.
+
+The pattern may contain a suffix which is used to automatically
+determine the format of the images contained in the files.
 
 For example the pattern "img-%03d.bmp" will match a sequence of
 filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@@ -67,6 +46,17 @@ filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
 sequence of filenames of the form @file{i%m%g-1.jpg},
 @file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
 
+The size, the pixel format, and the format of each image must be the
+same for all the files in the sequence.
+
+The following example shows how to use @command{ffmpeg} for creating a
+video from the images in the file sequence @file{img-001.jpeg},
+@file{img-002.jpeg}, ..., assuming an input frame rate of 10 frames per
+second:
+@example
+ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
+@end example
+
 Note that the pattern must not necessarily contain "%d" or
 "%0@var{N}d", for example to convert a single image file
 @file{img.jpeg} you can employ the command:
@@ -74,75 +64,6 @@ Note that the pattern must not necessarily contain "%d" or
 ffmpeg -i img.jpeg img.png
 @end example
 
-@item glob
-Select a glob wildcard pattern type.
-
-The pattern is interpreted like a @code{glob()} pattern. This is only
-selectable if libavformat was compiled with globbing support.
-
-@item glob_sequence @emph{(deprecated, will be removed)}
-Select a mixed glob wildcard/sequence pattern.
-
-If your version of libavformat was compiled with globbing support, and
-the provided pattern contains at least one glob meta character among
-@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
-interpreted like a @code{glob()} pattern, otherwise it is interpreted
-like a sequence pattern.
-
-All glob special characters @code{%*?[]@{@}} must be prefixed
-with "%". To escape a literal "%" you shall use "%%".
-
-For example the pattern @code{foo-%*.jpeg} will match all the
-filenames prefixed by "foo-" and terminating with ".jpeg", and
-@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
-"foo-", followed by a sequence of three characters, and terminating
-with ".jpeg".
-
-This pattern type is deprecated in favor of @var{glob} and
-@var{sequence}.
-@end table
-
-Default value is @var{glob_sequence}.
-@item pixel_format
-Set the pixel format of the images to read. If not specified the pixel
-format is guessed from the first image file in the sequence.
-@item start_number
-Set the index of the file matched by the image file pattern to start
-to read from. Default value is 0.
-@item start_number_range
-Set the index interval range to check when looking for the first image
-file in the sequence, starting from @var{start_number}. Default value
-is 5.
-@item video_size
-Set the video size of the images to read. If not specified the video
-size is guessed from the first image file in the sequence.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} for creating a video from the images in the file
-sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
-input frame rate of 10 frames per second:
-@example
-ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
-@end example
-
-@item
-As above, but start by reading from a file with index 100 in the sequence:
-@example
-ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
-@end example
-
-@item
-Read images matching the "*.png" glob pattern , that is all the files
-terminating with the ".png" suffix:
-@example
-ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
-@end example
-@end itemize
-
 @section applehttp
 
 Apple HTTP Live Streaming demuxer.

doc/developer.texi

@@ -14,13 +14,12 @@
 @section API
 @itemize @bullet
 @item libavcodec is the library containing the codecs (both encoding and
-decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
-it.
+decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
 
 @item libavformat is the library containing the file format handling (mux and
 demux code for several formats). Look at @file{ffplay.c} to use it in a
-player. See @file{doc/examples/muxing.c} to use it to generate audio or video
-streams.
+player. See @file{libavformat/output-example.c} to use it to generate
+audio or video streams.
 
 @end itemize
 
@@ -188,10 +187,8 @@ the following snippet into your @file{.vimrc}:
 set expandtab
 set shiftwidth=4
 set softtabstop=4
-set cindent
-set cinoptions=(0
-" allow tabs in Makefiles
-autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
+" Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
 " Trailing whitespace and tabs are forbidden, so highlight them.
 highlight ForbiddenWhitespace ctermbg=red guibg=red
 match ForbiddenWhitespace /\s\+$\|\t/
@@ -201,16 +198,10 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
 
 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 @example
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
+(setq c-default-style "k&r")
+(setq-default c-basic-offset 4)
+(setq-default indent-tabs-mode nil)
+(setq-default show-trailing-whitespace t)
 @end example
 
 @section Development Policy
@@ -393,9 +384,7 @@ send a reminder by email. Your patch should eventually be dealt with.
 @item
     Did you register it in @file{allcodecs.c} or @file{allformats.c}?
 @item
-    Did you add the AVCodecID to @file{avcodec.h}?
-    When adding new codec IDs, also add an entry to the codec descriptor
-    list in @file{libavcodec/codec_desc.c}.
+    Did you add the CodecID to @file{avcodec.h}?
 @item
     If it has a fourCC, did you add it to @file{libavformat/riff.c},
     even if it is only a decoder?
@@ -491,10 +480,6 @@ send a reminder by email. Your patch should eventually be dealt with.
     Consider to add a regression test for your code.
 @item
     If you added YASM code please check that things still work with --disable-yasm
-@item
-    Make sure you check the return values of function and return appropriate
-    error codes. Especially memory allocation functions like @code{av_malloc()}
-    are notoriously left unchecked, which is a serious problem.
 @end enumerate
 
 @section Patch review process
@@ -535,16 +520,4 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
 this case, the reference results of the regression tests shall be modified
 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 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 introductionary message for the patch series that you post to
-the ffmpeg-devel mailing list, a direct link to download the sample media.
-
-
 @bye

doc/encoders.texi

@@ -511,12 +511,6 @@ rc_2pass_vbr_minsection_pct
 @item slices
 @code{VP8E_SET_TOKEN_PARTITIONS}
 
-@item max-intra-rate
-@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
-
-@item force_key_frames
-@code{VPX_EFLAG_FORCE_KF}
-
 @item Alternate reference frame related
 @table @option
 @item vp8flags altref

doc/eval.texi

@@ -21,78 +21,30 @@ The following unary operators are available: @code{+}, @code{-}.
 The following functions are available:
 @table @option
 @item sinh(x)
-Compute hyperbolic sine of @var{x}.
-
 @item cosh(x)
-Compute hyperbolic cosine of @var{x}.
-
 @item tanh(x)
-Compute hyperbolic tangent of @var{x}.
-
 @item sin(x)
-Compute sine of @var{x}.
-
 @item cos(x)
-Compute cosine of @var{x}.
-
 @item tan(x)
-Compute tangent of @var{x}.
-
 @item atan(x)
-Compute arctangent of @var{x}.
-
 @item asin(x)
-Compute arcsine of @var{x}.
-
 @item acos(x)
-Compute arccosine of @var{x}.
-
 @item exp(x)
-Compute exponential of @var{x} (with base @code{e}, the Euler's number).
-
 @item log(x)
-Compute natural logarithm of @var{x}.
-
 @item abs(x)
-Compute absolute value of @var{x}.
-
 @item squish(x)
-Compute expression @code{1/(1 + exp(4*x))}.
-
 @item gauss(x)
-Compute Gauss function of @var{x}, corresponding to
-@code{exp(-x*x/2) / sqrt(2*PI)}.
-
-@item isinf(x)
-Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
-
 @item isnan(x)
 Return 1.0 if @var{x} is NAN, 0.0 otherwise.
 
 @item mod(x, y)
-Compute the remainder of division of @var{x} by @var{y}.
-
 @item max(x, y)
-Return the maximum between @var{x} and @var{y}.
-
 @item min(x, y)
-Return the maximum between @var{x} and @var{y}.
-
 @item eq(x, y)
-Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
-
 @item gte(x, y)
-Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
-
 @item gt(x, y)
-Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
-
 @item lte(x, y)
-Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
-
 @item lt(x, y)
-Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
-
 @item st(var, expr)
 Allow to store the value of the expression @var{expr} in an internal
 variable. @var{var} specifies the number of the variable where to
@@ -154,18 +106,6 @@ the evaluation of @var{y}, return 0 otherwise.
 @item ifnot(x, y)
 Evaluate @var{x}, and if the result is zero return the result of the
 evaluation of @var{y}, return 0 otherwise.
-
-@item taylor(expr, x) taylor(expr, x, id)
-Evaluate a taylor series at x.
-expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
-then 0 is assumed.
-note, when you have the derivatives at y instead of 0
-taylor(expr, x-y) can be used
-When the series does not converge the results are undefined.
-
-@item root(expr, max)
-Finds x where f(x)=0 in the interval 0..max.
-f() must be continuous or the result is undefined.
 @end table
 
 The following constants are available:

doc/examples/Makefile

@@ -1,36 +1,21 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
+# use pkg-config for getting CFLAGS abd LDFLAGS
+FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
+CFLAGS+=$(shell pkg-config  --cflags $(FFMPEG_LIBS))
+LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
 
-CFLAGS += -Wall -O2 -g
-CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
-
-EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
-                filtering_video                    \
-                filtering_audio                    \
-                metadata                           \
-                muxing                             \
-                scaling_video                      \
+EXAMPLES=decoding_encoding filtering metadata muxing
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
-# the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
-muxing:            LDLIBS += -lm
+%: %.o
+	$(CC) $< $(LDFLAGS) -o $@
 
-.phony: all clean-test clean
+%.o: %.c
+	$(CC) $< $(CFLAGS) -c -o $@
+
+.phony: all clean
 
 all: $(OBJS) $(EXAMPLES)
 
-clean-test:
-	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-
-clean: clean-test
-	$(RM) $(EXAMPLES) $(OBJS)
+clean:
+	rm -rf $(EXAMPLES) $(OBJS)

doc/examples/decoding_encoding.c

@@ -29,73 +29,16 @@
  * format handling
  */
 
-#include <math.h>
-
-#include <libavutil/opt.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/audioconvert.h>
-#include <libavutil/common.h>
-#include <libavutil/imgutils.h>
-#include <libavutil/mathematics.h>
-#include <libavutil/samplefmt.h>
+#include "libavutil/imgutils.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avcodec.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"
 
 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 
-/* check that a given sample format is supported by the encoder */
-static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
-{
-    const enum AVSampleFormat *p = codec->sample_fmts;
-
-    while (*p != AV_SAMPLE_FMT_NONE) {
-        if (*p == sample_fmt)
-            return 1;
-        p++;
-    }
-    return 0;
-}
-
-/* just pick the highest supported samplerate */
-static int select_sample_rate(AVCodec *codec)
-{
-    const int *p;
-    int best_samplerate = 0;
-
-    if (!codec->supported_samplerates)
-        return 44100;
-
-    p = codec->supported_samplerates;
-    while (*p) {
-        best_samplerate = FFMAX(*p, best_samplerate);
-        p++;
-    }
-    return best_samplerate;
-}
-
-/* select layout with the highest channel count */
-static int select_channel_layout(AVCodec *codec)
-{
-    const uint64_t *p;
-    uint64_t best_ch_layout = 0;
-    int best_nb_channells   = 0;
-
-    if (!codec->channel_layouts)
-        return AV_CH_LAYOUT_STEREO;
-
-    p = codec->channel_layouts;
-    while (*p) {
-        int nb_channels = av_get_channel_layout_nb_channels(*p);
-
-        if (nb_channels > best_nb_channells) {
-            best_ch_layout    = *p;
-            best_nb_channells = nb_channels;
-        }
-        p++;
-    }
-    return best_ch_layout;
-}
-
 /*
  * Audio encoding example
  */
@@ -103,20 +46,18 @@ static void audio_encode_example(const char *filename)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    AVFrame *frame;
-    AVPacket pkt;
-    int i, j, k, ret, got_output;
-    int buffer_size;
+    int frame_size, i, j, out_size, outbuf_size;
     FILE *f;
-    uint16_t *samples;
+    short *samples;
     float t, tincr;
+    uint8_t *outbuf;
 
-    printf("Encode audio file %s\n", filename);
+    printf("Audio encoding\n");
 
     /* find the MP2 encoder */
-    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+    codec = avcodec_find_encoder(CODEC_ID_MP2);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
@@ -124,58 +65,25 @@ static void audio_encode_example(const char *filename)
 
     /* put sample parameters */
     c->bit_rate = 64000;
-
-    /* check that the encoder supports s16 pcm input */
+    c->sample_rate = 44100;
+    c->channels = 2;
     c->sample_fmt = AV_SAMPLE_FMT_S16;
-    if (!check_sample_fmt(codec, c->sample_fmt)) {
-        fprintf(stderr, "Encoder does not support sample format %s",
-                av_get_sample_fmt_name(c->sample_fmt));
-        exit(1);
-    }
-
-    /* select other audio parameters supported by the encoder */
-    c->sample_rate    = select_sample_rate(codec);
-    c->channel_layout = select_channel_layout(codec);
-    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
+    /* the codec gives us the frame size, in samples */
+    frame_size = c->frame_size;
+    samples = malloc(frame_size * 2 * c->channels);
+    outbuf_size = 10000;
+    outbuf = malloc(outbuf_size);
+
     f = fopen(filename, "wb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    /* frame containing input raw audio */
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate audio frame\n");
-        exit(1);
-    }
-
-    frame->nb_samples     = c->frame_size;
-    frame->format         = c->sample_fmt;
-    frame->channel_layout = c->channel_layout;
-
-    /* the codec gives us the frame size, in samples,
-     * we calculate the size of the samples buffer in bytes */
-    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
-                                             c->sample_fmt, 0);
-    samples = av_malloc(buffer_size);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
-                buffer_size);
-        exit(1);
-    }
-    /* setup the data pointers in the AVFrame */
-    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                                   (const uint8_t*)samples, buffer_size, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not setup audio frame\n");
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
@@ -183,46 +91,19 @@ static void audio_encode_example(const char *filename)
     t = 0;
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
     for(i=0;i<200;i++) {
-        av_init_packet(&pkt);
-        pkt.data = NULL; // packet data will be allocated by the encoder
-        pkt.size = 0;
-
-        for (j = 0; j < c->frame_size; j++) {
+        for(j=0;j<frame_size;j++) {
             samples[2*j] = (int)(sin(t) * 10000);
-
-            for (k = 1; k < c->channels; k++)
-                samples[2*j + k] = samples[2*j];
+            samples[2*j+1] = samples[2*j];
             t += tincr;
         }
         /* encode the samples */
-        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding audio frame\n");
-            exit(1);
-        }
-        if (got_output) {
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
-    }
-
-    /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
-        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
+        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
+        fwrite(outbuf, 1, out_size, f);
     }
     fclose(f);
+    free(outbuf);
+    free(samples);
 
-    av_freep(&samples);
-    avcodec_free_frame(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -242,26 +123,26 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     av_init_packet(&avpkt);
 
-    printf("Decode audio file %s to %s\n", filename, outfilename);
+    printf("Audio decoding\n");
 
     /* find the mpeg audio decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+    codec = avcodec_find_decoder(CODEC_ID_MP2);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     f = fopen(filename, "rb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
     outfile = fopen(outfilename, "wb");
@@ -279,7 +160,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
         if (!decoded_frame) {
             if (!(decoded_frame = avcodec_alloc_frame())) {
-                fprintf(stderr, "Could not allocate audio frame\n");
+                fprintf(stderr, "out of memory\n");
                 exit(1);
             }
         } else
@@ -320,7 +201,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&decoded_frame);
+    av_free(decoded_frame);
 }
 
 /*
@@ -330,22 +211,22 @@ static void video_encode_example(const char *filename, int codec_id)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int i, ret, x, y, got_output;
+    int i, out_size, size, x, y, outbuf_size;
     FILE *f;
-    AVFrame *frame;
-    AVPacket pkt;
-    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
+    AVFrame *picture;
+    uint8_t *outbuf;
 
-    printf("Encode video file %s\n", filename);
+    printf("Video encoding\n");
 
     /* find the mpeg1 video encoder */
     codec = avcodec_find_encoder(codec_id);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
+    picture= avcodec_alloc_frame();
 
     /* put sample parameters */
     c->bit_rate = 400000;
@@ -358,103 +239,77 @@ static void video_encode_example(const char *filename, int codec_id)
     c->max_b_frames=1;
     c->pix_fmt = PIX_FMT_YUV420P;
 
-    if(codec_id == AV_CODEC_ID_H264)
+    if(codec_id == CODEC_ID_H264)
         av_opt_set(c->priv_data, "preset", "slow", 0);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     f = fopen(filename, "wb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-    frame->format = c->pix_fmt;
-    frame->width  = c->width;
-    frame->height = c->height;
+    /* alloc image and output buffer */
+    outbuf_size = 100000;
+    outbuf = malloc(outbuf_size);
 
     /* the image can be allocated by any means and av_image_alloc() is
      * just the most convenient way if av_malloc() is to be used */
-    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
-                         c->pix_fmt, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate raw picture buffer\n");
-        exit(1);
-    }
+    av_image_alloc(picture->data, picture->linesize,
+                   c->width, c->height, c->pix_fmt, 1);
 
     /* encode 1 second of video */
     for(i=0;i<25;i++) {
-        av_init_packet(&pkt);
-        pkt.data = NULL;    // packet data will be allocated by the encoder
-        pkt.size = 0;
-
         fflush(stdout);
         /* prepare a dummy image */
         /* Y */
         for(y=0;y<c->height;y++) {
             for(x=0;x<c->width;x++) {
-                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
+                picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
             }
         }
 
         /* Cb and Cr */
         for(y=0;y<c->height/2;y++) {
             for(x=0;x<c->width/2;x++) {
-                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
-                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
+                picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
+                picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
             }
         }
 
-        frame->pts = i;
-
         /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
+        out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
+        printf("encoding frame %3d (size=%5d)\n", i, out_size);
+        fwrite(outbuf, 1, out_size, f);
     }
 
     /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
+    for(; out_size; i++) {
         fflush(stdout);
 
-        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
+        out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
+        printf("write frame %3d (size=%5d)\n", i, out_size);
+        fwrite(outbuf, 1, out_size, f);
     }
 
     /* add sequence end code to have a real mpeg file */
-    fwrite(endcode, 1, sizeof(endcode), f);
+    outbuf[0] = 0x00;
+    outbuf[1] = 0x00;
+    outbuf[2] = 0x01;
+    outbuf[3] = 0xb7;
+    fwrite(outbuf, 1, 4, f);
     fclose(f);
+    free(outbuf);
 
     avcodec_close(c);
     av_free(c);
-    av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
+    av_free(picture->data[0]);
+    av_free(picture);
     printf("\n");
 }
 
@@ -491,16 +346,18 @@ static void video_decode_example(const char *outfilename, const char *filename)
     /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
     memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
 
-    printf("Decode video file %s to %s\n", filename, outfilename);
+    printf("Video decoding\n");
 
     /* find the mpeg1 video decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
+    codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
     if (!codec) {
-        fprintf(stderr, "Codec not found\n");
+        fprintf(stderr, "codec not found\n");
         exit(1);
     }
 
     c = avcodec_alloc_context3(codec);
+    picture= avcodec_alloc_frame();
+
     if(codec->capabilities&CODEC_CAP_TRUNCATED)
         c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
 
@@ -509,8 +366,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
        available in the bitstream. */
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
@@ -518,13 +375,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
 
     f = fopen(filename, "rb");
     if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    picture = avcodec_alloc_frame();
-    if (!picture) {
-        fprintf(stderr, "Could not allocate video frame\n");
+        fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
@@ -557,7 +408,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
                 exit(1);
             }
             if (got_picture) {
-                printf("Saving frame %3d\n", frame);
+                printf("saving frame %3d\n", frame);
                 fflush(stdout);
 
                 /* the picture is allocated by the decoder. no need to
@@ -579,7 +430,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
     avpkt.size = 0;
     len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
     if (got_picture) {
-        printf("Saving last frame %3d\n", frame);
+        printf("saving last frame %3d\n", frame);
         fflush(stdout);
 
         /* the picture is allocated by the decoder. no need to
@@ -594,42 +445,33 @@ static void video_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&picture);
+    av_free(picture);
     printf("\n");
 }
 
 int main(int argc, char **argv)
 {
-    const char *output_type;
+    const char *filename;
+
+    /* must be called before using avcodec lib */
+    avcodec_init();
 
     /* register all the codecs */
     avcodec_register_all();
 
-    if (argc < 2) {
-        printf("usage: %s output_type\n"
-               "API example program to decode/encode a media stream with libavcodec.\n"
-               "This program generates a synthetic stream and encodes it to a file\n"
-               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
-               "The encoded stream is then decoded and written to a raw data output\n."
-               "output_type must be choosen between 'h264', 'mp2', 'mpg'\n",
-               argv[0]);
-        return 1;
-    }
-    output_type = argv[1];
+    if (argc <= 1) {
+        audio_encode_example("/tmp/test.mp2");
+        audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
 
-    if (!strcmp(output_type, "h264")) {
-        video_encode_example("test.h264", AV_CODEC_ID_H264);
-    } else if (!strcmp(output_type, "mp2")) {
-        audio_encode_example("test.mp2");
-        audio_decode_example("test.sw", "test.mp2");
-    } else if (!strcmp(output_type, "mpg")) {
-        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
-        video_decode_example("test%02d.pgm", "test.mpg");
+        video_encode_example("/tmp/test.h264", CODEC_ID_H264);
+        video_encode_example("/tmp/test.mpg", CODEC_ID_MPEG1VIDEO);
+        filename = "/tmp/test.mpg";
     } else {
-        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
-                output_type);
-        return 1;
+        filename = argv[1];
     }
 
+    //    audio_decode_example("/tmp/test.sw", filename);
+    video_decode_example("/tmp/test%d.pgm", filename);
+
     return 0;
 }

doc/examples/demuxing.c

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

doc/examples/filtering.c

@@ -27,14 +27,11 @@
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
-#include <unistd.h>
 
 #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 <libavfilter/vsrc_buffer.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -48,7 +45,7 @@ static int64_t last_pts = AV_NOPTS_VALUE;
 
 static int open_input_file(const char *filename)
 {
-    int ret;
+    int ret, i;
     AVCodec *dec;
 
     if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
@@ -56,7 +53,7 @@ static int open_input_file(const char *filename)
         return ret;
     }
 
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+    if ((ret = av_find_stream_info(fmt_ctx)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
         return ret;
     }
@@ -71,7 +68,7 @@ static int open_input_file(const char *filename)
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
 
     /* init the video decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+    if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
         return ret;
     }
@@ -84,21 +81,17 @@ static int init_filters(const char *filters_descr)
     char args[512];
     int ret;
     AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
-    AVBufferSinkParams *buffersink_params;
-
     filter_graph = avfilter_graph_alloc();
 
     /* buffer video source: the decoded frames from the decoder will be inserted here. */
-    snprintf(args, sizeof(args),
-            "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
-            dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-            dec_ctx->time_base.num, dec_ctx->time_base.den,
-            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
-
+    snprintf(args, sizeof(args), "%d:%d:%d:%d:%d:%d:%d",
+             dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
+             dec_ctx->time_base.num, dec_ctx->time_base.den,
+             dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
     ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                                        args, NULL, filter_graph);
     if (ret < 0) {
@@ -107,11 +100,8 @@ static int init_filters(const char *filters_descr)
     }
 
     /* buffer video sink: to terminate the filter chain. */
-    buffersink_params = av_buffersink_params_alloc();
-    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, buffersink_params, filter_graph);
-    av_free(buffersink_params);
+                                       NULL, pix_fmts, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
         return ret;
@@ -128,13 +118,12 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
                                     &inputs, &outputs, NULL)) < 0)
         return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
         return ret;
-    return 0;
 }
 
 static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
@@ -199,42 +188,35 @@ int main(int argc, char **argv)
             avcodec_get_frame_defaults(&frame);
             got_frame = 0;
             ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
+            av_free_packet(&packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                 break;
             }
 
             if (got_frame) {
-                frame.pts = av_frame_get_best_effort_timestamp(&frame);
-
+                if (frame.pts == AV_NOPTS_VALUE)
+                    frame.pts = frame.pkt_dts == AV_NOPTS_VALUE ?
+                        frame.pkt_dts : frame.pkt_pts;
                 /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-                    break;
-                }
+                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);
 
                 /* pull filtered pictures from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if (ret < 0)
-                        goto end;
-
+                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
+                    av_vsink_buffer_get_video_buffer_ref(buffersink_ctx, &picref, 0);
                     if (picref) {
                         display_picref(picref, buffersink_ctx->inputs[0]->time_base);
-                        avfilter_unref_bufferp(&picref);
+                        avfilter_unref_buffer(picref);
                     }
                 }
             }
         }
-        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
     if (dec_ctx)
         avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
+    av_close_input_file(fmt_ctx);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         char buf[1024];

doc/examples/filtering_audio.c

@@ -1,240 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2012 Clément Bœsch
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * API example for audio decoding and filtering
- */
-
-#include <unistd.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-
-const char *filter_descr = "aresample=8000,aconvert=s16:mono";
-const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
-
-static AVFormatContext *fmt_ctx;
-static AVCodecContext *dec_ctx;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int audio_stream_index = -1;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *dec;
-
-    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    /* 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 a audio stream in the input file\n");
-        return ret;
-    }
-    audio_stream_index = ret;
-    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-
-    /* init the audio decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filters(const char *filters_descr)
-{
-    char args[512];
-    int ret;
-    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    AVABufferSinkParams *abuffersink_params;
-    const AVFilterLink *outlink;
-    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
-
-    filter_graph = avfilter_graph_alloc();
-
-    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (!dec_ctx->channel_layout)
-        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-             time_base.num, time_base.den, dec_ctx->sample_rate,
-             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
-    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
-                                       args, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
-    }
-
-    /* buffer audio sink: to terminate the filter chain. */
-    abuffersink_params = av_abuffersink_params_alloc();
-    abuffersink_params->sample_fmts     = sample_fmts;
-    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, abuffersink_params, filter_graph);
-    av_free(abuffersink_params);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        return ret;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        return ret;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-
-    /* Print summary of the sink buffer
-     * Note: args buffer is reused to store channel layout string */
-    outlink = buffersink_ctx->inputs[0];
-    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
-    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
-           (int)outlink->sample_rate,
-           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
-           args);
-
-    return 0;
-}
-
-static void print_samplesref(AVFilterBufferRef *samplesref)
-{
-    const AVFilterBufferRefAudioProps *props = samplesref->audio;
-    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
-    const uint16_t *p     = (uint16_t*)samplesref->data[0];
-    const uint16_t *p_end = p + n;
-
-    while (p < p_end) {
-        fputc(*p    & 0xff, stdout);
-        fputc(*p>>8 & 0xff, stdout);
-        p++;
-    }
-    fflush(stdout);
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet;
-    AVFrame frame;
-    int got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
-        exit(1);
-    }
-
-    avcodec_register_all();
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = init_filters(filter_descr)) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        AVFilterBufferRef *samplesref;
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
-
-        if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(&frame);
-            got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
-                continue;
-            }
-
-            if (got_frame) {
-                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
-                    break;
-                }
-
-                /* pull filtered audio from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
-                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if(ret < 0)
-                        goto end;
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_bufferp(&samplesref);
-                    }
-                }
-            }
-        }
-        av_free_packet(&packet);
-    }
-end:
-    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
-        exit(1);
-    }
-
-    exit(0);
-}

doc/examples/metadata.c

@@ -50,6 +50,6 @@ int main (int argc, char **argv)
     while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
         printf("%s=%s\n", tag->key, tag->value);
 
-    avformat_close_input(&fmt_ctx);
+    avformat_free_context(fmt_ctx);
     return 0;
 }

doc/examples/muxing.c

@@ -33,15 +33,17 @@
 #include <string.h>
 #include <math.h>
 
-#include <libavutil/mathematics.h>
-#include <libavformat/avformat.h>
-#include <libswscale/swscale.h>
+#include "libavutil/mathematics.h"
+#include "libavformat/avformat.h"
+#include "libswscale/swscale.h"
+
+#undef exit
 
 /* 5 seconds stream duration */
 #define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
 #define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
-#define STREAM_PIX_FMT    PIX_FMT_YUV420P /* default pix_fmt */
+#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */
 
 static int sws_flags = SWS_BICUBIC;
 
@@ -50,38 +52,34 @@ static int sws_flags = SWS_BICUBIC;
 
 static float t, tincr, tincr2;
 static int16_t *samples;
+static uint8_t *audio_outbuf;
+static int audio_outbuf_size;
 static int audio_input_frame_size;
 
 /*
  * add an audio output stream
  */
-static AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
-                                  enum AVCodecID codec_id)
+static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
     AVCodecContext *c;
     AVStream *st;
 
-    /* find the audio encoder */
-    *codec = avcodec_find_encoder(codec_id);
-    if (!(*codec)) {
-        fprintf(stderr, "Could not find codec\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, *codec);
+    st = avformat_new_stream(oc, NULL);
     if (!st) {
-        fprintf(stderr, "Could not allocate stream\n");
+        fprintf(stderr, "Could not alloc stream\n");
         exit(1);
     }
     st->id = 1;
 
     c = st->codec;
+    c->codec_id = codec_id;
+    c->codec_type = AVMEDIA_TYPE_AUDIO;
 
     /* put sample parameters */
-    c->sample_fmt  = AV_SAMPLE_FMT_S16;
-    c->bit_rate    = 64000;
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    c->bit_rate = 64000;
     c->sample_rate = 44100;
-    c->channels    = 2;
+    c->channels = 2;
 
     // some formats want stream headers to be separate
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
@@ -90,35 +88,57 @@ static AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
     return st;
 }
 
-static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+static void open_audio(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
+    AVCodec *codec;
 
     c = st->codec;
 
+    /* find the audio encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     /* init signal generator */
-    t     = 0;
+    t = 0;
     tincr = 2 * M_PI * 110.0 / c->sample_rate;
     /* increment frequency by 110 Hz per second */
     tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
+    audio_outbuf_size = 10000;
+    audio_outbuf = av_malloc(audio_outbuf_size);
+
+    /* ugly hack for PCM codecs (will be removed ASAP with new PCM
+       support to compute the input frame size in samples */
+    if (c->frame_size <= 1) {
+        audio_input_frame_size = audio_outbuf_size / c->channels;
+        switch(st->codec->codec_id) {
+        case CODEC_ID_PCM_S16LE:
+        case CODEC_ID_PCM_S16BE:
+        case CODEC_ID_PCM_U16LE:
+        case CODEC_ID_PCM_U16BE:
+            audio_input_frame_size >>= 1;
+            break;
+        default:
+            break;
+        }
+    } else {
         audio_input_frame_size = c->frame_size;
-    samples = av_malloc(audio_input_frame_size *
-                        av_get_bytes_per_sample(c->sample_fmt) *
-                        c->channels);
+    }
+    samples = av_malloc(audio_input_frame_size * 2 * c->channels);
 }
 
-/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
- * 'nb_channels' channels. */
+/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
+   'nb_channels' channels */
 static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
     int j, i, v;
@@ -127,9 +147,9 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
     q = samples;
     for (j = 0; j < frame_size; j++) {
         v = (int)(sin(t) * 10000);
-        for (i = 0; i < nb_channels; i++)
+        for(i = 0; i < nb_channels; i++)
             *q++ = v;
-        t     += tincr;
+        t += tincr;
         tincr += tincr2;
     }
 }
@@ -137,33 +157,26 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
-    AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet;
-
+    AVPacket pkt;
     av_init_packet(&pkt);
+
     c = st->codec;
 
     get_audio_frame(samples, audio_input_frame_size, c->channels);
-    frame->nb_samples = audio_input_frame_size;
-    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);
 
-    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
-    if (!got_packet)
-        return;
+    pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);
 
+    if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
+        pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+    pkt.flags |= AV_PKT_FLAG_KEY;
     pkt.stream_index = st->index;
+    pkt.data = audio_outbuf;
 
-    /* Write the compressed frame to the media file. */
+    /* write the compressed frame in the media file */
     if (av_interleaved_write_frame(oc, &pkt) != 0) {
         fprintf(stderr, "Error while writing audio frame\n");
         exit(1);
     }
-    avcodec_free_frame(&frame);
 }
 
 static void close_audio(AVFormatContext *oc, AVStream *st)
@@ -171,31 +184,24 @@ static void close_audio(AVFormatContext *oc, AVStream *st)
     avcodec_close(st->codec);
 
     av_free(samples);
+    av_free(audio_outbuf);
 }
 
 /**************************************************************/
 /* video output */
 
-static AVFrame *frame;
-static AVPicture src_picture, dst_picture;
+static AVFrame *picture, *tmp_picture;
 static uint8_t *video_outbuf;
 static int frame_count, video_outbuf_size;
 
-/* Add a video output stream. */
-static AVStream *add_video_stream(AVFormatContext *oc, AVCodec **codec,
-                                  enum AVCodecID codec_id)
+/* add a video output stream */
+static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
     AVCodecContext *c;
     AVStream *st;
+    AVCodec *codec;
 
-    /* find the video encoder */
-    *codec = avcodec_find_encoder(codec_id);
-    if (!(*codec)) {
-        fprintf(stderr, "codec not found\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, *codec);
+    st = avformat_new_stream(oc, NULL);
     if (!st) {
         fprintf(stderr, "Could not alloc stream\n");
         exit(1);
@@ -203,108 +209,135 @@ static AVStream *add_video_stream(AVFormatContext *oc, AVCodec **codec,
 
     c = st->codec;
 
-    avcodec_get_context_defaults3(c, *codec);
+    /* find the video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+    avcodec_get_context_defaults3(c, codec);
 
     c->codec_id = codec_id;
 
-    /* Put sample parameters. */
+    /* put sample parameters */
     c->bit_rate = 400000;
-    /* Resolution must be a multiple of two. */
-    c->width    = 352;
-    c->height   = 288;
-    /* timebase: This is the fundamental unit of time (in seconds) in terms
-     * of which frame timestamps are represented. For fixed-fps content,
-     * timebase should be 1/framerate and timestamp increments should be
-     * identical to 1. */
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* time base: this is the fundamental unit of time (in seconds) in terms
+       of which frame timestamps are represented. for fixed-fps content,
+       timebase should be 1/framerate and timestamp increments should be
+       identically 1. */
     c->time_base.den = STREAM_FRAME_RATE;
     c->time_base.num = 1;
-    c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
-    c->pix_fmt       = STREAM_PIX_FMT;
-    if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
+    c->gop_size = 12; /* emit one intra frame every twelve frames at most */
+    c->pix_fmt = STREAM_PIX_FMT;
+    if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
         /* just for testing, we also add B frames */
         c->max_b_frames = 2;
     }
-    if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
+    if (c->codec_id == CODEC_ID_MPEG1VIDEO){
         /* Needed to avoid using macroblocks in which some coeffs overflow.
-         * This does not happen with normal video, it just happens here as
-         * the motion of the chroma plane does not match the luma plane. */
-        c->mb_decision = 2;
+           This does not happen with normal video, it just happens here as
+           the motion of the chroma plane does not match the luma plane. */
+        c->mb_decision=2;
     }
-    /* Some formats want stream headers to be separate. */
+    // some formats want stream headers to be separate
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
         c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 
     return st;
 }
 
-static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
 {
-    int ret;
-    AVCodecContext *c = st->codec;
+    AVFrame *picture;
+    uint8_t *picture_buf;
+    int size;
+
+    picture = avcodec_alloc_frame();
+    if (!picture)
+        return NULL;
+    size = avpicture_get_size(pix_fmt, width, height);
+    picture_buf = av_malloc(size);
+    if (!picture_buf) {
+        av_free(picture);
+        return NULL;
+    }
+    avpicture_fill((AVPicture *)picture, picture_buf,
+                   pix_fmt, width, height);
+    return picture;
+}
+
+static void open_video(AVFormatContext *oc, AVStream *st)
+{
+    AVCodec *codec;
+    AVCodecContext *c;
+
+    c = st->codec;
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
 
     /* open the codec */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
+    if (avcodec_open(c, codec) < 0) {
+        fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     video_outbuf = NULL;
     if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
-        /* Allocate output buffer. */
-        /* XXX: API change will be done. */
-        /* Buffers passed into lav* can be allocated any way you prefer,
-         * as long as they're aligned enough for the architecture, and
-         * they're freed appropriately (such as using av_free for buffers
-         * allocated with av_malloc). */
+        /* allocate output buffer */
+        /* XXX: API change will be done */
+        /* buffers passed into lav* can be allocated any way you prefer,
+           as long as they're aligned enough for the architecture, and
+           they're freed appropriately (such as using av_free for buffers
+           allocated with av_malloc) */
         video_outbuf_size = 200000;
-        video_outbuf      = av_malloc(video_outbuf_size);
+        video_outbuf = av_malloc(video_outbuf_size);
     }
 
-    /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-
-    /* Allocate the encoded raw picture. */
-    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
-    if (ret < 0) {
+    /* allocate the encoded raw picture */
+    picture = alloc_picture(c->pix_fmt, c->width, c->height);
+    if (!picture) {
         fprintf(stderr, "Could not allocate picture\n");
         exit(1);
     }
 
-    /* If the output format is not YUV420P, then a temporary YUV420P
-     * picture is needed too. It is then converted to the required
-     * output format. */
+    /* if the output format is not YUV420P, then a temporary YUV420P
+       picture is needed too. It is then converted to the required
+       output format */
+    tmp_picture = NULL;
     if (c->pix_fmt != PIX_FMT_YUV420P) {
-        ret = avpicture_alloc(&src_picture, PIX_FMT_YUV420P, c->width, c->height);
-        if (ret < 0) {
+        tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
+        if (!tmp_picture) {
             fprintf(stderr, "Could not allocate temporary picture\n");
             exit(1);
         }
     }
-
-    /* copy data and linesize picture pointers to frame */
-    *((AVPicture *)frame) = dst_picture;
 }
 
-/* Prepare a dummy image. */
-static void fill_yuv_image(AVPicture *pict, int frame_index,
-                           int width, int height)
+/* prepare a dummy image */
+static void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height)
 {
     int x, y, i;
 
     i = frame_index;
 
     /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
+    for (y = 0; y < height; y++) {
+        for (x = 0; x < width; x++) {
             pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+        }
+    }
 
     /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
+    for (y = 0; y < height/2; y++) {
+        for (x = 0; x < width/2; x++) {
             pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
             pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
         }
@@ -313,75 +346,69 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
 
 static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
-    int ret;
-    static struct SwsContext *sws_ctx;
-    AVCodecContext *c = st->codec;
+    int out_size, ret;
+    AVCodecContext *c;
+    static struct SwsContext *img_convert_ctx;
+
+    c = st->codec;
 
     if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
+        /* no more frame to compress. The codec has a latency of a few
+           frames if using B frames, so we get the last frames by
+           passing the same picture again */
     } else {
         if (c->pix_fmt != PIX_FMT_YUV420P) {
             /* as we only generate a YUV420P picture, we must convert it
-             * to the codec pixel format if needed */
-            if (!sws_ctx) {
-                sws_ctx = sws_getContext(c->width, c->height, PIX_FMT_YUV420P,
-                                         c->width, c->height, c->pix_fmt,
-                                         sws_flags, NULL, NULL, NULL);
-                if (!sws_ctx) {
-                    fprintf(stderr,
-                            "Could not initialize the conversion context\n");
+               to the codec pixel format if needed */
+            if (img_convert_ctx == NULL) {
+                img_convert_ctx = sws_getContext(c->width, c->height,
+                                                 PIX_FMT_YUV420P,
+                                                 c->width, c->height,
+                                                 c->pix_fmt,
+                                                 sws_flags, NULL, NULL, NULL);
+                if (img_convert_ctx == NULL) {
+                    fprintf(stderr, "Cannot initialize the conversion context\n");
                     exit(1);
                 }
             }
-            fill_yuv_image(&src_picture, frame_count, c->width, c->height);
-            sws_scale(sws_ctx,
-                      (const uint8_t * const *)src_picture.data, src_picture.linesize,
-                      0, c->height, dst_picture.data, dst_picture.linesize);
+            fill_yuv_image(tmp_picture, frame_count, c->width, c->height);
+            sws_scale(img_convert_ctx, tmp_picture->data, tmp_picture->linesize,
+                      0, c->height, picture->data, picture->linesize);
         } else {
-            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
+            fill_yuv_image(picture, frame_count, c->width, c->height);
         }
     }
 
+
     if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* Raw video case - the API will change slightly in the near
-         * future for that. */
+        /* raw video case. The API will change slightly in the near
+           future for that. */
         AVPacket pkt;
         av_init_packet(&pkt);
 
-        pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = st->index;
-        pkt.data          = dst_picture.data[0];
-        pkt.size          = sizeof(AVPicture);
+        pkt.flags |= AV_PKT_FLAG_KEY;
+        pkt.stream_index = st->index;
+        pkt.data = (uint8_t *)picture;
+        pkt.size = sizeof(AVPicture);
 
         ret = av_interleaved_write_frame(oc, &pkt);
     } else {
         /* encode the image */
-        AVPacket pkt;
-        int got_output;
+        out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size, picture);
+        /* if zero size, it means the image was buffered */
+        if (out_size > 0) {
+            AVPacket pkt;
+            av_init_packet(&pkt);
 
-        av_init_packet(&pkt);
-        pkt.data = NULL;    // packet data will be allocated by the encoder
-        pkt.size = 0;
-
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding video frame\n");
-            exit(1);
-        }
-
-        /* If size is zero, it means the image was buffered. */
-        if (got_output) {
             if (c->coded_frame->pts != AV_NOPTS_VALUE)
-                pkt.pts = av_rescale_q(c->coded_frame->pts,
-                                       c->time_base, st->time_base);
-            if (c->coded_frame->key_frame)
+                pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+            if(c->coded_frame->key_frame)
                 pkt.flags |= AV_PKT_FLAG_KEY;
-
             pkt.stream_index = st->index;
+            pkt.data = video_outbuf;
+            pkt.size = out_size;
 
-            /* Write the compressed frame to the media file. */
+            /* write the compressed frame in the media file */
             ret = av_interleaved_write_frame(oc, &pkt);
         } else {
             ret = 0;
@@ -397,9 +424,12 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
 static void close_video(AVFormatContext *oc, AVStream *st)
 {
     avcodec_close(st->codec);
-    av_free(src_picture.data[0]);
-    av_free(dst_picture.data[0]);
-    av_free(frame);
+    av_free(picture->data[0]);
+    av_free(picture);
+    if (tmp_picture) {
+        av_free(tmp_picture->data[0]);
+        av_free(tmp_picture);
+    }
     av_free(video_outbuf);
 }
 
@@ -412,11 +442,10 @@ int main(int argc, char **argv)
     AVOutputFormat *fmt;
     AVFormatContext *oc;
     AVStream *audio_st, *video_st;
-    AVCodec *audio_codec, *video_codec;
     double audio_pts, video_pts;
     int i;
 
-    /* Initialize libavcodec, and register all codecs and formats. */
+    /* initialize libavcodec, and register all codecs and formats */
     av_register_all();
 
     if (argc != 2) {
@@ -441,26 +470,26 @@ int main(int argc, char **argv)
     }
     fmt = oc->oformat;
 
-    /* Add the audio and video streams using the default format codecs
-     * and initialize the codecs. */
+    /* add the audio and video streams using the default format codecs
+       and initialize the codecs */
     video_st = NULL;
     audio_st = NULL;
-    if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        video_st = add_video_stream(oc, &video_codec, fmt->video_codec);
+    if (fmt->video_codec != CODEC_ID_NONE) {
+        video_st = add_video_stream(oc, fmt->video_codec);
     }
-    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        audio_st = add_audio_stream(oc, &audio_codec, fmt->audio_codec);
+    if (fmt->audio_codec != CODEC_ID_NONE) {
+        audio_st = add_audio_stream(oc, fmt->audio_codec);
     }
 
-    /* Now that all the parameters are set, we can open the audio and
-     * video codecs and allocate the necessary encode buffers. */
-    if (video_st)
-        open_video(oc, video_codec, video_st);
-    if (audio_st)
-        open_audio(oc, audio_codec, audio_st);
-
     av_dump_format(oc, 0, filename, 1);
 
+    /* now that all the parameters are set, we can open the audio and
+       video codecs and allocate the necessary encode buffers */
+    if (video_st)
+        open_video(oc, video_st);
+    if (audio_st)
+        open_audio(oc, audio_st);
+
     /* open the output file, if needed */
     if (!(fmt->flags & AVFMT_NOFILE)) {
         if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
@@ -469,23 +498,18 @@ int main(int argc, char **argv)
         }
     }
 
-    /* Write the stream header, if any. */
-    if (avformat_write_header(oc, NULL) < 0) {
-        fprintf(stderr, "Error occurred when opening output file\n");
-        return 1;
-    }
-
-    frame->pts = 0;
-    for (;;) {
-        /* Compute current audio and video time. */
+    /* write the stream header, if any */
+    av_write_header(oc);
+    picture->pts = 0;
+    for(;;) {
+        /* compute current audio and video time */
         if (audio_st)
             audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
         else
             audio_pts = 0.0;
 
         if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
+            video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
         else
             video_pts = 0.0;
 
@@ -498,31 +522,32 @@ int main(int argc, char **argv)
             write_audio_frame(oc, audio_st);
         } else {
             write_video_frame(oc, video_st);
-            frame->pts++;
+            picture->pts++;
         }
     }
 
-    /* Write the trailer, if any. The trailer must be written before you
-     * close the CodecContexts open when you wrote the header; otherwise
-     * av_write_trailer() may try to use memory that was freed on
-     * av_codec_close(). */
+    /* write the trailer, if any.  the trailer must be written
+     * before you close the CodecContexts open when you wrote the
+     * header; otherwise write_trailer may try to use memory that
+     * was freed on av_codec_close() */
     av_write_trailer(oc);
 
-    /* Close each codec. */
+    /* close each codec */
     if (video_st)
         close_video(oc, video_st);
     if (audio_st)
         close_audio(oc, audio_st);
 
-    /* Free the streams. */
-    for (i = 0; i < oc->nb_streams; i++) {
+    /* free the streams */
+    for(i = 0; i < oc->nb_streams; i++) {
         av_freep(&oc->streams[i]->codec);
         av_freep(&oc->streams[i]);
     }
 
-    if (!(fmt->flags & AVFMT_NOFILE))
-        /* Close the output file. */
+    if (!(fmt->flags & AVFMT_NOFILE)) {
+        /* close the output file */
         avio_close(oc->pb);
+    }
 
     /* free the stream */
     av_free(oc);

doc/examples/scaling_video.c

@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libswscale API use example.
- */
-
-#include <libavutil/imgutils.h>
-#include <libavutil/parseutils.h>
-#include <libswscale/swscale.h>
-
-static void fill_yuv_image(uint8_t *data[4], int linesize[4],
-                           int width, int height, int frame_index)
-{
-    int x, y, i;
-
-    i = frame_index;
-
-    /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
-            data[0][y * linesize[0] + x] = x + y + i * 3;
-
-    /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
-            data[1][y * linesize[1] + x] = 128 + y + i * 2;
-            data[2][y * linesize[2] + x] = 64 + x + i * 5;
-        }
-    }
-}
-
-int main(int argc, char **argv)
-{
-    uint8_t *src_data[4], *dst_data[4];
-    int src_linesize[4], dst_linesize[4];
-    int src_w = 320, src_h = 240, dst_w, dst_h;
-    enum PixelFormat src_pix_fmt = PIX_FMT_YUV420P, dst_pix_fmt = PIX_FMT_RGB24;
-    const char *dst_size = NULL;
-    const char *dst_filename = NULL;
-    FILE *dst_file;
-    int dst_bufsize;
-    struct SwsContext *sws_ctx;
-    int i, ret;
-
-    if (argc != 3) {
-        fprintf(stderr, "Usage: %s output_file output_size\n"
-                "API example program to show how to scale an image with libswscale.\n"
-                "This program generates a series of pictures, rescales them to the given "
-                "output_size and saves them to an output file named output_file\n."
-                "\n", argv[0]);
-        exit(1);
-    }
-    dst_filename = argv[1];
-    dst_size     = argv[2];
-
-    if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
-        fprintf(stderr,
-                "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
-                dst_size);
-        exit(1);
-    }
-
-    dst_file = fopen(dst_filename, "wb");
-    if (!dst_file) {
-        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
-        exit(1);
-    }
-
-    /* create scaling context */
-    sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
-                             dst_w, dst_h, dst_pix_fmt,
-                             SWS_BILINEAR, NULL, NULL, NULL);
-    if (!sws_ctx) {
-        fprintf(stderr,
-                "Impossible to create scale context for the conversion "
-                "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
-                av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
-                av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
-        ret = AVERROR(EINVAL);
-        goto end;
-    }
-
-    /* allocate source and destination image buffers */
-    if ((ret = av_image_alloc(src_data, src_linesize,
-                              src_w, src_h, src_pix_fmt, 16)) < 0) {
-        fprintf(stderr, "Could not allocate source image\n");
-        goto end;
-    }
-
-    /* buffer is going to be written to rawvideo file, no alignmnet */
-    if ((ret = av_image_alloc(dst_data, dst_linesize,
-                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
-        fprintf(stderr, "Could not allocate destination image\n");
-        goto end;
-    }
-    dst_bufsize = ret;
-
-    for (i = 0; i < 100; i++) {
-        /* generate synthetic video */
-        fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
-
-        /* convert to destination format */
-        sws_scale(sws_ctx, (const uint8_t * const*)src_data,
-                  src_linesize, 0, src_h, dst_data, dst_linesize);
-
-        /* write scaled image to file */
-        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
-    }
-
-    fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
-           "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
-
-end:
-    if (dst_file)
-        fclose(dst_file);
-    av_freep(&src_data[0]);
-    av_freep(&dst_data[0]);
-    sws_freeContext(sws_ctx);
-    return ret < 0;
-}

doc/faq.texi

@@ -213,56 +213,8 @@ For ANY other help on Avisynth, please visit the
 
 @section How can I join video files?
 
