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175 Commits

Author SHA1 Message Date
Michael Niedermayer
d39b183d8d Update for 0.10.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-17 01:41:41 +01:00
Stefano Sabatini
dc8054128a lavfi: port MP swapuv filter
(cherry picked from commit fa35d880aa)

Conflicts:

	Changelog
	libavfilter/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-17 00:36:18 +01:00
Michael Niedermayer
001f4c7dc6 jpeglsdec: Prevent out of array write.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 00ab9cdae1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 16:46:30 +01:00
Michael Niedermayer
313ddbfe48 proresdec: Fix read via negative index in a global array.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0065080320)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 16:42:49 +01:00
Michael Niedermayer
7f5bd6c72b diracdec: Correct the bytestream end pointer.
This fixes some arith decoder overreads and a potential infinite loop.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0f13cc732b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 16:00:07 +01:00
Michael Niedermayer
0be85fd80f diracdec: Check for negative quants which would cause out of array reads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5cd8afee99)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 15:59:30 +01:00
Michael Niedermayer
9f253ebb41 diracdec: Fix integer overflow leading to out of global array read.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9729f140ae)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 15:59:21 +01:00
Michael Niedermayer
6242dae507 sonic: update to new API
Fixes Ticket1075

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6f9803e5e0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 15:23:10 +01:00
Michael Niedermayer
1749b0d74d mmvideo: restore initial y value.
This bug might have been exploitable (out of HEAP buffer writes)

Bug introduced by libav
	commit a55d5bdc6e
	Date:   Tue Mar 6 15:15:42 2012 -0800

	    algmm: convert to bytestream2 API.
(cherry picked from commit c2e3b564b3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 14:46:08 +01:00
Michael Niedermayer
568e9062bd Merge remote-tracking branch 'qatar/release/0.8' into release/0.10
* qatar/release/0.8: (154 commits)
  Update Changelog for the 0.8.1 Release
  dca: include libavutil/mathematics.h for possibly missing M_SQRT1_2
  dca: don't use av_clip_uintp2().
  snow: check reference frame indices.
  snow: reject unsupported chroma shifts.
  xa_adpcm: limit filter to prevent xa_adpcm_table[] array bounds overruns.
  h264: increase reference poc list from 16 to 32.
  h264: stricter reference limit enforcement.
  h264: improve parsing of broken AVC SPS
  Replace computations of remaining bits with calls to get_bits_left().
  png: convert to bytestream2 API.
  roqvideo: convert to bytestream2 API.
  smc: port to bytestream2 API.
  tgq: convert to bytestream2 API.
  algmm: convert to bytestream2 API.
  jvdec: unbreak video decoding
  h264: Fix invalid interlaced/progressive MB combinations for direct mode prediction.
  libx264: add 'stats' private option for setting 2pass stats filename.
  libx264: fix help text for slice-max-size option.
  avconv: reindent
  ...

Conflicts:
	Changelog
	RELEASE
	avconv.c
	doc/APIchanges
	ffplay.c
	libavcodec/Makefile
	libavcodec/aacdec.c
	libavcodec/alsdec.c
	libavcodec/atrac3.c
	libavcodec/avcodec.h
	libavcodec/dvdata.c
	libavcodec/fraps.c
	libavcodec/golomb.h
	libavcodec/h264.c
	libavcodec/h264.h
	libavcodec/h264_cabac.c
	libavcodec/h264_cavlc.c
	libavcodec/h264_direct.c
	libavcodec/h264_parser.c
	libavcodec/h264_ps.c
	libavcodec/h264idct_template.c
	libavcodec/indeo3.c
	libavcodec/kgv1dec.c
	libavcodec/kmvc.c
	libavcodec/mjpegbdec.c
	libavcodec/mmvideo.c
	libavcodec/mpegaudiodec.c
	libavcodec/mpegvideo.h
	libavcodec/options.c
	libavcodec/pngdec.c
	libavcodec/roqvideodec.c
	libavcodec/shorten.c
	libavcodec/svq3.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavcodec/xxan.c
	libavformat/Makefile
	libavformat/asfdec.c
	libavformat/dv.c
	libavformat/mov.c
	libavformat/nsvdec.c
	libavformat/utils.c
	libavformat/version.h
	libavutil/avutil.h
	libavutil/error.c
	libavutil/error.h
	libswscale/swscale.c
	libswscale/utils.c
	libswscale/x86/swscale_template.c
	tests/ref/acodec/g722

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 09:01:08 +01:00
Michael Niedermayer
5dbc75870f qpeg: Fix out of array writes.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 06:29:10 +01:00
Fabian Greffrath
c91a14638e srtdec: fix a format string vulnerability.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit aaa1173de7)
2012-03-16 06:29:10 +01:00
Nathan Caldwell
c00c380724 aacenc: Fix LONG_START windowing.
Forgot to add the equivalent amount to the incoming sample pointer as the output pointer.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 2e626dd513)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 06:29:10 +01:00
Nathan Caldwell
43625c5128 aacenc: Fix a bug where deinterleaved samples were stored in the wrong place.
10l: Forgot to adjust deinterleave for new location of incoming samples in 7946a5a.