-To "join" video files is quite ambiguous. The following list explains the
-different kinds of "joining" and points out how those are addressed in
-FFmpeg. To join video files may mean:
-
-@itemize
-
-@item
-To put them one after the other: this is called to @emph{concatenate} them
-(in short: concat) and is addressed
-@ref{How can I concatenate video files, in this very faq}.
-
-@item
-To put them together in the same file, to let the user choose between the
-different versions (example: different audio languages): this is called to
-@emph{multiplex} them together (in short: mux), and is done by simply
-invoking ffmpeg with several @option{-i} options.
-
-@item
-For audio, to put all channels together in a single stream (example: two
-mono streams into one stereo stream): this is sometimes called to
-@emph{merge} them, and can be done using the
-@url{http://ffmpeg.org/ffmpeg.html#amerge, @code{amerge}} filter.
-
-@item
-For audio, to play one on top of the other: this is called to @emph{mix}
-them, and can be done by first merging them into a single stream and then
-using the @url{http://ffmpeg.org/ffmpeg.html#pan, @code{pan}} filter to mix
-the channels at will.
-
-@item
-For video, to display both together, side by side or one on top of a part of
-the other; it can be done using the
-@url{http://ffmpeg.org/ffmpeg.html#overlay, @code{overlay}} video filter.
-
-@end itemize
-
-@anchor{How can I concatenate video files}
-@section How can I concatenate video files?
-
-There are several solutions, depending on the exact circumstances.
-
-@subsection Concatenating using filters
-
-FFmpeg has a @url{http://ffmpeg.org/ffmpeg.html#concat-1, @code{concat}}
-filter designed specifically for that, with examples in the documentation.
-
-@subsection Concatenating at the file level
-
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
-video by merely concatenating the files them.
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by
+merely concatenating them.
 
 Hence you may concatenate your multimedia files by first transcoding them to
 these privileged formats, then using the humble @code{cat} command (or the
@@ -270,38 +222,28 @@ equally humble @code{copy} under Windows), and finally transcoding back to your
 format of choice.
 
 @example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+ffmpeg -i input1.avi -same_quant intermediate1.mpg
+ffmpeg -i input2.avi -same_quant intermediate2.mpg
 cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+ffmpeg -i intermediate_all.mpg -same_quant output.avi
 @end example
 
-Additionally, you can use the @code{concat} protocol instead of @code{cat} or
-@code{copy} which will avoid creation of a potentially huge intermediate file.
+Notice that you should either use @code{-same_quant} or set a reasonably high
+bitrate for your intermediate and output files, if you want to preserve
+video quality.
 
-@example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
-ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
-@end example
-
-Note that you may need to escape the character "|" which is special for many
-shells.
-
-Another option is usage of named pipes, should your platform support it:
+Also notice that you may avoid the huge intermediate files by taking advantage
+of named pipes, should your platform support it:
 
 @example
 mkfifo intermediate1.mpg
 mkfifo intermediate2.mpg
-ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
-ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
+ffmpeg -i input1.avi -same_quant -y intermediate1.mpg < /dev/null &
+ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -qscale:v 2 -c:v mpeg4 -acodec libmp3lame -q:a 4 output.avi
+ffmpeg -f mpeg -i - -same_quant -c:v mpeg4 -acodec libmp3lame output.avi
 @end example
 
-@subsection Concatenating using raw audio and video
-
 Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
 allow concatenation, and the transcoding step is almost lossless.
 When using multiple yuv4mpegpipe(s), the first line needs to be discarded
@@ -309,8 +251,7 @@ from all but the first stream. This can be accomplished by piping through
 @code{tail} as seen below. Note that when piping through @code{tail} you
 must use command grouping, @code{@{  ;@}}, to background properly.
 
-For example, let's say we want to concatenate two FLV files into an
-output.flv file:
+For example, let's say we want to join two FLV files into an output.flv file:
 
 @example
 mkfifo temp1.a
@@ -327,7 +268,7 @@ cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
 ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
        -f yuv4mpegpipe -i all.v \
-       -qscale:v 2 -y output.flv
+       -same_quant -y output.flv
 rm temp[12].[av] all.[av]
 @end example
 
@@ -369,34 +310,11 @@ specifying the exact format.
 aconvert=s16:stereo:packed
 @end example
 
-@section Why does FFmpeg not see the subtitles in my VOB file?
-
-VOB and a few other formats do not have a global header that describes
-everything present in the file. Instead, applications are supposed to scan
-the file to see what it contains. Since VOB files are frequently large, only
-the beginning is scanned. If the subtitles happen only later in the file,
-they will not be initally detected.
-
-Some applications, including the @code{ffmpeg} command-line tool, can only
-work with streams that were detected during the initial scan; streams that
-are detected later are ignored.
-
-The size of the initial scan is controlled by two options: @code{probesize}
-(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
-the subtitle stream to be detected, both values must be large enough.
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
 
-Yes. Check the @file{doc/examples} directory in the source
-repository, also available online at:
-@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
-
-Examples are also installed by default, usually in
-@code{$PREFIX/share/ffmpeg/examples}.
-
-Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
+Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively,
 examine the source code for one of the many open source projects that
 already incorporate FFmpeg at (@url{projects.html}).
 
@@ -457,24 +375,6 @@ Yes, as long as the code is optional and can easily and cleanly be placed
 under #if CONFIG_GPL without breaking anything. So, for example, a new codec
 or filter would be OK under GPL while a bug fix to LGPL code would not.
 
-@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
-
-FFmpeg builds static libraries by default. In static libraries, dependencies
-are not handled. That has two consequences. First, you must specify the
-libraries in dependency order: @code{-lavdevice} must come before
-@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
-Second, external libraries that are used in FFmpeg have to be specified too.
-
-An easy way to get the full list of required libraries in dependency order
-is to use @code{pkg-config}.
-
-@example
-  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
-@end example
-
-See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
-more details.
-
 @section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
 
 FFmpeg is a pure C project, so to use the libraries within your C++ application

doc/fate.texi

@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle FFmpeg Automated Testing Environment
+@settitle FATE Automated Testing Environment
 @titlepage
-@center @titlefont{FFmpeg Automated Testing Environment}
+@center @titlefont{FATE Automated Testing Environment}
 @end titlepage
 
 @node Top
@@ -78,14 +78,11 @@ Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
 @end float
 
-To use a custom wrapper to run the test, pass @option{--target-exec} to
-@command{configure} or set the @var{TARGET_EXEC} Make variable.
-
 
 @chapter Submitting the results to the FFmpeg result aggregation server
 
   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
+shell script tests/fate.sh from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.
 
 @example
@@ -121,9 +118,8 @@ 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
-and send the public key to the FATE server administrator who can be contacted
-at the email address @email{fate-admin@@ffmpeg.org}.
+  When you have everything working properly you can create an SSH key and
+send its public part to the FATE server administrator.
 
   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
@@ -133,11 +129,6 @@ The FATE server's fingerprint is:
 
   b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
 
-  If you have problems connecting to the FATE server, it may help to try out
-the @command{ssh} command with one or more @option{-v} options. You should
-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 synchronisation of the samples directory.
 
@@ -175,20 +166,9 @@ the synchronisation of the samples directory.
 @item THREADS
     Specify how many threads to use while running regression tests, it is
     quite useful to detect thread-related regressions.
-@item THREAD_TYPE
-    Specify which threading strategy test, either @var{slice} or @var{frame},
-    by default @var{slice+frame}
-@item CPUFLAGS
-    Specify CPU flags.
-@item TARGET_EXEC
-    Specify or override the wrapper used to run the tests.
-    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-    through @command{ssh}.
 @end table
 
-@section Examples
-
+Example:
 @example
-make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
+make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
 @end example

doc/ffmpeg.texi

@@ -79,126 +79,6 @@ The format option may be needed for raw input files.
 
 @c man end DESCRIPTION
 
-@chapter Detailed description
-@c man begin DETAILED DESCRIPTION
-
-The transcoding process in @command{ffmpeg} for each output can be described by
-the following diagram:
-
-@example
- _______              ______________               _________              ______________            ________
-|       |            |              |             |         |            |              |          |        |
-| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
-| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
-|_______|            |______________|             |_________|            |______________|          |________|
-
-@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
-multiple input files, @command{ffmpeg} tries to keep them synchronized by
-tracking lowest timestamp on any active input stream.
-
-Encoded packets are then passed to the decoder (unless streamcopy is selected
-for the stream, see further for a description). The decoder produces
-uncompressed frames (raw video/PCM audio/...) which can be processed further by
-filtering (see next section). After filtering the frames are passed to the
-encoder, which encodes them and outputs encoded packets again. Finally those are
-passed to the muxer, which writes the encoded packets to the output file.
-
-@section Filtering
-Before encoding, @command{ffmpeg} can process raw audio and video frames using
-filters from the libavfilter library. Several chained filters form a filter
-graph.  @command{ffmpeg} distinguishes between two types of filtergraphs -
-simple and complex.
-
-@subsection Simple filtergraphs
-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:
-
-@example
- _________                        __________              ______________
-|         |                      |          |            |              |
-| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
-| frames  | -------------------> | frames   | ---------> | packets      |
-|_________|                      |__________|            |______________|
-
-@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:
-
-@example
- _______        _____________        _______        _____        ________
-|       |      |             |      |       |      |     |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
-|_______|      |_____________|      |_______|      |_____|      |________|
-
-@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
-touch the frame contents. Another example is the @code{setpts} filter, which
-only sets timestamps and otherwise passes the frames unchanged.
-
-@subsection Complex filtergraphs
-Complex filtergraphs are those which cannot be described as simply a linear
-processing chain applied to one stream. This is the case e.g. when the graph has
-more than one input and/or output, or when output stream type is different from
-input. They can be represented with the following diagram:
-
-@example
- _________
-|         |
-| input 0 |\                    __________
-|_________| \                  |          |
-             \   _________    /| output 0 |
-              \ |         |  / |__________|
- _________     \| complex | /
-|         |     |         |/
-| input 1 |---->| filter  |\
-|_________|     |         | \   __________
-               /| graph   |  \ |          |
-              / |         |   \| output 1 |
- _________   /  |_________|    |__________|
-|         | /
-| input 2 |/
-|_________|
-
-@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
-cannot be unambiguously associated with a single stream or file.
-
-A trivial example of a complex filtergraph is the @code{overlay} filter, which
-has two video inputs and one video output, containing one video overlaid on top
-of the other. Its audio counterpart is the @code{amix} filter.
-
-@section Stream copy
-Stream copy is a mode selected by supplying the @code{copy} parameter to the
-@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding
-step for the specified stream, so it does only demuxing and muxing. It is useful
-for changing the container format or modifying container-level metadata. The
-diagram above will in this case simplify to this:
-
-@example
- _______              ______________            ________
-|       |            |              |          |        |
-| input |  demuxer   | encoded data |  muxer   | output |
-| file  | ---------> | packets      | -------> | file   |
-|_______|            |______________|          |________|
-
-@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
-filters is obviously also impossible, since filters work on uncompressed data.
-
-@c man end DETAILED DESCRIPTION
-
 @chapter Stream selection
 @c man begin STREAM SELECTION
 
@@ -263,7 +143,7 @@ Stop writing the output after its duration reaches @var{duration}.
 @var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 
 @item -fs @var{limit_size} (@emph{output})
-Set the file size limit, expressed in bytes.
+Set the file size limit.
 
 @item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
@@ -284,7 +164,7 @@ streams are delayed by @var{offset} seconds.
 Set the recording timestamp in the container.
 The syntax for @var{time} is:
 @example
-now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z])
 @end example
 If the value is "now" it takes the current time.
 Time is local time unless 'Z' or 'z' is appended, in which case it is
@@ -344,41 +224,12 @@ codec-dependent.
 @var{filter_graph} is a description of the filter graph to apply to
 the stream. Use @code{-filters} to show all the available filters
 (including also sources and sinks).
-
-See also the @option{-filter_complex} option if you want to create filter graphs
-with multiple inputs and/or outputs.
 @item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).
 
 @item -stats (@emph{global})
 Print encoding progress/statistics. On by default.
 
-@item -progress @var{url} (@emph{global})
-Send program-friendly progress information to @var{url}.
-
-Progress information is written approximately every second and at the end of
-the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
-consists of only alphanumeric characters. The last key of a sequence of
-progress information is always "progress".
-
-@item -stdin
-Enable interaction on standard input. On by default unless standard input is
-used as an input. To explicitly disable interaction you need to specify
-@code{-nostdin}.
-
-Disabling interaction on standard input is useful, for example, if
-ffmpeg is in the background process group. Roughly the same result can
-be achieved with @code{ffmpeg ... < /dev/null} but it requires a
-shell.
-
-@item -debug_ts (@emph{global})
-Print timestamp information. It is off by default. This option is
-mostly useful for testing and debugging purposes, and the output
-format may change from one version to another, so it should not be
-employed by portable scripts.
-
-See also the option @code{-fdebug ts}.
-
 @item -attach @var{filename} (@emph{output})
 Add an attachment to the output file. This is supported by a few formats
 like Matroska for e.g. fonts used in rendering subtitles. Attachments
@@ -420,27 +271,70 @@ attachments.
 @item -vframes @var{number} (@emph{output})
 Set the number of video frames to record. This is an alias for @code{-frames:v}.
 @item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
-Set frame rate (Hz value, fraction or abbreviation).
-
-As an input option, ignore any timestamps stored in the file and instead
-generate timestamps assuming constant frame rate @var{fps}.
-
-As an output option, duplicate or drop input frames to achieve constant output
-frame rate @var{fps} (note that this actually causes the @code{fps} filter to be
-inserted to the end of the corresponding filtergraph).
-
+Set frame rate (Hz value, fraction or abbreviation), (default = 25).
 @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
-Set frame size.
-
-As an input option, this is a shortcut for the @option{video_size} private
-option, recognized by some demuxers for which the frame size is either not
-stored in the file or is configurable -- e.g. raw video or video grabbers.
-
-As an output option, this inserts the @code{scale} video filter to the
-@emph{end} of the corresponding filtergraph. Please use the @code{scale} filter
-directly to insert it at the beginning or some other place.
-
-The format is @samp{wxh} (default - same as source).
+Set frame size. The format is @samp{wxh} (default - same as source).
+The following abbreviations are recognized:
+@table @samp
+@item sqcif
+128x96
+@item qcif
+176x144
+@item cif
+352x288
+@item 4cif
+704x576
+@item 16cif
+1408x1152
+@item qqvga
+160x120
+@item qvga
+320x240
+@item vga
+640x480
+@item svga
+800x600
+@item xga
+1024x768
+@item uxga
+1600x1200
+@item qxga
+2048x1536
+@item sxga
+1280x1024
+@item qsxga
+2560x2048
+@item hsxga
+5120x4096
+@item wvga
+852x480
+@item wxga
+1366x768
+@item wsxga
+1600x1024
+@item wuxga
+1920x1200
+@item woxga
+2560x1600
+@item wqsxga
+3200x2048
+@item wquxga
+3840x2400
+@item whsxga
+6400x4096
+@item whuxga
+7680x4800
+@item cga
+320x200
+@item ega
+640x350
+@item hd480
+852x480
+@item hd720
+1280x720
+@item hd1080
+1920x1080
+@end table
 
 @item -aspect[:@var{stream_specifier}] @var{aspect} (@emph{output,per-stream})
 Set the video display aspect ratio specified by @var{aspect}.
@@ -467,7 +361,25 @@ pad=width:height:x:y:color instead.
 
 @item -vn (@emph{output})
 Disable video recording.
-
+@item -bt @var{tolerance}
+Set video bitrate tolerance (in bits, default 4000k).
+Has a minimum value of: (target_bitrate/target_framerate).
+In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
+willing to deviate from the target average bitrate value. This is
+not related to min/max bitrate. Lowering tolerance too much has
+an adverse effect on quality.
+@item -maxrate @var{bitrate}
+Set max video bitrate (in bit/s).
+Requires -bufsize to be set.
+@item -minrate @var{bitrate}
+Set min video bitrate (in bit/s).
+Most useful in setting up a CBR encode:
+@example
+ffmpeg -i myfile.avi -b:v 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
+@end example
+It is of little use elsewise.
+@item -bufsize @var{size}
+Set video buffer verifier buffer size (in bits).
 @item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
 @item -same_quant
@@ -476,7 +388,7 @@ Use same quantizer as source (implies VBR).
 Note that this is NOT SAME QUALITY. Do not use this option unless you know you
 need it.
 
-@item -pass[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
+@item -pass @var{n}
 Select the pass number (1 or 2). It is used to do two-pass
 video encoding. The statistics of the video are recorded in the first
 pass into a log file (see also the option -passlogfile),
@@ -489,12 +401,16 @@ ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL
 ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null
 @end example
 
-@item -passlogfile[:@var{stream_specifier}] @var{prefix} (@emph{output,per-stream})
+@item -passlogfile @var{prefix} (@emph{global})
 Set two-pass log file name prefix to @var{prefix}, the default file name
 prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
+Note that this option is overwritten by a local option of the same name
+when using @code{-vcodec libx264}. That option maps to the x264 option stats
+which has a different syntax.
+
 @item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.
 
@@ -512,29 +428,202 @@ also sources and sinks).  This is an alias for @code{-filter:v}.
 @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
 Set pixel format. Use @code{-pix_fmts} to show all the supported
 pixel formats.
-If the selected pixel format can not be selected, ffmpeg will print a
-warning and select the best pixel format supported by the encoder.
-If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error
-if the requested pixel format can not be selected, and automatic conversions
-inside filter graphs are disabled.
-If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
-as the input (or graph output) and automatic conversions are disabled.
-
 @item -sws_flags @var{flags} (@emph{input/output})
 Set SwScaler flags.
+@item -g @var{gop_size}
+Set the group of pictures size.
+@item -intra
+deprecated, use -g 1
 @item -vdt @var{n}
 Discard threshold.
+@item -qmin @var{q}
+minimum video quantizer scale (VBR)
+@item -qmax @var{q}
+maximum video quantizer scale (VBR)
+@item -qdiff @var{q}
+maximum difference between the quantizer scales (VBR)
+@item -qblur @var{blur}
+video quantizer scale blur (VBR) (range 0.0 - 1.0)
+@item -qcomp @var{compression}
+video quantizer scale compression (VBR) (default 0.5).
+Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
+
+@item -lmin @var{lambda}
+minimum video lagrange factor (VBR)
+@item -lmax @var{lambda}
+max video lagrange factor (VBR)
+@item -mblmin @var{lambda}
+minimum macroblock quantizer scale (VBR)
+@item -mblmax @var{lambda}
+maximum macroblock quantizer scale (VBR)
+
+These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
+but you may use the QP2LAMBDA constant to easily convert from 'q' units:
+@example
+ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
+@end example
+
+@item -rc_init_cplx @var{complexity}
+initial complexity for single pass encoding
+@item -b_qfactor @var{factor}
+qp factor between P- and B-frames
+@item -i_qfactor @var{factor}
+qp factor between P- and I-frames
+@item -b_qoffset @var{offset}
+qp offset between P- and B-frames
+@item -i_qoffset @var{offset}
+qp offset between P- and I-frames
+@item -rc_eq @var{equation}
+Set rate control equation (see section "Expression Evaluation")
+(default = @code{tex^qComp}).
+
+When computing the rate control equation expression, besides the
+standard functions defined in the section "Expression Evaluation", the
+following functions are available:
+@table @var
+@item bits2qp(bits)
+@item qp2bits(qp)
+@end table
+
+and the following constants are available:
+@table @var
+@item iTex
+@item pTex
+@item tex
+@item mv
+@item fCode
+@item iCount
+@item mcVar
+@item var
+@item isI
+@item isP
+@item isB
+@item avgQP
+@item qComp
+@item avgIITex
+@item avgPITex
+@item avgPPTex
+@item avgBPTex
+@item avgTex
+@end table
 
 @item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
 Rate control override for specific intervals, formatted as "int,int,int"
 list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.
+@item -me_method @var{method}
+Set motion estimation method to @var{method}.
+Available methods are (from lowest to best quality):
+@table @samp
+@item zero
+Try just the (0, 0) vector.
+@item phods
+@item log
+@item x1
+@item hex
+@item umh
+@item epzs
+(default method)
+@item full
+exhaustive search (slow and marginally better than epzs)
+@end table
+
+@item -dct_algo @var{algo}
+Set DCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_DCT_AUTO (default)
+@item 1
+FF_DCT_FASTINT
+@item 2
+FF_DCT_INT
+@item 3
+FF_DCT_MMX
+@item 4
+FF_DCT_MLIB
+@item 5
+FF_DCT_ALTIVEC
+@end table
+
+@item -idct_algo @var{algo}
+Set IDCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_IDCT_AUTO (default)
+@item 1
+FF_IDCT_INT
+@item 2
+FF_IDCT_SIMPLE
+@item 3
+FF_IDCT_SIMPLEMMX
+@item 4
+FF_IDCT_LIBMPEG2MMX
+@item 5
+FF_IDCT_PS2
+@item 6
+FF_IDCT_MLIB
+@item 7
+FF_IDCT_ARM
+@item 8
+FF_IDCT_ALTIVEC
+@item 9
+FF_IDCT_SH4
+@item 10
+FF_IDCT_SIMPLEARM
+@end table
+
+@item -er @var{n}
+Set error resilience to @var{n}.
+@table @samp
+@item 1
+FF_ER_CAREFUL (default)
+@item 2
+FF_ER_COMPLIANT
+@item 3
+FF_ER_AGGRESSIVE
+@item 4
+FF_ER_VERY_AGGRESSIVE
+@end table
+
+@item -ec @var{bit_mask}
+Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
+the following values:
+@table @samp
+@item 1
+FF_EC_GUESS_MVS (default = enabled)
+@item 2
+FF_EC_DEBLOCK (default = enabled)
+@end table
+
+@item -bf @var{frames}
+Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
+@item -mbd @var{mode}
+macroblock decision
+@table @samp
+@item 0
+FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in ffmpeg).
+@item 1
+FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
+@item 2
+FF_MB_DECISION_RD: rate distortion
+@end table
+
+@item -4mv
+Use four motion vector by macroblock (MPEG-4 only).
+@item -part
+Use data partitioning (MPEG-4 only).
+@item -bug @var{param}
+Work around encoder bugs that are not auto-detected.
+@item -strict @var{strictness}
+How strictly to follow the standards.
+@item -aic
+Enable Advanced intra coding (h263+).
+@item -umv
+Enable Unlimited Motion Vector (h263+)
 
 @item -deinterlace
 Deinterlace pictures.
-This option is deprecated since the deinterlacing is very low quality.
-Use the yadif filter with @code{-filter:v yadif}.
 @item -ilme
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
@@ -593,11 +682,6 @@ Set the audio codec. This is an alias for @code{-codec:a}.
 @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.
-@item -af @var{filter_graph} (@emph{output})
-@var{filter_graph} is a description of the filter graph to apply to
-the input audio.
-Use the option "-filters" to show all the available filters (including
-also sources and sinks).  This is an alias for @code{-filter:a}.
 @end table
 
 @section Advanced Audio options:
@@ -605,6 +689,28 @@ also sources and sinks).  This is an alias for @code{-filter:a}.
 @table @option
 @item -atag @var{fourcc/tag} (@emph{output})
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
+@item -audio_service_type @var{type}
+Set the type of service that the audio stream contains.
+@table @option
+@item ma
+Main Audio Service (default)
+@item ef
+Effects
+@item vi
+Visually Impaired
+@item hi
+Hearing Impaired
+@item di
+Dialogue
+@item co
+Commentary
+@item em
+Emergency
+@item vo
+Voice Over
+@item ka
+Karaoke
+@end table
 @item -absf @var{bitstream_filter}
 Deprecated, see -bsf
 @end table
@@ -622,29 +728,17 @@ Disable subtitle recording.
 Deprecated, see -bsf
 @end table
 
-@section Advanced Subtitle options:
+@section Audio/Video grab options
 
 @table @option
-
-@item -fix_sub_duration
-Fix subtitles durations. For each subtitle, wait for the next packet in the
-same stream and adjust the duration of the first to avoid overlap. This is
-necessary with some subtitles codecs, especially DVB subtitles, because the
-duration in the original packet is only a rough estimate and the end is
-actually marked by an empty subtitle frame. Failing to use this option when
-necessary can result in exaggerated durations or muxing failures due to
-non-monotonic timestamps.
-
-Note that this option will delay the output of all data until the next
-subtitle packet is decoded: it may increase memory consumption and latency a
-lot.
-
+@item -isync (@emph{global})
+Synchronize read on input.
 @end table
 
 @section Advanced options
 
 @table @option
-@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
+@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] (@emph{output})
 
 Designate one or more input streams as a source for the output file. Each input
 stream is identified by the input file index @var{input_file_id} and
@@ -660,10 +754,6 @@ the source for output stream 1, etc.
 A @code{-} character before the stream identifier creates a "negative" mapping.
 It disables matching streams from already created mappings.
 
-An alternative @var{[linklabel]} form will map outputs from complex filter
-graphs (see the @option{-filter_complex} option) to the output file.
-@var{linklabel} must correspond to a defined output link label in the graph.
-
 For example, to map ALL streams from the first input file to output
 @example
 ffmpeg -i INPUT -map 0 output
@@ -701,7 +791,7 @@ Note that using this option disables the default mappings for this output file.
 
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 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
+@var{output_file_id}.@var{stream_specifier} are not set, the audio channel will
 be mapped on all the audio streams.
 
 Using "-1" instead of
@@ -723,18 +813,18 @@ The order of the "-map_channel" option specifies the order of the channels in
 the output stream. The output channel layout is guessed from the number of
 channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac"
 in combination of "-map_channel" makes the channel gain levels to be updated if
-input and output channel layouts don't match (for instance two "-map_channel"
-options and "-ac 6").
+channel layouts don't match (for instance two "-map_channel" options and "-ac
+6").
 
-You can also extract each channel of an input to specific outputs; the following
-command extracts two channels of the @var{INPUT} audio stream (file 0, stream 0)
-to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1} outputs:
+You can also extract each channel of an @var{INPUT} to specific outputs; the
+following command extract each channel of the audio stream (file 0, stream 0)
+to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1}:
 @example
 ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1
 @end example
 
-The following example splits the channels of a stereo input into two separate
-streams, which are put into the same output file:
+The following example split the channels of a stereo input into streams:
+
 @example
 ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg
 @end example
@@ -744,17 +834,9 @@ input stream; you can't for example use "-map_channel" to pick multiple input
 audio channels contained in different streams (from the same or different files)
 and merge them into a single output stream. It is therefore not currently
 possible, for example, to turn two separate mono streams into a single stereo
-stream. However splitting a stereo stream into two single channel mono streams
+stream. However spliting a stereo stream into two single channel mono streams
 is possible.
 
-If you need this feature, a possible workaround is to use the @emph{amerge}
-filter. For example, if you need to merge a media (here @file{input.mkv}) with 2
-mono audio streams into one single stereo channel audio stream (and keep the
-video stream), you can use the following command:
-@example
-ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv
-@end example
-
 @item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata})
 Set metadata information of the next output file from @var{infile}. Note that
 those are file indices (zero-based), not filenames.
@@ -839,24 +921,16 @@ Show benchmarking information at the end of an encode.
 Shows CPU time used and maximum memory consumption.
 Maximum memory consumption is not supported on all systems,
 it will usually display as 0 if not supported.
-@item -benchmark_all (@emph{global})
-Show benchmarking information during the encode.
-Shows CPU time used in various steps (audio/video encode/decode).
 @item -timelimit @var{duration} (@emph{global})
 Exit after ffmpeg has been running for @var{duration} seconds.
 @item -dump (@emph{global})
 Dump each input packet to stderr.
 @item -hex (@emph{global})
 When dumping packets, also dump the payload.
+@item -ps @var{size}
+Set RTP payload size in bytes.
 @item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
-By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
-This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming). If
-your input(s) is coming from some other live streaming source (through HTTP or
-UDP for example) the server might already be in real-time, thus the option will
-likely not be required. On the other hand, this is meaningful if your input(s)
-is a file you are trying to push in real-time.
 @item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -865,10 +939,10 @@ This option is deprecated, use -loop 1.
 Repeatedly loop output for formats that support looping such as animated GIF
 (0 will loop the output infinitely).
 This option is deprecated, use -loop.
+@item -threads @var{count}
+Thread count.
 @item -vsync @var{parameter}
 Video sync method.
-For compatibility reasons old values can be specified as numbers.
-Newly added values will have to be specified as strings always.
 
 @table @option
 @item 0, passthrough
@@ -879,9 +953,6 @@ constant framerate.
 @item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
-@item drop
-As passthrough but destroys all timestamps, making the muxer generate
-fresh timestamps based on frame-rate.
 @item -1, auto
 Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@@ -896,34 +967,11 @@ Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps
 the parameter is the maximum samples per second by which the audio is changed.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
-This option has been deprecated. Use the @code{asyncts} audio filter instead.
 @item -copyts
 Copy timestamps from input to output.
-@item -copytb @var{mode}
-Specify how to set the encoder timebase when stream copying.  @var{mode} is an
-integer numeric value, and can assume one of the following values:
-
-@table @option
-@item 1
-Use the demuxer timebase.
-
-The time base is copied to the output encoder from the corresponding input
-demuxer. This is sometimes required to avoid non monotonically increasing
-timestamps when copying video streams with variable frame rate.
-
-@item 0
-Use the decoder timebase.
-
-The time base is copied to the output encoder from the corresponding input
-decoder.
-
-@item -1
-Try to make the choice automatically, in order to generate a sane output.
-@end table
-
-Default value is -1.
-
-@item -shortest (@emph{output})
+@item -copytb
+Copy input stream time base from input to output when stream copying.
+@item -shortest
 Finish encoding when the shortest input stream ends.
 @item -dts_delta_threshold
 Timestamp discontinuity delta threshold.
@@ -948,10 +996,10 @@ Set bitstream filters for matching streams. @var{bistream_filters} is
 a comma-separated list of bitstream filters. Use the @code{-bsfs} option
 to get the list of bitstream filters.
 @example
-ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
+ffmpeg -i h264.mp4 -c:v copy -vbsf h264_mp4toannexb -an out.h264
 @end example
 @example
-ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
+ffmpeg -i file.mov -an -vn -sbsf mov2textsub -c:s copy -f rawvideo sub.txt
 @end example
 
 @item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
@@ -963,70 +1011,8 @@ Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
 @example
 ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
 @end example
-
-@item -filter_complex @var{filtergraph} (@emph{global})
-Define a complex filter graph, i.e. one with arbitrary number of inputs and/or
-outputs. For simple graphs -- those with one input and one output of the same
-type -- see the @option{-filter} options. @var{filtergraph} is a description of
-the filter graph, as described in @ref{Filtergraph syntax}.
-
-Input link labels must refer to input streams using the
-@code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map}
-uses). If @var{stream_specifier} matches multiple streams, the first one will be
-used. An unlabeled input will be connected to the first unused input stream of
-the matching type.
-
-Output link labels are referred to with @option{-map}. Unlabeled outputs are
-added to the first output file.
-
-Note that with this option it is possible to use only lavfi sources without
-normal input files.
-
-For example, to overlay an image over video
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map
-'[out]' out.mkv
-@end example
-Here @code{[0:v]} refers to the first video stream in the first input file,
-which is linked to the first (main) input of the overlay filter. Similarly the
-first video stream in the second input is linked to the second (overlay) input
-of overlay.
-
-Assuming there is only one video stream in each input file, we can omit input
-labels, so the above is equivalent to
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map
-'[out]' out.mkv
-@end example
-
-Furthermore we can omit the output label and the single output from the filter
-graph will be added to the output file automatically, so we can simply write
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
-@end example
-
-To generate 5 seconds of pure red video using lavfi @code{color} source:
-@example
-ffmpeg -filter_complex 'color=red' -t 5 out.mkv
-@end example
 @end table
 
-As a special exception, you can use a bitmap subtitle stream as input: it
-will be converted into a video with the same size as the largest video in
-the file, or 720x576 if no video is present. Note that this is an
-experimental and temporary solution. It will be removed once libavfilter has
-proper support for subtitles.
-
-For example, to hardcode subtitles on top of a DVB-T recording stored in
-MPEG-TS format, delaying the subtitles by 1 second:
-@example
-ffmpeg -i input.ts -filter_complex \
-  '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
-  -sn -map '#0x2dc' output.mkv
-@end example
-(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
-audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
-
 @section Preset files
 A preset file contains a sequence of @var{option}=@var{value} pairs,
 one for each line, specifying a sequence of options which would be
@@ -1050,15 +1036,15 @@ First ffmpeg searches for a file named @var{arg}.ffpreset in the
 directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
 the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg})
 or in a @file{ffpresets} folder along the executable on win32,
-in that order. For example, if the argument is @code{libvpx-1080p}, it will
-search for the file @file{libvpx-1080p.ffpreset}.
+in that order. For example, if the argument is @code{libx264-max}, it will
+search for the file @file{libx264-max.ffpreset}.
 
 If no such file is found, then ffmpeg will search for a file named
 @var{codec_name}-@var{arg}.ffpreset 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{-vpre 1080p},
-then it will search for the file @file{libvpx-1080p.ffpreset}.
+the video codec with @code{-vcodec libx264} and use @code{-vpre max},
+then it will search for the file @file{libx264-max.ffpreset}.
 @c man end OPTIONS
 
 @chapter Tips
@@ -1086,7 +1072,7 @@ 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
+'-me zero' to speed up motion estimation, and '-intra' to disable
 motion estimation completely (you have only I-frames, which means it
 is about as good as JPEG compression).
 
@@ -1275,16 +1261,6 @@ composed of three digits padded with zeroes to express the sequence
 number. It is the same syntax supported by the C printf function, but
 only formats accepting a normal integer are suitable.
 
-When importing an image sequence, -i also supports expanding
-shell-like wildcard patterns (globbing) internally, by selecting the
-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 -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
-@end example
-
 @item
 You can put many streams of the same type in the output:
 
@@ -1295,23 +1271,9 @@ ffmpeg -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy tes
 The resulting output file @file{test12.avi} will contain first four streams from
 the input file in reverse order.
 
-@item
-To force CBR video output:
-@example
-ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
-@end example
-
-@item
-The four options lmin, lmax, mblmin and mblmax use 'lambda' units,
-but you may use the QP2LAMBDA constant to easily convert from 'q' units:
-@example
-ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
-@end example
-
 @end itemize
 @c man end EXAMPLES
 
-@include syntax.texi
 @include eval.texi
 @include decoders.texi
 @include encoders.texi

doc/ffmpeg.txt

@@ -29,7 +29,7 @@
                                                                           \                        /                                   :
                                                                     +======\======================/======+      ^                      :
                                                           ------> 0 |   : source_index  : st-:---        |      :                      :
-                    OutputFile output_files[]           /           +------------------------------------+      :                      :
+                    OuputFile output_files[]            /           +------------------------------------+      :                      :
                                                        /          1 |   :               :    :           |      :                      :
                  ^  +------+------------+-----+       /             +------------------------------------+      :                      :
                  :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :

doc/ffplay.texi

@@ -178,7 +178,6 @@ Seek to percentage in file corresponding to fraction of width.
 
 @c man end
 
-@include syntax.texi
 @include eval.texi
 @include decoders.texi
 @include demuxers.texi

doc/ffprobe.texi

@@ -80,7 +80,7 @@ Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
 Prettify the format of the displayed values, it corresponds to the
 options "-unit -prefix -byte_binary_prefix -sexagesimal".
 
-@item -of, -print_format @var{writer_name}[=@var{writer_options}]
+@item -print_format @var{writer_name}[=@var{writer_options}]
 Set the output printing format.
 
 @var{writer_name} specifies the name of the writer, and
@@ -94,13 +94,6 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.
 
-@item -show_data
-Show payload data, as an hexadecimal and ASCII dump. Coupled with
-@option{-show_packets}, it will dump the packets' data. Coupled with
-@option{-show_streams}, it will dump the codec extradata.
-
-The dump is printed as the "data" field. It may contain newlines.
-
 @item -show_error
 Show information about the error found when trying to probe the input.
 
@@ -113,11 +106,6 @@ stream.
 All the container format information is printed within a section with
 name "FORMAT".
 
-@item -show_format_entry @var{name}
-Like @option{-show_format}, but only prints the specified entry of the
-container format information, rather than all. This option may be given more
-than once, then all specified entries will be shown.
-
 @item -show_packets
 Show information about each packet contained in the input multimedia
 stream.
@@ -139,14 +127,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".
 
-@item -count_frames
-Count the number of frames per stream and report it in the
-corresponding stream section.
-
-@item -count_packets
-Count the number of packets per stream and report it in the
-corresponding stream section.
-
 @item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -170,10 +150,6 @@ Show information related to program and library versions. This is the
 equivalent of setting both @option{-show_program_version} and
 @option{-show_library_versions} options.
 
-@item -bitexact
-Force bitexact output, useful to produce output which is not dependent
-on the specific build.
-
 @item -i @var{input_file}
 Read @var{input_file}.
 
@@ -206,27 +182,8 @@ keyN=valN
 Metadata tags are printed as a line in the corresponding FORMAT or
 STREAM section, and are prefixed by the string "TAG:".
 
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-A description of the accepted options follows.
-
-@table @option
-
-@item nokey, nk
-If set to 1 specify not to print the key of each field. Default value
-is 0.
-
-@item noprint_wrappers, nw
-If set to 1 specify not to print the section header and footer.
-Default value is 0.
-@end table
-
-@section compact, csv
-Compact and CSV format.
-
-The @code{csv} writer is equivalent to @code{compact}, but supports
-different defaults.
+@section compact
+Compact format.
 
 Each section is printed on a single line.
 If no option is specifid, the output has the form:
@@ -247,23 +204,21 @@ The description of the accepted options follows.
 
 @item item_sep, s
 Specify the character to use for separating fields in the output line.
-It must be a single printable character, it is "|" by default ("," for
-the @code{csv} writer).
+It must be a single printable character, it is "|" by default.
 
 @item nokey, nk
 If set to 1 specify not to print the key of each field. Its default
-value is 0 (1 for the @code{csv} writer).
+value is 0.
 
 @item escape, e
-Set the escape mode to use, default to "c" ("csv" for the @code{csv}
-writer).
+Set the escape mode to use, default to "c".
 
 It can assume one of the following values:
 @table @option
 @item c
-Perform C-like escaping. Strings containing a newline ('\n'), carriage
-return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
-character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
+Perform C-like escaping. Strings containing a newline ('\n') or
+carriage return ('\r'), the escaping character ('\') or the item
+separator character @var{SEP} are escaped using C-like fashioned
 escaping, so that a newline is converted to the sequence "\n", a
 carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
 converted to "\@var{SEP}".
@@ -277,80 +232,13 @@ containing a newline ('\n'), a carriage return ('\r'), a double quote
 Perform no escaping.
 @end table
 
-@item print_section, p
-Print the section name at the begin of each line if the value is
-@code{1}, disable it with value set to @code{0}. Default value is
-@code{1}.
-
 @end table
 
-@section flat
-Flat format.
+@section csv
+CSV format.
 
-A free-form output where each line contains an explicit key=value, such as
-"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be
-directly embedded in sh scripts as long as the separator character is an
-alphanumeric character or an underscore (see @var{sep_char} option).
-
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-The description of the accepted options follows.
-
-@table @option
-@item sep_char, s
-Separator character used to separate the chapter, the section name, IDs and
-potential tags in the printed field key.
-
-Default value is '.'.
-
-@item hierarchical, h
-Specify if the section name specification should be hierarchical. If
-set to 1, and if there is more than one section in the current
-chapter, the section name will be prefixed by the name of the
-chapter. A value of 0 will disable this behavior.
-
-Default value is 1.
-@end table
-
-@section ini
-INI format output.
-
-Print output in an INI based format.
-
-The following conventions are adopted:
-
-@itemize
-@item
-all key and values are UTF-8
-@item
-'.' is the subgroup separator
-@item
-newline, '\t', '\f', '\b' and the following characters are escaped
-@item
-'\' is the escape character
-@item
-'#' is the comment indicator
-@item
-'=' is the key/value separator
-@item
-':' 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 ":".
-
-The description of the accepted options follows.
-
-@table @option
-@item hierarchical, h
-Specify if the section name specification should be hierarchical. If
-set to 1, and if there is more than one section in the current
-chapter, the section name will be prefixed by the name of the
-chapter. A value of 0 will disable this behavior.
-
-Default value is 1.
-@end table
+This writer is equivalent to
+@code{compact=item_sep=,:nokey=1:escape=csv}.
 
 @section json
 JSON based format.
@@ -377,10 +265,6 @@ XML based format.
 The XML output is described in the XML schema description file
 @file{ffprobe.xsd} installed in the FFmpeg datadir.
 
-An updated version of the schema can be retrieved at the url
-@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
-latest schema committed into the FFmpeg development source code tree.
-
 Note that the output issued will be compliant to the
 @file{ffprobe.xsd} schema only when no special global output options
 (@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@@ -407,31 +291,26 @@ This option automatically sets @option{fully_qualified} to 1.
 
 For more information about the XML format, see
 @url{http://www.w3.org/XML/}.
-@c man end WRITERS
 
 @chapter Timecode
-@c man begin TIMECODE
 
 @command{ffprobe} supports Timecode extraction:
 
 @itemize
 
-@item
-MPEG1/2 timecode is extracted from the GOP, and is available in the video
+@item MPEG1/2 timecode is extracted from the GOP, and is available in the video
 stream details (@option{-show_streams}, see @var{timecode}).
 
-@item
-MOV timecode is extracted from tmcd track, so is available in the tmcd
+@item MOV timecode is extracted from tmcd track, so is available in the tmcd
 stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
 
-@item
-DV, GXF and AVI timecodes are available in format metadata
+@item DV and GXF timecodes are available in format metadata
 (@option{-show_format}, see @var{TAG:timecode}).
 
 @end itemize
-@c man end TIMECODE
 
-@include syntax.texi
+@c man end WRITERS
+
 @include decoders.texi
 @include demuxers.texi
 @include protocols.texi

doc/ffprobe.xsd

@@ -39,12 +39,9 @@
       <xsd:attribute name="dts_time"      type="xsd:float" />
       <xsd:attribute name="duration"      type="xsd:long"  />
       <xsd:attribute name="duration_time" type="xsd:float" />
-      <xsd:attribute name="convergence_duration"      type="xsd:long"  />
-      <xsd:attribute name="convergence_duration_time" type="xsd:float" />
       <xsd:attribute name="size"          type="xsd:long" use="required" />
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
-      <xsd:attribute name="data"          type="xsd:string" />
     </xsd:complexType>
 
     <xsd:complexType name="frameType">
@@ -56,15 +53,11 @@
       <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_duration"  type="xsd:long" />
-      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
 
       <!-- audio attributes -->
       <xsd:attribute name="sample_fmt"             type="xsd:string"/>
       <xsd:attribute name="nb_samples"             type="xsd:long"  />
-      <xsd:attribute name="channels"               type="xsd:int"   />
-      <xsd:attribute name="channel_layout"         type="xsd:string"/>
 
       <!-- video attributes -->
       <xsd:attribute name="width"                  type="xsd:long"  />
@@ -87,19 +80,13 @@
     </xsd:complexType>
 
     <xsd:complexType name="streamType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
       <xsd:attribute name="index"            type="xsd:int" use="required"/>
       <xsd:attribute name="codec_name"       type="xsd:string" />
       <xsd:attribute name="codec_long_name"  type="xsd:string" />
-      <xsd:attribute name="profile"          type="xsd:string" />
       <xsd:attribute name="codec_type"       type="xsd:string" />
       <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
-      <xsd:attribute name="extradata"        type="xsd:string" />
 
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
@@ -121,14 +108,9 @@
       <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
       <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
       <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
-      <xsd:attribute name="start_pts"        type="xsd:long"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration_ts"      type="xsd:long"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="bit_rate"         type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
-      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
-      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
     </xsd:complexType>
 
     <xsd:complexType name="formatType">
@@ -139,7 +121,7 @@
       <xsd:attribute name="filename"         type="xsd:string" use="required"/>
       <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
       <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
-      <xsd:attribute name="format_long_name" type="xsd:string"/>
+      <xsd:attribute name="format_long_name" type="xsd:string" use="required"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="size"             type="xsd:long"/>

doc/ffserver.conf

@@ -373,3 +373,5 @@ ACL allow 192.168.0.0 192.168.255.255
 <Redirect index.html>
 URL http://www.ffmpeg.org/
 </Redirect>
+
+

doc/filter_design.txt

@@ -1,269 +0,0 @@
-Filter design
-=============
-
-This document explains guidelines that should be observed (or ignored with
-good reason) when writing filters for libavfilter.
-
-In this document, the word “frame” indicates either a video frame or a group
-of audio samples, as stored in an AVFilterBuffer structure.
-
-
-Format negotiation
-==================
-
-  The query_formats method should set, for each input and each output links,
-  the list of supported formats.
-
-  For video links, that means pixel format. For audio links, that means
-  channel layout, and sample format (the sample packing is implied by the
-  sample format).
-
-  The lists are not just lists, they are references to shared objects. When
-  the negotiation mechanism computes the intersection of the formats
-  supported at each ends of a link, all references to both lists are
-  replaced with a reference to the intersection. And when a single format is
-  eventually chosen for a link amongst the remaining list, again, all
-  references to the list are updated.
-
-  That means that if a filter requires that its input and output have the
-  same format amongst a supported list, all it has to do is use a reference
-  to the same list of formats.
-
-
-Buffer references ownership and permissions
-===========================================
-
-  Principle
-  ---------
-
-    Audio and video data are voluminous; the buffer and buffer reference
-    mechanism is intended to avoid, as much as possible, expensive copies of
-    that data while still allowing the filters to produce correct results.
-
-    The data is stored in buffers represented by AVFilterBuffer structures.
-    They must not be accessed directly, but through references stored in
-    AVFilterBufferRef structures. Several references can point to the
-    same buffer; the buffer is automatically deallocated once all
-    corresponding references have been destroyed.
-
-    The characteristics of the data (resolution, sample rate, etc.) are
-    stored in the reference; different references for the same buffer can
-    show different characteristics. In particular, a video reference can
-    point to only a part of a video buffer.
-
-    A reference is usually obtained as input to the start_frame or
-    filter_samples method or requested using the ff_get_video_buffer or
-    ff_get_audio_buffer functions. A new reference on an existing buffer can
-    be created with the avfilter_ref_buffer. A reference is destroyed using
-    the avfilter_unref_bufferp function.
-
-  Reference ownership
-  -------------------
-
-    At any time, a reference “belongs” to a particular piece of code,
-    usually a filter. With a few caveats that will be explained below, only
-    that piece of code is allowed to access it. It is also responsible for
-    destroying it, although this is sometimes done automatically (see the
-    section on link reference fields).
-
-    Here are the (fairly obvious) rules for reference ownership:
-
-    * A reference received by the start_frame or filter_samples method
-      belong to the corresponding filter.
-
-      Special exception: for video references: the reference may be used
-      internally for automatic copying and must not be destroyed before
-      end_frame; it can be given away to ff_start_frame.
-
-    * A reference passed to ff_start_frame or ff_filter_samples is given
-      away and must no longer be used.
-
-    * A reference created with avfilter_ref_buffer belongs to the code that
-      created it.
-
-    * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
-      belongs to the code that requested it.
-
-    * A reference given as return value by the get_video_buffer or
-      get_audio_buffer method is given away and must no longer be used.
-
-  Link reference fields
-  ---------------------
-
-    The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
-    the rules to handle them:
-
-    * cur_buf is set before the start_frame and filter_samples methods to
-      the same reference given as argument to the methods and belongs to the
-      destination filter of the link. If it has not been cleared after
-      end_frame or filter_samples, libavfilter will automatically destroy
-      the reference; therefore, any filter that needs to keep the reference
-      for longer must set cur_buf to NULL.
-
-    * out_buf belongs to the source filter of the link and can be used to
-      store a reference to the buffer that has been sent to the destination.
-      If it is not NULL after end_frame or filter_samples, libavfilter will
-      automatically destroy the reference.
-
-      If a video input pad does not have a start_frame method, the default
-      method will request a buffer on the first output of the filter, store
-      the reference in out_buf and push a second reference to the output.
-
-    * src_buf, cur_buf_copy and partial_buf are used by libavfilter
-      internally and must not be accessed by filters.
-
-  Reference permissions
-  ---------------------
-
-    The AVFilterBufferRef structure has a perms field that describes what
-    the code that owns the reference is allowed to do to the buffer data.
-    Different references for the same buffer can have different permissions.
-
-    For video filters, the permissions only apply to the parts of the buffer
-    that have already been covered by the draw_slice method.
-
-    The value is a binary OR of the following constants:
-
-    * AV_PERM_READ: the owner can read the buffer data; this is essentially
-      always true and is there for self-documentation.
-
-    * AV_PERM_WRITE: the owner can modify the buffer data.
-
-    * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data
-      will not be modified by previous filters.
-
-    * AV_PERM_REUSE: the owner can output the buffer several times, without
-      modifying the data in between.
-
-    * AV_PERM_REUSE2: the owner can output the buffer several times and
-      modify the data in between (useless without the WRITE permissions).
-
-    * AV_PERM_ALIGN: the owner can access the data using fast operations
-      that require data alignment.
-
-    The READ, WRITE and PRESERVE permissions are about sharing the same
-    buffer between several filters to avoid expensive copies without them
-    doing conflicting changes on the data.
-
-    The REUSE and REUSE2 permissions are about special memory for direct
-    rendering. For example a buffer directly allocated in video memory must
-    not modified once it is displayed on screen, or it will cause tearing;
-    it will therefore not have the REUSE2 permission.
-
-    The ALIGN permission is about extracting part of the buffer, for
-    copy-less padding or cropping for example.
-
-
-    References received on input pads are guaranteed to have all the
-    permissions stated in the min_perms field and none of the permissions
-    stated in the rej_perms.
-
-    References obtained by ff_get_video_buffer and ff_get_audio_buffer are
-    guaranteed to have at least all the permissions requested as argument.
-
-    References created by avfilter_ref_buffer have the same permissions as
-    the original reference minus the ones explicitly masked; the mask is
-    usually ~0 to keep the same permissions.
-
-    Filters should remove permissions on reference they give to output
-    whenever necessary. It can be automatically done by setting the
-    rej_perms field on the output pad.
-
-    Here are a few guidelines corresponding to common situations:
-
-    * Filters that modify and forward their frame (like drawtext) need the
-      WRITE permission.
-
-    * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and and must request a buffer
-      with the WRITE permission.
-
-    * Filters that intend to keep a reference after the filtering process
-      is finished (after end_frame or filter_samples returns) must have the
-      PRESERVE permission on it and remove the WRITE permission if they
-      create a new reference to give it away.
-
-    * Filters that intend to modify a reference they have kept after the end
-      of the filtering process need the REUSE2 permission and must remove
-      the PRESERVE permission if they create a new reference to give it
-      away.
-
-
-Frame scheduling
-================
-
-  The purpose of these rules is to ensure that frames flow in the filter
-  graph without getting stuck and accumulating somewhere.
-
-  Simple filters that output one frame for each input frame should not have
-  to worry about it.
-
-  start_frame / filter_samples
-  ----------------------------
-
-    These methods are called when a frame is pushed to the filter's input.
-    They can be called at any time except in a reentrant way.
-
-    If the input frame is enough to produce output, then the filter should
-    push the output frames on the output link immediately.
-
-    As an exception to the previous rule, if the input frame is enough to
-    produce several output frames, then the filter needs output only at
-    least one per link. The additional frames can be left buffered in the
-    filter; these buffered frames must be flushed immediately if a new input
-    produces new output.
-
-    (Example: framerate-doubling filter: start_frame must (1) flush the
-    second copy of the previous frame, if it is still there, (2) push the
-    first copy of the incoming frame, (3) keep the second copy for later.)
-
-    If the input frame is not enough to produce output, the filter must not
-    call request_frame to get more. It must just process the frame or queue
-    it. The task of requesting more frames is left to the filter's
-    request_frame method or the application.
-
-    If a filter has several inputs, the filter must be ready for frames
-    arriving randomly on any input. Therefore, any filter with several inputs
-    will most likely require some kind of queuing mechanism. It is perfectly
-    acceptable to have a limited queue and to drop frames when the inputs
-    are too unbalanced.
-
-  request_frame
-  -------------
-
-    This method is called when a frame is wanted on an output.
-
-    For an input, it should directly call start_frame or filter_samples on
-    the corresponding output.
-
-    For a filter, if there are queued frames already ready, one of these
-    frames should be pushed. If not, the filter should request a frame on
-    one of its inputs, repeatedly until at least one frame has been pushed.
-
-    Return values:
-    if request_frame could produce 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.
-
-    The typical implementation of request_frame for a filter with several
-    inputs will look like that:
-
-        if (frames_queued) {
-            push_one_frame();
-            return 0;
-        }
-        while (!frame_pushed) {
-            input = input_where_a_frame_is_most_needed();
-            ret = avfilter_request_frame(input);
-            if (ret == AVERROR_EOF) {
-                process_eof_on_input();
-            } 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 start_frame / filter_samples method will be called and do the work.

doc/general.texi

@@ -13,8 +13,7 @@
 
 FFmpeg can be hooked up with a number of external libraries to add support
 for more formats. None of them are used by default, their use has to be
-explicitly requested by passing the appropriate flags to
-@command{./configure}.
+explicitly requested by passing the appropriate flags to @file{./configure}.
 
 @section OpenJPEG
 
@@ -26,8 +25,8 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
 
 @section OpenCORE and VisualOn libraries
 
-Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
-libraries provide encoders for a number of audio codecs.
+Spun off Google Android sources, OpenCore and VisualOn libraries provide
+encoders for a number of audio codecs.
 
 @float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
@@ -63,14 +62,6 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
 
-@subsection Fraunhofer AAC library
-
-FFmpeg can make use of the Fraunhofer AAC 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-libfdk-aac} to configure to enable it.
-
 @section LAME
 
 FFmpeg can make use of the LAME library for MP3 encoding.
@@ -79,14 +70,6 @@ Go to @url{http://lame.sourceforge.net/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libmp3lame} to configure to enable it.
 
-@section TwoLAME
-
-FFmpeg can make use of the TwoLAME library for MP2 encoding.
-
-Go to @url{http://www.twolame.org/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libtwolame} to configure to enable it.
-
 @section libvpx
 
 FFmpeg can make use of the libvpx library for VP8 encoding.
@@ -109,17 +92,6 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section libilbc
-
-iLBC is a narrowband speech codec that has been made freely available
-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/dekkers/libilbc} and follow the instructions for
-installing the library. Then pass @code{--enable-libilbc} to configure to
-enable it.
-
 
 
 @chapter Supported File Formats, Codecs or Features
@@ -143,8 +115,6 @@ library:
 @item American Laser Games MM   @tab   @tab X
     @tab Multimedia format used in games like Mad Dog McCree.
 @item 3GPP AMR                  @tab X @tab X
-@item Amazing Studio Packed Animation File        @tab   @tab X
-    @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 ASF                       @tab X @tab X
@@ -173,8 +143,6 @@ library:
     @tab Multimedia format used by Delphine Software games.
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
-@item Commodore CDXL            @tab   @tab X
-    @tab Amiga CD video format
 @item Core Audio Format         @tab X @tab X
     @tab Apple Core Audio Format
 @item CRC testing format        @tab X @tab
@@ -212,7 +180,7 @@ library:
     @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
          playout servers.
 @item iCEDraw File              @tab   @tab X
-@item ICO                       @tab X @tab X
+@item ICO                       @tab   @tab X
     @tab Microsoft Windows ICO
 @item id Quake II CIN video     @tab   @tab X
 @item id RoQ                    @tab X @tab X
@@ -220,7 +188,6 @@ library:
 @item IEC61937 encapsulation @tab X @tab X
 @item IFF                       @tab   @tab X
     @tab Interchange File Format
-@item iLBC                      @tab X @tab X
 @item Interplay MVE             @tab   @tab X
     @tab Format used in various Interplay computer games.
 @item IV8                       @tab   @tab X
@@ -241,7 +208,6 @@ library:
 @item MAXIS XA                  @tab   @tab X
     @tab Used in Sim City 3000; file extension .xa.
 @item MD Studio                 @tab   @tab X
-@item Metal Gear Solid: The Twin Snakes @tab @tab X
 @item Mobotix .mxg              @tab   @tab X
 @item Monkey's Audio            @tab   @tab X
 @item Motion Pixels MVI         @tab   @tab X
@@ -353,8 +319,6 @@ library:
     @tab Multimedia format used by many games.
 @item SMJPEG                    @tab X @tab X
     @tab Used in certain Loki game ports.
-@item Smush                     @tab   @tab X
-    @tab Multimedia format used in some LucasArts games.
 @item Sony OpenMG (OMA)         @tab X @tab X
     @tab Audio format used in Sony Sonic Stage and Sony Vegas.
 @item Sony PlayStation STR      @tab   @tab X
@@ -369,7 +333,7 @@ library:
 @item True Audio                @tab   @tab X
 @item VC-1 test bitstream       @tab X @tab X
 @item WAV                       @tab X @tab X
-@item WavPack                   @tab X @tab X
+@item WavPack                   @tab   @tab X
 @item WebM                      @tab X @tab X
 @item Windows Televison (WTV)   @tab X @tab X
 @item Wing Commander III movie  @tab   @tab X
@@ -404,8 +368,6 @@ following image formats are supported:
     @tab Microsoft BMP image
 @item DPX          @tab X @tab X
     @tab Digital Picture Exchange
-@item EXR          @tab   @tab X
-    @tab OpenEXR
 @item JPEG         @tab X @tab X
     @tab Progressive JPEG is not supported.
 @item JPEG 2000    @tab X @tab X
@@ -431,14 +393,12 @@ following image formats are supported:
     @tab V.Flash PTX format
 @item SGI          @tab X @tab X
     @tab SGI RGB image format
-@item Sun Rasterfile  @tab X @tab X
+@item Sun Rasterfile  @tab   @tab X
     @tab Sun RAS image format
 @item TIFF         @tab X @tab X
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item XBM  @tab X @tab X
-    @tab X BitMap image format
 @item XWD  @tab X @tab X
     @tab X Window Dump image format
 @end multitable
@@ -458,7 +418,6 @@ following image formats are supported:
 @item 8SVX fibonacci         @tab     @tab  X
 @item A64 multicolor         @tab  X  @tab
     @tab Creates video suitable to be played on a commodore 64 (multicolor mode).
-@item Amazing Studio PAF Video @tab     @tab  X
 @item American Laser Games MM  @tab    @tab X
     @tab Used in games like Mad Dog McCree.
 @item AMV Video              @tab  X  @tab  X
@@ -485,8 +444,6 @@ following image formats are supported:
     @tab fourcc: AVrp
 @item AVS (Audio Video Standard) video  @tab     @tab  X
     @tab Video encoding used by the Creature Shock game.
-@item AYUV                   @tab  X  @tab  X
-    @tab Microsoft uncompressed packed 4:4:4:4
 @item Beam Software VB       @tab     @tab  X
 @item Bethesda VID video     @tab     @tab  X
     @tab Used in some games from Bethesda Softworks.
@@ -501,23 +458,19 @@ following image formats are supported:
     @tab fourcc: CSCD
 @item CD+G                   @tab     @tab  X
     @tab Video codec for CD+G karaoke disks
-@item CDXL                   @tab     @tab  X
-    @tab Amiga CD video codec
 @item Chinese AVS video      @tab  E  @tab  X
     @tab AVS1-P2, JiZhun profile, encoding through external library libxavs
 @item Delphine Software International CIN video  @tab     @tab  X
     @tab Codec used in Delphine Software International games.
 @item Discworld II BMV Video @tab     @tab  X
-@item Canopus Lossless Codec @tab     @tab  X
 @item Cinepak                @tab     @tab  X
 @item Cirrus Logic AccuPak   @tab  X  @tab  X
     @tab fourcc: CLJR
-@item CPiA Video Format      @tab     @tab  X
 @item Creative YUV (CYUV)    @tab     @tab  X
 @item DFA                    @tab     @tab  X
     @tab Codec used in Chronomaster game.
 @item Dirac                  @tab  E  @tab  X
-    @tab supported through external library libschroedinger
+    @tab supported through external libdirac/libschroedinger libraries
 @item Deluxe Paint Animation @tab     @tab  X
 @item DNxHD                  @tab   X @tab  X
     @tab aka SMPTE VC3
@@ -538,13 +491,12 @@ following image formats are supported:
 @item Escape 124             @tab     @tab  X
 @item Escape 130             @tab     @tab  X
 @item FFmpeg video codec #1  @tab  X  @tab  X
-    @tab lossless codec (fourcc: FFV1)
+    @tab experimental lossless codec (fourcc: FFV1)
 @item Flash Screen Video v1  @tab  X  @tab  X
     @tab fourcc: FSV1
 @item Flash Screen Video v2  @tab  X  @tab  X
 @item Flash Video (FLV)      @tab  X  @tab  X
     @tab Sorenson H.263 used in Flash
-@item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
 @item H.261                  @tab  X  @tab  X
 @item H.263 / H.263-1996     @tab  X  @tab  X
@@ -582,18 +534,8 @@ following image formats are supported:
 @item LCL (LossLess Codec Library) MSZH  @tab     @tab  X
 @item LCL (LossLess Codec Library) ZLIB  @tab  E  @tab  E
 @item LOCO                   @tab     @tab  X
-@item LucasArts Smush        @tab     @tab  X
-    @tab Used in LucasArts games.
 @item lossless MJPEG         @tab  X  @tab  X
-@item Microsoft ATC Screen   @tab     @tab  X
-    @tab Also known as Microsoft Screen 3.
-@item Microsoft Expression Encoder Screen  @tab     @tab  X
-    @tab Also known as Microsoft Titanium Screen 2.
 @item Microsoft RLE          @tab     @tab  X
-@item Microsoft Screen 1     @tab     @tab  X
-    @tab Also known as Windows Media Video V7 Screen.
-@item Microsoft Screen 2     @tab     @tab  X
-    @tab Also known as Windows Media Video V9 Screen.
 @item Microsoft Video 1      @tab     @tab  X
 @item Mimic                  @tab     @tab  X
     @tab Used in MSN Messenger Webcam streams.
@@ -622,6 +564,8 @@ following image formats are supported:
     @tab fourcc: VP60,VP61,VP62
 @item VP8                    @tab  E  @tab  X
     @tab fourcc: VP80, encoding supported through external library libvpx
+@item planar RGB             @tab     @tab  X
+    @tab fourcc: 8BPS
 @item Prores                 @tab     @tab  X
     @tab fourcc: apch,apcn,apcs,apco
 @item Q-team QPEG            @tab     @tab  X
@@ -661,16 +605,13 @@ following image formats are supported:
     @tab fourcc: SP5X
 @item TechSmith Screen Capture Codec  @tab     @tab  X
     @tab fourcc: TSCC
-@item TechSmith Screen Capture Codec 2  @tab     @tab  X
-    @tab fourcc: TSC2
 @item Theora                 @tab  E  @tab  X
     @tab encoding supported through external library libtheora
 @item Tiertex Limited SEQ video  @tab     @tab  X
     @tab Codec used in DOS CD-ROM FlashBack game.
-@item Ut Video               @tab  X  @tab  X
+@item Ut Video               @tab     @tab  X
 @item v210 QuickTime uncompressed 4:2:2 10-bit     @tab  X  @tab  X
 @item v308 QuickTime uncompressed 4:4:4            @tab  X  @tab  X
-@item v408 QuickTime uncompressed 4:4:4:4          @tab  X  @tab  X
 @item v410 QuickTime uncompressed 4:4:4 10-bit     @tab  X  @tab  X
 @item VBLE Lossless Codec    @tab     @tab  X
 @item VMware Screen Codec / VMware Video  @tab     @tab  X
@@ -691,7 +632,6 @@ following image formats are supported:
 @item Psygnosis YOP Video    @tab     @tab  X
 @item yuv4                   @tab  X  @tab  X
     @tab libquicktime uncompressed packed 4:2:0
-@item ZeroCodec Lossless Video @tab     @tab  X
 @item ZLIB                   @tab  X  @tab  X
     @tab part of LCL, encoder experimental
 @item Zip Motion Blocks Video  @tab   X @tab  X
@@ -760,7 +700,6 @@ following image formats are supported:
     @tab encoding supported through external library libopencore-amrnb
 @item AMR-WB                 @tab  E  @tab  X
     @tab encoding supported through external library libvo-amrwbenc
-@item Amazing Studio PAF Audio @tab     @tab  X
 @item Apple lossless audio   @tab  X  @tab  X
     @tab QuickTime fourcc 'alac'
 @item Atrac 1                @tab     @tab  X
@@ -794,9 +733,6 @@ following image formats are supported:
     @tab encoding supported through external library libgsm
 @item GSM Microsoft variant  @tab  E  @tab  X
     @tab encoding supported through external library libgsm
-@item IAC (Indeo Audio Coder)  @tab     @tab  X
-@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 MACE (Macintosh Audio Compression/Expansion) 3:1  @tab     @tab  X
 @item MACE (Macintosh Audio Compression/Expansion) 6:1  @tab     @tab  X
@@ -806,7 +742,6 @@ following image formats are supported:
     @tab Only versions 3.97-3.99 are supported.
 @item MP1 (MPEG audio layer 1)  @tab     @tab IX
 @item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
-    @tab libtwolame can be used alternatively for encoding.
 @item MP3 (MPEG audio layer 3)  @tab  E  @tab IX
     @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
 @item MPEG-4 Audio Lossless Coding (ALS)  @tab     @tab  X
@@ -846,7 +781,6 @@ following image formats are supported:
     @tab Real 28800 bit/s codec
 @item RealAudio 3.0 (dnet)   @tab IX  @tab  X
     @tab Real low bitrate AC-3 codec
-@item RealAudio Lossless     @tab     @tab  X
 @item RealAudio SIPR / ACELP.NET @tab     @tab  X
 @item Shorten                @tab     @tab  X
 @item Sierra VMD audio       @tab     @tab  X
@@ -863,15 +797,12 @@ following image formats are supported:
 @item TrueHD                 @tab     @tab  X
     @tab Used in HD-DVD and Blu-Ray discs.
 @item TwinVQ (VQF flavor)    @tab     @tab  X
-@item VIMA                   @tab     @tab  X
-    @tab Used in LucasArts SMUSH animations.
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
 @item WavPack                @tab     @tab  X
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
 @item Windows Media Audio 2  @tab  X  @tab  X
-@item Windows Media Audio Lossless @tab  @tab  X
 @item Windows Media Audio Pro @tab    @tab  X
 @item Windows Media Audio Voice @tab  @tab  X
 @end multitable
@@ -887,19 +818,13 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1 .1 .1 .1
 @item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
-@item SSA/ASS          @tab X @tab X @tab X @tab X
-@item DVB              @tab X @tab X @tab X @tab X
-@item DVD              @tab X @tab X @tab X @tab X
-@item JACOsub          @tab X @tab X @tab   @tab X
-@item MicroDVD         @tab X @tab X @tab   @tab X
-@item PGS              @tab   @tab   @tab   @tab X
-@item RealText         @tab   @tab X @tab   @tab X
-@item SAMI             @tab   @tab X @tab   @tab X
-@item SubRip (SRT)     @tab X @tab X @tab X @tab X
-@item SubViewer        @tab   @tab X @tab   @tab X
-@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
-@item WebVTT           @tab   @tab X @tab   @tab X
-@item XSUB             @tab   @tab   @tab X @tab X
+@item SSA/ASS      @tab X @tab X @tab X @tab X
+@item DVB          @tab X @tab X @tab X @tab X
+@item DVD          @tab X @tab X @tab X @tab X
+@item MicroDVD     @tab X @tab X @tab   @tab
+@item PGS          @tab   @tab   @tab   @tab X
+@item SubRip (SRT) @tab X @tab X @tab X @tab X
+@item XSUB         @tab   @tab   @tab X @tab X
 @end multitable
 
 @code{X} means that the feature is supported.
@@ -908,31 +833,19 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1
 @item Name         @tab Support
+@item Apple HTTP Live Streaming @tab X
 @item file         @tab X
 @item Gopher       @tab X
-@item HLS          @tab X
 @item HTTP         @tab X
-@item HTTPS        @tab X
-@item MMSH         @tab X
-@item MMST         @tab X
+@item MMS          @tab X
 @item pipe         @tab X
-@item RTMP         @tab X
-@item RTMPE        @tab X
-@item RTMPS        @tab X
-@item RTMPT        @tab X
-@item RTMPTE       @tab X
-@item RTMPTS       @tab X
 @item RTP          @tab X
-@item SCTP         @tab X
 @item TCP          @tab X
-@item TLS          @tab X
 @item UDP          @tab X
 @end multitable
 
 @code{X} means that the protocol is supported.
 
-@code{E} means that support is provided through an external library.
-
 
 @section Input/Output Devices
 
@@ -940,17 +853,12 @@ performance on systems without hardware floating point support).
 @item Name              @tab Input  @tab Output
 @item ALSA              @tab X      @tab X
 @item BKTR              @tab X      @tab
-@item caca              @tab        @tab X
 @item DV1394            @tab X      @tab
-@item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab
 @item JACK              @tab X      @tab
-@item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
-@item OpenAL            @tab X
 @item OSS               @tab X      @tab X
 @item Pulseaudio        @tab X      @tab
-@item SDL               @tab        @tab X
+@item Video4Linux       @tab X      @tab
 @item Video4Linux2      @tab X      @tab
 @item VfW capture       @tab X      @tab
 @item X11 grabbing      @tab X      @tab
@@ -962,12 +870,11 @@ performance on systems without hardware floating point support).
 
 @multitable @columnfractions .4 .1 .1
 @item Codec/format      @tab Read   @tab Write
-@item AVI               @tab X      @tab X
 @item DV                @tab X      @tab X
 @item GXF               @tab X      @tab X
-@item MOV               @tab X      @tab X
+@item MOV               @tab X      @tab
 @item MPEG1/2           @tab X      @tab X
-@item MXF               @tab X      @tab X
+@item MXF               @tab        @tab X
 @end multitable
 
 @bye

doc/git-howto.texi

@@ -65,14 +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.
 
-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
-
-@example
-git config --global core.autocrlf false
-@end example
-
 
 @section Updating the source tree to the latest revision
 
@@ -258,32 +250,6 @@ git commit
 @end example
 
 
-@chapter Git configuration
-
-In order to simplify a few workflows, it is advisable to configure both
-your personal Git installation and your local FFmpeg repository.
-
-@section Personal Git installation
-
-Add the following to your @file{~/.gitconfig} to help @command{git send-email}
-and @command{git format-patch} detect renames:
-
-@example
-[diff]
-        renames = copy
-@end example
-
-@section Repository configuration
-
-In order to have @command{git send-email} automatically send patches
-to the ffmpeg-devel mailing list, add the following stanza
-to @file{/path/to/ffmpeg/repository/.git/config}:
-
-@example
-[sendemail]
-        to = ffmpeg-devel@@ffmpeg.org
-@end example
-
 @chapter FFmpeg specific
 
 @section Reverting broken commits
@@ -372,43 +338,6 @@ git checkout -b svn_23456 $SHA1
 
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
-
-@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
-proper order. This list tries to be exhaustive. In case you are just
-pushing a typo in a comment, some of the steps may be unnecessary.
-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 --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 testsuite.
-
-@itemize
-@item @command{make distclean}
-@item @command{/path/to/ffmpeg/configure}
-@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
-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).
-
-Also note that every single commit should pass the test suite, not just
-the result of a series of patches.
-
-Once everything passed, push the changes to your public ffmpeg clone and post a
-merge request to ffmpeg-devel. You can also push them directly but this is not
-recommended.
-
 @chapter Server Issues
 
 Contact the project admins @email{root@@ffmpeg.org} if you have technical

doc/indevs.texi

@@ -59,7 +59,7 @@ BSD video input device.
 
 Windows DirectShow input device.
 
-DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
+DirectShow support is enabled when FFmpeg is built with mingw-w64.
 Currently only audio and video devices are supported.
 
 Multiple devices may be opened as separate inputs, but they may also be
@@ -112,19 +112,6 @@ defaults to 0).
 Set audio device number for devices with same name (starts at 0,
 defaults to 0).
 
-@item pixel_format
-Select pixel format to be used by DirectShow. This may only be set when
-the video codec is not set or set to rawvideo.
-
-@item audio_buffer_size
-Set audio device buffer size in milliseconds (which can directly
-impact latency, depending on the device).
-Defaults to using the audio device's
-default buffer size (typically some multiple of 500ms).
-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}
-
 @end table
 
 @subsection Examples
@@ -192,59 +179,6 @@ ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
-@section iec61883
-
-FireWire DV/HDV input device using libiec61883.
-
-To enable this input device, you need libiec61883, libraw1394 and
-libavc1394 installed on your system. Use the configure option
-@code{--enable-libiec61883} to compile with the device enabled.
-
-The iec61883 capture device supports capturing from a video device
-connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
-FireWire stack (juju). This is the default DV/HDV input method in Linux
-Kernel 2.6.37 and later, since the old FireWire stack was removed.
-
-Specify the FireWire port to be used as input file, or "auto"
-to choose the first port connected.
-
-@subsection Options
-
-@table @option
-
-@item dvtype
-Override autodetection of DV/HDV. This should only be used if auto
-detection does not work, or if usage of a different device type
-should be prohibited. Treating a DV device as HDV (or vice versa) will
-not work and result in undefined behavior.
-The values @option{auto}, @option{dv} and @option{hdv} are supported.
-
-@item dvbuffer
-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.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Grab and show the input of a FireWire DV/HDV device.
-@example
-ffplay -f iec61883 -i auto
-@end example
-
-@item
-Grab and record the input of a FireWire DV/HDV device,
-using a packet buffer of 100000 packets if the source is HDV.
-@example
-ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
-@end example
-
-@end itemize
-
 @section jack
 
 JACK input device.
@@ -570,9 +504,9 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
 @end example
 
-@section video4linux2
+@section video4linux and video4linux2
 
-Video4Linux2 input video device.
+Video4Linux and Video4Linux2 input video devices.
 
 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
@@ -580,28 +514,36 @@ systems tend to automatically create such nodes when the device
 kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.
 
-Video4Linux2 devices usually support a limited set of
+Video4Linux and Video4Linux2 devices only support a limited set of
 @var{width}x@var{height} sizes and framerates. You can check which are
-supported using @command{-list_formats all} for Video4Linux2 devices.
+supported for example with the command @command{dov4l} for Video4Linux
+devices and using @command{-list_formats all} for Video4Linux2 devices.
 
-Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
+If the size for the device is set to 0x0, the input device will
+try to auto-detect the size to use.
+Only for the video4linux2 device, if the frame rate is set to 0/0 the
+input device will use the frame rate value already set in the driver.
 
-The time base for the timestamps is 1 microsecond. Depending on the kernel
-version and configuration, the timestamps may be derived from the real time
-clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
-boot time, unaffected by NTP or manual changes to the clock). The
-@option{-timestamps abs} or @option{-ts abs} option can be used to force
-conversion into the real time clock.
+Video4Linux support is deprecated since Linux 2.6.30, and will be
+dropped in later versions.
 
 Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
 option), it will always be used.
-@example
-# Grab and show the input of a video4linux2 device.
-ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
 
-# Grab and record the input of a video4linux2 device, leave the
-framerate and size as previously set.
-ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
+Follow some usage examples of the video4linux devices with the ff*
+tools.
+@example
+# Grab and show the input of a video4linux device, frame rate is set
+# to the default of 25/1.
+ffplay -s 320x240 -f video4linux /dev/video0
+
+# Grab and show the input of a video4linux2 device, auto-adjust size.
+ffplay -f video4linux2 /dev/video0
+
+# Grab and record the input of a video4linux2 device, auto-adjust size,
+# frame rate value defaults to 0/0 so it is read from the video4linux2
+# driver.
+ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
 @end example
 
 "v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
@@ -643,23 +585,17 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").
 For example to grab from @file{:0.0} using @command{ffmpeg}:
 @example
 ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
-@end example
 
-Grab at position @code{10,20}:
-@example
+# Grab at position 10,20.
 ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
 @end example
 
-@subsection Options
+@subsection @var{follow_mouse} AVOption
 
-@table @option
-@item draw_mouse
-Specify whether to draw the mouse pointer. A value of @code{0} specify
-not to draw the pointer. Default value is @code{1}.
-
-@item follow_mouse
-Make the grabbed area follow the mouse. The argument can be
-@code{centered} or a number of pixels @var{PIXELS}.
+The syntax is:
+@example
+-follow_mouse centered|@var{PIXELS}
+@end example
 
 When it is specified with "centered", the grabbing region follows the mouse
 pointer and keeps the pointer at the center of region; otherwise, the region
@@ -669,36 +605,28 @@ zero) to the edge of region.
 For example:
 @example
 ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
-@end example
 
-To follow only when the mouse pointer reaches within 100 pixels to edge:
-@example
+# Follows only when the mouse pointer reaches within 100 pixels to edge
 ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
 @end example
 
-@item framerate
-Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a framerate of @code{30000/1001}.
+@subsection @var{show_region} AVOption
 
-@item show_region
-Show grabbed region on screen.
+The syntax is:
+@example
+-show_region 1
+@end example
 
-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.
+If @var{show_region} AVOption is specified with @var{1}, then the grabbing
+region will be indicated on screen. With this option, it's easy to know what is
+being grabbed if only a portion of the screen is grabbed.
 
 For example:
 @example
 ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
-@end example
 
-With @var{follow_mouse}:
-@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
+# With follow_mouse
+ffmpeg -f x11grab -follow_mouse centered -show_region 1  -r 25 -s cif -i :0.0 out.mpg
 @end example
 
-@item video_size
-Set the video frame size. Default value is @code{vga}.
-@end table
-
 @c man end INPUT DEVICES

doc/issue_tracker.txt

@@ -24,7 +24,7 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://ffmpeg.org/trac/ffmpeg
+http(s)://trac.ffmpeg.org
 
 Type:
 -----

doc/libavfilter.texi

@@ -0,0 +1,92 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle Libavfilter Documentation
+@titlepage
+@center @titlefont{Libavfilter Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Introduction
+
+Libavfilter is the filtering API of FFmpeg. It is the substitute of the
+now deprecated 'vhooks' and started as a Google Summer of Code project.
+
+Audio filtering integration into the main FFmpeg repository is a work in
+progress, so audio API and ABI should not be considered stable yet.
+
+@chapter Tutorial
+
+In libavfilter, it is possible for filters to have multiple inputs and
+multiple outputs.
+To illustrate the sorts of things that are possible, we can
+use a complex filter graph. For example, the following one:
+
+@example
+input --> split --> fifo -----------------------> overlay --> output
+            |                                        ^
+            |                                        |
+            +------> fifo --> crop --> vflip --------+
+@end example
+
+splits the stream in two streams, sends one stream through the crop filter
+and the vflip filter before merging it back with the other stream by
+overlaying it on top. You can use the following command to achieve this:
+
+@example
+ffmpeg -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output
+@end example
+
+The result will be that in output the top half of the video is mirrored
+onto the bottom half.
+
+Video filters are loaded using the @var{-vf} option passed to
+@command{ffmpeg} or to @command{ffplay}. Filters in the same linear
+chain are separated by commas. In our example, @var{split, fifo,
+overlay} are in one linear chain, and @var{fifo, crop, vflip} are in
+another. The points where the linear chains join are labeled by names
+enclosed in square brackets. In our example, that is @var{[T1]} and
+@var{[T2]}. The magic labels @var{[in]} and @var{[out]} are the points
+where video is input and output.
+
+Some filters take in input a list of parameters: they are specified
+after the filter name and an equal sign, and are separated each other
+by a semicolon.
+
+There exist so-called @var{source filters} that do not have a video
+input, and we expect in the future some @var{sink filters} that will
+not have video output.
+
+@chapter graph2dot
+
+The @file{graph2dot} program included in the FFmpeg @file{tools}
+directory can be used to parse a filter graph description and issue a
+corresponding textual representation in the dot language.
+
+Invoke the command:
+@example
+graph2dot -h
+@end example
+
+to see how to use @file{graph2dot}.
+
+You can then pass the dot description to the @file{dot} program (from
+the graphviz suite of programs) and obtain a graphical representation
+of the filter graph.
+
+For example the sequence of commands:
+@example
+echo @var{GRAPH_DESCRIPTION} | \
+tools/graph2dot -o graph.tmp && \
+dot -Tpng graph.tmp -o graph.png && \
+display graph.png
+@end example
+
+can be used to create and display an image representing the graph
+described by the @var{GRAPH_DESCRIPTION} string.
+
+@include filters.texi
+
+@bye

doc/mips.txt

@@ -1,65 +0,0 @@
-MIPS optimizations info
-===============================================
-
-MIPS optimizations of codecs are targeting MIPS 74k family of
-CPUs. Some of these optimizations are relying more on properties of
-this architecture and some are relying less (and can be used on most
-MIPS architectures without degradation in performance).
-
-Along with FFMPEG copyright notice, there is MIPS copyright notice in
-all the files that are created by people from MIPS Technologies.
-
-Example of copyright notice:
-===============================================
-/*
- * Copyright (c) 2012
- *      MIPS Technologies, Inc., California.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
- * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
- *    contributors may be used to endorse or promote products derived from
- *    this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
- * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
- * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
- * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
- * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
- * SUCH DAMAGE.
- *
- * Author:  Author Name (author_name@@mips.com)
- */
-
-Files that have MIPS copyright notice in them:
-===============================================
-* libavutil/mips/
-      libm_mips.h
-* libavcodec/mips/
-      acelp_filters_mips.c
-      acelp_vectors_mips.c
-      amrwbdec_mips.c
-      amrwbdec_mips.h
-      celp_filters_mips.c
-      celp_math_mips.c
-      compute_antialias_fixed.h
-      compute_antialias_float.h
-      lsp_mips.h
-      dsputil_mips.c
-      fft_mips.c
-      fft_table.h
-      fft_init_table.c
-      fmtconvert_mips.c
-      mpegaudiodsp_mips_fixed.c
-      mpegaudiodsp_mips_float.c

doc/muxers.texi

@@ -56,37 +56,31 @@ See also the @ref{framecrc} muxer.
 @anchor{framecrc}
 @section framecrc
 
-Per-packet CRC (Cyclic Redundancy Check) testing format.
+Per-frame CRC (Cyclic Redundancy Check) testing format.
 
-This muxer computes and prints the Adler-32 CRC for each audio
-and video packet. By default audio frames are converted to signed
+This muxer computes and prints the Adler-32 CRC for each decoded audio
+and video frame. By default audio frames are converted to signed
 16-bit raw audio and video frames to raw video before computing the
 CRC.
 
 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}, 0x@var{CRC}
-@end example
+frame of the form: @var{stream_index}, @var{frame_dts},
+@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal
+number 0-padded to 8 digits containing the CRC of the decoded frame.
 
-@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
-CRC of the packet.
-
-For example to compute the CRC of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.crc}:
+For example to compute the CRC of each decoded frame in the input, and
+store it in the file @file{out.crc}:
 @example
 ffmpeg -i INPUT -f framecrc out.crc
 @end example
 
-To print the information to stdout, use the command:
+You can print the CRC of each decoded frame to stdout with the command:
 @example
 ffmpeg -i INPUT -f framecrc -
 @end example
 
-With @command{ffmpeg}, you can select the output format to which the
-audio and video frames are encoded before computing the CRC for each
-packet by specifying the audio and video codec. For example, to
+You can select the output format of each frame with @command{ffmpeg} by
+specifying the audio and video codec and format. For example, to
 compute the CRC of each decoded input audio frame converted to PCM
 unsigned 8-bit and of each decoded input video frame converted to
 MPEG-2 video, use the command:
@@ -96,72 +90,6 @@ ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
 
 See also the @ref{crc} muxer.
 
-@anchor{framemd5}
-@section framemd5
-
-Per-packet MD5 testing format.
-
-This muxer computes and prints the MD5 hash for each audio
-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.
-
-For example to compute the MD5 of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f framemd5 out.md5
-@end example
-
-To print the information to stdout, use the command:
-@example
-ffmpeg -i INPUT -f framemd5 -
-@end example
-
-See also the @ref{md5} muxer.
-
-@anchor{ico}
-@section ico
-
-ICO file muxer.
-
-Microsoft's icon file format (ICO) has some strict limitations that should be noted:
-
-@itemize
-@item
-Size cannot exceed 256 pixels in any dimension
-
-@item
-Only BMP and PNG images can be stored
-
-@item
-If a BMP image is used, it must be one of the following pixel formats:
-@example
-BMP Bit Depth      FFmpeg Pixel Format
-1bit               pal8
-4bit               pal8
-8bit               pal8
-16bit              rgb555le
-24bit              bgr24
-32bit              bgra
-@end example
-
-@item
-If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
-
-@item
-If a PNG image is used, it must use the rgba pixel format
-@end itemize
-
 @anchor{image2}
 @section image2
 
@@ -220,104 +148,18 @@ each of the YUV420P components. To read or write this image file format,
 specify the name of the '.Y' file. The muxer will automatically open the
 '.U' and '.V' files as required.
 
-@anchor{md5}
-@section md5
+@section mov
 
-MD5 testing format.
+MOV / MP4 muxer
 
-This muxer computes and prints the MD5 hash of all the input audio
-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.
-
-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.
-
-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
-@end example
-
-You can print the MD5 to stdout with the command:
-@example
-ffmpeg -i INPUT -f md5 -
-@end example
-
-See also the @ref{framemd5} muxer.
-
-@section MOV/MP4/ISMV
-
-The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
-file has all the metadata about all packets stored in one location
-(written at the end of the file, it can be moved to the start for
-better playback using the @command{qt-faststart} tool). A fragmented
-file consists of a number of fragments, where packets and metadata
-about these packets are stored together. Writing a fragmented
-file has the advantage that the file is decodable even if the
-writing is interrupted (while a normal MOV/MP4 is undecodable if
-it is not properly finished), and it requires less memory when writing
-very long files (since writing normal MOV/MP4 files stores info about
-every single packet in memory until the file is closed). The downside
-is that it is less compatible with other applications.
-
-Fragmentation is enabled by setting one of the AVOptions that define
-how to cut the file into fragments:
+The muxer options are:
 
 @table @option
 @item -moov_size @var{bytes}
 Reserves space for the moov atom at the beginning of the file instead of placing the
 moov atom at the end. If the space reserved is insufficient, muxing will fail.
-@item -movflags frag_keyframe
-Start a new fragment at each video keyframe.
-@item -frag_duration @var{duration}
-Create fragments that are @var{duration} microseconds long.
-@item -frag_size @var{size}
-Create fragments that contain up to @var{size} bytes of payload data.
-@item -movflags frag_custom
-Allow the caller to manually choose when to cut fragments, by
-calling @code{av_write_frame(ctx, NULL)} to write a fragment with
-the packets written so far. (This is only useful with other
-applications integrating libavformat, not from @command{ffmpeg}.)
-@item -min_frag_duration @var{duration}
-Don't create fragments that are shorter than @var{duration} microseconds long.
 @end table
 
-If more than one condition is specified, fragments are cut when
-one of the specified conditions is fulfilled. The exception to this is
-@code{-min_frag_duration}, which has to be fulfilled for any of the other
-conditions to apply.
-
-Additionally, the way the output file is written can be adjusted
-through a few other options:
-
-@table @option
-@item -movflags empty_moov
-Write an initial moov atom directly at the start of the file, without
-describing any samples in it. Generally, an mdat/moov pair is written
-at the start of the file, as a normal MOV/MP4 file, containing only
-a short portion of the file. With this option set, there is no initial
-mdat atom, and the moov atom only describes the tracks but has
-a zero duration.
-
-Files written with this option set do not work in QuickTime.
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags separate_moof
-Write a separate moof (movie fragment) atom for each track. Normally,
-packets for all tracks are written in a moof atom (which is slightly
-more efficient), but with this option set, the muxer writes one moof/mdat
-pair for each track, making it easier to separate tracks.
-
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@end table
-
-Smooth Streaming content can be pushed in real time to a publishing
-point on IIS with this muxer. Example:
-@example
-ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
-@end example
-
 @section mpegts
 
 MPEG transport stream muxer.
@@ -444,7 +286,7 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
 @end example
 
-@section segment, stream_segment, ssegment
+@section segment
 
 Basic stream segmenter.
 