This produced incorrect, but surprisingly listenable results.

Thanks to Justin Ruggles for the report.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit dc7e7d4dd9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-16 06:29:10 +01:00
Reinhard Tartler
5effcfa767 Update Changelog for the 0.8.1 Release 2012-03-15 08:58:14 +01:00
Kostya Shishkov
1ee0cd1ad7 dca: include libavutil/mathematics.h for possibly missing M_SQRT1_2
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
2012-03-14 23:32:15 +01:00
Ronald S. Bultje
b594732475 dca: don't use av_clip_uintp2().
The argument is not a literal, thus causing the ARM v6 or later
builds to break.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
2012-03-14 23:30:19 +01:00
Michael Niedermayer
ce15406e78 snow: check reference frame indices.
Fixes NULL ptr dereference

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1f8ff2b13c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:35:09 +01:00
Michael Niedermayer
c9e95636a8 snow: reject unsupported chroma shifts.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit c9837954e7)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:34:55 +01:00
Ronald S. Bultje
6e5c07f4c8 xa_adpcm: limit filter to prevent xa_adpcm_table[] array bounds overruns.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 86020073db)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:34:36 +01:00
Ronald S. Bultje
c999a8ed65 h264: increase reference poc list from 16 to 32.
Interlaced images can have 32 references (16 per field), so limiting the
array size to 16 leads to invalid writes.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 48cbe4b092)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:34:13 +01:00
Ronald S. Bultje
4d343a6f47 h264: stricter reference limit enforcement.
Progressive images can have only 16 references, error out if there are
more, since the data is almost certainly corrupt, and the invalid value
will lead to random crashes or invalid writes later on.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit e0febda22d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:33:15 +01:00
Michael Niedermayer
a81a6d9c80 h264: improve parsing of broken AVC SPS
Parsing the entire NAL as SPS fixes decoding of some AVC bitstreams
with broken escaping. Since the size of the NAL unit is known and
checked against the buffer end we can parse it entirely without buffer
overreads.

Fixes playback of
http://streams.videolan.org/streams/mp4/Mr_MrsSmith-h264_aac.mp4

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit 3aa661ec56)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:27:22 +01:00
Alex Converse
48f0eeb2e5 Replace computations of remaining bits with calls to get_bits_left().
(cherry picked from commit 3574a85ce5)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:27:16 +01:00
Ronald S. Bultje
d26e47bf6c png: convert to bytestream2 API.
Protects against overreads in the input buffer.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 4c25269ced)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:14:28 +01:00
Ronald S. Bultje
568a474a08 roqvideo: convert to bytestream2 API.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit cdf1577162)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:09:40 +01:00
Ronald S. Bultje
9a66cdbc16 smc: port to bytestream2 API.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 8febcb9fc1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:09:28 +01:00
Ronald S. Bultje
ddb1149e25 tgq: convert to bytestream2 API.
This protects against input buffer overreads.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 1255eed533)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:09:19 +01:00
Ronald S. Bultje
f6778f58d4 algmm: convert to bytestream2 API.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit a55d5bdc6e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:09:19 +01:00
Paul B Mahol
e4e4d92641 jvdec: unbreak video decoding
The safe bitstream reader broke it since the buffer size was specified
in bytes instead of bits.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
CC: libav-stable@libav.org
(cherry picked from commit a1c036e961)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:02:23 +01:00
Michael Niedermayer
de0ff4ce69 h264: Fix invalid interlaced/progressive MB combinations for direct mode prediction.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 758ec11153)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:00:52 +01:00
Anton Khirnov
6548cb2578 libx264: add 'stats' private option for setting 2pass stats filename.
x264 always opens the file itself with fopen, so we cannot use the
standard lavc stats mechanism.