@@ -452,200 +294,27 @@ The segmenter muxer outputs streams to a number of separate files of nearly
 fixed duration. Output filename pattern can be set in a fashion similar to
 @ref{image2}.
 
-@code{stream_segment} is a variant of the muxer used to write to
-streaming output formats, i.e. which do not require global headers,
-and is recommended for outputting e.g. to MPEG transport stream segments.
-@code{ssegment} is a shorter alias for @code{stream_segment}.
-
 Every segment starts with a video keyframe, if a video stream is present.
-Note that if you want accurate splitting for a video file, you need to
-make the input key frames correspond to the exact splitting times
-expected by the segmenter, or the segment muxer will start the new
-segment with the key frame found next after the specified start
-time.
-
 The segment muxer works best with a single constant frame rate video.
 
-Optionally it can generate a list of the created segments, by setting
-the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option.
-
-The segment muxer supports the following options:
+Optionally it can generate a flat list of the created segments, one segment
+per line.
 
 @table @option
 @item segment_format @var{format}
 Override the inner container format, by default it is guessed by the filename
 extension.
+@item segment_time @var{t}
+Set segment duration to @var{t} seconds.
 @item segment_list @var{name}
-Generate also a listfile named @var{name}. If not specified no
-listfile is generated.
-@item segment_list_flags @var{flags}
-Set flags affecting the segment list generation.
-
-It currently supports the following flags:
-@table @var
-@item cache
-Allow caching (only affects M3U8 list files).
-
-@item live
-Allow live-friendly file generation.
-
-This currently only affects M3U8 lists. In particular, write a fake
-EXT-X-TARGETDURATION duration field at the top of the file, based on
-the specified @var{segment_time}.
-@end table
-
-Default value is @code{cache}.
-
+Generate also a listfile named @var{name}.
 @item segment_list_size @var{size}
-Overwrite the listfile once it reaches @var{size} entries. If 0
-the listfile is never overwritten. Default value is 0.
-@item segment_list type @var{type}
-Specify the format for the segment list file.
-
-The following values are recognized:
-@table @option
-@item flat
-Generate a flat list for the created segments, one segment per line.
-
-@item csv, ext
-Generate a list for the created segments, one segment per line,
-each line matching the format (comma-separated values):
-@example
-@var{segment_filename},@var{segment_start_time},@var{segment_end_time}
-@end example
-
-@var{segment_filename} is the name of the output file generated by the
-muxer according to the provided pattern. CSV escaping (according to
-RFC4180) is applied if required.
-
-@var{segment_start_time} and @var{segment_end_time} specify
-the segment start and end time expressed in seconds.
-
-A list file with the suffix @code{".csv"} or @code{".ext"} will
-auto-select this format.
-
-@code{ext} is deprecated in favor or @code{csv}.
-
-@item m3u8
-Generate an extended M3U8 file, version 4, compliant with
-@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt}.
-
-A list file with the suffix @code{".m3u8"} will auto-select this format.
+Overwrite the listfile once it reaches @var{size} entries.
 @end table
 
-If not specified the type is guessed from the list file name suffix.
-@item segment_time @var{time}
-Set segment duration to @var{time}. Default value is "2".
-@item segment_time_delta @var{delta}
-Specify the accuracy time when selecting the start time for a
-segment. Default value is "0".
-
-When delta is specified a key-frame will start a new segment if its
-PTS satisfies the relation:
 @example
-PTS >= start_time - time_delta
+ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
 @end example
 
-This option is useful when splitting video content, which is always
-split at GOP boundaries, in case a key frame is found just before the
-specified split time.
-
-In particular may be used in combination with the @file{ffmpeg} option
-@var{force_key_frames}. The key frame times specified by
-@var{force_key_frames} may not be set accurately because of rounding
-issues, with the consequence that a key frame time may result set just
-before the specified time. For constant frame rate videos a value of
-1/2*@var{frame_rate} should address the worst case mismatch between
-the specified time and the time set by @var{force_key_frames}.
-
-@item segment_times @var{times}
-Specify a list of split points. @var{times} contains a list of comma
-separated duration specifications, in increasing order.
-@item segment_wrap @var{limit}
-Wrap around segment index once it reaches @var{limit}.
-@end table
-
-Some examples follow.
-
-@itemize
-@item
-To remux the content of file @file{in.mkv} to a list of segments
-@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
-generated segments to @file{out.list}:
-@example
-ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut
-@end example
-
-@item
-As the example above, but segment the input file according to the split
-points specified by the @var{segment_times} option:
-@example
-ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
-@end example
-
-@item
-As the example above, but use the @code{ffmpeg} @var{force_key_frames}
-option to force key frames in the input at the specified location, together
-with the segment option @var{segment_time_delta} to account for
-possible roundings operated when setting key frame times.
-@example
-ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -vcodec mpeg4 -acodec pcm_s16le -map 0 \
--f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut
-@end example
-In order to force key frames on the input file, transcoding is
-required.
-
-@item
-To convert the @file{in.mkv} to TS segments using the @code{libx264}
-and @code{libfaac} encoders:
-@example
-ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
-@end example
-
-@item
-Segment the input file, and create an M3U8 live playlist (can be used
-as live HLS source):
-@example
-ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
--segment_list_flags +live -segment_time 10 out%03d.mkv
-@end example
-@end itemize
-
-@section mp3
-
-The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
-optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
-not written by default, but may be enabled with the @code{write_id3v1} option.
-
-For seekable output the muxer also writes a Xing frame at the beginning, which
-contains the number of frames in the file. It is useful for computing duration
-of VBR files.
-
-The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
-are supplied to the muxer in form of a video stream with a single packet. There
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
-@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
-
-Note that the APIC frames must be written at the beginning, so the muxer will
-buffer the audio frames until it gets all the pictures. It is therefore advised
-to provide the pictures as soon as possible to avoid excessive buffering.
-
-Examples:
-
-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
-@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
-@end example
-
-To attach a picture to an mp3 file select both the audio and the picture stream
-with @code{map}:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
--metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
-@end example
 
 @c man end MUXERS

doc/outdevs.texi

@@ -22,88 +22,6 @@ A description of the currently available output devices follows.
 
 ALSA (Advanced Linux Sound Architecture) output device.
 
-@section caca
-
-CACA output device.
-
-This output devices allows to show a video stream in CACA window.
-Only one CACA window is allowed per application, so you can
-have only one instance of this output device in an application.
-
-To enable this output device you need to configure FFmpeg with
-@code{--enable-libcaca}.
-libcaca is a graphics library that outputs text instead of pixels.
-
-For more information about libcaca, check:
-@url{http://caca.zoy.org/wiki/libcaca}
-
-@subsection Options
-
-@table @option
-
-@item window_title
-Set the CACA window title, if not specified default to the filename
-specified for the output device.
-
-@item window_size
-Set the CACA window size, can be a string of the form
-@var{width}x@var{height} or a video size abbreviation.
-If not specified it defaults to the size of the input video.
-
-@item driver
-Set display driver.
-
-@item algorithm
-Set dithering algorithm. Dithering is necessary
-because the picture being rendered has usually far more colours than
-the available palette.
-The accepted values are listed with @code{-list_dither algorithms}.
-
-@item antialias
-Set antialias method. Antialiasing smoothens the rendered
-image and avoids the commonly seen staircase effect.
-The accepted values are listed with @code{-list_dither antialiases}.
-
-@item charset
-Set which characters are going to be used when rendering text.
-The accepted values are listed with @code{-list_dither charsets}.
-
-@item color
-Set color to be used when rendering text.
-The accepted values are listed with @code{-list_dither colors}.
-
-@item list_drivers
-If set to @option{true}, print a list of available drivers and exit.
-
-@item list_dither
-List available dither options related to the argument.
-The argument must be one of @code{algorithms}, @code{antialiases},
-@code{charsets}, @code{colors}.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-The following command shows the @command{ffmpeg} output is an
-CACA window, forcing its size to 80x25:
-@example
-ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
-@end example
-
-@item
-Show the list of available drivers and exit:
-@example
-ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
-@end example
-
-@item
-Show the list of available dither colors and exit:
-@example
-ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
-@end example
-@end itemize
-
 @section oss
 
 OSS (Open Sound System) output device.
@@ -137,8 +55,7 @@ to the same value of @var{window_title}.
 @item window_size
 Set the SDL window size, can be a string of the form
 @var{width}x@var{height} or a video size abbreviation.
-If not specified it defaults to the size of the input video,
-downscaled according to the aspect ratio.
+If not specified it defaults to the size of the input video.
 @end table
 
 @subsection Examples

doc/platform.texi

@@ -27,11 +27,11 @@ to configure.
 @section BSD
 
 BSD make will not build FFmpeg, you need to install and use GNU Make
-(@command{gmake}).
+(@file{gmake}).
 
 @section (Open)Solaris
 
-GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
+GNU Make is required to build FFmpeg, so you have to invoke (@file{gmake}),
 standard Solaris Make will not work. When building with a non-c99 front-end
 (gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
 or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
@@ -51,14 +51,15 @@ The toolchain provided with Xcode is sufficient to build the basic
 unacelerated code.
 
 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
-@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
-assembler functions. Just download the Perl script and put it somewhere
+@url{https://github.com/FFmpeg/gas-preprocessor} or
+@url{https://github.com/yuvi/gas-preprocessor} to build the optimized
+assembler functions. Put the Perl script somewhere
 in your PATH, FFmpeg's configure will pick it up automatically.
 
 Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
 optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
 @uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
-@uref{http://mxcl.github.com/homebrew/, Homebrew}
+@uref{https://mxcl.github.com/homebrew/, Homebrew}
 or @uref{http://www.macports.org, MacPorts} can easily provide it.
 
 
@@ -89,7 +90,7 @@ section and the FAQ.
 
 FFmpeg does not build out-of-the-box with the packages the automated MinGW
 installer provides. It also requires coreutils to be installed and many other
-packages updated to the latest version. The minimum versions for some packages
+packages updated to the latest version. The minimum version for some packages
 are listed below:
 
 @itemize
@@ -109,11 +110,14 @@ Notes:
 @item Building natively using MSYS 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
+noticeable when running make for a second time (for example in
 @code{make install}).
 
 @item In order to compile FFplay, you must have the MinGW development library
-of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
+of @uref{http://www.libsdl.org/, SDL}.
+Edit the @file{bin/sdl-config} script so that it points to the correct prefix
+where SDL was installed. Verify that @file{sdl-config} can be launched from
+the MSYS command line.
 
 @item By using @code{./configure --enable-shared} when configuring FFmpeg,
 you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
@@ -137,7 +141,7 @@ you might have to modify the procedures slightly.
 
 @subsection Using static libraries
 
-Assuming you have just built and installed FFmpeg in @file{/usr/local}:
+Assuming you have just built and installed FFmpeg in @file{/usr/local}.
 
 @enumerate
 
@@ -272,7 +276,7 @@ To create import libraries that work with the @code{/OPT:REF} option
 
 @enumerate
 
-@item Open @emph{Visual Studio 2005 Command Prompt}.
+@item Open @file{Visual Studio 2005 Command Prompt}.
 
 Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
 which sets up the environment variables for the Visual C++ tools
@@ -282,14 +286,17 @@ which sets up the environment variables for the Visual C++ tools
 @item Enter the @file{bin} directory where the created LIB and DLL files
 are stored.
 
-@item Generate new import libraries with @command{lib.exe}:
+@item Generate new import libraries with @file{lib.exe}:
 
 @example
-lib /machine:i386 /def:..\lib\foo-version.def  /out:foo.lib
+lib /machine:i386 /def:..\lib\avcodec-53.def  /out:avcodec.lib
+lib /machine:i386 /def:..\lib\avdevice-53.def /out:avdevice.lib
+lib /machine:i386 /def:..\lib\avfilter-2.def  /out:avfilter.lib
+lib /machine:i386 /def:..\lib\avformat-53.def /out:avformat.lib
+lib /machine:i386 /def:..\lib\avutil-51.def   /out:avutil.lib
+lib /machine:i386 /def:..\lib\swscale-2.def   /out:swscale.lib
 @end example
 
-Replace @code{foo-version} and @code{foo} with the respective library names.
-
 @end enumerate
 
 @anchor{Cross compilation for Windows with Linux}
@@ -318,9 +325,24 @@ following "Devel" ones:
 binutils, gcc4-core, make, git, mingw-runtime, texi2html
 @end example
 
-In order to run FATE you will also need the following "Utils" packages:
+And the following "Utils" one:
 @example
-bc, diffutils
+diffutils
+@end example
+
+Then run
+
+@example
+./configure
+@end example
+
+to make a static build.
+
+The current @code{gcc4-core} package is buggy and needs this flag to build
+shared libraries:
+
+@example
+./configure --enable-shared --disable-static --extra-cflags=-fno-reorder-functions
 @end example
 
 If you want to build FFmpeg with additional libraries, download Cygwin
@@ -333,12 +355,16 @@ These library packages are only available from
 @uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
 
 @example
-yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
-libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
+yasm, libSDL-devel, libdirac-devel, libfaac-devel, libaacplus-devel, libgsm-devel,
+libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel,
+libxvidcore-devel
 @end example
 
-The recommendation for x264 is to build it from source, as it evolves too
-quickly for Cygwin Ports to be up to date.
+The recommendation for libnut and x264 is to build them from source by
+yourself, as they evolve too quickly for Cygwin Ports to be up to date.
+
+Cygwin 1.7.x has IPv6 support. You can add IPv6 to Cygwin 1.5.x by means
+of the @code{libgetaddrinfo-devel} package, available at Cygwin Ports.
 
 @section Crosscompilation for Windows under Cygwin
 

doc/print_options.c

@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2012 Anton Khirnov
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * generate texinfo manpages for avoptions
- */
-
-#include <stddef.h>
-#include <string.h>
-#include <float.h>
-
-#include "libavformat/avformat.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/opt.h"
-
-static void print_usage(void)
-{
-    fprintf(stderr, "Usage: enum_options type\n"
-            "type: format codec\n");
-    exit(1);
-}
-
-static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
-{
-    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
-    switch (o->type) {
-    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
-    case AV_OPT_TYPE_STRING:   printf("string");             break;
-    case AV_OPT_TYPE_INT:
-    case AV_OPT_TYPE_INT64:    printf("integer");            break;
-    case AV_OPT_TYPE_FLOAT:
-    case AV_OPT_TYPE_DOUBLE:   printf("float");              break;
-    case AV_OPT_TYPE_RATIONAL: printf("rational number");    break;
-    case AV_OPT_TYPE_FLAGS:    printf("flags");              break;
-    default:                   printf("value");              break;
-    }
-    printf("} (@emph{");
-
-    if (o->flags & AV_OPT_FLAG_DECODING_PARAM) {
-        printf("input");
-        if (o->flags & AV_OPT_FLAG_ENCODING_PARAM)
-            printf("/");
-    }
-    if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output");
-    if (o->flags & AV_OPT_FLAG_AUDIO_PARAM)    printf(",audio");
-    if (o->flags & AV_OPT_FLAG_VIDEO_PARAM)    printf(",video");
-    if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles");
-
-    printf("})\n");
-    if (o->help)
-        printf("%s\n", o->help);
-
-    if (o->unit) {
-        const AVOption *u;
-        printf("\nPossible values:\n@table @samp\n");
-
-        for (u = opts; u->name; u++) {
-            if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit))
-                printf("@item %s\n%s\n", u->name, u->help ? u->help : "");
-        }
-        printf("@end table\n");
-    }
-}
-
-static void show_opts(const AVOption *opts, int per_stream)
-{
-    const AVOption *o;
-
-    printf("@table @option\n");
-    for (o = opts; o->name; o++) {
-        if (o->type != AV_OPT_TYPE_CONST)
-            print_option(opts, o, per_stream);
-    }
-    printf("@end table\n");
-}
-
-static void show_format_opts(void)
-{
-#include "libavformat/options_table.h"
-
-    printf("@section Format AVOptions\n");
-    show_opts(options, 0);
-}
-
-static void show_codec_opts(void)
-{
-#include "libavcodec/options_table.h"
-
-    printf("@section Codec AVOptions\n");
-    show_opts(options, 1);
-}
-
-int main(int argc, char **argv)
-{
-    if (argc < 2)
-        print_usage();
-
-    printf("@c DO NOT EDIT THIS FILE!\n"
-           "@c It was generated by print_options.\n\n");
-    if (!strcmp(argv[1], "format"))
-        show_format_opts();
-    else if (!strcmp(argv[1], "codec"))
-        show_codec_opts();
-    else
-        print_usage();
-
-    return 0;
-}

doc/protocols.texi

@@ -19,34 +19,20 @@ supported protocols.
 
 A description of the currently available protocols follows.
 
-@section bluray
+@section applehttp
 
-Read BluRay playlist.
+Read Apple HTTP Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote HTTP resources or local files, accessed using the standard
+file protocol.
+HTTP is default, specific protocol can be declared by specifying
+"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
+is either "file" or "http".
 
-The accepted options are:
-@table @option
-
-@item angle
-BluRay angle
-
-@item chapter
-Start chapter (1...N)
-
-@item playlist
-Playlist to read (BDMV/PLAYLIST/?????.mpls)
-
-@end table
-
-Examples:
-
-Read longest playlist from BluRay mounted to /mnt/bluray:
 @example
-bluray:/mnt/bluray
-@end example
-
-Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
-@example
--playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
+applehttp://host/path/to/remote/resource.m3u8
+applehttp+http://host/path/to/remote/resource.m3u8
+applehttp+file://path/to/local/resource.m3u8
 @end example
 
 @section concat
@@ -95,26 +81,6 @@ specified with the name "FILE.mpeg" is interpreted as the URL
 
 Gopher protocol.
 
-@section hls
-
-Read Apple HTTP Live Streaming compliant segmented stream as
-a uniform one. The M3U8 playlists describing the segments can be
-remote HTTP resources or local files, accessed using the standard
-file protocol.
-The nested protocol is declared by specifying
-"+@var{proto}" after the hls URI scheme name, where @var{proto}
-is either "file" or "http".
-
-@example
-hls+http://host/path/to/remote/resource.m3u8
-hls+file://path/to/local/resource.m3u8
-@end example
-
-Using this protocol is discouraged - the hls demuxer should work
-just as well (if not, please report the issues) and is more complete.
-To use the hls demuxer instead, simply use the direct URLs to the
-m3u8 files.
-
 @section http
 
 HTTP (Hyper Text Transfer Protocol).
@@ -194,7 +160,7 @@ content across a TCP/IP network.
 
 The required syntax is:
 @example
-rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
+rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
 @end example
 
 The accepted parameters are:
@@ -209,88 +175,11 @@ The number of the TCP port to use (by default is 1935).
 @item app
 It is the name of the application to access. It usually corresponds to
 the path where the application is installed on the RTMP server
-(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
-the value parsed from the URI through the @code{rtmp_app} option, too.
+(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
 
 @item playpath
 It is the path or name of the resource to play with reference to the
-application specified in @var{app}, may be prefixed by "mp4:". You
-can override the value parsed from the URI through the @code{rtmp_playpath}
-option, too.
-
-@item listen
-Act as a server, listening for an incoming connection.
-
-@item timeout
-Maximum time to wait for the incoming connection. Implies listen.
-@end table
-
-Additionally, the following parameters can be set via command line options
-(or in code via @code{AVOption}s):
-@table @option
-
-@item rtmp_app
-Name of application to connect on the RTMP server. This option
-overrides the parameter specified in the URI.
-
-@item rtmp_buffer
-Set the client buffer time in milliseconds. The default is 3000.
-
-@item rtmp_conn
-Extra arbitrary AMF connection parameters, parsed from a string,
-e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
-Each value is prefixed by a single character denoting the type,
-B for Boolean, N for number, S for string, O for object, or Z for null,
-followed by a colon. For Booleans the data must be either 0 or 1 for
-FALSE or TRUE, respectively.  Likewise for Objects the data must be 0 or
-1 to end or begin an object, respectively. Data items in subobjects may
-be named, by prefixing the type with 'N' and specifying the name before
-the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
-times to construct arbitrary AMF sequences.
-
-@item rtmp_flashver
-Version of the Flash plugin used to run the SWF player. The default
-is LNX 9,0,124,2.
-
-@item rtmp_flush_interval
-Number of packets flushed in the same request (RTMPT only). The default
-is 10.
-
-@item rtmp_live
-Specify that the media is a live stream. No resuming or seeking in
-live streams is possible. The default value is @code{any}, which means the
-subscriber first tries to play the live stream specified in the
-playpath. If a live stream of that name is not found, it plays the
-recorded stream. The other possible values are @code{live} and
-@code{recorded}.
-
-@item rtmp_pageurl
-URL of the web page in which the media was embedded. By default no
-value will be sent.
-
-@item rtmp_playpath
-Stream identifier to play or to publish. This option overrides the
-parameter specified in the URI.
-
-@item rtmp_subscribe
-Name of live stream to subscribe to. By default no value will be sent.
-It is only sent if the option is specified or if rtmp_live
-is set to live.
-
-@item rtmp_swfhash
-SHA256 hash of the decompressed SWF file (32 bytes).
-
-@item rtmp_swfsize
-Size of the decompressed SWF file, required for SWFVerification.
-
-@item rtmp_swfurl
-URL of the SWF player for the media. By default no value will be sent.
-
-@item rtmp_swfverify
-URL to player swf file, compute hash/size automatically.
-
-@item rtmp_tcurl
-URL of the target stream. Defaults to proto://host[:port]/app.
+application specified in @var{app}, may be prefixed by "mp4:".
 
 @end table
 
@@ -300,46 +189,6 @@ For example to read with @command{ffplay} a multimedia resource named
 ffplay rtmp://myserver/vod/sample
 @end example
 
-@section rtmpe
-
-Encrypted Real-Time Messaging Protocol.
-
-The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
-streaming multimedia content within standard cryptographic primitives,
-consisting of Diffie-Hellman key exchange and HMACSHA256, generating
-a pair of RC4 keys.
-
-@section rtmps
-
-Real-Time Messaging Protocol over a secure SSL connection.
-
-The Real-Time Messaging Protocol (RTMPS) is used for streaming
-multimedia content across an encrypted connection.
-
-@section rtmpt
-
-Real-Time Messaging Protocol tunneled through HTTP.
-
-The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
-for streaming multimedia content within HTTP requests to traverse
-firewalls.
-
-@section rtmpte
-
-Encrypted Real-Time Messaging Protocol tunneled through HTTP.
-
-The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
-is used for streaming multimedia content within HTTP requests to traverse
-firewalls.
-
-@section rtmpts
-
-Real-Time Messaging Protocol tunneled through HTTPS.
-
-The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
-for streaming multimedia content within HTTPS requests to traverse
-firewalls.
-
 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
 
 Real-Time Messaging Protocol and its variants supported through
@@ -393,7 +242,7 @@ data transferred over RDT).
 
 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
-@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
+@uref{https://github.com/revmischa/rtsp-server, RTSP server}).
 
 The required syntax for a RTSP url is:
 @example
@@ -432,14 +281,12 @@ Flags for @code{rtsp_flags}:
 @table @option
 @item filter_src
 Accept packets only from negotiated peer address and port.
-@item listen
-Act as a server, listening for an incoming connection.
 @end table
 
 When receiving data over UDP, the demuxer tries to reorder received packets
-(since they may arrive out of order, or packets may get lost totally). This
-can be disabled by setting the maximum demuxing delay to zero (via
-the @code{max_delay} field of AVFormatContext).
+(since they may arrive out of order, or packets may get lost totally). In
+order for this to be enabled, a maximum delay must be specified in the
+@code{max_delay} field of AVFormatContext.
 
 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
 streams to display can be chosen with @code{-vst} @var{n} and
@@ -466,12 +313,6 @@ To send a stream in realtime to a RTSP server, for others to watch:
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
 @end example
 
-To receive a stream in realtime:
-
-@example
-ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
-@end example
-
 @section sap
 
 Session Announcement Protocol (RFC 2974). This is not technically a
@@ -578,11 +419,6 @@ tcp://@var{hostname}:@var{port}[?@var{options}]
 @item listen
 Listen for an incoming connection
 
-@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
-In write mode: if socket cannot be written in more than this time interval, raise error.
-This also sets timeout on TCP connection establishing.
-
 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
@@ -590,48 +426,6 @@ ffplay tcp://@var{hostname}:@var{port}
 
 @end table
 
-@section tls
-
-Transport Layer Security/Secure Sockets Layer
-
-The required syntax for a TLS/SSL url is:
-@example
-tls://@var{hostname}:@var{port}[?@var{options}]
-@end example
-
-@table @option
-
-@item listen
-Act as a server, listening for an incoming connection.
-
-@item cafile=@var{filename}
-Certificate authority file. The file must be in OpenSSL PEM format.
-
-@item cert=@var{filename}
-Certificate file. The file must be in OpenSSL PEM format.
-
-@item key=@var{filename}
-Private key file.
-
-@item verify=@var{0|1}
-Verify the peer's certificate.
-
-@end table
-
-Example command lines:
-
-To create a TLS/SSL server that serves an input stream.
-
-@example
-ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
-@end example
-
-To play back a stream from the TLS/SSL server using @command{ffplay}:
-
-@example
-ffplay tls://@var{hostname}:@var{port}
-@end example
-
 @section udp
 
 User Datagram Protocol.
@@ -641,23 +435,16 @@ The required syntax for a UDP url is:
 udp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
-@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
-
-In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows to reduce loss of data due to
-UDP socket buffer overruns. The @var{fifo_size} and
-@var{overrun_nonfatal} options are related to this buffer.
-
-The list of supported options follows.
+@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
+Follow the list of supported options.
 
 @table @option
 
 @item buffer_size=@var{size}
-Set the UDP socket buffer size in bytes. This is used both for the
-receiving and the sending buffer size.
+set the UDP buffer size in bytes
 
 @item localport=@var{port}
-Override the local UDP port to bind with.
+override the local UDP port to bind with
 
 @item localaddr=@var{addr}
 Choose the local IP address. This is useful e.g. if sending multicast
@@ -665,13 +452,13 @@ and the host has multiple interfaces, where the user can choose
 which interface to send on by specifying the IP address of that interface.
 
 @item pkt_size=@var{size}
-Set the size in bytes of UDP packets.
+set the size in bytes of UDP packets
 
 @item reuse=@var{1|0}
-Explicitly allow or disallow reusing UDP sockets.
+explicitly allow or disallow reusing UDP sockets
 
 @item ttl=@var{ttl}
-Set the time to live value (for multicast only).
+set the time to live value (for multicast only)
 
 @item connect=@var{1|0}
 Initialize the UDP socket with @code{connect()}. In this case, the
@@ -683,28 +470,9 @@ and makes writes return with AVERROR(ECONNREFUSED) if "destination
 unreachable" is received.
 For receiving, this gives the benefit of only receiving packets from
 the specified peer address/port.
-
-@item sources=@var{address}[,@var{address}]
-Only receive packets sent to the multicast group from one of the
-specified sender IP addresses.
-
-@item block=@var{address}[,@var{address}]
-Ignore packets sent to the multicast group from the specified
-sender IP addresses.
-
-@item fifo_size=@var{units}
-Set the UDP receiving circular buffer size, expressed as a number of
-packets with size of 188 bytes. If not specified defaults to 7*4096.
-
-@item overrun_nonfatal=@var{1|0}
-Survive in case of UDP receiving circular buffer overrun. Default
-value is 0.
-
-@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
 @end table
 
-Some usage examples of the UDP protocol with @command{ffmpeg} follow.
+Some usage examples of the udp protocol with @command{ffmpeg} follow.
 
 To stream over UDP to a remote endpoint:
 @example

doc/swresample.txt

@@ -32,7 +32,7 @@ Special Converter         v
                         Output
 
 Planar/Packed convertion is done when needed during sample format convertion
-Every step can be skipped without memcpy when its not needed.
+Every step can be skiped without memcpy when its not needed.
 Either Resampling and Rematrixing can be performed first depending on which
 way its faster.
 The Buffers are needed for resampling due to resamplng being a process that

doc/swscale.txt

@@ -96,3 +96,4 @@ would benefit from it.
 Also, as already hinted at, initFilter() accepts an optional convolutional
 filter as input that can be used for contrast, saturation, blur, sharpening
 shift, chroma vs. luma shift, ...
+

doc/syntax.texi

@@ -1,158 +0,0 @@
-@chapter Syntax
-@c man begin SYNTAX
-
-When evaluating specific formats, FFmpeg uses internal library parsing
-functions, shared by the tools. This section documents the syntax of
-some of these formats.
-
-@anchor{date syntax}
-@section Date
-
-The accepted syntax is:
-@example
-[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
-now
-@end example
-
-If the value is "now" it takes the current time.
-
-Time is local time unless Z is appended, in which case it is
-interpreted as UTC.
-If the year-month-day part is not specified it takes the current
-year-month-day.
-
-@anchor{time duration syntax}
-@section Time duration
-
-The accepted syntax is:
-@example
-[-]HH:MM:SS[.m...]
-[-]S+[.m...]
-@end example
-
-@var{HH} expresses the number of hours, @var{MM} the number a of minutes
-and @var{SS} the number of seconds.
-
-@anchor{video size syntax}
-@section Video size
-Specify the size of the sourced video, it may be a string of the form
-@var{width}x@var{height}, or the name of a size abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item sqcif
-128x96
-@item qcif
-176x144
-@item cif
-352x288
-@item 4cif
-704x576
-@item 16cif
-1408x1152
-@item qqvga
-160x120
-@item qvga
-320x240
-@item vga
-640x480
-@item svga
-800x600
-@item xga
-1024x768
-@item uxga
-1600x1200
-@item qxga
-2048x1536
-@item sxga
-1280x1024
-@item qsxga
-2560x2048
-@item hsxga
-5120x4096
-@item wvga
-852x480
-@item wxga
-1366x768
-@item wsxga
-1600x1024
-@item wuxga
-1920x1200
-@item woxga
-2560x1600
-@item wqsxga
-3200x2048
-@item wquxga
-3840x2400
-@item whsxga
-6400x4096
-@item whuxga
-7680x4800
-@item cga
-320x200
-@item ega
-640x350
-@item hd480
-852x480
-@item hd720
-1280x720
-@item hd1080
-1920x1080
-@end table
-
-@anchor{video rate syntax}
-@section Video rate
-
-Specify the frame rate of a video, expressed as the number of frames
-generated per second. It has to be a string in the format
-@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
-number or a valid video frame rate abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-30000/1001
-@item pal
-25/1
-@item qntsc
-30000/1
-@item qpal
-25/1
-@item sntsc
-30000/1
-@item spal
-25/1
-@item film
-24/1
-@item ntsc-film
-24000/1
-@end table
-
-@anchor{ratio syntax}
-@section Ratio
-
-A ratio can be expressed as an expression, or in the form
-@var{numerator}:@var{denominator}.
-
-Note that a ratio with infinite (1/0) or negative value is
-considered valid, so you should check on the returned value if you
-want to exclude those values.
-
-The undefined value can be expressed using the "0:0" string.
-
-@anchor{color syntax}
-@section Color
-
-It can be the name of a color (case insensitive match) or a
-[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
-representing the alpha component.
-
-The alpha component may be a string composed by "0x" followed by an
-hexadecimal number or a decimal number between 0.0 and 1.0, which
-represents the opacity value (0x00/0.0 means completely transparent,
-0xff/1.0 completely opaque).
-If the alpha component is not specified then 0xff is assumed.
-
-The string "random" will result in a random color.
-
-@c man end SYNTAX

doc/t2h.init

@@ -158,9 +158,6 @@ $AFTER_BODY_OPEN
 EOT
 }
 
-# declare encoding in header
-$IN_ENCODING = $ENCODING = "utf-8";
-
 # no navigation elements
 $SECTION_NAVIGATION = 0;
 # the same for texi2html 5.0

doc/texi2pod.pl

@@ -1,4 +1,4 @@
-#! /usr/bin/perl
+#! /usr/bin/perl -w
 
 #   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
 
@@ -23,8 +23,6 @@
 # markup to Perl POD format.  It's intended to be used to extract
 # something suitable for a manpage from a Texinfo document.
 
-use warnings;
-
 $output = 0;
 $skipping = 0;
 %sects = ();
@@ -38,7 +36,7 @@ $shift = "";
 %defs = ();
 $fnno = 1;
 $inf = "";
-@ibase = ();
+$ibase = "";
 
 while ($_ = shift) {
     if (/^-D(.*)$/) {
@@ -54,8 +52,6 @@ while ($_ = shift) {
         die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
             unless $flag =~ /^[a-zA-Z0-9_-]+$/;
         $defs{$flag} = $value;
-    } elsif (/^-I(.*)$/) {
-        push @ibase, $1 ne "" ? $1 : shift;
     } elsif (/^-/) {
         usage();
     } else {
@@ -65,12 +61,10 @@ while ($_ = shift) {
     }
 }
 
-push @ibase, ".";
-
 if (defined $in) {
     $inf = gensym();
     open($inf, "<$in") or die "opening \"$in\": $!\n";
-    push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|;
+    $ibase = $1 if $in =~ m|^(.+)/[^/]+$|;
 } else {
     $inf = \*STDIN;
 }
@@ -80,7 +74,7 @@ if (defined $out) {
 }
 
 while(defined $inf) {
-INF: while(<$inf>) {
+while(<$inf>) {
     # Certain commands are discarded without further processing.
     /^\@(?:
          [a-z]+index            # @*index: useful only in complete manual
@@ -110,10 +104,11 @@ INF: while(<$inf>) {
         push @instack, $inf;
         $inf = gensym();
 
-        for (@ibase) {
-            open($inf, "<" . $_ . "/" . $1) and next INF;
-        }
-        die "cannot open $1: $!\n";
+        # Try cwd and $ibase.
+        open($inf, "<" . $1)
+            or open($inf, "<" . $ibase . "/" . $1)
+                or die "cannot open $1 or $ibase/$1: $!\n";
+        next;
     };
 
     # Look for blocks surrounded by @c man begin SECTION ... @c man end.

doc/viterbi.txt

@@ -107,3 +107,4 @@ one with score 3)
 
 Author: Michael niedermayer
 Copyright LGPL
+

ffmpeg.h

@@ -1,409 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_H
-#define FFMPEG_H
-
-#include "config.h"
-
-#include <stdint.h>
-#include <stdio.h>
-#include <signal.h>
-
-#if HAVE_PTHREADS
-#include <pthread.h>
-#endif
-
-#include "cmdutils.h"
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/avfiltergraph.h"
-
-#include "libavutil/avutil.h"
-#include "libavutil/dict.h"
-#include "libavutil/fifo.h"
-#include "libavutil/pixfmt.h"
-#include "libavutil/rational.h"
-
-#include "libswresample/swresample.h"
-
-#define VSYNC_AUTO       -1
-#define VSYNC_PASSTHROUGH 0
-#define VSYNC_CFR         1
-#define VSYNC_VFR         2
-#define VSYNC_DROP        0xff
-
-#define MAX_STREAMS 1024    /* arbitrary sanity check value */
-
-/* select an input stream for an output stream */
-typedef struct StreamMap {
-    int disabled;           /** 1 is this mapping is disabled by a negative map */
-    int file_index;
-    int stream_index;
-    int sync_file_index;
-    int sync_stream_index;
-    char *linklabel;       /** name of an output link, for mapping lavfi outputs */
-} StreamMap;
-
-typedef struct {
-    int  file_idx,  stream_idx,  channel_idx; // input
-    int ofile_idx, ostream_idx;               // output
-} AudioChannelMap;
-
-typedef struct OptionsContext {
-    /* input/output options */
-    int64_t start_time;
-    const char *format;
-
-    SpecifierOpt *codec_names;
-    int        nb_codec_names;
-    SpecifierOpt *audio_channels;
-    int        nb_audio_channels;
-    SpecifierOpt *audio_sample_rate;
-    int        nb_audio_sample_rate;
-    SpecifierOpt *frame_rates;
-    int        nb_frame_rates;
-    SpecifierOpt *frame_sizes;
-    int        nb_frame_sizes;
-    SpecifierOpt *frame_pix_fmts;
-    int        nb_frame_pix_fmts;
-
-    /* input options */
-    int64_t input_ts_offset;
-    int rate_emu;
-
-    SpecifierOpt *ts_scale;
-    int        nb_ts_scale;
-    SpecifierOpt *dump_attachment;
-    int        nb_dump_attachment;
-
-    /* output options */
-    StreamMap *stream_maps;
-    int     nb_stream_maps;
-    AudioChannelMap *audio_channel_maps; /* one info entry per -map_channel */
-    int           nb_audio_channel_maps; /* number of (valid) -map_channel settings */
-    int metadata_global_manual;
-    int metadata_streams_manual;
-    int metadata_chapters_manual;
-    const char **attachments;
-    int       nb_attachments;
-
-    int chapters_input_file;
-
-    int64_t recording_time;
-    uint64_t limit_filesize;
-    float mux_preload;
-    float mux_max_delay;
-    int shortest;
-
-    int video_disable;
-    int audio_disable;
-    int subtitle_disable;
-    int data_disable;
-
-    /* indexed by output file stream index */
-    int   *streamid_map;
-    int nb_streamid_map;
-
-    SpecifierOpt *metadata;
-    int        nb_metadata;
-    SpecifierOpt *max_frames;
-    int        nb_max_frames;
-    SpecifierOpt *bitstream_filters;
-    int        nb_bitstream_filters;
-    SpecifierOpt *codec_tags;
-    int        nb_codec_tags;
-    SpecifierOpt *sample_fmts;
-    int        nb_sample_fmts;
-    SpecifierOpt *qscale;
-    int        nb_qscale;
-    SpecifierOpt *forced_key_frames;
-    int        nb_forced_key_frames;
-    SpecifierOpt *force_fps;
-    int        nb_force_fps;
-    SpecifierOpt *frame_aspect_ratios;
-    int        nb_frame_aspect_ratios;
-    SpecifierOpt *rc_overrides;
-    int        nb_rc_overrides;
-    SpecifierOpt *intra_matrices;
-    int        nb_intra_matrices;
-    SpecifierOpt *inter_matrices;
-    int        nb_inter_matrices;
-    SpecifierOpt *top_field_first;
-    int        nb_top_field_first;
-    SpecifierOpt *metadata_map;
-    int        nb_metadata_map;
-    SpecifierOpt *presets;
-    int        nb_presets;
-    SpecifierOpt *copy_initial_nonkeyframes;
-    int        nb_copy_initial_nonkeyframes;
-    SpecifierOpt *copy_prior_start;
-    int        nb_copy_prior_start;
-    SpecifierOpt *filters;
-    int        nb_filters;
-    SpecifierOpt *fix_sub_duration;
-    int        nb_fix_sub_duration;
-    SpecifierOpt *pass;
-    int        nb_pass;
-    SpecifierOpt *passlogfiles;
-    int        nb_passlogfiles;
-} OptionsContext;
-
-typedef struct InputFilter {
-    AVFilterContext    *filter;
-    struct InputStream *ist;
-    struct FilterGraph *graph;
-    uint8_t            *name;
-} InputFilter;
-
-typedef struct OutputFilter {
-    AVFilterContext     *filter;
-    struct OutputStream *ost;
-    struct FilterGraph  *graph;
-    uint8_t             *name;
-
-    /* temporary storage until stream maps are processed */
-    AVFilterInOut       *out_tmp;
-} OutputFilter;
-
-typedef struct FilterGraph {
-    int            index;
-    const char    *graph_desc;
-
-    AVFilterGraph *graph;
-
-    InputFilter   **inputs;
-    int          nb_inputs;
-    OutputFilter **outputs;
-    int         nb_outputs;
-} FilterGraph;
-
-typedef struct InputStream {
-    int file_index;
-    AVStream *st;
-    int discard;             /* true if stream data should be discarded */
-    int decoding_needed;     /* true if the packets must be decoded in 'raw_fifo' */
-    AVCodec *dec;
-    AVFrame *decoded_frame;
-
-    int64_t       start;     /* time when read started */
-    /* predicted dts of the next packet read for this stream or (when there are
-     * several frames in a packet) of the next frame in current packet (in AV_TIME_BASE units) */
-    int64_t       next_dts;
-    int64_t       dts;       ///< dts of the last packet read for this stream (in AV_TIME_BASE units)
-
-    int64_t       next_pts;  ///< synthetic pts for the next decode frame (in AV_TIME_BASE units)
-    int64_t       pts;       ///< current pts of the decoded frame  (in AV_TIME_BASE units)
-    int           wrap_correction_done;
-    double ts_scale;
-    int is_start;            /* is 1 at the start and after a discontinuity */
-    int saw_first_ts;
-    int showed_multi_packet_warning;
-    AVDictionary *opts;
-    AVRational framerate;               /* framerate forced with -r */
-    int top_field_first;
-
-    int resample_height;
-    int resample_width;
-    int resample_pix_fmt;
-
-    int      resample_sample_fmt;
-    int      resample_sample_rate;
-    int      resample_channels;
-    uint64_t resample_channel_layout;
-
-    int fix_sub_duration;
-    struct { /* previous decoded subtitle and related variables */
-        int got_output;
-        int ret;
-        AVSubtitle subtitle;
-    } prev_sub;
-
-    struct sub2video {
-        int64_t last_pts;
-        AVFilterBufferRef *ref;
-        int w, h;
-    } sub2video;
-
-    /* a pool of free buffers for decoded data */
-    FrameBuffer *buffer_pool;
-    int dr1;
-
-    /* decoded data from this stream goes into all those filters
-     * currently video and audio only */
-    InputFilter **filters;
-    int        nb_filters;
-} InputStream;
-
-typedef struct InputFile {
-    AVFormatContext *ctx;
-    int eof_reached;      /* true if eof reached */
-    int eagain;           /* true if last read attempt returned EAGAIN */
-    int ist_index;        /* index of first stream in input_streams */
-    int64_t ts_offset;
-    int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
-                             from ctx.nb_streams if new streams appear during av_read_frame() */
-    int nb_streams_warn;  /* number of streams that the user was warned of */
-    int rate_emu;
-
-#if HAVE_PTHREADS
-    pthread_t thread;           /* thread reading from this file */
-    int finished;               /* the thread has exited */
-    int joined;                 /* the thread has been joined */
-    pthread_mutex_t fifo_lock;  /* lock for access to fifo */
-    pthread_cond_t  fifo_cond;  /* the main thread will signal on this cond after reading from fifo */
-    AVFifoBuffer *fifo;         /* demuxed packets are stored here; freed by the main thread */
-#endif
-} InputFile;
-
-typedef struct OutputStream {
-    int file_index;          /* file index */
-    int index;               /* stream index in the output file */
-    int source_index;        /* InputStream index */
-    AVStream *st;            /* stream in the output file */
-    int encoding_needed;     /* true if encoding needed for this stream */
-    int frame_number;
-    /* input pts and corresponding output pts
-       for A/V sync */
-    struct InputStream *sync_ist; /* input stream to sync against */
-    int64_t sync_opts;       /* output frame counter, could be changed to some true timestamp */ // FIXME look at frame_number
-    /* pts of the first frame encoded for this stream, used for limiting
-     * recording time */
-    int64_t first_pts;
-    AVBitStreamFilterContext *bitstream_filters;
-    AVCodec *enc;
-    int64_t max_frames;
-    AVFrame *filtered_frame;
-
-    /* video only */
-    AVRational frame_rate;
-    int force_fps;
-    int top_field_first;
-
-    float frame_aspect_ratio;
-    float last_quality;
-
-    /* forced key frames */
-    int64_t *forced_kf_pts;
-    int forced_kf_count;
-    int forced_kf_index;
-    char *forced_keyframes;
-
-    /* audio only */
-    int audio_channels_map[SWR_CH_MAX];  /* list of the channels id to pick from the source stream */
-    int audio_channels_mapped;           /* number of channels in audio_channels_map */
-
-    char *logfile_prefix;
-    FILE *logfile;
-
-    OutputFilter *filter;
-    char *avfilter;
-
-    int64_t sws_flags;
-    int64_t swr_dither_method;
-    double swr_dither_scale;
-    AVDictionary *opts;
-    int finished;        /* no more packets should be written for this stream */
-    int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
-    int stream_copy;
-    const char *attachment_filename;
-    int copy_initial_nonkeyframes;
-    int copy_prior_start;
-
-    int keep_pix_fmt;
-} OutputStream;
-
-typedef struct OutputFile {
-    AVFormatContext *ctx;
-    AVDictionary *opts;
-    int ost_index;       /* index of the first stream in output_streams */
-    int64_t recording_time;  ///< desired length of the resulting file in microseconds == AV_TIME_BASE units
-    int64_t start_time;      ///< start time in microseconds == AV_TIME_BASE units
-    uint64_t limit_filesize; /* filesize limit expressed in bytes */
-
-    int shortest;
-} OutputFile;
-
-extern InputStream **input_streams;
-extern int        nb_input_streams;
-extern InputFile   **input_files;
-extern int        nb_input_files;
-
-extern OutputStream **output_streams;
-extern int         nb_output_streams;
-extern OutputFile   **output_files;
-extern int         nb_output_files;
-
-extern FilterGraph **filtergraphs;
-extern int        nb_filtergraphs;
-
-extern char *vstats_filename;
-
-extern float audio_drift_threshold;
-extern float dts_delta_threshold;
-extern float dts_error_threshold;
-
-extern int audio_volume;
-extern int audio_sync_method;
-extern int video_sync_method;
-extern int do_benchmark;
-extern int do_benchmark_all;
-extern int do_deinterlace;
-extern int do_hex_dump;
-extern int do_pkt_dump;
-extern int copy_ts;
-extern int copy_tb;
-extern int debug_ts;
-extern int exit_on_error;
-extern int print_stats;
-extern int qp_hist;
-extern int same_quant;
-extern int stdin_interaction;
-extern int frame_bits_per_raw_sample;
-extern AVIOContext *progress_avio;
-
-extern const AVIOInterruptCB int_cb;
-
-extern const OptionDef options[];
-
-void term_init(void);
-void term_exit(void);
-
-void reset_options(OptionsContext *o, int is_input);
-void show_usage(void);
-
-void opt_output_file(void *optctx, const char *filename);
-
-void assert_avoptions(AVDictionary *m);
-
-int guess_input_channel_layout(InputStream *ist);
-
-enum PixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum PixelFormat target);
-void choose_sample_fmt(AVStream *st, AVCodec *codec);
-
-int configure_filtergraph(FilterGraph *fg);
-int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out);
-int ist_in_filtergraph(FilterGraph *fg, InputStream *ist);
-FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
-
-#endif /* FFMPEG_H */

ffmpeg_filter.c

@@ -1,774 +0,0 @@
-/*
- * ffmpeg filter configuration
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "ffmpeg.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/avfiltergraph.h"
-#include "libavfilter/buffersink.h"
-
-#include "libavutil/audioconvert.h"
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/bprint.h"
-#include "libavutil/pixdesc.h"
-#include "libavutil/pixfmt.h"
-#include "libavutil/imgutils.h"
-#include "libavutil/samplefmt.h"
-
-enum PixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum PixelFormat target)
-{
-    if (codec && codec->pix_fmts) {
-        const enum PixelFormat *p = codec->pix_fmts;
-        int has_alpha= av_pix_fmt_descriptors[target].nb_components % 2 == 0;
-        enum PixelFormat best= PIX_FMT_NONE;
-        if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
-            if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_NONE };
-            } else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, PIX_FMT_YUV420P,
-                                                 PIX_FMT_YUV422P, PIX_FMT_YUV444P, PIX_FMT_BGRA, PIX_FMT_NONE };
-            }
-        }
-        for (; *p != PIX_FMT_NONE; p++) {
-            best= avcodec_find_best_pix_fmt_of_2(best, *p, target, has_alpha, NULL);
-            if (*p == target)
-                break;
-        }
-        if (*p == PIX_FMT_NONE) {
-            if (target != PIX_FMT_NONE)
-                av_log(NULL, AV_LOG_WARNING,
-                       "Incompatible pixel format '%s' for codec '%s', auto-selecting format '%s'\n",
-                       av_pix_fmt_descriptors[target].name,
-                       codec->name,
-                       av_pix_fmt_descriptors[best].name);
-            return best;
-        }
-    }
-    return target;
-}
-
-void choose_sample_fmt(AVStream *st, AVCodec *codec)
-{
-    if (codec && codec->sample_fmts) {
-        const enum AVSampleFormat *p = codec->sample_fmts;
-        for (; *p != -1; p++) {
-            if (*p == st->codec->sample_fmt)
-                break;
-        }
-        if (*p == -1) {
-            if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
-                av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
-            if(av_get_sample_fmt_name(st->codec->sample_fmt))
-            av_log(NULL, AV_LOG_WARNING,
-                   "Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n",
-                   av_get_sample_fmt_name(st->codec->sample_fmt),
-                   codec->name,
-                   av_get_sample_fmt_name(codec->sample_fmts[0]));
-            st->codec->sample_fmt = codec->sample_fmts[0];
-        }
-    }
-}
-
-static char *choose_pix_fmts(OutputStream *ost)
-{
-     if (ost->keep_pix_fmt) {
-        if (ost->filter)
-            avfilter_graph_set_auto_convert(ost->filter->graph->graph,
-                                            AVFILTER_AUTO_CONVERT_NONE);
-        if (ost->st->codec->pix_fmt == PIX_FMT_NONE)
-            return NULL;
-        return av_strdup(av_get_pix_fmt_name(ost->st->codec->pix_fmt));
-    }
-    if (ost->st->codec->pix_fmt != PIX_FMT_NONE) {
-        return av_strdup(av_get_pix_fmt_name(choose_pixel_fmt(ost->st, ost->enc, ost->st->codec->pix_fmt)));
-    } else if (ost->enc && ost->enc->pix_fmts) {
-        const enum PixelFormat *p;
-        AVIOContext *s = NULL;
-        uint8_t *ret;
-        int len;
-
-        if (avio_open_dyn_buf(&s) < 0)
-            exit_program(1);
-
-        p = ost->enc->pix_fmts;
-        if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
-            if (ost->st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_NONE };
-            } else if (ost->st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, PIX_FMT_YUV420P,
-                                                    PIX_FMT_YUV422P, PIX_FMT_YUV444P, PIX_FMT_BGRA, PIX_FMT_NONE };
-            }
-        }
-
-        for (; *p != PIX_FMT_NONE; p++) {
-            const char *name = av_get_pix_fmt_name(*p);
-            avio_printf(s, "%s:", name);
-        }
-        len = avio_close_dyn_buf(s, &ret);
-        ret[len - 1] = 0;
-        return ret;
-    } else
-        return NULL;
-}
-
-/**
- * Define a function for building a string containing a list of
- * allowed formats,
- */
-#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name, separator)\
-static char *choose_ ## var ## s(OutputStream *ost)                            \
-{                                                                              \
-    if (ost->st->codec->var != none) {                                         \
-        get_name(ost->st->codec->var);                                         \
-        return av_strdup(name);                                                \
-    } else if (ost->enc->supported_list) {                                     \
-        const type *p;                                                         \
-        AVIOContext *s = NULL;                                                 \
-        uint8_t *ret;                                                          \
-        int len;                                                               \
-                                                                               \
-        if (avio_open_dyn_buf(&s) < 0)                                         \
-            exit_program(1);                                                   \
-                                                                               \
-        for (p = ost->enc->supported_list; *p != none; p++) {                  \
-            get_name(*p);                                                      \
-            avio_printf(s, "%s" separator, name);                              \
-        }                                                                      \
-        len = avio_close_dyn_buf(s, &ret);                                     \
-        ret[len - 1] = 0;                                                      \
-        return ret;                                                            \
-    } else                                                                     \
-        return NULL;                                                           \
-}
-
-// DEF_CHOOSE_FORMAT(enum PixelFormat, pix_fmt, pix_fmts, PIX_FMT_NONE,
-//                   GET_PIX_FMT_NAME, ":")
-
-DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts,
-                  AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME, ",")
-
-DEF_CHOOSE_FORMAT(int, sample_rate, supported_samplerates, 0,
-                  GET_SAMPLE_RATE_NAME, ",")
-
-DEF_CHOOSE_FORMAT(uint64_t, channel_layout, channel_layouts, 0,
-                  GET_CH_LAYOUT_NAME, ",")
-
-FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)
-{
-    FilterGraph *fg = av_mallocz(sizeof(*fg));
-
-    if (!fg)
-        exit_program(1);
-    fg->index = nb_filtergraphs;
-
-    fg->outputs = grow_array(fg->outputs, sizeof(*fg->outputs), &fg->nb_outputs,
-                             fg->nb_outputs + 1);
-    if (!(fg->outputs[0] = av_mallocz(sizeof(*fg->outputs[0]))))
-        exit_program(1);
-    fg->outputs[0]->ost   = ost;
-    fg->outputs[0]->graph = fg;
-
-    ost->filter = fg->outputs[0];
-
-    fg->inputs = grow_array(fg->inputs, sizeof(*fg->inputs), &fg->nb_inputs,
-                            fg->nb_inputs + 1);
-    if (!(fg->inputs[0] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit_program(1);
-    fg->inputs[0]->ist   = ist;
-    fg->inputs[0]->graph = fg;
-
-    ist->filters = grow_array(ist->filters, sizeof(*ist->filters),
-                              &ist->nb_filters, ist->nb_filters + 1);
-    ist->filters[ist->nb_filters - 1] = fg->inputs[0];
-
-    filtergraphs = grow_array(filtergraphs, sizeof(*filtergraphs),
-                              &nb_filtergraphs, nb_filtergraphs + 1);
-    filtergraphs[nb_filtergraphs - 1] = fg;
-
-    return fg;
-}
-
-static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
-{
-    InputStream *ist = NULL;
-    enum AVMediaType type = avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx);
-    int i;
-
-    // TODO: support other filter types
-    if (type != AVMEDIA_TYPE_VIDEO && type != AVMEDIA_TYPE_AUDIO) {
-        av_log(NULL, AV_LOG_FATAL, "Only video and audio filters supported "
-               "currently.\n");
-        exit_program(1);
-    }
-
-    if (in->name) {
-        AVFormatContext *s;
-        AVStream       *st = NULL;
-        char *p;
-        int file_idx = strtol(in->name, &p, 0);
-
-        if (file_idx < 0 || file_idx >= nb_input_files) {
-            av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtergraph description %s.\n",
-                   file_idx, fg->graph_desc);
-            exit_program(1);
-        }
-        s = input_files[file_idx]->ctx;
-
-        for (i = 0; i < s->nb_streams; i++) {
-            enum AVMediaType stream_type = s->streams[i]->codec->codec_type;
-            if (stream_type != type &&
-                !(stream_type == AVMEDIA_TYPE_SUBTITLE &&
-                  type == AVMEDIA_TYPE_VIDEO /* sub2video hack */))
-                continue;
-            if (check_stream_specifier(s, s->streams[i], *p == ':' ? p + 1 : p) == 1) {
-                st = s->streams[i];
-                break;
-            }
-        }
-        if (!st) {
-            av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
-                   "matches no streams.\n", p, fg->graph_desc);
-            exit_program(1);
-        }
-        ist = input_streams[input_files[file_idx]->ist_index + st->index];
-    } else {
-        /* find the first unused stream of corresponding type */
-        for (i = 0; i < nb_input_streams; i++) {
-            ist = input_streams[i];
-            if (ist->st->codec->codec_type == type && ist->discard)
-                break;
-        }
-        if (i == nb_input_streams) {
-            av_log(NULL, AV_LOG_FATAL, "Cannot find a matching stream for "
-                   "unlabeled input pad %d on filter %s\n", in->pad_idx,
-                   in->filter_ctx->name);
-            exit_program(1);
-        }
-    }
-    av_assert0(ist);
-
-    ist->discard         = 0;
-    ist->decoding_needed++;
-    ist->st->discard = AVDISCARD_NONE;
-
-    fg->inputs = grow_array(fg->inputs, sizeof(*fg->inputs),
-                            &fg->nb_inputs, fg->nb_inputs + 1);
-    if (!(fg->inputs[fg->nb_inputs - 1] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit_program(1);
-    fg->inputs[fg->nb_inputs - 1]->ist   = ist;
-    fg->inputs[fg->nb_inputs - 1]->graph = fg;
-
-    ist->filters = grow_array(ist->filters, sizeof(*ist->filters),
-                              &ist->nb_filters, ist->nb_filters + 1);
-    ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
-}
-
-static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
-{
-    char *pix_fmts;
-    OutputStream *ost = ofilter->ost;
-    AVCodecContext *codec = ost->st->codec;
-    AVFilterContext *last_filter = out->filter_ctx;
-    int pad_idx = out->pad_idx;
-    int ret;
-    char name[255];
-    AVBufferSinkParams *buffersink_params = av_buffersink_params_alloc();
-
-    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
-    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("ffbuffersink"),
-                                       name, NULL, NULL, fg->graph);
-    av_freep(&buffersink_params);
-
-    if (ret < 0)
-        return ret;
-
-    if (codec->width || codec->height) {
-        char args[255];
-        AVFilterContext *filter;
-
-        snprintf(args, sizeof(args), "%d:%d:flags=0x%X",
-                 codec->width,
-                 codec->height,
-                 (unsigned)ost->sws_flags);
-        snprintf(name, sizeof(name), "scaler for output stream %d:%d",
-                 ost->file_index, ost->index);
-        if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
-                                                name, args, NULL, fg->graph)) < 0)
-            return ret;
-        if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
-            return ret;
-
-        last_filter = filter;
-        pad_idx = 0;
-    }
-
-    if ((pix_fmts = choose_pix_fmts(ost))) {
-        AVFilterContext *filter;
-        snprintf(name, sizeof(name), "pixel format for output stream %d:%d",
-                 ost->file_index, ost->index);
-        if ((ret = avfilter_graph_create_filter(&filter,
-                                                avfilter_get_by_name("format"),
-                                                "format", pix_fmts, NULL,
-                                                fg->graph)) < 0)
-            return ret;
-        if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
-            return ret;
-
-        last_filter = filter;
-        pad_idx     = 0;
-        av_freep(&pix_fmts);
-    }
-
-    if (ost->frame_rate.num && 0) {
-        AVFilterContext *fps;
-        char args[255];
-
-        snprintf(args, sizeof(args), "fps=%d/%d", ost->frame_rate.num,
-                 ost->frame_rate.den);
-        snprintf(name, sizeof(name), "fps for output stream %d:%d",
-                 ost->file_index, ost->index);
-        ret = avfilter_graph_create_filter(&fps, avfilter_get_by_name("fps"),
-                                           name, args, NULL, fg->graph);
-        if (ret < 0)
-            return ret;
-
-        ret = avfilter_link(last_filter, pad_idx, fps, 0);
-        if (ret < 0)
-            return ret;
-        last_filter = fps;
-        pad_idx = 0;
-    }
-
-    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
-        return ret;
-
-    return 0;
-}
-
-static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
-{
-    OutputStream *ost = ofilter->ost;
-    AVCodecContext *codec  = ost->st->codec;
-    AVFilterContext *last_filter = out->filter_ctx;
-    int pad_idx = out->pad_idx;
-    char *sample_fmts, *sample_rates, *channel_layouts;
-    char name[255];
-    int ret;
-
-
-    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
-    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("ffabuffersink"),
-                                       name, NULL, NULL, fg->graph);
-    if (ret < 0)
-        return ret;
-
-#define AUTO_INSERT_FILTER(opt_name, filter_name, arg) do {                 \
-    AVFilterContext *filt_ctx;                                              \
-                                                                            \
-    av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi "            \
-           "similarly to -af " filter_name "=%s.\n", arg);                  \
-                                                                            \
-    ret = avfilter_graph_create_filter(&filt_ctx,                           \
-                                       avfilter_get_by_name(filter_name),   \
-                                       filter_name, arg, NULL, fg->graph);  \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    ret = avfilter_link(last_filter, pad_idx, filt_ctx, 0);                 \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    last_filter = filt_ctx;                                                 \
-    pad_idx = 0;                                                            \
-} while (0)
-    if (ost->audio_channels_mapped) {
-        int i;
-        AVBPrint pan_buf;
-        av_bprint_init(&pan_buf, 256, 8192);
-        av_bprintf(&pan_buf, "0x%"PRIx64,
-                   av_get_default_channel_layout(ost->audio_channels_mapped));
-        for (i = 0; i < ost->audio_channels_mapped; i++)
-            if (ost->audio_channels_map[i] != -1)
-                av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
-
-        AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str);
-        av_bprint_finalize(&pan_buf, NULL);
-    }
-
-    if (codec->channels && !codec->channel_layout)
-        codec->channel_layout = av_get_default_channel_layout(codec->channels);
-
-    sample_fmts     = choose_sample_fmts(ost);
-    sample_rates    = choose_sample_rates(ost);
-    channel_layouts = choose_channel_layouts(ost);
-    if (sample_fmts || sample_rates || channel_layouts) {
-        AVFilterContext *format;
-        char args[256];
-        args[0] = 0;
-
-        if (sample_fmts)
-            av_strlcatf(args, sizeof(args), "sample_fmts=%s:",
-                            sample_fmts);
-        if (sample_rates)
-            av_strlcatf(args, sizeof(args), "sample_rates=%s:",
-                            sample_rates);
-        if (channel_layouts)
-            av_strlcatf(args, sizeof(args), "channel_layouts=%s:",
-                            channel_layouts);
-
-        av_freep(&sample_fmts);
-        av_freep(&sample_rates);
-        av_freep(&channel_layouts);
-
-        snprintf(name, sizeof(name), "audio format for output stream %d:%d",
-                 ost->file_index, ost->index);
-        ret = avfilter_graph_create_filter(&format,
-                                           avfilter_get_by_name("aformat"),
-                                           name, args, NULL, fg->graph);
-        if (ret < 0)
-            return ret;
-
-        ret = avfilter_link(last_filter, pad_idx, format, 0);
-        if (ret < 0)
-            return ret;
-
-        last_filter = format;
-        pad_idx = 0;
-    }
-
-    if (audio_volume != 256 && 0) {
-        char args[256];
-
-        snprintf(args, sizeof(args), "%f", audio_volume / 256.);
-        AUTO_INSERT_FILTER("-vol", "volume", args);
-    }
-
-    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
-        return ret;
-
-    return 0;
-}
-
-#define DESCRIBE_FILTER_LINK(f, inout, in)                         \
-{                                                                  \
-    AVFilterContext *ctx = inout->filter_ctx;                      \
-    AVFilterPad *pads = in ? ctx->input_pads  : ctx->output_pads;  \
-    int       nb_pads = in ? ctx->input_count : ctx->output_count; \
-    AVIOContext *pb;                                               \
-                                                                   \
-    if (avio_open_dyn_buf(&pb) < 0)                                \
-        exit_program(1);                                           \
-                                                                   \
-    avio_printf(pb, "%s", ctx->filter->name);                      \
-    if (nb_pads > 1)                                               \
-        avio_printf(pb, ":%s", avfilter_pad_get_name(pads, inout->pad_idx));\
-    avio_w8(pb, 0);                                                \
-    avio_close_dyn_buf(pb, &f->name);                              \
-}
-
-int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
-{
-    av_freep(&ofilter->name);
-    DESCRIBE_FILTER_LINK(ofilter, out, 0);
-
-    switch (avfilter_pad_get_type(out->filter_ctx->output_pads, out->pad_idx)) {
-    case AVMEDIA_TYPE_VIDEO: return configure_output_video_filter(fg, ofilter, out);
-    case AVMEDIA_TYPE_AUDIO: return configure_output_audio_filter(fg, ofilter, out);
-    default: av_assert0(0);
-    }
-}
-
-static int sub2video_prepare(InputStream *ist)
-{
-    AVFormatContext *avf = input_files[ist->file_index]->ctx;
-    int i, ret, w, h;
-    uint8_t *image[4];
-    int linesize[4];
-
-    /* Compute the size of the canvas for the subtitles stream.
-       If the subtitles codec has set a size, use it. Otherwise use the
-       maximum dimensions of the video streams in the same file. */
-    w = ist->st->codec->width;
-    h = ist->st->codec->height;
-    if (!(w && h)) {
-        for (i = 0; i < avf->nb_streams; i++) {
-            if (avf->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
-                w = FFMAX(w, avf->streams[i]->codec->width);
-                h = FFMAX(h, avf->streams[i]->codec->height);
-            }
-        }
-        if (!(w && h)) {
-            w = FFMAX(w, 720);
-            h = FFMAX(h, 576);
-        }
-        av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
-    }
-    ist->sub2video.w = ist->st->codec->width  = w;
-    ist->sub2video.h = ist->st->codec->height = h;
-
-    /* rectangles are PIX_FMT_PAL8, but we have no guarantee that the
-       palettes for all rectangles are identical or compatible */
-    ist->st->codec->pix_fmt = PIX_FMT_RGB32;
-
-    ret = av_image_alloc(image, linesize, w, h, PIX_FMT_RGB32, 32);
-    if (ret < 0)
-        return ret;
-    memset(image[0], 0, h * linesize[0]);
-    ist->sub2video.ref = avfilter_get_video_buffer_ref_from_arrays(
-            image, linesize, AV_PERM_READ | AV_PERM_PRESERVE,
-            w, h, PIX_FMT_RGB32);
-    if (!ist->sub2video.ref) {
-        av_free(image[0]);
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
-                                        AVFilterInOut *in)
-{
-    AVFilterContext *first_filter = in->filter_ctx;
-    AVFilter *filter = avfilter_get_by_name("buffer");
-    InputStream *ist = ifilter->ist;
-    AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
-                                         ist->st->time_base;
-    AVRational fr = ist->framerate.num ? ist->framerate :
-                                         ist->st->r_frame_rate;
-    AVRational sar;
-    AVBPrint args;
-    char name[255];
-    int pad_idx = in->pad_idx;
-    int ret;
-
-    if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
-        ret = sub2video_prepare(ist);
-        if (ret < 0)
-            return ret;
-    }
-
-    sar = ist->st->sample_aspect_ratio.num ?
-          ist->st->sample_aspect_ratio :
-          ist->st->codec->sample_aspect_ratio;
-    if(!sar.den)
-        sar = (AVRational){0,1};
-    av_bprint_init(&args, 0, 1);
-    av_bprintf(&args,
-             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
-             "pixel_aspect=%d/%d:sws_param=flags=%d", ist->st->codec->width,
-             ist->st->codec->height, ist->st->codec->pix_fmt,
-             tb.num, tb.den, sar.num, sar.den,
-             SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
-    if (fr.num && fr.den)
-        av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
-    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
-             ist->file_index, ist->st->index);
-
-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter, name,
-                                            args.str, NULL, fg->graph)) < 0)
-        return ret;
-
-    if (ist->framerate.num) {
-        AVFilterContext *setpts;
-
-        snprintf(name, sizeof(name), "force CFR for input from stream %d:%d",
-                 ist->file_index, ist->st->index);
-        if ((ret = avfilter_graph_create_filter(&setpts,
-                                                avfilter_get_by_name("setpts"),
-                                                name, "N", NULL,
-                                                fg->graph)) < 0)
-            return ret;
-
-        if ((ret = avfilter_link(setpts, 0, first_filter, pad_idx)) < 0)
-            return ret;
-
-        first_filter = setpts;
-        pad_idx = 0;
-    }
-
-    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
-        return ret;
-    return 0;
-}
-
-static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
-                                        AVFilterInOut *in)
-{
-    AVFilterContext *first_filter = in->filter_ctx;
-    AVFilter *filter = avfilter_get_by_name("abuffer");
-    InputStream *ist = ifilter->ist;
-    int pad_idx = in->pad_idx;
-    char args[255], name[255];
-    int ret;
-
-    snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s"
-             ":channel_layout=0x%"PRIx64,
-             1, ist->st->codec->sample_rate,
-             ist->st->codec->sample_rate,
-             av_get_sample_fmt_name(ist->st->codec->sample_fmt),
-             ist->st->codec->channel_layout);
-    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
-             ist->file_index, ist->st->index);
-
-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
-                                            name, args, NULL,
-                                            fg->graph)) < 0)
-        return ret;
-
-#define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do {                 \
-    AVFilterContext *filt_ctx;                                              \
-                                                                            \
-    av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi "            \
-           "similarly to -af " filter_name "=%s.\n", arg);                  \
-                                                                            \
-    snprintf(name, sizeof(name), "graph %d %s for input stream %d:%d",      \
-                fg->index, filter_name, ist->file_index, ist->st->index);   \
-    ret = avfilter_graph_create_filter(&filt_ctx,                           \
-                                       avfilter_get_by_name(filter_name),   \
-                                       name, arg, NULL, fg->graph);         \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    ret = avfilter_link(filt_ctx, 0, first_filter, pad_idx);                \
-    if (ret < 0)                                                            \
-        return ret;                                                         \
-                                                                            \
-    first_filter = filt_ctx;                                                  \
-} while (0)
-
-    if (audio_sync_method > 0) {
-        char args[256] = {0};
-
-        av_strlcatf(args, sizeof(args), "min_comp=0.001:min_hard_comp=%f", audio_drift_threshold);
-        if (audio_sync_method > 1)
-            av_strlcatf(args, sizeof(args), ":max_soft_comp=%f", audio_sync_method/(double)ist->st->codec->sample_rate);
-        AUTO_INSERT_FILTER_INPUT("-async", "aresample", args);
-    }
-
-//     if (ost->audio_channels_mapped) {
-//         int i;
-//         AVBPrint pan_buf;
-//         av_bprint_init(&pan_buf, 256, 8192);
-//         av_bprintf(&pan_buf, "0x%"PRIx64,
-//                    av_get_default_channel_layout(ost->audio_channels_mapped));
-//         for (i = 0; i < ost->audio_channels_mapped; i++)
-//             if (ost->audio_channels_map[i] != -1)
-//                 av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
-//         AUTO_INSERT_FILTER_INPUT("-map_channel", "pan", pan_buf.str);
-//         av_bprint_finalize(&pan_buf, NULL);
-//     }
-
-    if (audio_volume != 256) {
-        char args[256];
-
-        snprintf(args, sizeof(args), "%f", audio_volume / 256.);
-        AUTO_INSERT_FILTER_INPUT("-vol", "volume", args);
-    }
-    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
-        return ret;
-
-    return 0;
-}
-
-static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
-                                  AVFilterInOut *in)
-{
-    av_freep(&ifilter->name);
-    DESCRIBE_FILTER_LINK(ifilter, in, 1);
-
-    switch (avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx)) {
-    case AVMEDIA_TYPE_VIDEO: return configure_input_video_filter(fg, ifilter, in);
-    case AVMEDIA_TYPE_AUDIO: return configure_input_audio_filter(fg, ifilter, in);
-    default: av_assert0(0);
-    }
-}
-
-int configure_filtergraph(FilterGraph *fg)
-{
-    AVFilterInOut *inputs, *outputs, *cur;
-    int ret, i, init = !fg->graph, simple = !fg->graph_desc;
-    const char *graph_desc = simple ? fg->outputs[0]->ost->avfilter :
-                                      fg->graph_desc;
-
-    avfilter_graph_free(&fg->graph);
-    if (!(fg->graph = avfilter_graph_alloc()))
-        return AVERROR(ENOMEM);
-
-    if (simple) {
-        OutputStream *ost = fg->outputs[0]->ost;
-        char args[255];
-        snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
-        fg->graph->scale_sws_opts = av_strdup(args);
-    }
-
-    if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
-        return ret;
-
-    if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
-        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' does not have "
-               "exactly one input and output.\n", graph_desc);
-        return AVERROR(EINVAL);
-    }
-
-    for (cur = inputs; !simple && init && cur; cur = cur->next)
-        init_input_filter(fg, cur);
-
-    for (cur = inputs, i = 0; cur; cur = cur->next, i++)
-        if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0)
-            return ret;
-    avfilter_inout_free(&inputs);
-
-    if (!init || simple) {
-        /* we already know the mappings between lavfi outputs and output streams,
-         * so we can finish the setup */
-        for (cur = outputs, i = 0; cur; cur = cur->next, i++)
-            configure_output_filter(fg, fg->outputs[i], cur);
-        avfilter_inout_free(&outputs);
-
-        if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
-            return ret;
-    } else {
-        /* wait until output mappings are processed */
-        for (cur = outputs; cur;) {
-            fg->outputs = grow_array(fg->outputs, sizeof(*fg->outputs),
-                                     &fg->nb_outputs, fg->nb_outputs + 1);
-            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
-                exit_program(1);
-            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
-            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
-            cur = cur->next;
-            fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
-        }
-    }
-
-    return 0;
-}
-
-int ist_in_filtergraph(FilterGraph *fg, InputStream *ist)
-{
-    int i;
-    for (i = 0; i < fg->nb_inputs; i++)
-        if (fg->inputs[i]->ist == ist)
-            return 1;
-    return 0;
-}
-

ffpresets/libvpx-1080p.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=1
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-1080p50_60.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=25
+quality=good
+speed=0
+profile=1
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-360p.ffpreset

@@ -0,0 +1,16 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=0
+qmax=63
+qmin=0
+vb=768k
+
+#ignored unless using -pass 2
+maxrate=1.5M
+minrate=40k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-720p.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=0
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-720p50_60.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=25
+quality=good
+speed=0
+profile=0
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffserver.c

@@ -30,7 +30,7 @@
 #include <string.h>
 #include <stdlib.h>
 #include "libavformat/avformat.h"
-// FIXME those are internal headers, ffserver _really_ shouldn't use them
+// FIXME those are internal headers, avserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
@@ -40,7 +40,6 @@
 #include "libavformat/internal.h"
 #include "libavformat/url.h"
 
-#include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/lfg.h"
 #include "libavutil/dict.h"
@@ -48,8 +47,6 @@
 #include "libavutil/random_seed.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/opt.h"
-#include "libavutil/time.h"
-
 #include <stdarg.h>
 #include <unistd.h>
 #include <fcntl.h>
@@ -58,6 +55,7 @@
 #include <poll.h>
 #endif
 #include <errno.h>
+#include <sys/time.h>
 #include <time.h>
 #include <sys/wait.h>
 #include <signal.h>
@@ -341,7 +339,8 @@ static int resolve_host(struct in_addr *sin_addr, const char *hostname)
     if (!ff_inet_aton(hostname, sin_addr)) {
 #if HAVE_GETADDRINFO
         struct addrinfo *ai, *cur;
-        struct addrinfo hints = { 0 };
+        struct addrinfo hints;
+        memset(&hints, 0, sizeof(hints));
         hints.ai_family = AF_INET;
         if (getaddrinfo(hostname, NULL, &hints, &ai))
             return -1;
@@ -765,7 +764,7 @@ static void start_wait_request(HTTPContext *c, int is_rtsp)
 
 static void http_send_too_busy_reply(int fd)
 {
-    char buffer[400];
+    char buffer[300];
     int len = snprintf(buffer, sizeof(buffer),
                        "HTTP/1.0 503 Server too busy\r\n"
                        "Content-type: text/html\r\n"
@@ -775,7 +774,6 @@ static void http_send_too_busy_reply(int fd)
                        "<p>The number of current connections is %d, and this exceeds the limit of %d.</p>\r\n"
                        "</body></html>\r\n",
                        nb_connections, nb_max_connections);
-    av_assert0(len < sizeof(buffer));
     send(fd, buffer, len, 0);
 }
 
@@ -1568,7 +1566,7 @@ static int http_parse_request(HTTPContext *c)
     if (stream->stream_type == STREAM_TYPE_REDIRECT) {
         c->http_error = 301;
         q = c->buffer;
-        snprintf(q, c->buffer_size,
+        q += snprintf(q, c->buffer_size,
                       "HTTP/1.0 301 Moved\r\n"
                       "Location: %s\r\n"
                       "Content-type: text/html\r\n"
@@ -1576,7 +1574,6 @@ static int http_parse_request(HTTPContext *c)
                       "<html><head><title>Moved</title></head><body>\r\n"
                       "You should be <a href=\"%s\">redirected</a>.\r\n"
                       "</body></html>\r\n", stream->feed_filename, stream->feed_filename);
-        q += strlen(q);
         /* prepare output buffer */
         c->buffer_ptr = c->buffer;
         c->buffer_end = q;
@@ -1607,7 +1604,7 @@ static int http_parse_request(HTTPContext *c)
     if (c->post == 0 && max_bandwidth < current_bandwidth) {
         c->http_error = 503;
         q = c->buffer;
-        snprintf(q, c->buffer_size,
+        q += snprintf(q, c->buffer_size,
                       "HTTP/1.0 503 Server too busy\r\n"
                       "Content-type: text/html\r\n"
                       "\r\n"
@@ -1616,7 +1613,6 @@ static int http_parse_request(HTTPContext *c)
                       "<p>The bandwidth being served (including your stream) is %"PRIu64"kbit/sec, "
                       "and this exceeds the limit of %"PRIu64"kbit/sec.</p>\r\n"
                       "</body></html>\r\n", current_bandwidth, max_bandwidth);
-        q += strlen(q);
         /* prepare output buffer */
         c->buffer_ptr = c->buffer;
         c->buffer_end = q;
@@ -1659,7 +1655,7 @@ static int http_parse_request(HTTPContext *c)
                     q = c->buffer;
                     switch(redir_type) {
                     case REDIR_ASX:
-                        snprintf(q, c->buffer_size,
+                        q += snprintf(q, c->buffer_size,
                                       "HTTP/1.0 200 ASX Follows\r\n"
                                       "Content-type: video/x-ms-asf\r\n"
                                       "\r\n"
@@ -1667,25 +1663,22 @@ static int http_parse_request(HTTPContext *c)
                                       //"<!-- Autogenerated by ffserver -->\r\n"
                                       "<ENTRY><REF HREF=\"http://%s/%s%s\"/></ENTRY>\r\n"
                                       "</ASX>\r\n", hostbuf, filename, info);
-                        q += strlen(q);
                         break;
                     case REDIR_RAM:
-                        snprintf(q, c->buffer_size,
+                        q += snprintf(q, c->buffer_size,
                                       "HTTP/1.0 200 RAM Follows\r\n"
                                       "Content-type: audio/x-pn-realaudio\r\n"
                                       "\r\n"
                                       "# Autogenerated by ffserver\r\n"
                                       "http://%s/%s%s\r\n", hostbuf, filename, info);
-                        q += strlen(q);
                         break;
                     case REDIR_ASF:
-                        snprintf(q, c->buffer_size,
+                        q += snprintf(q, c->buffer_size,
                                       "HTTP/1.0 200 ASF Redirect follows\r\n"
                                       "Content-type: video/x-ms-asf\r\n"
                                       "\r\n"
                                       "[Reference]\r\n"
                                       "Ref1=http://%s/%s%s\r\n", hostbuf, filename, info);
-                        q += strlen(q);
                         break;
                     case REDIR_RTSP:
                         {
@@ -1695,13 +1688,12 @@ static int http_parse_request(HTTPContext *c)
                             p = strrchr(hostname, ':');
                             if (p)
                                 *p = '\0';
-                            snprintf(q, c->buffer_size,
+                            q += snprintf(q, c->buffer_size,
                                           "HTTP/1.0 200 RTSP Redirect follows\r\n"
                                           /* XXX: incorrect mime type ? */
                                           "Content-type: application/x-rtsp\r\n"
                                           "\r\n"
                                           "rtsp://%s:%d/%s\r\n", hostname, ntohs(my_rtsp_addr.sin_port), filename);
-                            q += strlen(q);
                         }
                         break;
                     case REDIR_SDP:
@@ -1710,11 +1702,10 @@ static int http_parse_request(HTTPContext *c)
                             int sdp_data_size, len;
                             struct sockaddr_in my_addr;
 
-                            snprintf(q, c->buffer_size,
+                            q += snprintf(q, c->buffer_size,
                                           "HTTP/1.0 200 OK\r\n"
                                           "Content-type: application/sdp\r\n"
                                           "\r\n");
-                            q += strlen(q);
 
                             len = sizeof(my_addr);
                             getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
@@ -1833,12 +1824,12 @@ static int http_parse_request(HTTPContext *c)
     }
 
     /* prepare http header */
-    c->buffer[0] = 0;
-    av_strlcatf(c->buffer, c->buffer_size, "HTTP/1.0 200 OK\r\n");
+    q = c->buffer;
+    q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 OK\r\n");
     mime_type = c->stream->fmt->mime_type;
     if (!mime_type)
         mime_type = "application/x-octet-stream";
-    av_strlcatf(c->buffer, c->buffer_size, "Pragma: no-cache\r\n");
+    q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Pragma: no-cache\r\n");
 
     /* for asf, we need extra headers */
     if (!strcmp(c->stream->fmt->name,"asf_stream")) {
@@ -1846,11 +1837,10 @@ static int http_parse_request(HTTPContext *c)
 
         c->wmp_client_id = av_lfg_get(&random_state);
 
-        av_strlcatf(c->buffer, c->buffer_size, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id);
+        q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id);
     }
-    av_strlcatf(c->buffer, c->buffer_size, "Content-Type: %s\r\n", mime_type);
-    av_strlcatf(c->buffer, c->buffer_size, "\r\n");
-    q = c->buffer + strlen(c->buffer);
+    q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-Type: %s\r\n", mime_type);
+    q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
 
     /* prepare output buffer */
     c->http_error = 0;
@@ -1861,7 +1851,7 @@ static int http_parse_request(HTTPContext *c)
  send_error:
     c->http_error = 404;
     q = c->buffer;
-    snprintf(q, c->buffer_size,
+    q += snprintf(q, c->buffer_size,
                   "HTTP/1.0 404 Not Found\r\n"
                   "Content-type: text/html\r\n"
                   "\r\n"
@@ -1869,7 +1859,6 @@ static int http_parse_request(HTTPContext *c)
                   "<head><title>404 Not Found</title></head>\n"
                   "<body>%s</body>\n"
                   "</html>\n", msg);
-    q += strlen(q);
     /* prepare output buffer */
     c->buffer_ptr = c->buffer;
     c->buffer_end = q;
@@ -1885,7 +1874,7 @@ static int http_parse_request(HTTPContext *c)
 
 static void fmt_bytecount(AVIOContext *pb, int64_t count)
 {
-    static const char suffix[] = " kMGTP";
+    static const char *suffix = " kMGTP";
     const char *s;
 
     for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++);
@@ -2034,7 +2023,7 @@ static void compute_status(HTTPContext *c)
                         char cpuperc[10];
                         char cpuused[64];
 
-                        if (fscanf(pid_stat, "%9s %63s", cpuperc,
+                        if (fscanf(pid_stat, "%10s %64s", cpuperc,
                                    cpuused) == 2) {
                             avio_printf(pb, "Currently using %s%% of the cpu. Total time used %s.\n",
                                          cpuperc, cpuused);
@@ -2140,13 +2129,12 @@ static int open_input_stream(HTTPContext *c, const char *info)
     char buf[128];
     char input_filename[1024];
     AVFormatContext *s = NULL;
-    int buf_size, i, ret;
+    int i, ret;
     int64_t stream_pos;
 
     /* find file name */
     if (c->stream->feed) {
         strcpy(input_filename, c->stream->feed->feed_filename);
-        buf_size = FFM_PACKET_SIZE;
         /* compute position (absolute time) */
         if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
             if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0)
@@ -2158,7 +2146,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
             stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000;
     } else {
         strcpy(input_filename, c->stream->feed_filename);
-        buf_size = 0;
         /* compute position (relative time) */
         if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
             if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0)
@@ -2174,10 +2161,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
         http_log("could not open %s: %d\n", input_filename, ret);
         return -1;
     }
-
-    /* set buffer size */
-    if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size);
-
     s->flags |= AVFMT_FLAG_GENPTS;
     c->fmt_in = s;
     if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
@@ -2599,11 +2582,8 @@ static int http_start_receive_data(HTTPContext *c)
     if (c->stream->truncate) {
         /* truncate feed file */
         ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE);
+        ftruncate(c->feed_fd, FFM_PACKET_SIZE);
         http_log("Truncating feed file '%s'\n", c->stream->feed_filename);
-        if (ftruncate(c->feed_fd, FFM_PACKET_SIZE) < 0) {
-            http_log("Error truncating feed file: %s\n", strerror(errno));
-            return -1;
-        }
     } else {
         if ((c->stream->feed_write_index = ffm_read_write_index(fd)) < 0) {
             http_log("Error reading write index from feed file: %s\n", strerror(errno));
@@ -2851,7 +2831,7 @@ static int rtsp_parse_request(HTTPContext *c)
     char protocol[32];
     char line[1024];
     int len;
-    RTSPMessageHeader header1 = { 0 }, *header = &header1;
+    RTSPMessageHeader header1, *header = &header1;
 
     c->buffer_ptr[0] = '\0';
     p = c->buffer;
@@ -2877,6 +2857,7 @@ static int rtsp_parse_request(HTTPContext *c)
     }
 
     /* parse each header line */
+    memset(header, 0, sizeof(*header));
     /* skip to next line */
     while (*p != '\n' && *p != '\0')
         p++;
@@ -2937,14 +2918,12 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
 {
     AVFormatContext *avc;
     AVStream *avs = NULL;
-    AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
     int i;
 
     avc =  avformat_alloc_context();
-    if (avc == NULL || !rtp_format) {
+    if (avc == NULL) {
         return -1;
     }
-    avc->oformat = rtp_format;
     av_dict_set(&avc->metadata, "title",
                stream->title[0] ? stream->title : "No Title", 0);
     avc->nb_streams = stream->nb_streams;
@@ -3483,6 +3462,9 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
 {
     AVStream *fst;
 
+    if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams))
+        return NULL;
+
     fst = av_mallocz(sizeof(AVStream));
     if (!fst)
         return NULL;
@@ -3572,7 +3554,7 @@ static void extract_mpeg4_header(AVFormatContext *infile)
     mpeg4_count = 0;
     for(i=0;i<infile->nb_streams;i++) {
         st = infile->streams[i];
-        if (st->codec->codec_id == AV_CODEC_ID_MPEG4 &&
+        if (st->codec->codec_id == CODEC_ID_MPEG4 &&
             st->codec->extradata_size == 0) {
             mpeg4_count++;
         }
@@ -3585,7 +3567,7 @@ static void extract_mpeg4_header(AVFormatContext *infile)
         if (av_read_packet(infile, &pkt) < 0)
             break;
         st = infile->streams[pkt.stream_index];
-        if (st->codec->codec_id == AV_CODEC_ID_MPEG4 &&
+        if (st->codec->codec_id == CODEC_ID_MPEG4 &&
             st->codec->extradata_size == 0) {
             av_freep(&st->codec->extradata);
             /* fill extradata with the header */
@@ -3688,8 +3670,6 @@ static void build_feed_streams(void)
             int matches = 0;
 
             if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) {
-                /* set buffer size */
-                ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
                 /* Now see if it matches */
                 if (s->nb_streams == feed->nb_streams) {
                     matches = 1;
@@ -3776,7 +3756,7 @@ static void build_feed_streams(void)
             s->nb_streams = feed->nb_streams;
             s->streams = feed->streams;
             if (avformat_write_header(s, NULL) < 0) {
-                http_log("Container doesn't support the required parameters\n");
+                http_log("Container doesn't supports the required parameters\n");
                 exit(1);
             }
             /* XXX: need better api */
@@ -3830,6 +3810,9 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
 {
     AVStream *st;
 
+    if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams))
+        return;
+
     /* compute default parameters */
     switch(av->codec_type) {
     case AVMEDIA_TYPE_AUDIO:
@@ -3901,22 +3884,22 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
     memcpy(st->codec, av, sizeof(AVCodecContext));
 }
 
-static enum AVCodecID opt_audio_codec(const char *arg)
+static enum CodecID opt_audio_codec(const char *arg)
 {
     AVCodec *p= avcodec_find_encoder_by_name(arg);
 
     if (p == NULL || p->type != AVMEDIA_TYPE_AUDIO)
-        return AV_CODEC_ID_NONE;
+        return CODEC_ID_NONE;
 
     return p->id;
 }
 
-static enum AVCodecID opt_video_codec(const char *arg)
+static enum CodecID opt_video_codec(const char *arg)
 {
     AVCodec *p= avcodec_find_encoder_by_name(arg);
 
     if (p == NULL || p->type != AVMEDIA_TYPE_VIDEO)
-        return AV_CODEC_ID_NONE;
+        return CODEC_ID_NONE;
 
     return p->id;
 }
@@ -3959,7 +3942,7 @@ static int ffserver_opt_default(const char *opt, const char *arg,
 
 static int ffserver_opt_preset(const char *arg,
                        AVCodecContext *avctx, int type,
-                       enum AVCodecID *audio_id, enum AVCodecID *video_id)
+                       enum CodecID *audio_id, enum CodecID *video_id)
 {
     FILE *f=NULL;
     char filename[1000], tmp[1000], tmp2[1000], line[1000];
@@ -4041,7 +4024,7 @@ static int parse_ffconfig(const char *filename)
     FFStream **last_stream, *stream, *redirect;
     FFStream **last_feed, *feed, *s;
     AVCodecContext audio_enc, video_enc;
-    enum AVCodecID audio_id, video_id;
+    enum CodecID audio_id, video_id;
 
     f = fopen(filename, "r");
     if (!f) {
@@ -4058,8 +4041,8 @@ static int parse_ffconfig(const char *filename)
     stream = NULL;
     feed = NULL;
     redirect = NULL;
-    audio_id = AV_CODEC_ID_NONE;
-    video_id = AV_CODEC_ID_NONE;
+    audio_id = CODEC_ID_NONE;
+    video_id = CODEC_ID_NONE;
 
 #define ERROR(...) report_config_error(filename, line_num, &errors, __VA_ARGS__)
     for(;;) {
@@ -4249,11 +4232,11 @@ static int parse_ffconfig(const char *filename)
                 }
 
                 stream->fmt = ffserver_guess_format(NULL, stream->filename, NULL);
-                avcodec_get_context_defaults3(&video_enc, NULL);
-                avcodec_get_context_defaults3(&audio_enc, NULL);
+                avcodec_get_context_defaults2(&video_enc, AVMEDIA_TYPE_VIDEO);
+                avcodec_get_context_defaults2(&audio_enc, AVMEDIA_TYPE_AUDIO);
 
-                audio_id = AV_CODEC_ID_NONE;
-                video_id = AV_CODEC_ID_NONE;
+                audio_id = CODEC_ID_NONE;
+                video_id = CODEC_ID_NONE;
                 if (stream->fmt) {
                     audio_id = stream->fmt->audio_codec;
                     video_id = stream->fmt->video_codec;
@@ -4335,13 +4318,13 @@ static int parse_ffconfig(const char *filename)
         } else if (!av_strcasecmp(cmd, "AudioCodec")) {
             get_arg(arg, sizeof(arg), &p);
             audio_id = opt_audio_codec(arg);
-            if (audio_id == AV_CODEC_ID_NONE) {
+            if (audio_id == CODEC_ID_NONE) {
                 ERROR("Unknown AudioCodec: %s\n", arg);
             }
         } else if (!av_strcasecmp(cmd, "VideoCodec")) {
             get_arg(arg, sizeof(arg), &p);
             video_id = opt_video_codec(arg);
-            if (video_id == AV_CODEC_ID_NONE) {
+            if (video_id == CODEC_ID_NONE) {
                 ERROR("Unknown VideoCodec: %s\n", arg);
             }
         } else if (!av_strcasecmp(cmd, "MaxTime")) {
@@ -4532,9 +4515,9 @@ static int parse_ffconfig(const char *filename)
             if (stream)
                 video_enc.dark_masking = atof(arg);
         } else if (!av_strcasecmp(cmd, "NoVideo")) {
-            video_id = AV_CODEC_ID_NONE;
+            video_id = CODEC_ID_NONE;
         } else if (!av_strcasecmp(cmd, "NoAudio")) {
-            audio_id = AV_CODEC_ID_NONE;
+            audio_id = CODEC_ID_NONE;
         } else if (!av_strcasecmp(cmd, "ACL")) {
             parse_acl_row(stream, feed, NULL, p, filename, line_num);
         } else if (!av_strcasecmp(cmd, "DynamicACL")) {
@@ -4572,12 +4555,12 @@ static int parse_ffconfig(const char *filename)
                 ERROR("No corresponding <Stream> for </Stream>\n");
             } else {
                 if (stream->feed && stream->fmt && strcmp(stream->fmt->name, "ffm") != 0) {
-                    if (audio_id != AV_CODEC_ID_NONE) {
+                    if (audio_id != CODEC_ID_NONE) {
                         audio_enc.codec_type = AVMEDIA_TYPE_AUDIO;
                         audio_enc.codec_id = audio_id;
                         add_codec(stream, &audio_enc);
                     }
-                    if (video_id != AV_CODEC_ID_NONE) {
+                    if (video_id != CODEC_ID_NONE) {
                         video_enc.codec_type = AVMEDIA_TYPE_VIDEO;
                         video_enc.codec_id = video_id;
                         add_codec(stream, &video_enc);
@@ -4665,12 +4648,13 @@ static void opt_debug(void)
     logfilename[0] = '-';
 }
 
-void show_help_default(const char *opt, const char *arg)
+static int opt_help(const char *opt, const char *arg)
 {
     printf("usage: ffserver [options]\n"
            "Hyper fast multi format Audio/Video streaming server\n");
     printf("\n");
-    show_help_options(options, "Main options:", 0, 0, 0);
+    show_help_options(options, "Main options:\n", 0, 0);
+    return 0;
 }
 
 static const OptionDef options[] = {
@@ -4683,7 +4667,7 @@ static const OptionDef options[] = {
 
 int main(int argc, char **argv)
 {
-    struct sigaction sigact = { { 0 } };
+    struct sigaction sigact;
 
     parse_loglevel(argc, argv, options);
     av_register_all();
@@ -4701,6 +4685,7 @@ int main(int argc, char **argv)
 
     av_lfg_init(&random_state, av_get_random_seed());
 
+    memset(&sigact, 0, sizeof(sigact));
     sigact.sa_handler = handle_child_exit;
     sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
     sigaction(SIGCHLD, &sigact, 0);

libavcodec/8bps.c

@@ -33,27 +33,25 @@
 
 #include <stdio.h>
 #include <stdlib.h>
-#include <string.h>
 
-#include "libavutil/internal.h"
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 
 
-static const enum PixelFormat pixfmt_rgb24[] = {
-    PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE };
+static const enum PixelFormat pixfmt_rgb24[] = {PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE};
 
 /*
  * Decoder context
  */
 typedef struct EightBpsContext {
-    AVCodecContext *avctx;
-    AVFrame pic;
 
-    unsigned char planes;
-    unsigned char planemap[4];
+        AVCodecContext *avctx;
+        AVFrame pic;
 
-    uint32_t pal[256];
+        unsigned char planes;
+        unsigned char planemap[4];
+
+        uint32_t pal[256];
 } EightBpsContext;
 
 
@@ -62,90 +60,91 @@ typedef struct EightBpsContext {
  * Decode a frame
  *
  */
-static int decode_frame(AVCodecContext *avctx, void *data,
-                        int *data_size, AVPacket *avpkt)
+static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
 {
-    const uint8_t *buf = avpkt->data;
-    int buf_size       = avpkt->size;
-    EightBpsContext * const c = avctx->priv_data;
-    const unsigned char *encoded = buf;
-    unsigned char *pixptr, *pixptr_end;
-    unsigned int height = avctx->height; // Real image height
-    unsigned int dlen, p, row;
-    const unsigned char *lp, *dp;
-    unsigned char count;
-    unsigned int planes     = c->planes;
-    unsigned char *planemap = c->planemap;
+        const uint8_t *buf = avpkt->data;
+        int buf_size = avpkt->size;
+        EightBpsContext * const c = avctx->priv_data;
+        const unsigned char *encoded = buf;
+        unsigned char *pixptr, *pixptr_end;
+        unsigned int height = avctx->height; // Real image height
+        unsigned int dlen, p, row;
+        const unsigned char *lp, *dp, *ep;
+        unsigned char count;
+        unsigned int planes = c->planes;
+        unsigned char *planemap = c->planemap;
 
-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
+        if(c->pic.data[0])
+                avctx->release_buffer(avctx, &c->pic);
 
-    c->pic.reference    = 0;
-    c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
-    if (avctx->get_buffer(avctx, &c->pic) < 0){
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-        return -1;
-    }
+        c->pic.reference = 0;
+        c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
+        if(avctx->get_buffer(avctx, &c->pic) < 0){
+                av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+                return -1;
+        }
 
-    /* Set data pointer after line lengths */
-    dp = encoded + planes * (height << 1);
+        ep = encoded + buf_size;
 
-    for (p = 0; p < planes; p++) {
-        /* Lines length pointer for this plane */
-        lp = encoded + p * (height << 1);
+        /* Set data pointer after line lengths */
+        dp = encoded + planes * (height << 1);
 
-        /* Decode a plane */
-        for (row = 0; row < height; row++) {
-            pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
-            pixptr_end = pixptr + c->pic.linesize[0];
-            dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
-            /* Decode a row of this plane */
-            while (dlen > 0) {
-                if (dp + 1 >= buf + buf_size)
-                    return -1;
-                if ((count = *dp++) <= 127) {
-                    count++;
-                    dlen -= count + 1;
-                    if (pixptr + count * planes > pixptr_end)
-                        break;
-                    if (dp + count > buf + buf_size)
-                        return -1;
-                    while (count--) {
-                        *pixptr = *dp++;
-                        pixptr += planes;
-                    }
-                } else {
-                    count = 257 - count;
-                    if (pixptr + count * planes > pixptr_end)
-                        break;
-                    while (count--) {
-                        *pixptr = *dp;
-                        pixptr += planes;
-                    }
-                    dp++;
-                    dlen -= 2;
+        for (p = 0; p < planes; p++) {
+                /* Lines length pointer for this plane */
+                lp = encoded + p * (height << 1);
+
+                /* Decode a plane */
+                for(row = 0; row < height; row++) {
+                        pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
+                        pixptr_end = pixptr + c->pic.linesize[0];
+                        if (ep - lp < row * 2 + 2)
+                            return AVERROR_INVALIDDATA;
+                        dlen = av_be2ne16(*(const unsigned short *)(lp+row*2));
+                        /* Decode a row of this plane */
+                        while(dlen > 0) {
+                                if(ep - dp <= 1) return -1;
+                                if ((count = *dp++) <= 127) {
+                                        count++;
+                                        dlen -= count + 1;
+                                        if (pixptr + count * planes > pixptr_end)
+                                            break;
+                                        if(ep - dp < count) return -1;
+                                        while(count--) {
+                                                *pixptr = *dp++;
+                                                pixptr += planes;
+                                        }
+                                } else {
+                                        count = 257 - count;
+                                        if (pixptr + count * planes > pixptr_end)
+                                            break;
+                                        while(count--) {
+                                                *pixptr = *dp;
+                                                pixptr += planes;
+                                        }
+                                        dp++;
+                                        dlen -= 2;
+                                }
+                        }
                 }
-            }
-        }
-    }
-
-    if (avctx->bits_per_coded_sample <= 8) {
-        const uint8_t *pal = av_packet_get_side_data(avpkt,
-                                                     AV_PKT_DATA_PALETTE,
-                                                     NULL);
-        if (pal) {
-            c->pic.palette_has_changed = 1;
-            memcpy(c->pal, pal, AVPALETTE_SIZE);
         }
 
-        memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
-    }
+        if (avctx->bits_per_coded_sample <= 8) {
+                const uint8_t *pal = av_packet_get_side_data(avpkt,
+                                                             AV_PKT_DATA_PALETTE,
+                                                             NULL);
+                if (pal) {
+                        c->pic.palette_has_changed = 1;
+                        memcpy(c->pal, pal, AVPALETTE_SIZE);
+                }
 
-    *data_size = sizeof(AVFrame);
-    *(AVFrame*)data = c->pic;
+                memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
+        }
 
-    /* always report that the buffer was completely consumed */
-    return buf_size;
+        *data_size = sizeof(AVFrame);
+        *(AVFrame*)data = c->pic;
+
+        /* always report that the buffer was completely consumed */
+        return buf_size;
 }
 
 
@@ -156,47 +155,47 @@ static int decode_frame(AVCodecContext *avctx, void *data,
  */
 static av_cold int decode_init(AVCodecContext *avctx)
 {
-    EightBpsContext * const c = avctx->priv_data;
+        EightBpsContext * const c = avctx->priv_data;
 
-    c->avctx       = avctx;
-    c->pic.data[0] = NULL;
+        c->avctx = avctx;
 
-    avcodec_get_frame_defaults(&c->pic);
-    switch (avctx->bits_per_coded_sample) {
-    case 8:
-        avctx->pix_fmt = PIX_FMT_PAL8;
-        c->planes      = 1;
-        c->planemap[0] = 0; // 1st plane is palette indexes
-        break;
-    case 24:
-        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
-        c->planes      = 3;
-        c->planemap[0] = 2; // 1st plane is red
-        c->planemap[1] = 1; // 2nd plane is green
-        c->planemap[2] = 0; // 3rd plane is blue
-        break;
-    case 32:
-        avctx->pix_fmt = PIX_FMT_RGB32;
-        c->planes      = 4;
+        avcodec_get_frame_defaults(&c->pic);
+        c->pic.data[0] = NULL;
+
+        switch (avctx->bits_per_coded_sample) {
+                case 8:
+                        avctx->pix_fmt = PIX_FMT_PAL8;
+                        c->planes = 1;
+                        c->planemap[0] = 0; // 1st plane is palette indexes
+                        break;
+                case 24:
+                        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
+                        c->planes = 3;
+                        c->planemap[0] = 2; // 1st plane is red
+                        c->planemap[1] = 1; // 2nd plane is green
+                        c->planemap[2] = 0; // 3rd plane is blue
+                        break;
+                case 32:
+                        avctx->pix_fmt = PIX_FMT_RGB32;
+                        c->planes = 4;
 #if HAVE_BIGENDIAN
-        c->planemap[0] = 1; // 1st plane is red
-        c->planemap[1] = 2; // 2nd plane is green
-        c->planemap[2] = 3; // 3rd plane is blue
-        c->planemap[3] = 0; // 4th plane is alpha
+                        c->planemap[0] = 1; // 1st plane is red
+                        c->planemap[1] = 2; // 2nd plane is green
+                        c->planemap[2] = 3; // 3rd plane is blue
+                        c->planemap[3] = 0; // 4th plane is alpha
 #else
-        c->planemap[0] = 2; // 1st plane is red
-        c->planemap[1] = 1; // 2nd plane is green
-        c->planemap[2] = 0; // 3rd plane is blue
-        c->planemap[3] = 3; // 4th plane is alpha
+                        c->planemap[0] = 2; // 1st plane is red
+                        c->planemap[1] = 1; // 2nd plane is green
+                        c->planemap[2] = 0; // 3rd plane is blue
+                        c->planemap[3] = 3; // 4th plane is alpha
 #endif
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n",
-               avctx->bits_per_coded_sample);
-        return -1;
-    }
+                        break;
+                default:
+                        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n", avctx->bits_per_coded_sample);
+                        return -1;
+        }
 
-    return 0;
+  return 0;
 }
 
 
@@ -209,12 +208,12 @@ static av_cold int decode_init(AVCodecContext *avctx)
  */
 static av_cold int decode_end(AVCodecContext *avctx)
 {
-    EightBpsContext * const c = avctx->priv_data;
+        EightBpsContext * const c = avctx->priv_data;
 
-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
+        if (c->pic.data[0])
+                avctx->release_buffer(avctx, &c->pic);
 
-    return 0;
+        return 0;
 }
 
 
@@ -222,7 +221,7 @@ static av_cold int decode_end(AVCodecContext *avctx)
 AVCodec ff_eightbps_decoder = {
     .name           = "8bps",
     .type           = AVMEDIA_TYPE_VIDEO,
-    .id             = AV_CODEC_ID_8BPS,
+    .id             = CODEC_ID_8BPS,
     .priv_data_size = sizeof(EightBpsContext),
     .init           = decode_init,
     .close          = decode_end,

libavcodec/8svx.c

@@ -37,9 +37,8 @@
  * http://aminet.net/mods/smpl/
  */
 
-#include "libavutil/avassert.h"
 #include "avcodec.h"
-#include "libavutil/common.h"
+#include "internal.h"
 
 /** decoder context */
 typedef struct EightSvxContext {
@@ -113,25 +112,17 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
     /* decode and interleave the first packet */
     if (!esc->samples && avpkt) {
         uint8_t *deinterleaved_samples, *p = NULL;
-        int packet_size = avpkt->size;
 
-        if (packet_size % avctx->channels) {
-            av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
-            if (packet_size < avctx->channels)
-                return packet_size;
-            packet_size -= packet_size % avctx->channels;
-        }
-        esc->samples_size = !esc->table ?
-            packet_size : avctx->channels + (packet_size-avctx->channels) * 2;
+        esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
+            avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
         if (!(esc->samples = av_malloc(esc->samples_size)))
             return AVERROR(ENOMEM);
 
         /* decompress */
-        if (esc->table) {
+        if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
             const uint8_t *buf = avpkt->data;
-            uint8_t *dst;
             int buf_size = avpkt->size;
-            int i, n = esc->samples_size;
+            int n = esc->samples_size;
 
             if (buf_size < 2) {
                 av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
@@ -139,15 +130,15 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
             }
             if (!(deinterleaved_samples = av_mallocz(n)))
                 return AVERROR(ENOMEM);
-            dst = p = deinterleaved_samples;
+            p = deinterleaved_samples;
 
             /* the uncompressed starting value is contained in the first byte */
-            dst = deinterleaved_samples;
-            for (i = 0; i < avctx->channels; i++) {
-                delta_decode(dst, buf + 1, buf_size / avctx->channels - 1, buf[0], esc->table);
-                buf += buf_size / avctx->channels;
-                dst += n / avctx->channels - 1;
-            }
+            if (avctx->channels == 2) {
+                delta_decode(deinterleaved_samples      , buf+1, buf_size/2-1, buf[0], esc->table);
+                buf += buf_size/2;
+                delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
+            } else
+                delta_decode(deinterleaved_samples      , buf+1, buf_size-1  , buf[0], esc->table);
         } else {
             deinterleaved_samples = avpkt->data;
         }
@@ -160,9 +151,8 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     /* get output buffer */
-    av_assert1(!(esc->samples_size % avctx->channels || esc->samples_idx % avctx->channels));
-    esc->frame.nb_samples = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx)  / avctx->channels;
-    if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
+    esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1)  / avctx->channels;
+    if ((ret = ff_get_buffer(avctx, &esc->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
@@ -177,7 +167,7 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
         *dst++ = *src++ + 128;
     esc->samples_idx += out_data_size;
 
-    return esc->table ?
+    return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
         (avctx->frame_number == 0)*2 + out_data_size / 2 :
         out_data_size;
 }
@@ -192,10 +182,10 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
     }
 
     switch (avctx->codec->id) {
-    case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci;    break;
-    case AV_CODEC_ID_8SVX_EXP: esc->table = exponential;  break;
-    case AV_CODEC_ID_PCM_S8_PLANAR:
-    case AV_CODEC_ID_8SVX_RAW: esc->table = NULL;         break;
+    case CODEC_ID_8SVX_FIB: esc->table = fibonacci;    break;
+    case CODEC_ID_8SVX_EXP: esc->table = exponential;  break;
+    case CODEC_ID_PCM_S8_PLANAR:
+    case CODEC_ID_8SVX_RAW: esc->table = NULL;         break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
         return AVERROR_INVALIDDATA;
@@ -219,11 +209,10 @@ static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
     return 0;
 }
 
-#if CONFIG_EIGHTSVX_FIB_DECODER
 AVCodec ff_eightsvx_fib_decoder = {
   .name           = "8svx_fib",
   .type           = AVMEDIA_TYPE_AUDIO,
-  .id             = AV_CODEC_ID_8SVX_FIB,
+  .id             = CODEC_ID_8SVX_FIB,
   .priv_data_size = sizeof (EightSvxContext),
   .init           = eightsvx_decode_init,
   .decode         = eightsvx_decode_frame,
@@ -231,12 +220,11 @@ AVCodec ff_eightsvx_fib_decoder = {
   .capabilities   = CODEC_CAP_DR1,
   .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
 };
-#endif
-#if CONFIG_EIGHTSVX_EXP_DECODER
+
 AVCodec ff_eightsvx_exp_decoder = {
   .name           = "8svx_exp",
   .type           = AVMEDIA_TYPE_AUDIO,
-  .id             = AV_CODEC_ID_8SVX_EXP,
+  .id             = CODEC_ID_8SVX_EXP,
   .priv_data_size = sizeof (EightSvxContext),
   .init           = eightsvx_decode_init,
   .decode         = eightsvx_decode_frame,
@@ -244,12 +232,11 @@ AVCodec ff_eightsvx_exp_decoder = {
   .capabilities   = CODEC_CAP_DR1,
   .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
 };
-#endif
-#if CONFIG_PCM_S8_PLANAR_DECODER
+
 AVCodec ff_pcm_s8_planar_decoder = {
     .name           = "pcm_s8_planar",
     .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_PCM_S8_PLANAR,
+    .id             = CODEC_ID_PCM_S8_PLANAR,
     .priv_data_size = sizeof(EightSvxContext),
     .init           = eightsvx_decode_init,
     .close          = eightsvx_decode_close,
@@ -257,4 +244,3 @@ AVCodec ff_pcm_s8_planar_decoder = {
     .capabilities   = CODEC_CAP_DR1,
     .long_name      = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
 };
-#endif

libavcodec/Makefile

@@ -3,22 +3,13 @@ include $(SUBDIR)../config.mak
 NAME = avcodec
 FFLIBS = avutil
 
-HEADERS = avcodec.h                                                     \
-          avfft.h                                                       \
-          dxva2.h                                                       \
-          old_codec_ids.h                                               \
-          vaapi.h                                                       \
-          vda.h                                                         \
-          vdpau.h                                                       \
-          version.h                                                     \
-          xvmc.h                                                        \
+HEADERS = avcodec.h avfft.h dxva2.h opt.h vaapi.h vda.h vdpau.h version.h xvmc.h
 
 OBJS = allcodecs.o                                                      \
        audioconvert.o                                                   \
        avpacket.o                                                       \
        bitstream.o                                                      \
        bitstream_filter.o                                               \
-       codec_desc.o                                                     \
        dsputil.o                                                        \
        faanidct.o                                                       \
        fmtconvert.o                                                     \
@@ -34,14 +25,13 @@ OBJS = allcodecs.o                                                      \
        utils.o                                                          \
 
 # parts needed for many different codecs
-OBJS-$(CONFIG_AANDCTTABLES)            += aandcttab.o
+OBJS-$(CONFIG_AANDCT)                  += aandcttab.o
 OBJS-$(CONFIG_AC3DSP)                  += ac3dsp.o
 OBJS-$(CONFIG_CRYSTALHD)               += crystalhd.o
-OBJS-$(CONFIG_DCT)                     += dct.o dct32_fixed.o dct32_float.o
-OBJS-$(CONFIG_DWT)                     += dwt.o snow.o
-OBJS-$(CONFIG_DXVA2)                   += dxva2.o
 OBJS-$(CONFIG_ENCODERS)                += faandct.o jfdctfst.o jfdctint.o
-OBJS-$(CONFIG_ERROR_RESILIENCE)        += error_resilience.o
+OBJS-$(CONFIG_DCT)                     += dct.o dct32_fixed.o dct32_float.o
+OBJS-$(CONFIG_DWT)                     += dwt.o
+OBJS-$(CONFIG_DXVA2)                   += dxva2.o
 FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
 OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
                                           $(FFT-OBJS-yes)
@@ -49,36 +39,28 @@ OBJS-$(CONFIG_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
-OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
 OBJS-$(CONFIG_LSP)                     += lsp.o
 OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o
 OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
-                                          mpegaudiodsp_data.o           \
                                           mpegaudiodsp_fixed.o          \
                                           mpegaudiodsp_float.o
-OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideo_motion.o
-OBJS-$(CONFIG_MPEGVIDEOENC)            += mpegvideo_enc.o mpeg12data.o  \
-                                          motion_est.o ratecontrol.o
-OBJS-$(CONFIG_RANGECODER)              += rangecoder.o
 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o
 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)
 OBJS-$(CONFIG_SINEWIN)                 += sinewin.o
 OBJS-$(CONFIG_VAAPI)                   += vaapi.o
+OBJS-$(CONFIG_VDA)                     += vda.o
 OBJS-$(CONFIG_VDPAU)                   += vdpau.o
-OBJS-$(CONFIG_VP3DSP)                  += vp3dsp.o
 
 # decoders/encoders/hardware accelerators
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
 OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \
-                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp.o aacpsdsp.o
+                                          aacadtsdec.o mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                           aacpsy.o aactab.o      \
                                           psymodel.o iirfilter.o \
-                                          mpeg4audio.o kbdwin.o  \
-                                          audio_frame_queue.o
+                                          mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
@@ -94,12 +76,8 @@ OBJS-$(CONFIG_AMRNB_DECODER)           += amrnbdec.o celp_filters.o   \
 OBJS-$(CONFIG_AMRWB_DECODER)           += amrwbdec.o celp_filters.o   \
                                           celp_math.o acelp_filters.o \
                                           acelp_vectors.o             \
-                                          acelp_pitch_delay.o
+                                          acelp_pitch_delay.o lsp.o
 OBJS-$(CONFIG_AMV_DECODER)             += sp5xdec.o mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpeg.o           \
-                                          mpegvideo_enc.o motion_est.o \
-                                          ratecontrol.o mpeg12data.o   \
-                                          mpegvideo.o
 OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
@@ -113,19 +91,14 @@ OBJS-$(CONFIG_ATRAC1_DECODER)          += atrac1.o atrac.o
 OBJS-$(CONFIG_ATRAC3_DECODER)          += atrac3.o atrac.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
-OBJS-$(CONFIG_AVRP_DECODER)            += avrndec.o
 OBJS-$(CONFIG_AVRP_DECODER)            += r210dec.o
 OBJS-$(CONFIG_AVRP_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
-OBJS-$(CONFIG_AVUI_DECODER)            += avuidec.o
-OBJS-$(CONFIG_AVUI_ENCODER)            += avuienc.o
-OBJS-$(CONFIG_AYUV_DECODER)            += v408dec.o
-OBJS-$(CONFIG_AYUV_ENCODER)            += v408enc.o
 OBJS-$(CONFIG_BETHSOFTVID_DECODER)     += bethsoftvideo.o
 OBJS-$(CONFIG_BFI_DECODER)             += bfi.o
 OBJS-$(CONFIG_BINK_DECODER)            += bink.o binkdsp.o
-OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o wma_common.o
-OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o wma_common.o
+OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o
+OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o
 OBJS-$(CONFIG_BINTEXT_DECODER)         += bintext.o cga_data.o
 OBJS-$(CONFIG_BMP_DECODER)             += bmp.o msrledec.o
 OBJS-$(CONFIG_BMP_ENCODER)             += bmpenc.o
@@ -133,25 +106,24 @@ OBJS-$(CONFIG_BMV_VIDEO_DECODER)       += bmv.o
 OBJS-$(CONFIG_BMV_AUDIO_DECODER)       += bmv.o
 OBJS-$(CONFIG_C93_DECODER)             += c93.o
 OBJS-$(CONFIG_CAVS_DECODER)            += cavs.o cavsdec.o cavsdsp.o \
-                                          cavsdata.o mpeg12data.o
+                                          mpeg12data.o mpegvideo.o
 OBJS-$(CONFIG_CDGRAPHICS_DECODER)      += cdgraphics.o
-OBJS-$(CONFIG_CDXL_DECODER)            += cdxl.o
 OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
 OBJS-$(CONFIG_CLJR_DECODER)            += cljr.o
 OBJS-$(CONFIG_CLJR_ENCODER)            += cljr.o
-OBJS-$(CONFIG_CLLC_DECODER)            += cllc.o
 OBJS-$(CONFIG_COOK_DECODER)            += cook.o
-OBJS-$(CONFIG_CPIA_DECODER)            += cpia.o
 OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
-OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
-                                          dca_parser.o synth_filter.o
+OBJS-$(CONFIG_DCA_DECODER)             += dca.o synth_filter.o dcadsp.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                           dirac_arith.o mpeg12data.o dwt.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
 OBJS-$(CONFIG_DNXHD_DECODER)           += dnxhddec.o dnxhddata.o
-OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o
+OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o       \
+                                          mpegvideo_enc.o motion_est.o \
+                                          ratecontrol.o mpeg12data.o   \
+                                          mpegvideo.o
 OBJS-$(CONFIG_DPX_DECODER)             += dpx.o
 OBJS-$(CONFIG_DPX_ENCODER)             += dpxenc.o
 OBJS-$(CONFIG_DSICINAUDIO_DECODER)     += dsicinav.o
@@ -160,31 +132,34 @@ OBJS-$(CONFIG_DVBSUB_DECODER)          += dvbsubdec.o
 OBJS-$(CONFIG_DVBSUB_ENCODER)          += dvbsub.o
 OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o
 OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o
-OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o dv_profile.o
-OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o dv_profile.o
+OBJS-$(CONFIG_DVVIDEO_DECODER)         += dv.o dvdata.o
+OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_DXTORY_DECODER)          += dxtory.o
 OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3_data.o
-OBJS-$(CONFIG_EAC3_ENCODER)            += eac3enc.o eac3_data.o
+OBJS-$(CONFIG_EAC3_ENCODER)            += eac3enc.o ac3enc.o ac3enc_float.o \
+                                          ac3tab.o ac3.o kbdwin.o eac3_data.o
 OBJS-$(CONFIG_EACMV_DECODER)           += eacmv.o
 OBJS-$(CONFIG_EAMAD_DECODER)           += eamad.o eaidct.o mpeg12.o \
-                                          mpeg12data.o
+                                          mpeg12data.o mpegvideo.o  \
+                                          error_resilience.o
 OBJS-$(CONFIG_EATGQ_DECODER)           += eatgq.o eaidct.o
 OBJS-$(CONFIG_EATGV_DECODER)           += eatgv.o
 OBJS-$(CONFIG_EATQI_DECODER)           += eatqi.o eaidct.o mpeg12.o \
-                                          mpeg12data.o
+                                          mpeg12data.o mpegvideo.o  \
+                                          error_resilience.o
 OBJS-$(CONFIG_EIGHTBPS_DECODER)        += 8bps.o
 OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER)    += 8svx.o
 OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
+OBJS-$(CONFIG_EIGHTSVX_RAW_DECODER)    += 8svx.o
 OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
 OBJS-$(CONFIG_ESCAPE130_DECODER)       += escape130.o
-OBJS-$(CONFIG_EXR_DECODER)             += exr.o
-OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o
-OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o
+OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o rangecoder.o
+OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o
 OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o flacdsp.o
+OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
 OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
@@ -195,32 +170,39 @@ OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
-                                          celp_filters.o
-OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
+                                          celp_filters.o celp_math.o
+OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o
 OBJS-$(CONFIG_G729_DECODER)            += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
 OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o
 OBJS-$(CONFIG_GSM_DECODER)             += gsmdec.o gsmdec_data.o msgsmdec.o
 OBJS-$(CONFIG_GSM_MS_DECODER)          += gsmdec.o gsmdec_data.o msgsmdec.o
-OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o h261data.o
-OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o h261data.o
+OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o \
+                                          mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o             \
+                                          mpegvideo_enc.o motion_est.o \
+                                          ratecontrol.o mpeg12data.o   \
+                                          mpegvideo.o
 OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                           mpeg4video.o mpeg4videodec.o flvdec.o\
-                                          intelh263dec.o
+                                          intelh263dec.o mpegvideo.o           \
+                                          error_resilience.o
 OBJS-$(CONFIG_H263_VAAPI_HWACCEL)      += vaapi_mpeg4.o
-OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
-                                          h263.o ituh263enc.o flvenc.o
+OBJS-$(CONFIG_H263_ENCODER)            += mpegvideo_enc.o mpeg4video.o      \
+                                          mpeg4videoenc.o motion_est.o      \
+                                          ratecontrol.o h263.o ituh263enc.o \
+                                          flvenc.o mpeg12data.o             \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o                               \
                                           h264_loopfilter.o h264_direct.o      \
                                           cabac.o h264_sei.o h264_ps.o         \
-                                          h264_refs.o h264_cavlc.o h264_cabac.o
+                                          h264_refs.o h264_cavlc.o h264_cabac.o\
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_H264_DXVA2_HWACCEL)      += dxva2_h264.o
 OBJS-$(CONFIG_H264_VAAPI_HWACCEL)      += vaapi_h264.o
 OBJS-$(CONFIG_H264_VDA_HWACCEL)        += vda_h264.o
-OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o
-OBJS-$(CONFIG_IAC_DECODER)             += imc.o
 OBJS-$(CONFIG_IDCIN_DECODER)           += idcinvideo.o
 OBJS-$(CONFIG_IDF_DECODER)             += bintext.o cga_data.o
 OBJS-$(CONFIG_IFF_BYTERUN1_DECODER)    += iff.o
@@ -232,7 +214,6 @@ OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
-OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
 OBJS-$(CONFIG_JPEG2000_DECODER)        += j2kdec.o mqcdec.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEGLS_DECODER)          += jpeglsdec.o jpegls.o \
@@ -242,21 +223,25 @@ OBJS-$(CONFIG_JV_DECODER)              += jvdec.o
 OBJS-$(CONFIG_KGV1_DECODER)            += kgv1dec.o
 OBJS-$(CONFIG_KMVC_DECODER)            += kmvc.o
 OBJS-$(CONFIG_LAGARITH_DECODER)        += lagarith.o lagarithrac.o
-OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpegenc.o mjpeg.o
+OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpegenc.o mjpeg.o \
+                                          mpegvideo_enc.o motion_est.o  \
+                                          ratecontrol.o mpeg12data.o    \
+                                          mpegvideo.o
 OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
 OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
-OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
+OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o
+OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o           \
+                                          mpegvideo_enc.o motion_est.o \
+                                          ratecontrol.o mpeg12data.o   \
+                                          mpegvideo.o
 OBJS-$(CONFIG_MJPEGB_DECODER)          += mjpegbdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_MLP_DECODER)             += mlpdec.o mlpdsp.o
 OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
 OBJS-$(CONFIG_MOTIONPIXELS_DECODER)    += motionpixels.o
-OBJS-$(CONFIG_MOVTEXT_DECODER)         += movtextdec.o ass.o
-OBJS-$(CONFIG_MOVTEXT_ENCODER)         += movtextenc.o ass_split.o
 OBJS-$(CONFIG_MP1_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o \
                                           mpegaudio.o mpegaudiodata.o
 OBJS-$(CONFIG_MP1FLOAT_DECODER)        += mpegaudiodec_float.o mpegaudiodecheader.o \
@@ -264,7 +249,7 @@ OBJS-$(CONFIG_MP1FLOAT_DECODER)        += mpegaudiodec_float.o mpegaudiodecheade
 OBJS-$(CONFIG_MP2_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o \
                                           mpegaudio.o mpegaudiodata.o
 OBJS-$(CONFIG_MP2_ENCODER)             += mpegaudioenc.o mpegaudio.o \
-                                          mpegaudiodata.o mpegaudiodsp_data.o
+                                          mpegaudiodata.o
 OBJS-$(CONFIG_MP2FLOAT_DECODER)        += mpegaudiodec_float.o mpegaudiodecheader.o \
                                           mpegaudio.o mpegaudiodata.o
 OBJS-$(CONFIG_MP3ADU_DECODER)          += mpegaudiodec.o mpegaudiodecheader.o \
@@ -281,46 +266,49 @@ OBJS-$(CONFIG_MP3_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o \
                                           mpegaudio.o mpegaudiodata.o
 OBJS-$(CONFIG_MP3FLOAT_DECODER)        += mpegaudiodec_float.o mpegaudiodecheader.o \
                                           mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MPC7_DECODER)            += mpc7.o mpc.o
-OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o
+OBJS-$(CONFIG_MPC7_DECODER)            += mpc7.o mpc.o mpegaudiodec.o      \
+                                          mpegaudiodecheader.o mpegaudio.o \
+                                          mpegaudiodata.o
+OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o mpegaudiodec.o      \
+                                          mpegaudiodecheader.o mpegaudio.o \
+                                          mpegaudiodata.o
 OBJS-$(CONFIG_MPEGVIDEO_DECODER)       += mpeg12.o mpeg12data.o \
                                           mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEG_XVMC_DECODER)       += mpegvideo_xvmc.o
-OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
-                                          timecode.o
+OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12.o mpeg12data.o \
+                                          mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpegvideo_enc.o \
+                                          timecode.o                  \
+                                          motion_est.o ratecontrol.o  \
+                                          mpeg12.o mpeg12data.o       \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)     += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)     += vaapi_mpeg2.o
-OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
-                                          timecode.o
+OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12.o mpeg12data.o \
+                                          mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpegvideo_enc.o \
+                                          timecode.o                  \
+                                          motion_est.o ratecontrol.o  \
+                                          mpeg12.o mpeg12data.o       \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)     += vaapi_mpeg4.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
                                           h263.o ituh263dec.o mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
                                           h263.o ituh263dec.o mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
-OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
-OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
-OBJS-$(CONFIG_MSS2_DECODER)            += mss2.o mss12.o mss2dsp.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
-OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o mss34dsp.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
-OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o \
-                                          audio_frame_queue.o
+OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
 OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
-OBJS-$(CONFIG_PAF_VIDEO_DECODER)       += paf.o
-OBJS-$(CONFIG_PAF_AUDIO_DECODER)       += paf.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o pnm.o
 OBJS-$(CONFIG_PBM_DECODER)             += pnmdec.o pnm.o
@@ -333,20 +321,20 @@ OBJS-$(CONFIG_PGMYUV_DECODER)          += pnmdec.o pnm.o
 OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o pnm.o
 OBJS-$(CONFIG_PGSSUB_DECODER)          += pgssubdec.o
 OBJS-$(CONFIG_PICTOR_DECODER)          += pictordec.o cga_data.o
-OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o pngdsp.o
+OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o
 OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
 OBJS-$(CONFIG_PPM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o
-OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
-OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_ANATOLIY_ENCODER) += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_KOSTYA_ENCODER)   += proresenc_kostya.o proresdata.o proresdsp.o
+OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o
+OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o
+OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
-OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \
+OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o celp_math.o         \
                                           celp_filters.o acelp_vectors.o \
                                           acelp_filters.o
-OBJS-$(CONFIG_QDM2_DECODER)            += qdm2.o
+OBJS-$(CONFIG_QDM2_DECODER)            += qdm2.o mpegaudiodec.o            \
+                                          mpegaudiodecheader.o mpegaudio.o \
+                                          mpegaudiodata.o
 OBJS-$(CONFIG_QDRAW_DECODER)           += qdrw.o
 OBJS-$(CONFIG_QPEG_DECODER)            += qpeg.o
 OBJS-$(CONFIG_QTRLE_DECODER)           += qtrle.o
@@ -356,13 +344,10 @@ OBJS-$(CONFIG_R10K_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_R210_DECODER)            += r210dec.o
 OBJS-$(CONFIG_R210_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
-OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o \
-                                          audio_frame_queue.o
-OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_filters.o
-OBJS-$(CONFIG_RALF_DECODER)            += ralf.o
+OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o
+OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
 OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o
-OBJS-$(CONFIG_REALTEXT_DECODER)        += realtextdec.o ass.o
 OBJS-$(CONFIG_RL2_DECODER)             += rl2.o
 OBJS-$(CONFIG_ROQ_DECODER)             += roqvideodec.o roqvideo.o
 OBJS-$(CONFIG_ROQ_ENCODER)             += roqvideoenc.o roqvideo.o elbg.o
@@ -373,11 +358,11 @@ OBJS-$(CONFIG_RV10_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV10_ENCODER)            += rv10enc.o
 OBJS-$(CONFIG_RV20_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV20_ENCODER)            += rv20enc.o
-OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o
-OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o
-OBJS-$(CONFIG_SAMI_DECODER)            += samidec.o ass.o
+OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o \
+                                          mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_S302M_DECODER)           += s302m.o
-OBJS-$(CONFIG_SANM_DECODER)            += sanm.o
 OBJS-$(CONFIG_SGI_DECODER)             += sgidec.o
 OBJS-$(CONFIG_SGI_ENCODER)             += sgienc.o rle.o
 OBJS-$(CONFIG_SHORTEN_DECODER)         += shorten.o
@@ -388,9 +373,12 @@ OBJS-$(CONFIG_SIPR_DECODER)            += sipr.o acelp_pitch_delay.o \
 OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
 OBJS-$(CONFIG_SMACKER_DECODER)         += smacker.o
 OBJS-$(CONFIG_SMC_DECODER)             += smc.o
-OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o
-OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o              \
-                                          h263.o ituh263enc.o
+OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o rangecoder.o
+OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o rangecoder.o    \
+                                          motion_est.o ratecontrol.o       \
+                                          h263.o mpegvideo.o               \
+                                          error_resilience.o ituh263enc.o  \
+                                          mpegvideo_enc.o mpeg12data.o
 OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
 OBJS-$(CONFIG_SONIC_ENCODER)           += sonic.o
@@ -398,44 +386,41 @@ OBJS-$(CONFIG_SONIC_LS_ENCODER)        += sonic.o
 OBJS-$(CONFIG_SP5X_DECODER)            += sp5xdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_SRT_DECODER)             += srtdec.o ass.o
 OBJS-$(CONFIG_SRT_ENCODER)             += srtenc.o ass_split.o
-OBJS-$(CONFIG_SUBRIP_DECODER)          += srtdec.o ass.o
-OBJS-$(CONFIG_SUBRIP_ENCODER)          += srtenc.o ass_split.o
-OBJS-$(CONFIG_SUBVIEWER_DECODER)       += subviewerdec.o ass.o
 OBJS-$(CONFIG_SUNRAST_DECODER)         += sunrast.o
-OBJS-$(CONFIG_SUNRAST_ENCODER)         += sunrastenc.o
-OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o svq13.o h263.o
+OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o h263.o \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_SVQ1_ENCODER)            += svq1enc.o svq1.o    \
-                                          h263.o ituh263enc.o
-OBJS-$(CONFIG_SVQ3_DECODER)            += svq3.o svq13.o h263.o h264.o        \
+                                          motion_est.o h263.o \
+                                          mpegvideo.o error_resilience.o \
+                                          ituh263enc.o mpegvideo_enc.o   \
+                                          ratecontrol.o mpeg12data.o
+OBJS-$(CONFIG_SVQ3_DECODER)            += h264.o svq3.o                       \
                                           h264_loopfilter.o h264_direct.o     \
                                           h264_sei.o h264_ps.o h264_refs.o    \
-                                          h264_cavlc.o h264_cabac.o cabac.o
+                                          h264_cavlc.o h264_cabac.o cabac.o   \
+                                          mpegvideo.o error_resilience.o      \
+                                          svq1dec.o svq1.o h263.o
 OBJS-$(CONFIG_TARGA_DECODER)           += targa.o
 OBJS-$(CONFIG_TARGA_ENCODER)           += targaenc.o rle.o
 OBJS-$(CONFIG_THEORA_DECODER)          += xiph.o
 OBJS-$(CONFIG_THP_DECODER)             += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
-OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o
-OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o tiff_data.o
+OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o
+OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o
 OBJS-$(CONFIG_TMV_DECODER)             += tmv.o cga_data.o
-OBJS-$(CONFIG_TRUEHD_DECODER)          += mlpdec.o mlpdsp.o
 OBJS-$(CONFIG_TRUEMOTION1_DECODER)     += truemotion1.o
 OBJS-$(CONFIG_TRUEMOTION2_DECODER)     += truemotion2.o
 OBJS-$(CONFIG_TRUESPEECH_DECODER)      += truespeech.o
 OBJS-$(CONFIG_TSCC_DECODER)            += tscc.o msrledec.o
-OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
 OBJS-$(CONFIG_TTA_DECODER)             += tta.o
-OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvq.o
+OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvq.o celp_math.o
 OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
-OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideodec.o utvideo.o
-OBJS-$(CONFIG_UTVIDEO_ENCODER)         += utvideoenc.o utvideo.o
+OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideo.o
 OBJS-$(CONFIG_V210_DECODER)            += v210dec.o
 OBJS-$(CONFIG_V210_ENCODER)            += v210enc.o
 OBJS-$(CONFIG_V308_DECODER)            += v308dec.o
 OBJS-$(CONFIG_V308_ENCODER)            += v308enc.o
-OBJS-$(CONFIG_V408_DECODER)            += v408dec.o
-OBJS-$(CONFIG_V408_ENCODER)            += v408enc.o
 OBJS-$(CONFIG_V410_DECODER)            += v410dec.o
 OBJS-$(CONFIG_V410_ENCODER)            += v410enc.o
 OBJS-$(CONFIG_V210X_DECODER)           += v210x.o
@@ -455,30 +440,29 @@ OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbis.o \
                                           vorbis_data.o xiph.o
 OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                           vorbis_data.o
-OBJS-$(CONFIG_VP3_DECODER)             += vp3.o
+OBJS-$(CONFIG_VP3_DECODER)             += vp3.o vp3dsp.o
 OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
-                                          vp56rac.o
+                                          vp3dsp.o vp56rac.o
 OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
-                                          vp6dsp.o vp56rac.o
+                                          vp3dsp.o vp6dsp.o vp56rac.o
 OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
-OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o
-OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
-OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
-OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
+OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma.o
+OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o
+OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o aactab.o
 OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
-                                          celp_filters.o \
+                                          celp_math.o celp_filters.o \
                                           acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o        \
                                           msmpeg4.o msmpeg4data.o \
                                           intrax8.o intrax8dsp.o
 OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
-                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o \
+                                          msmpeg4.o msmpeg4data.o \
                                           mpeg4videodec.o ituh263dec.o h263dec.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
@@ -486,8 +470,6 @@ OBJS-$(CONFIG_XAN_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_XAN_WC3_DECODER)         += xan.o
 OBJS-$(CONFIG_XAN_WC4_DECODER)         += xxan.o
 OBJS-$(CONFIG_XBIN_DECODER)            += bintext.o cga_data.o
-OBJS-$(CONFIG_XBM_DECODER)             += xbmdec.o
-OBJS-$(CONFIG_XBM_ENCODER)             += xbmenc.o
 OBJS-$(CONFIG_XL_DECODER)              += xl.o
 OBJS-$(CONFIG_XSUB_DECODER)            += xsubdec.o
 OBJS-$(CONFIG_XSUB_ENCODER)            += xsubenc.o
@@ -498,7 +480,6 @@ OBJS-$(CONFIG_Y41P_ENCODER)            += y41penc.o
 OBJS-$(CONFIG_YOP_DECODER)             += yop.o
 OBJS-$(CONFIG_YUV4_DECODER)            += yuv4dec.o
 OBJS-$(CONFIG_YUV4_ENCODER)            += yuv4enc.o
-OBJS-$(CONFIG_ZEROCODEC_DECODER)       += zerocodec.o
 OBJS-$(CONFIG_ZLIB_DECODER)            += lcldec.o
 OBJS-$(CONFIG_ZLIB_ENCODER)            += lclenc.o
 OBJS-$(CONFIG_ZMBV_DECODER)            += zmbv.o
@@ -593,105 +574,88 @@ OBJS-$(CONFIG_ADPCM_THP_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o
 
 # libavformat dependencies
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
 OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o  \
                                           ac3tab.o
-OBJS-$(CONFIG_DV_DEMUXER)              += dv_profile.o
-OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o timecode.o
-OBJS-$(CONFIG_FLAC_DEMUXER)            += flac.o flacdata.o vorbis_data.o \
-                                          vorbis_parser.o xiph.o
-OBJS-$(CONFIG_FLAC_MUXER)              += flac.o flacdata.o vorbis_data.o
+OBJS-$(CONFIG_DV_DEMUXER)              += dvdata.o
+OBJS-$(CONFIG_DV_MUXER)                += dvdata.o timecode.o
+OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o
+OBJS-$(CONFIG_FLAC_MUXER)              += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLV_DEMUXER)             += mpeg4audio.o
 OBJS-$(CONFIG_GXF_DEMUXER)             += mpeg12data.o
 OBJS-$(CONFIG_IFF_DEMUXER)             += iff.o
-OBJS-$(CONFIG_ISMV_MUXER)              += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_LATM_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER)    += xiph.o mpeg4audio.o vorbis_data.o \
-                                          flac.o flacdata.o
+                                          flacdec.o flacdata.o flac.o
 OBJS-$(CONFIG_MATROSKA_DEMUXER)        += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_MATROSKA_MUXER)          += mpeg4audio.o mpegaudiodata.o  \
-                                          flac.o flacdata.o vorbis_data.o xiph.o
-OBJS-$(CONFIG_MP2_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
+OBJS-$(CONFIG_MATROSKA_MUXER)          += xiph.o mpeg4audio.o \
+                                          flacdec.o flacdata.o flac.o \
+                                          mpegaudiodata.o vorbis_data.o
 OBJS-$(CONFIG_MP3_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
 OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o ac3tab.o timecode.o
 OBJS-$(CONFIG_MOV_MUXER)               += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_MPEGTS_MUXER)            += mpeg4audio.o
+OBJS-$(CONFIG_MPEGTS_MUXER)            += mpegvideo.o mpeg4audio.o
 OBJS-$(CONFIG_MPEGTS_DEMUXER)          += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MXF_MUXER)               += timecode.o
 OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
-OBJS-$(CONFIG_OGG_DEMUXER)             += xiph.o flac.o flacdata.o     \
-                                          mpeg12data.o vorbis_parser.o \
-                                          dirac.o vorbis_data.o
-OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flac.o flacdata.o \
+OBJS-$(CONFIG_OGG_DEMUXER)             += flacdec.o flacdata.o flac.o \
+                                          dirac.o mpeg12data.o vorbis_data.o
+OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flacdec.o flacdata.o flac.o \
                                           vorbis_data.o
-OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o xiph.o
-OBJS-$(CONFIG_RTPDEC)                  += mjpeg.o
+OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o mpegvideo.o xiph.o
 OBJS-$(CONFIG_SPDIF_DEMUXER)           += aacadtsdec.o mpeg4audio.o
-OBJS-$(CONFIG_SPDIF_MUXER)             += dca.o
-OBJS-$(CONFIG_WEBM_MUXER)              += mpeg4audio.o mpegaudiodata.o  \
-                                          xiph.o flac.o flacdata.o \
-                                          vorbis_data.o
+OBJS-$(CONFIG_WEBM_MUXER)              += xiph.o mpeg4audio.o \
+                                          flacdec.o flacdata.o flac.o \
+                                          mpegaudiodata.o vorbis_data.o
 OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
-OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o audio_frame_queue.o
-OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBDIRAC_DECODER)           += libdiracdec.o
+OBJS-$(CONFIG_LIBDIRAC_ENCODER)           += libdiracenc.o libdirac_libschro.o
+OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsm.o
-OBJS-$(CONFIG_LIBILBC_DECODER)            += libilbc.o
-OBJS-$(CONFIG_LIBILBC_ENCODER)            += libilbc.o
-OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o \
-                                             audio_frame_queue.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o \
-                                             audio_frame_queue.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o \
-                                             audio_frame_queue.o
+OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o
+OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o
+OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENJPEG_DECODER)        += libopenjpegdec.o
 OBJS-$(CONFIG_LIBOPENJPEG_ENCODER)        += libopenjpegenc.o
-OBJS-$(CONFIG_LIBOPUS_DECODER)            += libopus_dec.o vorbis_data.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \
-                                             libschroedinger.o
+                                             libschroedinger.o    \
+                                             libdirac_libschro.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \
-                                             libschroedinger.o
+                                             libschroedinger.o    \
+                                             libdirac_libschro.o
 OBJS-$(CONFIG_LIBSPEEX_DECODER)           += libspeexdec.o
-OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o
 OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
 OBJS-$(CONFIG_LIBTHEORA_ENCODER)          += libtheoraenc.o
-OBJS-$(CONFIG_LIBTWOLAME_ENCODER)         += libtwolame.o
-OBJS-$(CONFIG_LIBUTVIDEO_DECODER)         += libutvideodec.o
-OBJS-$(CONFIG_LIBUTVIDEO_ENCODER)         += libutvideoenc.o
-OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o \
-                                             audio_frame_queue.o
+OBJS-$(CONFIG_LIBUTVIDEO_DECODER)         += libutvideo.o
+OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o
 OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)     += libvo-amrwbenc.o
-OBJS-$(CONFIG_LIBVORBIS_DECODER)          += libvorbisdec.o
-OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbisenc.o audio_frame_queue.o \
-                                             vorbis_data.o vorbis_parser.o xiph.o
+OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbis.o vorbis_data.o
 OBJS-$(CONFIG_LIBVPX_DECODER)             += libvpxdec.o
 OBJS-$(CONFIG_LIBVPX_ENCODER)             += libvpxenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
-OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
+OBJS-$(CONFIG_LIBXVID)                    += libxvidff.o libxvid_rc.o
 
 # parsers
 OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o \
                                           aacadtsdec.o mpeg4audio.o
-OBJS-$(CONFIG_AAC_LATM_PARSER)         += latm_parser.o
 OBJS-$(CONFIG_AC3_PARSER)              += ac3_parser.o ac3tab.o \
                                           aac_ac3_parser.o
 OBJS-$(CONFIG_ADX_PARSER)              += adx_parser.o adx.o
-OBJS-$(CONFIG_BMP_PARSER)              += bmp_parser.o
 OBJS-$(CONFIG_CAVSVIDEO_PARSER)        += cavs_parser.o
-OBJS-$(CONFIG_COOK_PARSER)             += cook_parser.o
-OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o dca.o
+OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o
 OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
 OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
 OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
@@ -705,24 +669,26 @@ OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264.o            \
                                           cabac.o                         \
                                           h264_refs.o h264_sei.o h264_direct.o \
                                           h264_loopfilter.o h264_cabac.o \
-                                          h264_cavlc.o h264_ps.o
+                                          h264_cavlc.o h264_ps.o \
+                                          mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_AAC_LATM_PARSER)         += latm_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
+                                          mpegvideo.o error_resilience.o \
                                           mpeg4videodec.o mpeg4video.o \
                                           ituh263dec.o h263dec.o
-OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                           mpegaudiodecheader.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o    \
-                                          mpeg12.o mpeg12data.o
+                                          mpeg12.o mpeg12data.o \
+                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_PNM_PARSER)              += pnm_parser.o pnm.o
 OBJS-$(CONFIG_RV30_PARSER)             += rv34_parser.o
 OBJS-$(CONFIG_RV40_PARSER)             += rv34_parser.o
 OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o vc1.o vc1data.o \
                                           msmpeg4.o msmpeg4data.o mpeg4video.o \
-                                          h263.o
-OBJS-$(CONFIG_VORBIS_PARSER)           += vorbis_parser.o xiph.o
+                                          h263.o mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
 OBJS-$(CONFIG_VP8_PARSER)              += vp8_parser.o
 
@@ -744,9 +710,11 @@ OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
 OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o
 
 # thread libraries
-OBJS-$(HAVE_PTHREADS)                  += pthread.o frame_thread_encoder.o
-OBJS-$(HAVE_W32THREADS)                += pthread.o frame_thread_encoder.o
-OBJS-$(HAVE_OS2THREADS)                += pthread.o frame_thread_encoder.o
+OBJS-$(HAVE_PTHREADS)                  += pthread.o
+OBJS-$(HAVE_W32THREADS)                += pthread.o
+OBJS-$(HAVE_OS2THREADS)                += pthread.o
+
+OBJS-$(CONFIG_MLIB)                    += mlib/dsputil_mlib.o           \
 
 # inverse.o contains the ff_inverse table definition, which is used by
 # the FASTDIV macro (from libavutil); since referencing the external
@@ -758,48 +726,30 @@ SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           aac_tablegen_decl.h           \
                                           fft-internal.h                \
-                                          old_codec_ids.h               \
                                           tableprint.h                  \
-                                          $(ARCH)/vp56_arith.h          \
-
+                                          $(ARCH)/vp56_arith.h
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
+SKIPHEADERS-$(CONFIG_LIBDIRAC)         += libdirac.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
-SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
-SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
-SKIPHEADERS-$(CONFIG_VDA)              += vda.h
+SKIPHEADERS-$(CONFIG_VDA)              += vda_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
-SKIPHEADERS-$(HAVE_OS2THREADS)         += os2threads.h
+SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(HAVE_W32THREADS)         += w32pthreads.h
 
-TESTPROGS = cabac                                                       \
-            dct                                                         \
-            fft                                                         \
-            fft-fixed                                                   \
-            golomb                                                      \
-            iirfilter                                                   \
-            rangecoder                                                  \
-            snowenc                                                     \
-
+TESTPROGS = cabac dct fft fft-fixed h264 iirfilter rangecoder snowenc
 TESTPROGS-$(HAVE_MMX) += motion
 TESTOBJS = dctref.o
 
-TOOLS = fourcc2pixfmt
+HOSTPROGS = aac_tablegen aacps_tablegen cbrt_tablegen cos_tablegen      \
+            dv_tablegen motionpixels_tablegen mpegaudio_tablegen        \
+            pcm_tablegen qdm2_tablegen sinewin_tablegen
 
-HOSTPROGS = aac_tablegen                                                \
-            aacps_tablegen                                              \
-            cbrt_tablegen                                               \
-            cos_tablegen                                                \
-            dv_tablegen                                                 \
-            motionpixels_tablegen                                       \
-            mpegaudio_tablegen                                          \
-            pcm_tablegen                                                \
-            qdm2_tablegen                                               \
-            sinewin_tablegen                                            \
+DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
 
 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 
-$(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o $(SUBDIR)aandcttab.o
+$(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o
 
 TRIG_TABLES  = cos cos_fixed sin
 TRIG_TABLES := $(TRIG_TABLES:%=$(SUBDIR)%_tables.c)
@@ -833,3 +783,10 @@ $(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
 $(SUBDIR)pcm.o: $(SUBDIR)pcm_tables.h
 $(SUBDIR)qdm2.o: $(SUBDIR)qdm2_tables.h
 endif
+
+CODEC_NAMES_SH := $(SRC_PATH)/$(SUBDIR)codec_names.sh
+AVCODEC_H      := $(SRC_PATH)/$(SUBDIR)avcodec.h
+$(SUBDIR)codec_names.h: $(CODEC_NAMES_SH) config.h $(AVCODEC_H)
+	$(CC) $(CPPFLAGS) $(CFLAGS) -E $(AVCODEC_H) | \
+	$(CODEC_NAMES_SH) config.h $@
+$(SUBDIR)utils.o: $(SUBDIR)codec_names.h

libavcodec/a64enc.h

@@ -50,9 +50,6 @@ typedef struct A64Context {
     uint8_t *mc_colram;
     uint8_t *mc_palette;
     int mc_pal_size;
-
-    /* pts of the next packet that will be output */
-    int64_t next_pts;
 } A64Context;
 
 #endif /* AVCODEC_A64ENC_H */

libavcodec/a64multienc.c

@@ -28,8 +28,6 @@
 #include "a64colors.h"
 #include "a64tables.h"
 #include "elbg.h"
-#include "internal.h"
-#include "libavutil/common.h"
 #include "libavutil/intreadwrite.h"
 
 #define DITHERSTEPS   8
@@ -188,7 +186,7 @@ static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
     av_log(avctx, AV_LOG_INFO, "charset lifetime set to %d frame(s)\n", c->mc_lifetime);
 
     c->mc_frame_counter = 0;
-    c->mc_use_5col      = avctx->codec->id == AV_CODEC_ID_A64_MULTI5;
+    c->mc_use_5col      = avctx->codec->id == CODEC_ID_A64_MULTI5;
     c->mc_pal_size      = 4 + c->mc_use_5col;
 
     /* precalc luma values for later use */
@@ -223,8 +221,6 @@ static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
     if (!avctx->codec_tag)
          avctx->codec_tag = AV_RL32("a64m");
 
-    c->next_pts = AV_NOPTS_VALUE;
-
     return 0;
 }
 
@@ -243,19 +239,19 @@ static void a64_compress_colram(unsigned char *buf, int *charmap, uint8_t *colra
     }
 }
 
-static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
-                                 const AVFrame *pict, int *got_packet)
+static int a64multi_encode_frame(AVCodecContext *avctx, unsigned char *buf,
+                                 int buf_size, void *data)
 {
     A64Context *c = avctx->priv_data;
-    AVFrame *const p = &c->picture;
+    AVFrame *pict = data;
+    AVFrame *const p = (AVFrame *) & c->picture;
 
     int frame;
     int x, y;
     int b_height;
     int b_width;
 
-    int req_size, ret;
-    uint8_t *buf = NULL;
+    int req_size;
 
     int *charmap     = c->mc_charmap;
     uint8_t *colram  = c->mc_colram;
@@ -278,7 +274,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
     }
 
     /* no data, means end encoding asap */
-    if (!pict) {
+    if (!data) {
         /* all done, end encoding */
         if (!c->mc_lifetime) return 0;
         /* no more frames in queue, prepare to flush remaining frames */
@@ -296,8 +292,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
             p->key_frame = 1;
             to_meta_with_crop(avctx, p, meta + 32000 * c->mc_frame_counter);
             c->mc_frame_counter++;
-            if (c->next_pts == AV_NOPTS_VALUE)
-                c->next_pts = pict->pts;
             /* lifetime is not reached so wait for next frame first */
             return 0;
         }
@@ -308,11 +302,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
         req_size = 0;
         /* any frames to encode? */
         if (c->mc_lifetime) {
-            req_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
-            if ((ret = ff_alloc_packet2(avctx, pkt, req_size)) < 0)
-                return ret;
-            buf = pkt->data;
-
             /* calc optimal new charset + charmaps */
             ff_init_elbg(meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
             ff_do_elbg  (meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
@@ -321,12 +310,15 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
             render_charset(avctx, charset, colram);
 
             /* copy charset to buf */
-            memcpy(buf, charset, charset_size);
+            memcpy(buf,charset, charset_size);
 
             /* advance pointers */
             buf      += charset_size;
             charset  += charset_size;
+            req_size += charset_size;
         }
+        /* no charset so clean buf */
+        else memset(buf, 0, charset_size);
 
         /* write x frames to buf */
         for (frame = 0; frame < c->mc_lifetime; frame++) {
@@ -359,41 +351,37 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
         /* reset counter */
         c->mc_frame_counter = 0;
 
-        pkt->pts = pkt->dts = c->next_pts;
-        c->next_pts         = AV_NOPTS_VALUE;
-
-        pkt->size   = req_size;
-        pkt->flags |= AV_PKT_FLAG_KEY;
-        *got_packet = !!req_size;
+        if (req_size > buf_size) {
+            av_log(avctx, AV_LOG_ERROR, "buf size too small (need %d, got %d)\n", req_size, buf_size);
+            return -1;
+        }
+        return req_size;
     }
     return 0;
 }
 
-#if CONFIG_A64MULTI_ENCODER
 AVCodec ff_a64multi_encoder = {
     .name           = "a64multi",
     .type           = AVMEDIA_TYPE_VIDEO,
-    .id             = AV_CODEC_ID_A64_MULTI,
+    .id             = CODEC_ID_A64_MULTI,
     .priv_data_size = sizeof(A64Context),
     .init           = a64multi_init_encoder,
-    .encode2        = a64multi_encode_frame,
+    .encode         = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
     .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
     .capabilities   = CODEC_CAP_DELAY,
 };
-#endif
-#if CONFIG_A64MULTI5_ENCODER
+
 AVCodec ff_a64multi5_encoder = {
     .name           = "a64multi5",
     .type           = AVMEDIA_TYPE_VIDEO,
-    .id             = AV_CODEC_ID_A64_MULTI5,
+    .id             = CODEC_ID_A64_MULTI5,
     .priv_data_size = sizeof(A64Context),
     .init           = a64multi_init_encoder,
-    .encode2        = a64multi_encode_frame,
+    .encode         = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
     .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
     .capabilities   = CODEC_CAP_DELAY,
 };
-#endif

libavcodec/aac.h

@@ -30,7 +30,6 @@
 #ifndef AVCODEC_AAC_H
 #define AVCODEC_AAC_H
 
-#include "libavutil/float_dsp.h"
 #include "avcodec.h"
 #include "dsputil.h"
 #include "fft.h"
@@ -113,15 +112,6 @@ enum OCStatus {
     OC_LOCKED,      ///< Output configuration locked in place
 };
 
-typedef struct {
-    MPEG4AudioConfig m4ac;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int layout_map_tags;
-    int channels;
-    uint64_t channel_layout;
-    enum OCStatus status;
-} OutputConfiguration;
-
 /**
  * Predictor State
  */
@@ -264,6 +254,8 @@ typedef struct {
     AVCodecContext *avctx;
     AVFrame frame;
 
+    MPEG4AudioConfig m4ac;
+
     int is_saved;                 ///< Set if elements have stored overlap from previous frame.
     DynamicRangeControl che_drc;
 
@@ -271,6 +263,9 @@ typedef struct {
      * @name Channel element related data
      * @{
      */
+    enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
+                                                   *   first index as the first 4 raw data block types
+                                                   */
     ChannelElement          *che[4][MAX_ELEM_ID];
     ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
     int tags_mapped;
@@ -293,7 +288,6 @@ typedef struct {
     FFTContext mdct_ltp;
     DSPContext dsp;
     FmtConvertContext fmt_conv;
-    AVFloatDSPContext fdsp;
     int random_state;
     /** @} */
 
@@ -304,18 +298,9 @@ typedef struct {
     float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
     /** @} */
 
-
-    /**
-     * @name Japanese DTV specific extension
-     * @{
-     */
-    int enable_jp_dmono; ///< enable japanese DTV specific 'dual mono'
-    int dmono_mode;      ///< select the channel to decode in dual mono.
-    /** @} */
-
     DECLARE_ALIGNED(32, float, temp)[128];
 
-    OutputConfiguration oc[2];
+    enum OCStatus output_configured;
     int warned_num_aac_frames;
 } AACContext;
 

libavcodec/aac_ac3_parser.c

@@ -20,7 +20,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "libavutil/common.h"
 #include "parser.h"
 #include "aac_ac3_parser.h"
 
@@ -79,7 +78,7 @@ get_next:
        and total number of samples found in an AAC ADTS header are not
        reliable. Bit rate is still accurate because the total frame duration in
        seconds is still correct (as is the number of bits in the frame). */
-    if (avctx->codec_id != AV_CODEC_ID_AAC) {
+    if (avctx->codec_id != CODEC_ID_AAC) {
         avctx->sample_rate = s->sample_rate;
 
         /* allow downmixing to stereo (or mono for AC-3) */
@@ -87,14 +86,14 @@ get_next:
                 avctx->request_channels < s->channels &&
                 (avctx->request_channels <= 2 ||
                 (avctx->request_channels == 1 &&
-                (avctx->codec_id == AV_CODEC_ID_AC3 ||
-                 avctx->codec_id == AV_CODEC_ID_EAC3)))) {
+                (avctx->codec_id == CODEC_ID_AC3 ||
+                 avctx->codec_id == CODEC_ID_EAC3)))) {
             avctx->channels = avctx->request_channels;
         } else {
             avctx->channels = s->channels;
             avctx->channel_layout = s->channel_layout;
         }
-        s1->duration = s->samples;
+        avctx->frame_size = s->samples;
         avctx->audio_service_type = s->service_type;
     }
 

libavcodec/aac_ac3_parser.h

@@ -28,13 +28,13 @@
 #include "parser.h"
 
 typedef enum {
-    AAC_AC3_PARSE_ERROR_SYNC        = -1,
-    AAC_AC3_PARSE_ERROR_BSID        = -2,
-    AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -3,
-    AAC_AC3_PARSE_ERROR_FRAME_SIZE  = -4,
-    AAC_AC3_PARSE_ERROR_FRAME_TYPE  = -5,
-    AAC_AC3_PARSE_ERROR_CRC         = -6,
-    AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -7,
+    AAC_AC3_PARSE_ERROR_SYNC        = -0x1030c0a,
+    AAC_AC3_PARSE_ERROR_BSID        = -0x2030c0a,
+    AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -0x3030c0a,
+    AAC_AC3_PARSE_ERROR_FRAME_SIZE  = -0x4030c0a,
+    AAC_AC3_PARSE_ERROR_FRAME_TYPE  = -0x5030c0a,
+    AAC_AC3_PARSE_ERROR_CRC         = -0x6030c0a,
+    AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -0x7030c0a,
 } AACAC3ParseError;
 
 typedef struct AACAC3ParseContext {
@@ -55,7 +55,7 @@ typedef struct AACAC3ParseContext {
     uint64_t state;
 
     int need_next_header;
-    enum AVCodecID codec_id;
+    enum CodecID codec_id;
 } AACAC3ParseContext;
 
 int ff_aac_ac3_parse(AVCodecParserContext *s1,

libavcodec/aac_parser.c

@@ -61,7 +61,7 @@ static av_cold int aac_parse_init(AVCodecParserContext *s1)
 
 
 AVCodecParser ff_aac_parser = {
-    .codec_ids      = { AV_CODEC_ID_AAC },
+    .codec_ids      = { CODEC_ID_AAC },
     .priv_data_size = sizeof(AACAC3ParseContext),
     .parser_init    = aac_parse_init,
     .parser_parse   = ff_aac_ac3_parse,

libavcodec/aaccoder.c

@@ -506,7 +506,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             idx = cb;
     ppos = max_sfb;
     while (ppos > 0) {
-        av_assert1(idx >= 0);
+        assert(idx >= 0);
         cb = idx;
         stackrun[stack_len] = path[ppos][cb].run;
         stackcb [stack_len] = cb;
@@ -713,18 +713,16 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                                           const float lambda)
 {
     int start = 0, i, w, w2, g;
-    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
-    float dists[128] = { 0 }, uplims[128];
+    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
+    float dists[128], uplims[128];
     float maxvals[128];
     int fflag, minscaler;
     int its  = 0;
     int allz = 0;
     float minthr = INFINITY;
 
-    // for values above this the decoder might end up in an endless loop
-    // due to always having more bits than what can be encoded.
-    destbits = FFMIN(destbits, 5800);
     //XXX: some heuristic to determine initial quantizers will reduce search time
+    memset(dists, 0, sizeof(dists));
     //determine zero bands and upper limits
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         for (g = 0;  g < sce->ics.num_swb; g++) {
@@ -878,7 +876,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
     } else {
         for (w = 0; w < 8; w++) {
             const float *coeffs = sce->coeffs + w*128;
-            curband = start = 0;
+            start = 0;
             for (i = 0; i < 128; i++) {
                 if (i - start >= sce->ics.swb_sizes[curband]) {
                     start += sce->ics.swb_sizes[curband];

libavcodec/aacdectab.h

@@ -80,14 +80,14 @@ static const float * const tns_tmp2_map[4] = {
 
 static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
 
-static const uint8_t aac_channel_layout_map[7][5][3] = {
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+static const uint8_t aac_channel_layout_map[7][5][2] = {
+    { { TYPE_SCE, 0 }, },
+    { { TYPE_CPE, 0 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_SCE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_CPE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 2 }, { TYPE_CPE, 1 }, },
 };
 
 static const uint64_t aac_channel_layout[8] = {
@@ -97,7 +97,7 @@ static const uint64_t aac_channel_layout[8] = {
     AV_CH_LAYOUT_4POINT0,
     AV_CH_LAYOUT_5POINT0_BACK,
     AV_CH_LAYOUT_5POINT1_BACK,
-    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
+    AV_CH_LAYOUT_7POINT1_WIDE,
     0,
 };
 

libavcodec/aacenc.c

@@ -30,12 +30,10 @@
  * add temporal noise shaping
  ***********************************/
 
-#include "libavutil/float_dsp.h"
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "put_bits.h"
 #include "dsputil.h"
-#include "internal.h"
 #include "mpeg4audio.h"
 #include "kbdwin.h"
 #include "sinewin.h"
@@ -146,7 +144,7 @@ static const uint8_t aac_chan_configs[6][5] = {
 };
 
 /**
- * Table to remap channels from libavcodec's default order to AAC order.
+ * Table to remap channels from Libav's default order to AAC order.
  */
 static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
     { 0 },
@@ -183,9 +181,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
 }
 
 #define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, AVFloatDSPContext *fdsp, \
-                                    SingleChannelElement *sce, \
-                                    const float *audio)
+static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio)
 
 WINDOW_FUNC(only_long)
 {
@@ -193,7 +189,7 @@ WINDOW_FUNC(only_long)
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     float *out = sce->ret;
 
-    fdsp->vector_fmul       (out,        audio,        lwindow, 1024);
+    dsp->vector_fmul        (out,        audio,        lwindow, 1024);
     dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
 }
 
@@ -203,7 +199,7 @@ WINDOW_FUNC(long_start)
     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
     float *out = sce->ret;
 
-    fdsp->vector_fmul(out, audio, lwindow, 1024);
+    dsp->vector_fmul(out, audio, lwindow, 1024);
     memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
     dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
     memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
@@ -216,7 +212,7 @@ WINDOW_FUNC(long_stop)
     float *out = sce->ret;
 
     memset(out, 0, sizeof(out[0]) * 448);
-    fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
+    dsp->vector_fmul(out + 448, audio + 448, swindow, 128);
     memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
     dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
 }
@@ -227,10 +223,9 @@ WINDOW_FUNC(eight_short)
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *in = audio + 448;
     float *out = sce->ret;
-    int w;
 
-    for (w = 0; w < 8; w++) {
-        fdsp->vector_fmul       (out, in, w ? pwindow : swindow, 128);
+    for (int w = 0; w < 8; w++) {
+        dsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
         out += 128;
         in  += 128;
         dsp->vector_fmul_reverse(out, in, swindow, 128);
@@ -238,9 +233,7 @@ WINDOW_FUNC(eight_short)
     }
 }
 
-static void (*const apply_window[4])(DSPContext *dsp, AVFloatDSPContext *fdsp,
-                                     SingleChannelElement *sce,
-                                     const float *audio) = {
+static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = {
     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
     [LONG_START_SEQUENCE]  = apply_long_start_window,
     [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
@@ -253,7 +246,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
     int i;
     float *output = sce->ret;
 
-    apply_window[sce->ics.window_sequence[0]](&s->dsp, &s->fdsp, sce, audio);
+    apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio);
 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
@@ -480,9 +473,10 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
 
 /*
  * Deinterleave input samples.
- * Channels are reordered from libavcodec's default order to AAC order.
+ * Channels are reordered from Libav's default order to AAC order.
  */
-static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
+static void deinterleave_input_samples(AACEncContext *s,
+                                       const float *samples)
 {
     int ch, i;
     const int sinc = s->channels;
@@ -490,46 +484,38 @@ static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
 
     /* deinterleave and remap input samples */
     for (ch = 0; ch < sinc; ch++) {
+        const float *sptr = samples + channel_map[ch];
+
         /* copy last 1024 samples of previous frame to the start of the current frame */
         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
 
         /* deinterleave */
-        i = 2048;
-        if (frame) {
-            const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
-            for (; i < 2048 + frame->nb_samples; i++) {
-                s->planar_samples[ch][i] = *sptr;
-                sptr += sinc;
-            }
+        for (i = 2048; i < 3072; i++) {
+            s->planar_samples[ch][i] = *sptr;
+            sptr += sinc;
         }
-        memset(&s->planar_samples[ch][i], 0,
-               (3072 - i) * sizeof(s->planar_samples[0][0]));
     }
 }
 
-static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                            const AVFrame *frame, int *got_packet_ptr)
+static int aac_encode_frame(AVCodecContext *avctx,
+                            uint8_t *frame, int buf_size, void *data)
 {
     AACEncContext *s = avctx->priv_data;
     float **samples = s->planar_samples, *samples2, *la, *overlap;
     ChannelElement *cpe;
-    int i, ch, w, g, chans, tag, start_ch, ret;
+    int i, ch, w, g, chans, tag, start_ch;
     int chan_el_counter[4];
     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
 
-    if (s->last_frame == 2)
+    if (s->last_frame)
         return 0;
 
-    /* add current frame to queue */
-    if (frame) {
-        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
-            return ret;
+    if (data) {
+        deinterleave_input_samples(s, data);
+        if (s->psypp)
+            ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
     }
 
-    deinterleave_input_samples(s, frame);
-    if (s->psypp)
-        ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
-
     if (!avctx->frame_number)
         return 0;
 
@@ -545,7 +531,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             overlap  = &samples[cur_channel][0];
             samples2 = overlap + 1024;
             la       = samples2 + (448+64);
-            if (!frame)
+            if (!data)
                 la = NULL;
             if (tag == TYPE_LFE) {
                 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
@@ -576,15 +562,9 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
         start_ch += chans;
     }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels))) {
-        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
-        return ret;
-    }
     do {
         int frame_bits;
-
-        init_put_bits(&s->pb, avpkt->data, avpkt->size);
-
+        init_put_bits(&s->pb, frame, buf_size*8);
         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
             put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
         start_ch = 0;
@@ -664,15 +644,10 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         s->lambda = FFMIN(s->lambda, 65536.f);
     }
 
-    if (!frame)
-        s->last_frame++;
+    if (!data)
+        s->last_frame = 1;
 
-    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
-                       &avpkt->duration);
-
-    avpkt->size = put_bits_count(&s->pb) >> 3;
-    *got_packet_ptr = 1;
-    return 0;
+    return put_bits_count(&s->pb)>>3;
 }
 
 static av_cold int aac_encode_end(AVCodecContext *avctx)
@@ -686,10 +661,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
         ff_psy_preprocess_end(s->psypp);
     av_freep(&s->buffer.samples);
     av_freep(&s->cpe);
-    ff_af_queue_close(&s->afq);
-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
     return 0;
 }
 
@@ -697,8 +668,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 
-    ff_dsputil_init(&s->dsp, avctx);
-    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+    dsputil_init(&s->dsp, avctx);
 
     // window init
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
@@ -716,19 +686,13 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 
 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
 {
-    int ch;
     FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
     FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
     FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
 
-    for(ch = 0; ch < s->channels; ch++)
+    for(int ch = 0; ch < s->channels; ch++)
         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
 
-#if FF_API_OLD_ENCODE_AUDIO
-    if (!(avctx->coded_frame = avcodec_alloc_frame()))
-        goto alloc_fail;
-#endif
-
     return 0;
 alloc_fail:
     return AVERROR(ENOMEM);
@@ -790,9 +754,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     for (i = 0; i < 428; i++)
         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
 
-    avctx->delay = 1024;
-    ff_af_queue_init(avctx, &s->afq);
-
     return 0;
 fail:
     aac_encode_end(avctx);
@@ -801,11 +762,11 @@ fail:
 
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
-    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
-        {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-        {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-        {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},
+    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
+        {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.dbl = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+        {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.dbl =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+        {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.dbl =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.dbl = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},
     {NULL}
 };
 
@@ -819,16 +780,13 @@ static const AVClass aacenc_class = {
 AVCodec ff_aac_encoder = {
     .name           = "aac",
     .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_AAC,
+    .id             = CODEC_ID_AAC,
     .priv_data_size = sizeof(AACEncContext),
     .init           = aac_encode_init,
-    .encode2        = aac_encode_frame,
+    .encode         = aac_encode_frame,
     .close          = aac_encode_end,
-    .supported_samplerates = avpriv_mpeg4audio_sample_rates,
-    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
-                      CODEC_CAP_EXPERIMENTAL,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
-                                                     AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
-    .priv_class     = &aacenc_class,
+    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+    .priv_class = &aacenc_class,
 };

libavcodec/aacenc.h

@@ -22,13 +22,12 @@
 #ifndef AVCODEC_AACENC_H
 #define AVCODEC_AACENC_H
 
-#include "libavutil/float_dsp.h"
 #include "avcodec.h"
 #include "put_bits.h"
 #include "dsputil.h"
 
 #include "aac.h"
-#include "audio_frame_queue.h"
+
 #include "psymodel.h"
 
 #define AAC_CODER_NB 4
@@ -62,7 +61,6 @@ typedef struct AACEncContext {
     FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
     FFTContext mdct128;                          ///< short (128 samples) frame transform context
     DSPContext  dsp;
-    AVFloatDSPContext fdsp;
     float *planar_samples[6];                    ///< saved preprocessed input
 
     int samplerate_index;                        ///< MPEG-4 samplerate index
@@ -76,7 +74,6 @@ typedef struct AACEncContext {
     int cur_channel;
     int last_frame;
     float lambda;
-    AudioFrameQueue afq;
     DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
     DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients
 

libavcodec/aacps.c

@@ -27,7 +27,6 @@
 #include "aacps.h"
 #include "aacps_tablegen.h"
 #include "aacpsdata.c"
-#include "dsputil.h"
 
 #define PS_BASELINE 0  ///< Operate in Baseline PS mode
                        ///< Baseline implies 10 or 20 stereo bands,
@@ -285,7 +284,7 @@ err:
 
 /** Split one subband into 2 subsubbands with a symmetric real filter.
  * The filter must have its non-center even coefficients equal to zero. */
-static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8], int len, int reverse)
+static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[7], int len, int reverse)
 {
     int i, j;
     for (i = 0; i < len; i++, in++) {
@@ -305,14 +304,26 @@ static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8
 }
 
 /** Split one subband into 6 subsubbands with a complex filter */
-static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int len)
+static void hybrid6_cx(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int len)
 {
-    int i;
+    int i, j, ssb;
     int N = 8;
-    LOCAL_ALIGNED_16(float, temp, [8], [2]);
+    float temp[8][2];
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, filter, 1, N);
+        for (ssb = 0; ssb < N; ssb++) {
+            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
+            for (j = 0; j < 6; j++) {
+                float in0_re = in[j][0];
+                float in0_im = in[j][1];
+                float in1_re = in[12-j][0];
+                float in1_im = in[12-j][1];
+                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
+                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
+            }
+            temp[ssb][0] = sum_re;
+            temp[ssb][1] = sum_im;
+        }
         out[0][i][0] = temp[6][0];
         out[0][i][1] = temp[6][1];
         out[1][i][0] = temp[7][0];
@@ -328,18 +339,28 @@ static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], c
     }
 }
 
-static void hybrid4_8_12_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int N, int len)
+static void hybrid4_8_12_cx(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int N, int len)
 {
-    int i;
+    int i, j, ssb;
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, filter, 32, N);
+        for (ssb = 0; ssb < N; ssb++) {
+            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
+            for (j = 0; j < 6; j++) {
+                float in0_re = in[j][0];
+                float in0_im = in[j][1];
+                float in1_re = in[12-j][0];
+                float in1_im = in[12-j][1];
+                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
+                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
+            }
+            out[ssb][i][0] = sum_re;
+            out[ssb][i][1] = sum_im;
+        }
     }
 }
 
-static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
-                            float in[5][44][2], float L[2][38][64],
-                            int is34, int len)
+static void hybrid_analysis(float out[91][32][2], float in[5][44][2], float L[2][38][64], int is34, int len)
 {
     int i, j;
     for (i = 0; i < 5; i++) {
@@ -349,17 +370,27 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
         }
     }
     if (is34) {
-        hybrid4_8_12_cx(dsp, in[0], out,    f34_0_12, 12, len);
-        hybrid4_8_12_cx(dsp, in[1], out+12, f34_1_8,   8, len);
-        hybrid4_8_12_cx(dsp, in[2], out+20, f34_2_4,   4, len);
-        hybrid4_8_12_cx(dsp, in[3], out+24, f34_2_4,   4, len);
-        hybrid4_8_12_cx(dsp, in[4], out+28, f34_2_4,   4, len);
-        dsp->hybrid_analysis_ileave(out + 27, L, 5, len);
+        hybrid4_8_12_cx(in[0], out,    f34_0_12, 12, len);
+        hybrid4_8_12_cx(in[1], out+12, f34_1_8,   8, len);
+        hybrid4_8_12_cx(in[2], out+20, f34_2_4,   4, len);
+        hybrid4_8_12_cx(in[3], out+24, f34_2_4,   4, len);
+        hybrid4_8_12_cx(in[4], out+28, f34_2_4,   4, len);
+        for (i = 0; i < 59; i++) {
+            for (j = 0; j < len; j++) {
+                out[i+32][j][0] = L[0][j][i+5];
+                out[i+32][j][1] = L[1][j][i+5];
+            }
+        }
     } else {
-        hybrid6_cx(dsp, in[0], out, f20_0_8, len);
+        hybrid6_cx(in[0], out, f20_0_8, len);
         hybrid2_re(in[1], out+6, g1_Q2, len, 1);
         hybrid2_re(in[2], out+8, g1_Q2, len, 0);
-        dsp->hybrid_analysis_ileave(out + 7, L, 3, len);
+        for (i = 0; i < 61; i++) {
+            for (j = 0; j < len; j++) {
+                out[i+10][j][0] = L[0][j][i+3];
+                out[i+10][j][1] = L[1][j][i+3];
+            }
+        }
     }
     //update in_buf
     for (i = 0; i < 5; i++) {
@@ -367,8 +398,7 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
     }
 }
 
-static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
-                             float in[91][32][2], int is34, int len)
+static void hybrid_synthesis(float out[2][38][64], float in[91][32][2], int is34, int len)
 {
     int i, n;
     if (is34) {
@@ -392,7 +422,12 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
                 out[1][n][4] += in[28+i][n][1];
             }
         }
-        dsp->hybrid_synthesis_deint(out, in + 27, 5, len);
+        for (i = 0; i < 59; i++) {
+            for (n = 0; n < len; n++) {
+                out[0][n][i+5] = in[i+32][n][0];
+                out[1][n][i+5] = in[i+32][n][1];
+            }
+        }
     } else {
         for (n = 0; n < len; n++) {
             out[0][n][0] = in[0][n][0] + in[1][n][0] + in[2][n][0] +
@@ -404,7 +439,12 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
             out[0][n][2] = in[8][n][0] + in[9][n][0];
             out[1][n][2] = in[8][n][1] + in[9][n][1];
         }
-        dsp->hybrid_synthesis_deint(out, in + 7, 3, len);
+        for (i = 0; i < 61; i++) {
+            for (n = 0; n < len; n++) {
+                out[0][n][i+3] = in[i+10][n][0];
+                out[1][n][i+3] = in[i+10][n][1];
+            }
+        }
     }
 }
 
@@ -608,8 +648,8 @@ static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
 
 static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
 {
-    LOCAL_ALIGNED_16(float, power, [34], [PS_QMF_TIME_SLOTS]);
-    LOCAL_ALIGNED_16(float, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
+    float power[34][PS_QMF_TIME_SLOTS] = {{0}};
+    float transient_gain[34][PS_QMF_TIME_SLOTS];
     float *peak_decay_nrg = ps->peak_decay_nrg;
     float *power_smooth = ps->power_smooth;
     float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
@@ -621,8 +661,10 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
     const float a_smooth          = 0.25f; ///< Smoothing coefficient
     int i, k, m, n;
     int n0 = 0, nL = 32;
-
-    memset(power, 0, 34 * sizeof(*power));
+    static const int link_delay[] = { 3, 4, 5 };
+    static const float a[] = { 0.65143905753106f,
+                               0.56471812200776f,
+                               0.48954165955695f };
 
     if (is34 != ps->is34bands_old) {
         memset(ps->peak_decay_nrg,         0, sizeof(ps->peak_decay_nrg));
@@ -632,9 +674,11 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
         memset(ps->ap_delay,               0, sizeof(ps->ap_delay));
     }
 
-    for (k = 0; k < NR_BANDS[is34]; k++) {
-        int i = k_to_i[k];
-        ps->dsp.add_squares(power[i], s[k], nL - n0);
+    for (n = n0; n < nL; n++) {
+        for (k = 0; k < NR_BANDS[is34]; k++) {
+            int i = k_to_i[k];
+            power[i][n] += s[k][n][0] * s[k][n][0] + s[k][n][1] * s[k][n][1];
+        }
     }
 
     //Transient detection
@@ -662,31 +706,54 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
     for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
         int b = k_to_i[k];
         float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
+        float ag[PS_AP_LINKS];
         g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
         for (m = 0; m < PS_AP_LINKS; m++) {
             memcpy(ap_delay[k][m],   ap_delay[k][m]+numQMFSlots,           5*sizeof(ap_delay[k][m][0]));
+            ag[m] = a[m] * g_decay_slope;
+        }
+        for (n = n0; n < nL; n++) {
+            float in_re = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][0] -
+                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][1];
+            float in_im = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][1] +
+                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][0];
+            for (m = 0; m < PS_AP_LINKS; m++) {
+                float a_re                = ag[m] * in_re;
+                float a_im                = ag[m] * in_im;
+                float link_delay_re       = ap_delay[k][m][n+5-link_delay[m]][0];
+                float link_delay_im       = ap_delay[k][m][n+5-link_delay[m]][1];
+                float fractional_delay_re = Q_fract_allpass[is34][k][m][0];
+                float fractional_delay_im = Q_fract_allpass[is34][k][m][1];
+                ap_delay[k][m][n+5][0] = in_re;
+                ap_delay[k][m][n+5][1] = in_im;
+                in_re = link_delay_re * fractional_delay_re - link_delay_im * fractional_delay_im - a_re;
+                in_im = link_delay_re * fractional_delay_im + link_delay_im * fractional_delay_re - a_im;
+                ap_delay[k][m][n+5][0] += ag[m] * in_re;
+                ap_delay[k][m][n+5][1] += ag[m] * in_im;
+            }
+            out[k][n][0] = transient_gain[b][n] * in_re;
+            out[k][n][1] = transient_gain[b][n] * in_im;
         }
-        ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
-                            phi_fract[is34][k], Q_fract_allpass[is34][k],
-                            transient_gain[b], g_decay_slope, nL - n0);
     }
     for (; k < SHORT_DELAY_BAND[is34]; k++) {
-        int i = k_to_i[k];
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
-        //H = delay 14
-        ps->dsp.mul_pair_single(out[k], delay[k] + PS_MAX_DELAY - 14,
-                                transient_gain[i], nL - n0);
+        for (n = n0; n < nL; n++) {
+            //H = delay 14
+            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][0];
+            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][1];
+        }
     }
     for (; k < NR_BANDS[is34]; k++) {
-        int i = k_to_i[k];
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
-        //H = delay 1
-        ps->dsp.mul_pair_single(out[k], delay[k] + PS_MAX_DELAY - 1,
-                                transient_gain[i], nL - n0);
+        for (n = n0; n < nL; n++) {
+            //H = delay 1
+            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][0];
+            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][1];
+        }
     }
 }
 
@@ -730,7 +797,7 @@ static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
 
 static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
 {
-    int e, b, k;
+    int e, b, k, n;
 
     float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
     float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
@@ -842,52 +909,78 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             H22[0][e+1][b] = h22;
         }
         for (k = 0; k < NR_BANDS[is34]; k++) {
-            float h[2][4];
-            float h_step[2][4];
+            float h11r, h12r, h21r, h22r;
+            float h11i, h12i, h21i, h22i;
+            float h11r_step, h12r_step, h21r_step, h22r_step;
+            float h11i_step, h12i_step, h21i_step, h22i_step;
             int start = ps->border_position[e];
             int stop  = ps->border_position[e+1];
             float width = 1.f / (stop - start);
             b = k_to_i[k];
-            h[0][0] = H11[0][e][b];
-            h[0][1] = H12[0][e][b];
-            h[0][2] = H21[0][e][b];
-            h[0][3] = H22[0][e][b];
+            h11r = H11[0][e][b];
+            h12r = H12[0][e][b];
+            h21r = H21[0][e][b];
+            h22r = H22[0][e][b];
             if (!PS_BASELINE && ps->enable_ipdopd) {
             //Is this necessary? ps_04_new seems unchanged
             if ((is34 && k <= 13 && k >= 9) || (!is34 && k <= 1)) {
-                h[1][0] = -H11[1][e][b];
-                h[1][1] = -H12[1][e][b];
-                h[1][2] = -H21[1][e][b];
-                h[1][3] = -H22[1][e][b];
+                h11i = -H11[1][e][b];
+                h12i = -H12[1][e][b];
+                h21i = -H21[1][e][b];
+                h22i = -H22[1][e][b];
             } else {
-                h[1][0] = H11[1][e][b];
-                h[1][1] = H12[1][e][b];
-                h[1][2] = H21[1][e][b];
-                h[1][3] = H22[1][e][b];
+                h11i = H11[1][e][b];
+                h12i = H12[1][e][b];
+                h21i = H21[1][e][b];
+                h22i = H22[1][e][b];
             }
             }
             //Interpolation
-            h_step[0][0] = (H11[0][e+1][b] - h[0][0]) * width;
-            h_step[0][1] = (H12[0][e+1][b] - h[0][1]) * width;
-            h_step[0][2] = (H21[0][e+1][b] - h[0][2]) * width;
-            h_step[0][3] = (H22[0][e+1][b] - h[0][3]) * width;
+            h11r_step = (H11[0][e+1][b] - h11r) * width;
+            h12r_step = (H12[0][e+1][b] - h12r) * width;
+            h21r_step = (H21[0][e+1][b] - h21r) * width;
+            h22r_step = (H22[0][e+1][b] - h22r) * width;
             if (!PS_BASELINE && ps->enable_ipdopd) {
-                h_step[1][0] = (H11[1][e+1][b] - h[1][0]) * width;
-                h_step[1][1] = (H12[1][e+1][b] - h[1][1]) * width;
-                h_step[1][2] = (H21[1][e+1][b] - h[1][2]) * width;
-                h_step[1][3] = (H22[1][e+1][b] - h[1][3]) * width;
+                h11i_step = (H11[1][e+1][b] - h11i) * width;
+                h12i_step = (H12[1][e+1][b] - h12i) * width;
+                h21i_step = (H21[1][e+1][b] - h21i) * width;
+                h22i_step = (H22[1][e+1][b] - h22i) * width;
+            }
+            for (n = start + 1; n <= stop; n++) {
+                //l is s, r is d
+                float l_re = l[k][n][0];
+                float l_im = l[k][n][1];
+                float r_re = r[k][n][0];
+                float r_im = r[k][n][1];
+                h11r += h11r_step;
+                h12r += h12r_step;
+                h21r += h21r_step;
+                h22r += h22r_step;
+                if (!PS_BASELINE && ps->enable_ipdopd) {
+                    h11i += h11i_step;
+                    h12i += h12i_step;
+                    h21i += h21i_step;
+                    h22i += h22i_step;
+
+                    l[k][n][0] = h11r*l_re + h21r*r_re - h11i*l_im - h21i*r_im;
+                    l[k][n][1] = h11r*l_im + h21r*r_im + h11i*l_re + h21i*r_re;
+                    r[k][n][0] = h12r*l_re + h22r*r_re - h12i*l_im - h22i*r_im;
+                    r[k][n][1] = h12r*l_im + h22r*r_im + h12i*l_re + h22i*r_re;
+                } else {
+                    l[k][n][0] = h11r*l_re + h21r*r_re;
+                    l[k][n][1] = h11r*l_im + h21r*r_im;
+                    r[k][n][0] = h12r*l_re + h22r*r_re;
+                    r[k][n][1] = h12r*l_im + h22r*r_im;
+                }
             }
-            ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
-                l[k] + start + 1, r[k] + start + 1,
-                h, h_step, stop - start);
         }
     }
 }
 
 int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
 {
-    LOCAL_ALIGNED_16(float, Lbuf, [91], [32][2]);
-    LOCAL_ALIGNED_16(float, Rbuf, [91], [32][2]);
+    float Lbuf[91][32][2];
+    float Rbuf[91][32][2];
     const int len = 32;
     int is34 = ps->is34bands;
 
@@ -896,11 +989,11 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
     if (top < NR_ALLPASS_BANDS[is34])
         memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));
 
-    hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
+    hybrid_analysis(Lbuf, ps->in_buf, L, is34, len);
     decorrelation(ps, Rbuf, Lbuf, is34);
     stereo_processing(ps, Lbuf, Rbuf, is34);
-    hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
-    hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);
+    hybrid_synthesis(L, Lbuf, is34, len);
+    hybrid_synthesis(R, Rbuf, is34, len);
 
     return 0;
 }
@@ -948,5 +1041,4 @@ av_cold void ff_ps_init(void) {
 
 av_cold void ff_ps_ctx_init(PSContext *ps)
 {
-    ff_psdsp_init(&ps->dsp);
 }

libavcodec/aacps.h

@@ -24,7 +24,6 @@
 
 #include <stdint.h>
 
-#include "aacpsdsp.h"
 #include "avcodec.h"
 #include "get_bits.h"
 
@@ -61,19 +60,18 @@ typedef struct {
     int    is34bands;
     int    is34bands_old;
 
-    DECLARE_ALIGNED(16, float, in_buf)[5][44][2];
-    DECLARE_ALIGNED(16, float, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
-    DECLARE_ALIGNED(16, float, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
-    DECLARE_ALIGNED(16, float, peak_decay_nrg)[34];
-    DECLARE_ALIGNED(16, float, power_smooth)[34];
-    DECLARE_ALIGNED(16, float, peak_decay_diff_smooth)[34];
-    DECLARE_ALIGNED(16, float, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  in_buf[5][44][2];
+    float  delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    float  ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    float  peak_decay_nrg[34];
+    float  power_smooth[34];
+    float  peak_decay_diff_smooth[34];
+    float  H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
     int8_t opd_hist[PS_MAX_NR_IIDICC];
     int8_t ipd_hist[PS_MAX_NR_IIDICC];
-    PSDSPContext dsp;
 } PSContext;
 
 void ff_ps_init(void);

libavcodec/aacps_tablegen.c

@@ -69,23 +69,23 @@ int main(void)
     write_float_3d_array(HB, 46, 8, 4);
     printf("};\n");
 
-    printf("static const DECLARE_ALIGNED(16, float, f20_0_8)[8][8][2] = {\n");
-    write_float_3d_array(f20_0_8, 8, 8, 2);
+    printf("static const float f20_0_8[8][7][2] = {\n");
+    write_float_3d_array(f20_0_8, 8, 7, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2] = {\n");
-    write_float_3d_array(f34_0_12, 12, 8, 2);
+    printf("static const float f34_0_12[12][7][2] = {\n");
+    write_float_3d_array(f34_0_12, 12, 7, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_1_8)[8][8][2] = {\n");
-    write_float_3d_array(f34_1_8, 8, 8, 2);
+    printf("static const float f34_1_8[8][7][2] = {\n");
+    write_float_3d_array(f34_1_8, 8, 7, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_2_4)[4][8][2] = {\n");
-    write_float_3d_array(f34_2_4, 4, 8, 2);
+    printf("static const float f34_2_4[4][7][2] = {\n");
+    write_float_3d_array(f34_2_4, 4, 7, 2);
     printf("};\n");
 
-    printf("static const DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
+    printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
     write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, phi_fract)[2][50][2] = {\n");
+    printf("static const float phi_fract[2][50][2] = {\n");
     write_float_3d_array(phi_fract, 2, 50, 2);
     printf("};\n");
 

libavcodec/aacps_tablegen.h

@@ -23,7 +23,6 @@
 #ifndef AACPS_TABLEGEN_H
 #define AACPS_TABLEGEN_H
 
-#include <math.h>
 #include <stdint.h>
 
 #if CONFIG_HARDCODED_TABLES
@@ -32,7 +31,6 @@
 #else
 #include "libavutil/common.h"
 #include "libavutil/mathematics.h"
-#include "libavutil/mem.h"
 #define NR_ALLPASS_BANDS20 30
 #define NR_ALLPASS_BANDS34 50
 #define PS_AP_LINKS 3
@@ -40,12 +38,12 @@ static float pd_re_smooth[8*8*8];
 static float pd_im_smooth[8*8*8];
 static float HA[46][8][4];
 static float HB[46][8][4];
-static DECLARE_ALIGNED(16, float, f20_0_8) [ 8][8][2];
-static DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2];
-static DECLARE_ALIGNED(16, float, f34_1_8) [ 8][8][2];
-static DECLARE_ALIGNED(16, float, f34_2_4) [ 4][8][2];
-static DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
-static DECLARE_ALIGNED(16, float, phi_fract)[2][50][2];
+static float f20_0_8 [ 8][7][2];
+static float f34_0_12[12][7][2];
+static float f34_1_8 [ 8][7][2];
+static float f34_2_4 [ 4][7][2];
+static float Q_fract_allpass[2][50][3][2];
+static float phi_fract[2][50][2];
 
 static const float g0_Q8[] = {
     0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
@@ -67,7 +65,7 @@ static const float g2_Q4[] = {
      0.16486303567403f,  0.23279856662996f, 0.25f
 };
 
-static void make_filters_from_proto(float (*filter)[8][2], const float *proto, int bands)
+static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
 {
     int q, n;
     for (q = 0; q < bands; q++) {

libavcodec/aacpsdsp.c

@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "config.h"
-#include "libavutil/attributes.h"
-#include "aacpsdsp.h"
-
-static void ps_add_squares_c(float *dst, const float (*src)[2], int n)
-{
-    int i;
-    for (i = 0; i < n; i++)
-        dst[i] += src[i][0] * src[i][0] + src[i][1] * src[i][1];
-}
-
-static void ps_mul_pair_single_c(float (*dst)[2], float (*src0)[2], float *src1,
-                                 int n)
-{
-    int i;
-    for (i = 0; i < n; i++) {
-        dst[i][0] = src0[i][0] * src1[i];
-        dst[i][1] = src0[i][1] * src1[i];
-    }
-}
-
-static void ps_hybrid_analysis_c(float (*out)[2], float (*in)[2],
-                                 const float (*filter)[8][2],
-                                 int stride, int n)
-{
-    int i, j;
-
-    for (i = 0; i < n; i++) {
-        float sum_re = filter[i][6][0] * in[6][0];
-        float sum_im = filter[i][6][0] * in[6][1];
-
-        for (j = 0; j < 6; j++) {
-            float in0_re = in[j][0];
-            float in0_im = in[j][1];
-            float in1_re = in[12-j][0];
-            float in1_im = in[12-j][1];
-            sum_re += filter[i][j][0] * (in0_re + in1_re) -
-                      filter[i][j][1] * (in0_im - in1_im);
-            sum_im += filter[i][j][0] * (in0_im + in1_im) +
-                      filter[i][j][1] * (in0_re - in1_re);
-        }
-        out[i * stride][0] = sum_re;
-        out[i * stride][1] = sum_im;
-    }
-}
-
-static void ps_hybrid_analysis_ileave_c(float (*out)[32][2], float L[2][38][64],
-                                        int i, int len)
-{
-    int j;
-
-    for (; i < 64; i++) {
-        for (j = 0; j < len; j++) {
-            out[i][j][0] = L[0][j][i];
-            out[i][j][1] = L[1][j][i];
-        }
-    }
-}
-
-static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
-                                        float (*in)[32][2],
-                                        int i, int len)
-{
-    int n;
-
-    for (; i < 64; i++) {
-        for (n = 0; n < len; n++) {
-            out[0][n][i] = in[i][n][0];
-            out[1][n][i] = in[i][n][1];
-        }
-    }
-}
-
-static void ps_decorrelate_c(float (*out)[2], float (*delay)[2],
-                             float (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
-                             const float phi_fract[2], float (*Q_fract)[2],
-                             const float *transient_gain,
-                             float g_decay_slope,
-                             int len)
-{
-    static const float a[] = { 0.65143905753106f,
-                               0.56471812200776f,
-                               0.48954165955695f };
-    float ag[PS_AP_LINKS];
-    int m, n;
-
-    for (m = 0; m < PS_AP_LINKS; m++)
-        ag[m] = a[m] * g_decay_slope;
-
-    for (n = 0; n < len; n++) {
-        float in_re = delay[n][0] * phi_fract[0] - delay[n][1] * phi_fract[1];
-        float in_im = delay[n][0] * phi_fract[1] + delay[n][1] * phi_fract[0];
-        for (m = 0; m < PS_AP_LINKS; m++) {
-            float a_re                = ag[m] * in_re;
-            float a_im                = ag[m] * in_im;
-            float link_delay_re       = ap_delay[m][n+2-m][0];
-            float link_delay_im       = ap_delay[m][n+2-m][1];
-            float fractional_delay_re = Q_fract[m][0];
-            float fractional_delay_im = Q_fract[m][1];
-            float apd_re = in_re;
-            float apd_im = in_im;
-            in_re = link_delay_re * fractional_delay_re -
-                    link_delay_im * fractional_delay_im - a_re;
-            in_im = link_delay_re * fractional_delay_im +
-                    link_delay_im * fractional_delay_re - a_im;
-            ap_delay[m][n+5][0] = apd_re + ag[m] * in_re;
-            ap_delay[m][n+5][1] = apd_im + ag[m] * in_im;
-        }
-        out[n][0] = transient_gain[n] * in_re;
-        out[n][1] = transient_gain[n] * in_im;
-    }
-}
-
-static void ps_stereo_interpolate_c(float (*l)[2], float (*r)[2],
-                                    float h[2][4], float h_step[2][4],
-                                    int len)
-{
-    float h0 = h[0][0];
-    float h1 = h[0][1];
-    float h2 = h[0][2];
-    float h3 = h[0][3];
-    float hs0 = h_step[0][0];
-    float hs1 = h_step[0][1];
-    float hs2 = h_step[0][2];
-    float hs3 = h_step[0][3];
-    int n;
-
-    for (n = 0; n < len; n++) {
-        //l is s, r is d
-        float l_re = l[n][0];
-        float l_im = l[n][1];
-        float r_re = r[n][0];
-        float r_im = r[n][1];
-        h0 += hs0;
-        h1 += hs1;
-        h2 += hs2;
-        h3 += hs3;
-        l[n][0] = h0 * l_re + h2 * r_re;
-        l[n][1] = h0 * l_im + h2 * r_im;
-        r[n][0] = h1 * l_re + h3 * r_re;
-        r[n][1] = h1 * l_im + h3 * r_im;
-    }
-}
-
-static void ps_stereo_interpolate_ipdopd_c(float (*l)[2], float (*r)[2],
-                                           float h[2][4], float h_step[2][4],
-                                           int len)
-{
-    float h00  = h[0][0],      h10  = h[1][0];
-    float h01  = h[0][1],      h11  = h[1][1];
-    float h02  = h[0][2],      h12  = h[1][2];
-    float h03  = h[0][3],      h13  = h[1][3];
-    float hs00 = h_step[0][0], hs10 = h_step[1][0];
-    float hs01 = h_step[0][1], hs11 = h_step[1][1];
-    float hs02 = h_step[0][2], hs12 = h_step[1][2];
-    float hs03 = h_step[0][3], hs13 = h_step[1][3];
-    int n;
-
-    for (n = 0; n < len; n++) {
-        //l is s, r is d
-        float l_re = l[n][0];
-        float l_im = l[n][1];
-        float r_re = r[n][0];
-        float r_im = r[n][1];
-        h00 += hs00;
-        h01 += hs01;
-        h02 += hs02;
-        h03 += hs03;
-        h10 += hs10;
-        h11 += hs11;
-        h12 += hs12;
-        h13 += hs13;
-
-        l[n][0] = h00 * l_re + h02 * r_re - h10 * l_im - h12 * r_im;
-        l[n][1] = h00 * l_im + h02 * r_im + h10 * l_re + h12 * r_re;
-        r[n][0] = h01 * l_re + h03 * r_re - h11 * l_im - h13 * r_im;
-        r[n][1] = h01 * l_im + h03 * r_im + h11 * l_re + h13 * r_re;
-    }
-}
-
-av_cold void ff_psdsp_init(PSDSPContext *s)
-{
-    s->add_squares            = ps_add_squares_c;
-    s->mul_pair_single        = ps_mul_pair_single_c;
-    s->hybrid_analysis        = ps_hybrid_analysis_c;
-    s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
-    s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
-    s->decorrelate            = ps_decorrelate_c;
-    s->stereo_interpolate[0]  = ps_stereo_interpolate_c;
-    s->stereo_interpolate[1]  = ps_stereo_interpolate_ipdopd_c;
-
-    if (ARCH_ARM)
-        ff_psdsp_init_arm(s);
-}

libavcodec/aacpsdsp.h

@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2012 Mans Rullgard
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef LIBAVCODEC_AACPSDSP_H
-#define LIBAVCODEC_AACPSDSP_H
-
-#define PS_QMF_TIME_SLOTS 32
-#define PS_AP_LINKS 3
-#define PS_MAX_AP_DELAY 5
-
-typedef struct PSDSPContext {
-    void (*add_squares)(float *dst, const float (*src)[2], int n);
-    void (*mul_pair_single)(float (*dst)[2], float (*src0)[2], float *src1,
-                            int n);
-    void (*hybrid_analysis)(float (*out)[2], float (*in)[2],
-                            const float (*filter)[8][2],
-                            int stride, int n);
-    void (*hybrid_analysis_ileave)(float (*out)[32][2], float L[2][38][64],
-                                   int i, int len);
-    void (*hybrid_synthesis_deint)(float out[2][38][64], float (*in)[32][2],
-                                   int i, int len);
-    void (*decorrelate)(float (*out)[2], float (*delay)[2],
-                        float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
-                        const float phi_fract[2], float (*Q_fract)[2],
-                        const float *transient_gain,
-                        float g_decay_slope,
-                        int len);
-    void (*stereo_interpolate[2])(float (*l)[2], float (*r)[2],
-                                  float h[2][4], float h_step[2][4],
-                                  int len);
-} PSDSPContext;
-
-void ff_psdsp_init(PSDSPContext *s);
-void ff_psdsp_init_arm(PSDSPContext *s);
-
-#endif /* LIBAVCODEC_AACPSDSP_H */

libavcodec/aacpsy.c

@@ -24,8 +24,6 @@
  * AAC encoder psychoacoustic model
  */
 
-#include "libavutil/libm.h"
-
 #include "avcodec.h"
 #include "aactab.h"
 #include "psymodel.h"
@@ -294,7 +292,7 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
     int i, j, g, start;
     float prev, minscale, minath, minsnr, pe_min;
     const int chan_bitrate = ctx->avctx->bit_rate / ctx->avctx->channels;
-    const int bandwidth    = ctx->avctx->cutoff ? ctx->avctx->cutoff : AAC_CUTOFF(ctx->avctx);
+    const int bandwidth    = ctx->avctx->cutoff ? ctx->avctx->cutoff : ctx->avctx->sample_rate / 2;
     const float num_bark   = calc_bark((float)bandwidth);
 
     ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
@@ -335,7 +333,7 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
             coeff->spread_low[1] = pow(10.0, -bark_width * en_spread_low);
             coeff->spread_hi [1] = pow(10.0, -bark_width * en_spread_hi);
             pe_min = bark_pe * bark_width;
-            minsnr = exp2(pe_min / band_sizes[g]) - 1.5f;
+            minsnr = pow(2.0f, pe_min / band_sizes[g]) - 1.5f;
             coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB);
         }
         start = 0;
@@ -391,8 +389,9 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
     AacPsyChannel *pch  = &pctx->ch[channel];
     uint8_t grouping     = 0;
     int next_type        = pch->next_window_seq;
-    FFPsyWindowInfo wi  = { { 0 } };
+    FFPsyWindowInfo wi;
 
+    memset(&wi, 0, sizeof(wi));
     if (la) {
         float s[8], v;
         int switch_to_eight = 0;
@@ -526,11 +525,8 @@ static float calc_reduction_3gpp(float a, float desired_pe, float pe,
 {
     float thr_avg, reduction;
 
-    if(active_lines == 0.0)
-        return 0;
-
-    thr_avg   = exp2f((a - pe) / (4.0f * active_lines));
-    reduction = exp2f((a - desired_pe) / (4.0f * active_lines)) - thr_avg;
+    thr_avg   = powf(2.0f, (a - pe) / (4.0f * active_lines));
+    reduction = powf(2.0f, (a - desired_pe) / (4.0f * active_lines)) - thr_avg;
 
     return FFMAX(reduction, 0.0f);
 }
@@ -568,7 +564,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
     AacPsyChannel *pch  = &pctx->ch[channel];
     int start = 0;
     int i, w, g;
-    float desired_bits, desired_pe, delta_pe, reduction= NAN, spread_en[128] = {0};
+    float desired_bits, desired_pe, delta_pe, reduction, spread_en[128] = {0};
     float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
     float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
     const int      num_bands   = ctx->num_bands[wi->num_windows == 8];
@@ -588,7 +584,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
                 form_factor  += sqrtf(fabs(coefs[start+i]));
             }
             band->thr      = band->energy * 0.001258925f;
-            band->nz_lines = band->energy>0 ? form_factor / powf(band->energy / band_sizes[g], 0.25f) : 0;
+            band->nz_lines = form_factor / powf(band->energy / band_sizes[g], 0.25f);
 
             start += band_sizes[g];
         }
@@ -708,7 +704,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
                         float delta_sfb_pe = band->norm_fac * norm_fac * delta_pe;
                         float thr = band->thr;
 
-                        thr *= exp2f(delta_sfb_pe / band->active_lines);
+                        thr *= powf(2.0f, delta_sfb_pe / band->active_lines);
                         if (thr > coeffs[g].min_snr * band->energy && band->avoid_holes == PSY_3GPP_AH_INACTIVE)
                             thr = FFMAX(band->thr, coeffs[g].min_snr * band->energy);
                         band->thr = thr;
@@ -789,8 +785,9 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
     int uselongblock = 1;
     int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
     int i;
-    FFPsyWindowInfo wi = { { 0 } };
+    FFPsyWindowInfo wi;
 
+    memset(&wi, 0, sizeof(wi));
     if (la) {
         float hpfsmpl[AAC_BLOCK_SIZE_LONG];
         float const *pf = hpfsmpl;

libavcodec/aacsbr.c

@@ -32,7 +32,6 @@
 #include "aacsbrdata.h"
 #include "fft.h"
 #include "aacps.h"
-#include "sbrdsp.h"
 #include "libavutil/libm.h"
 #include "libavutil/avassert.h"
 
@@ -129,24 +128,13 @@ av_cold void ff_aac_sbr_init(void)
     ff_ps_init();
 }
 
-/** Places SBR in pure upsampling mode. */
-static void sbr_turnoff(SpectralBandReplication *sbr) {
-    sbr->start = 0;
-    // Init defults used in pure upsampling mode
-    sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
-    sbr->m[1] = 0;
-    // Reset values for first SBR header
-    sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
-    memset(&sbr->spectrum_params, -1, sizeof(SpectrumParameters));
-}
-
 av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
 {
     float mdct_scale;
     if(sbr->mdct.mdct_bits)
         return;
-    sbr->kx[0] = sbr->kx[1];
-    sbr_turnoff(sbr);
+    sbr->kx[0] = sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
+    sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
     sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     /* SBR requires samples to be scaled to +/-32768.0 to work correctly.
@@ -156,7 +144,6 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
     ff_mdct_init(&sbr->mdct,     7, 1, 1.0 / (64 * mdct_scale));
     ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
     ff_ps_ctx_init(&sbr->ps);
-    ff_sbrdsp_init(&sbr->dsp);
 }
 
 av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
@@ -555,7 +542,7 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
             k = sbr->n_master;
     } while (sb != sbr->kx[1] + sbr->m[1]);
 
-    if (sbr->patch_num_subbands[sbr->num_patches-1] < 3 && sbr->num_patches > 1)
+    if (sbr->num_patches > 1 && sbr->patch_num_subbands[sbr->num_patches-1] < 3)
         sbr->num_patches--;
 
     return 0;
@@ -918,7 +905,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
 {
     switch (bs_extension_id) {
     case EXTENSION_ID_PS:
-        if (!ac->oc[1].m4ac.ps) {
+        if (!ac->m4ac.ps) {
             av_log(ac->avctx, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
             *num_bits_left = 0;
@@ -933,9 +920,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
         }
         break;
     default:
-        // some files contain 0-padding
-        if (bs_extension_id || *num_bits_left > 16 || show_bits(gb, *num_bits_left))
-            av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
+        av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
         skip_bits_long(gb, *num_bits_left); // bs_fill_bits
         *num_bits_left = 0;
         break;
@@ -1011,18 +996,18 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
 
     if (id_aac == TYPE_SCE || id_aac == TYPE_CCE) {
         if (read_sbr_single_channel_element(ac, sbr, gb)) {
-            sbr_turnoff(sbr);
+            sbr->start = 0;
             return get_bits_count(gb) - cnt;
         }
     } else if (id_aac == TYPE_CPE) {
         if (read_sbr_channel_pair_element(ac, sbr, gb)) {
-            sbr_turnoff(sbr);
+            sbr->start = 0;
             return get_bits_count(gb) - cnt;
         }
     } else {
         av_log(ac->avctx, AV_LOG_ERROR,
             "Invalid bitstream - cannot apply SBR to element type %d\n", id_aac);
-        sbr_turnoff(sbr);
+        sbr->start = 0;
         return get_bits_count(gb) - cnt;
     }
     if (get_bits1(gb)) { // bs_extended_data
@@ -1054,7 +1039,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
     if (err < 0) {
         av_log(ac->avctx, AV_LOG_ERROR,
                "SBR reset failed. Switching SBR to pure upsampling mode.\n");
-        sbr_turnoff(sbr);
+        sbr->start = 0;
     }
 }
 
@@ -1077,9 +1062,9 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
     sbr->reset = 0;
 
     if (!sbr->sample_rate)
-        sbr->sample_rate = 2 * ac->oc[1].m4ac.sample_rate; //TODO use the nominal sample rate for arbitrary sample rate support
-    if (!ac->oc[1].m4ac.ext_sample_rate)
-        ac->oc[1].m4ac.ext_sample_rate = 2 * ac->oc[1].m4ac.sample_rate;
+        sbr->sample_rate = 2 * ac->m4ac.sample_rate; //TODO use the nominal sample rate for arbitrary sample rate support
+    if (!ac->m4ac.ext_sample_rate)
+        ac->m4ac.ext_sample_rate = 2 * ac->m4ac.sample_rate;
 
     if (crc) {
         skip_bits(gb, 10); // bs_sbr_crc_bits; TODO - implement CRC check
@@ -1089,7 +1074,6 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
     //Save some state from the previous frame.
     sbr->kx[0] = sbr->kx[1];
     sbr->m[0] = sbr->m[1];
-    sbr->kx_and_m_pushed = 1;
 
     num_sbr_bits++;
     if (get_bits1(gb)) // bs_header_flag
@@ -1124,12 +1108,7 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
             for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
                 float temp1 = exp2f(sbr->data[0].env_facs[e][k] * alpha + 7.0f);
                 float temp2 = exp2f((pan_offset - sbr->data[1].env_facs[e][k]) * alpha);
-                float fac;
-                if (temp1 > 1E20) {
-                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
-                    temp1 = 1;
-                }
-                fac   = temp1 / (1.0f + temp2);
+                float fac   = temp1 / (1.0f + temp2);
                 sbr->data[0].env_facs[e][k] = fac;
                 sbr->data[1].env_facs[e][k] = fac * temp2;
             }
@@ -1138,12 +1117,7 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
             for (k = 0; k < sbr->n_q; k++) {
                 float temp1 = exp2f(NOISE_FLOOR_OFFSET - sbr->data[0].noise_facs[e][k] + 1);
                 float temp2 = exp2f(12 - sbr->data[1].noise_facs[e][k]);
-                float fac;
-                if (temp1 > 1E20) {
-                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
-                    temp1 = 1;
-                }
-                fac = temp1 / (1.0f + temp2);
+                float fac   = temp1 / (1.0f + temp2);
                 sbr->data[0].noise_facs[e][k] = fac;
                 sbr->data[1].noise_facs[e][k] = fac * temp2;
             }
@@ -1152,15 +1126,9 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
         for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
             float alpha = sbr->data[ch].bs_amp_res ? 1.0f : 0.5f;
             for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
-                for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
+                for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++)
                     sbr->data[ch].env_facs[e][k] =
                         exp2f(alpha * sbr->data[ch].env_facs[e][k] + 6.0f);
-                    if (sbr->data[ch].env_facs[e][k] > 1E20) {
-                        av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
-                        sbr->data[ch].env_facs[e][k] = 1;
-                    }
-                }
-
             for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
                 for (k = 0; k < sbr->n_q; k++)
                     sbr->data[ch].noise_facs[e][k] =
@@ -1175,21 +1143,33 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
  * @param   x       pointer to the beginning of the first sample window
  * @param   W       array of complex-valued samples split into subbands
  */
-static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
-                             SBRDSPContext *sbrdsp, const float *in, float *x,
+static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in, float *x,
                              float z[320], float W[2][32][32][2])
 {
-    int i;
+    int i, k;
     memcpy(W[0], W[1], sizeof(W[0]));
     memcpy(x    , x+1024, (320-32)*sizeof(x[0]));
     memcpy(x+288, in,         1024*sizeof(x[0]));
     for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
                                // are not supported
         dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
-        sbrdsp->sum64x5(z);
-        sbrdsp->qmf_pre_shuffle(z);
+        for (k = 0; k < 64; k++) {
+            float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
+            z[k] = f;
+        }
+        //Shuffle to IMDCT
+        z[64] = z[0];
+        for (k = 1; k < 32; k++) {
+            z[64+2*k-1] =  z[   k];
+            z[64+2*k  ] = -z[64-k];
+        }
+        z[64+63] = z[32];
+
         mdct->imdct_half(mdct, z, z+64);
-        sbrdsp->qmf_post_shuffle(W[1][i], z);
+        for (k = 0; k < 32; k++) {
+            W[1][i][k][0] = -z[63-k];
+            W[1][i][k][1] = z[k];
+        }
         x += 32;
     }
 }
@@ -1199,7 +1179,6 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
  * (14496-3 sp04 p206)
  */
 static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
-                              SBRDSPContext *sbrdsp,
                               float *out, float X[2][38][64],
                               float mdct_buf[2][64],
                               float *v0, int *v_off, const unsigned int div)
@@ -1223,12 +1202,20 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
                 X[0][i][32+n] =  X[1][i][31-n];
             }
             mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
-            sbrdsp->qmf_deint_neg(v, mdct_buf[0]);
+            for (n = 0; n < 32; n++) {
+                v[     n] =  mdct_buf[0][63 - 2*n];
+                v[63 - n] = -mdct_buf[0][62 - 2*n];
+            }
         } else {
-            sbrdsp->neg_odd_64(X[1][i]);
+            for (n = 1; n < 64; n+=2) {
+                X[1][i][n] = -X[1][i][n];
+            }
             mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
             mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);
-            sbrdsp->qmf_deint_bfly(v, mdct_buf[1], mdct_buf[0]);
+            for (n = 0; n < 64; n++) {
+                v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+                v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+            }
         }
         dsp->vector_fmul_add(out, v                , sbr_qmf_window               , zero64, 64 >> div);
         dsp->vector_fmul_add(out, v + ( 192 >> div), sbr_qmf_window + ( 64 >> div), out   , 64 >> div);
@@ -1244,20 +1231,45 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
     }
 }
 
+static void autocorrelate(const float x[40][2], float phi[3][2][2], int lag)
+{
+    int i;
+    float real_sum = 0.0f;
+    float imag_sum = 0.0f;
+    if (lag) {
+        for (i = 1; i < 38; i++) {
+            real_sum += x[i][0] * x[i+lag][0] + x[i][1] * x[i+lag][1];
+            imag_sum += x[i][0] * x[i+lag][1] - x[i][1] * x[i+lag][0];
+        }
+        phi[2-lag][1][0] = real_sum + x[ 0][0] * x[lag][0] + x[ 0][1] * x[lag][1];
+        phi[2-lag][1][1] = imag_sum + x[ 0][0] * x[lag][1] - x[ 0][1] * x[lag][0];
+        if (lag == 1) {
+            phi[0][0][0] = real_sum + x[38][0] * x[39][0] + x[38][1] * x[39][1];
+            phi[0][0][1] = imag_sum + x[38][0] * x[39][1] - x[38][1] * x[39][0];
+        }
+    } else {
+        for (i = 1; i < 38; i++) {
+            real_sum += x[i][0] * x[i][0] + x[i][1] * x[i][1];
+        }
+        phi[2][1][0] = real_sum + x[ 0][0] * x[ 0][0] + x[ 0][1] * x[ 0][1];
+        phi[1][0][0] = real_sum + x[38][0] * x[38][0] + x[38][1] * x[38][1];
+    }
+}
+
 /** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
  * (14496-3 sp04 p214)
  * Warning: This routine does not seem numerically stable.
  */
-static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
-                                  float (*alpha0)[2], float (*alpha1)[2],
+static void sbr_hf_inverse_filter(float (*alpha0)[2], float (*alpha1)[2],
                                   const float X_low[32][40][2], int k0)
 {
     int k;
     for (k = 0; k < k0; k++) {
-        LOCAL_ALIGNED_16(float, phi, [3], [2][2]);
-        float dk;
+        float phi[3][2][2], dk;
 
-        dsp->autocorrelate(X_low[k], phi);
+        autocorrelate(X_low[k], phi, 0);
+        autocorrelate(X_low[k], phi, 1);
+        autocorrelate(X_low[k], phi, 2);
 
         dk =  phi[2][1][0] * phi[1][0][0] -
              (phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
@@ -1353,11 +1365,12 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
                       const float bw_array[5], const uint8_t *t_env,
                       int bs_num_env)
 {
-    int j, x;
+    int i, j, x;
     int g = 0;
     int k = sbr->kx[1];
     for (j = 0; j < sbr->num_patches; j++) {
         for (x = 0; x < sbr->patch_num_subbands[j]; x++, k++) {
+            float alpha[4];
             const int p = sbr->patch_start_subband[j] + x;
             while (g <= sbr->n_q && k >= sbr->f_tablenoise[g])
                 g++;
@@ -1369,10 +1382,26 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
                 return -1;
             }
 
-            sbr->dsp.hf_gen(X_high[k] + ENVELOPE_ADJUSTMENT_OFFSET,
-                            X_low[p]  + ENVELOPE_ADJUSTMENT_OFFSET,
-                            alpha0[p], alpha1[p], bw_array[g],
-                            2 * t_env[0], 2 * t_env[bs_num_env]);
+            alpha[0] = alpha1[p][0] * bw_array[g] * bw_array[g];
+            alpha[1] = alpha1[p][1] * bw_array[g] * bw_array[g];
+            alpha[2] = alpha0[p][0] * bw_array[g];
+            alpha[3] = alpha0[p][1] * bw_array[g];
+
+            for (i = 2 * t_env[0]; i < 2 * t_env[bs_num_env]; i++) {
+                const int idx = i + ENVELOPE_ADJUSTMENT_OFFSET;
+                X_high[k][idx][0] =
+                    X_low[p][idx - 2][0] * alpha[0] -
+                    X_low[p][idx - 2][1] * alpha[1] +
+                    X_low[p][idx - 1][0] * alpha[2] -
+                    X_low[p][idx - 1][1] * alpha[3] +
+                    X_low[p][idx][0];
+                X_high[k][idx][1] =
+                    X_low[p][idx - 2][1] * alpha[0] +
+                    X_low[p][idx - 2][0] * alpha[1] +
+                    X_low[p][idx - 1][1] * alpha[2] +
+                    X_low[p][idx - 1][0] * alpha[3] +
+                    X_low[p][idx][1];
+            }
         }
     }
     if (k < sbr->m[1] + sbr->kx[1])
@@ -1383,8 +1412,8 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
 
 /// Generate the subband filtered lowband
 static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
-                     const float Y0[38][64][2], const float Y1[38][64][2],
-                     const float X_low[32][40][2], int ch)
+                     const float X_low[32][40][2], const float Y[2][38][64][2],
+                     int ch)
 {
     int k, i;
     const int i_f = 32;
@@ -1398,8 +1427,8 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
     }
     for (; k < sbr->kx[0] + sbr->m[0]; k++) {
         for (i = 0; i < i_Temp; i++) {
-            X[0][i][k] = Y0[i + i_f][k][0];
-            X[1][i][k] = Y0[i + i_f][k][1];
+            X[0][i][k] = Y[0][i + i_f][k][0];
+            X[1][i][k] = Y[0][i + i_f][k][1];
         }
     }
 
@@ -1411,8 +1440,8 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
     }
     for (; k < sbr->kx[1] + sbr->m[1]; k++) {
         for (i = i_Temp; i < i_f; i++) {
-            X[0][i][k] = Y1[i][k][0];
-            X[1][i][k] = Y1[i][k][1];
+            X[0][i][k] = Y[1][i][k][0];
+            X[1][i][k] = Y[1][i][k][1];
         }
     }
     return 0;
@@ -1421,7 +1450,7 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
 /** High Frequency Adjustment (14496-3 sp04 p217) and Mapping
  * (14496-3 sp04 p217)
  */
-static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
+static void sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
                         SBRData *ch_data, int e_a[2])
 {
     int e, i, m;
@@ -1432,12 +1461,7 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
         uint16_t *table = ch_data->bs_freq_res[e + 1] ? sbr->f_tablehigh : sbr->f_tablelow;
         int k;
 
-        if (sbr->kx[1] != table[0]) {
-            av_log(ac->avctx, AV_LOG_ERROR, "kx != f_table{high,low}[0]. "
-                   "Derived frequency tables were not regenerated.\n");
-            sbr_turnoff(sbr);
-            return AVERROR_BUG;
-        }
+        av_assert0(sbr->kx[1] <= table[0]);
         for (i = 0; i < ilim; i++)
             for (m = table[i]; m < table[i + 1]; m++)
                 sbr->e_origmapped[e][m - sbr->kx[1]] = ch_data->env_facs[e+1][i];
@@ -1472,15 +1496,13 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
     }
 
     memcpy(ch_data->s_indexmapped[0], ch_data->s_indexmapped[ch_data->bs_num_env], sizeof(ch_data->s_indexmapped[0]));
-    return 0;
 }
 
 /// Estimation of current envelope (14496-3 sp04 p218)
 static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
                              SpectralBandReplication *sbr, SBRData *ch_data)
 {
-    int e, m;
-    int kx1 = sbr->kx[1];
+    int e, i, m;
 
     if (sbr->bs_interpol_freq) {
         for (e = 0; e < ch_data->bs_num_env; e++) {
@@ -1489,7 +1511,12 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
             int iub = ch_data->t_env[e + 1] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;
 
             for (m = 0; m < sbr->m[1]; m++) {
-                float sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
+                float sum = 0.0f;
+
+                for (i = ilb; i < iub; i++) {
+                    sum += X_high[m + sbr->kx[1]][i][0] * X_high[m + sbr->kx[1]][i][0] +
+                           X_high[m + sbr->kx[1]][i][1] * X_high[m + sbr->kx[1]][i][1];
+                }
                 e_curr[e][m] = sum * recip_env_size;
             }
         }
@@ -1507,11 +1534,14 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
                 const int den = env_size * (table[p + 1] - table[p]);
 
                 for (k = table[p]; k < table[p + 1]; k++) {
-                    sum += sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb);
+                    for (i = ilb; i < iub; i++) {
+                        sum += X_high[k][i][0] * X_high[k][i][0] +
+                               X_high[k][i][1] * X_high[k][i][1];
+                    }
                 }
                 sum /= den;
                 for (k = table[p]; k < table[p + 1]; k++) {
-                    e_curr[e][k - kx1] = sum;
+                    e_curr[e][k - sbr->kx[1]] = sum;
                 }
             }
         }
@@ -1578,8 +1608,7 @@ static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
 }
 
 /// Assembling HF Signals (14496-3 sp04 p220)
-static void sbr_hf_assemble(float Y1[38][64][2],
-                            const float X_high[64][40][2],
+static void sbr_hf_assemble(float Y[2][38][64][2], const float X_high[64][40][2],
                             SpectralBandReplication *sbr, SBRData *ch_data,
                             const int e_a[2])
 {
@@ -1594,9 +1623,14 @@ static void sbr_hf_assemble(float Y1[38][64][2],
         0.11516383427084,
         0.03183050093751,
     };
+    static const int8_t phi[2][4] = {
+        {  1,  0, -1,  0}, // real
+        {  0,  1,  0, -1}, // imaginary
+    };
     float (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
     int indexnoise = ch_data->f_indexnoise;
     int indexsine  = ch_data->f_indexsine;
+    memcpy(Y[0], Y[1], sizeof(Y[0]));
 
     if (sbr->reset) {
         for (i = 0; i < h_SL; i++) {
@@ -1617,48 +1651,64 @@ static void sbr_hf_assemble(float Y1[38][64][2],
 
     for (e = 0; e < ch_data->bs_num_env; e++) {
         for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
-            LOCAL_ALIGNED_16(float, g_filt_tab, [48]);
-            LOCAL_ALIGNED_16(float, q_filt_tab, [48]);
-            float *g_filt, *q_filt;
+            int phi_sign = (1 - 2*(kx & 1));
 
             if (h_SL && e != e_a[0] && e != e_a[1]) {
-                g_filt = g_filt_tab;
-                q_filt = q_filt_tab;
                 for (m = 0; m < m_max; m++) {
                     const int idx1 = i + h_SL;
-                    g_filt[m] = 0.0f;
-                    q_filt[m] = 0.0f;
-                    for (j = 0; j <= h_SL; j++) {
-                        g_filt[m] += g_temp[idx1 - j][m] * h_smooth[j];
-                        q_filt[m] += q_temp[idx1 - j][m] * h_smooth[j];
-                    }
+                    float g_filt = 0.0f;
+                    for (j = 0; j <= h_SL; j++)
+                        g_filt += g_temp[idx1 - j][m] * h_smooth[j];
+                    Y[1][i][m + kx][0] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][0] * g_filt;
+                    Y[1][i][m + kx][1] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][1] * g_filt;
                 }
             } else {
-                g_filt = g_temp[i + h_SL];
-                q_filt = q_temp[i];
+                for (m = 0; m < m_max; m++) {
+                    const float g_filt = g_temp[i + h_SL][m];
+                    Y[1][i][m + kx][0] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][0] * g_filt;
+                    Y[1][i][m + kx][1] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][1] * g_filt;
+                }
             }
 
-            sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
-                               i + ENVELOPE_ADJUSTMENT_OFFSET);
-
             if (e != e_a[0] && e != e_a[1]) {
-                sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
-                                                   q_filt, indexnoise,
-                                                   kx, m_max);
-            } else {
-                int idx = indexsine&1;
-                int A = (1-((indexsine+(kx & 1))&2));
-                int B = (A^(-idx)) + idx;
-                float *out = &Y1[i][kx][idx];
-                float *in  = sbr->s_m[e];
-                for (m = 0; m+1 < m_max; m+=2) {
-                    out[2*m  ] += in[m  ] * A;
-                    out[2*m+2] += in[m+1] * B;
+                for (m = 0; m < m_max; m++) {
+                    indexnoise = (indexnoise + 1) & 0x1ff;
+                    if (sbr->s_m[e][m]) {
+                        Y[1][i][m + kx][0] +=
+                            sbr->s_m[e][m] * phi[0][indexsine];
+                        Y[1][i][m + kx][1] +=
+                            sbr->s_m[e][m] * (phi[1][indexsine] * phi_sign);
+                    } else {
+                        float q_filt;
+                        if (h_SL) {
+                            const int idx1 = i + h_SL;
+                            q_filt = 0.0f;
+                            for (j = 0; j <= h_SL; j++)
+                                q_filt += q_temp[idx1 - j][m] * h_smooth[j];
+                        } else {
+                            q_filt = q_temp[i][m];
+                        }
+                        Y[1][i][m + kx][0] +=
+                            q_filt * sbr_noise_table[indexnoise][0];
+                        Y[1][i][m + kx][1] +=
+                            q_filt * sbr_noise_table[indexnoise][1];
+                    }
+                    phi_sign = -phi_sign;
+                }
+            } else {
+                indexnoise = (indexnoise + m_max) & 0x1ff;
+                for (m = 0; m < m_max; m++) {
+                    Y[1][i][m + kx][0] +=
+                        sbr->s_m[e][m] * phi[0][indexsine];
+                    Y[1][i][m + kx][1] +=
+                        sbr->s_m[e][m] * (phi[1][indexsine] * phi_sign);
+                    phi_sign = -phi_sign;
                 }
-                if(m_max&1)
-                    out[2*m  ] += in[m  ] * A;
             }
-            indexnoise = (indexnoise + m_max) & 0x1ff;
             indexsine = (indexsine + 1) & 3;
         }
     }
@@ -1669,54 +1719,39 @@ static void sbr_hf_assemble(float Y1[38][64][2],
 void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
                   float* L, float* R)
 {
-    int downsampled = ac->oc[1].m4ac.ext_sample_rate < sbr->sample_rate;
+    int downsampled = ac->m4ac.ext_sample_rate < sbr->sample_rate;
     int ch;
     int nch = (id_aac == TYPE_CPE) ? 2 : 1;
-    int err;
-
-    if (!sbr->kx_and_m_pushed) {
-        sbr->kx[0] = sbr->kx[1];
-        sbr->m[0] = sbr->m[1];
-    } else {
-        sbr->kx_and_m_pushed = 0;
-    }
 
     if (sbr->start) {
         sbr_dequant(sbr, id_aac);
     }
     for (ch = 0; ch < nch; ch++) {
         /* decode channel */
-        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
+        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
                          (float*)sbr->qmf_filter_scratch,
                          sbr->data[ch].W);
         sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W);
-        sbr->data[ch].Ypos ^= 1;
         if (sbr->start) {
-            sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1, sbr->X_low, sbr->k[0]);
+            sbr_hf_inverse_filter(sbr->alpha0, sbr->alpha1, sbr->X_low, sbr->k[0]);
             sbr_chirp(sbr, &sbr->data[ch]);
             sbr_hf_gen(ac, sbr, sbr->X_high, sbr->X_low, sbr->alpha0, sbr->alpha1,
                        sbr->data[ch].bw_array, sbr->data[ch].t_env,
                        sbr->data[ch].bs_num_env);
 
             // hf_adj
-            err = sbr_mapping(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
-            if (!err) {
-                sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
-                sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
-                sbr_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
-                                sbr->X_high, sbr, &sbr->data[ch],
-                                sbr->data[ch].e_a);
-            }
+            sbr_mapping(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
+            sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
+            sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
+            sbr_hf_assemble(sbr->data[ch].Y, sbr->X_high, sbr, &sbr->data[ch],
+                            sbr->data[ch].e_a);
         }
 
         /* synthesis */
-        sbr_x_gen(sbr, sbr->X[ch],
-                  sbr->data[ch].Y[1-sbr->data[ch].Ypos],
-                  sbr->data[ch].Y[  sbr->data[ch].Ypos],
-                  sbr->X_low, ch);
+        sbr_x_gen(sbr, sbr->X[ch], sbr->X_low, sbr->data[ch].Y, ch);
     }
 
-    if (ac->oc[1].m4ac.ps == 1) {
+    if (ac->m4ac.ps == 1) {
         if (sbr->ps.start) {
             ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
         } else {
@@ -1725,12 +1760,12 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
         nch = 2;
     }
 
-    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, L, sbr->X[0], sbr->qmf_filter_scratch,
+    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
                       sbr->data[0].synthesis_filterbank_samples,
                       &sbr->data[0].synthesis_filterbank_samples_offset,
                       downsampled);
     if (nch == 2)
-        sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, R, sbr->X[1], sbr->qmf_filter_scratch,
+        sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, R, sbr->X[1], sbr->qmf_filter_scratch,
                           sbr->data[1].synthesis_filterbank_samples,
                           &sbr->data[1].synthesis_filterbank_samples_offset,
                           downsampled);