CC: libav-stable@libav.org
(cherry picked from commit d533e395e1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:00:12 +01:00
Anton Khirnov
f6257cf4b7 libx264: fix help text for slice-max-size option.
CC: libav-stable@libav.org
(cherry picked from commit 9d5c131ece)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 21:00:02 +01:00
Anton Khirnov
a15adb18fa avconv: reindent
CC: libav-stable@libav.org
(cherry picked from commit 64334ddbbc)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:59:00 +01:00
Anton Khirnov
666bd5848a avconv: link '-passlogfile' option to libx264 'stats' AVOption.
Fixes bug 204.

CC: libav-stable@libav.org
(cherry picked from commit 6e8be949f1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:57:11 +01:00
Janne Grunau
d94256d36c Revert "h264: clear trailing bits in partially parsed NAL units"
This reverts commit 729ebb2f18.

There was an off-by-one error in the bit mask calculation clearing
actually the last valid bit and causing
http://bugzilla.libav.org/show_bug.cgi?id=227

The broken sample (Mr_MrsSmith-h264_aac.mp4) the commit was fixing
does not work after correcting the off-by-one error.

CC: libav-stable@libav.org
(cherry picked from commit 8a6037c390)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:56:55 +01:00
Ronald S. Bultje
7bb97a61df mpc: pad mpc_CC/SCF[] tables to allow for negative indices.
MPC8 allows indices of mpc_CC up to -1, and mpc_SCF up to -6, thus pad
the tables by that much on the left end.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit d7eabd5042)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:48:29 +01:00
Ronald S. Bultje
c65eadee5d xxan: protect against chroma LUT overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit f77bfa8376)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:47:19 +01:00
Ronald S. Bultje
a43f4bd601 xxan: convert to bytestream2 API.
Protects against overreads.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 5518827816)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:47:19 +01:00
Ronald S. Bultje
8f881885c2 xxan: don't read before start of buffer in av_memcpy_backptr().
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit f1279e286b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:47:19 +01:00
Ronald S. Bultje
26521d87ba dsicinvideo: validate buffer offset before copying pixels.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit c95fefa042)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:47:19 +01:00
Ronald S. Bultje
e1a4143793 cook: error out on quant_index values outside [-63, 63] range.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 97e48b2f54)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:47:19 +01:00
Ronald S. Bultje
b9482a6efd cook: extend channel uncoupling tables so the full bit range is covered.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 37cc8600d0)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-14 20:47:19 +01:00
Ronald S. Bultje
88c3cc019c cook: expand dither_tab[], and make sure indexes into it don't overflow.
Fixes overflows in accessing dither_tab[].

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 442c3a8cb1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:40:29 +01:00
Ronald S. Bultje
9980e4df3b huffyuv: add padding to classic (v1) huffman tables.
We slightly overread the input buffer, so we require
padding at the end of the buffer, as is documented in the
get_bits API. Without padding, we'll read uninitialized
data or beyond the end of the .rodata, which may crash.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 4ffe5e2aa5)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:36:39 +01:00
Ronald S. Bultje
d4f2786cda avs: fix infinite loop on end-of-stream.
The codec would keep returning the last decoded frame if the stream
contains B-frames, since it wouldn't clear that frame from the list of
frames to be returned to the user.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 83f15a1228)

Conflicts:

	libavcodec/cavsdec.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:36:02 +01:00
Alex Converse
2744fdbd9e tiffdec: Prevent illegal memory access caused by recycled pointers.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit fd0be63049)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:30:55 +01:00
Ronald S. Bultje
1fcc2c6091 wma: fix off-by-one in array bounds check.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit b4bccf3e4e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:30:39 +01:00
Ronald S. Bultje
74871ac70a dv: check buffer size before reading profile.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit e97efecec8)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:30:21 +01:00
Ronald S. Bultje
9cb7f6e54a raw: move buffer size check up.
This way, it protects against overreads for 4bpp/2bpp content also.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit cc5dd632ce)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:23:58 +01:00
Ronald S. Bultje
ed6aaf579d dca: prevent accessing static arrays with invalid indexes.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit e6ffd997cb)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:22:32 +01:00
Ronald S. Bultje
e1b4614ab4 lpcm: fix sample size calculation for 20bit LCPM.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit f1320dc3be)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:12:00 +01:00
Ronald S. Bultje
c3bf08d04c smacker: error out if palette copy-with-offset overruns palette size.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit a93b572ae4)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-13 23:12:00 +01:00
Ronald S. Bultje
12247a13e0 Don't use ff_cropTbl[] for IDCT.
Results of IDCT can by far outreach the range of ff_cropTbl[], leading
to overreads and potentially crashes.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit c23acbaed4)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-08 22:07:55 +01:00
Ronald S. Bultje
7503861b42 swscale: make filterPos 32bit.
Fixes overflows for large image sizes.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 2254b559cb)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-08 22:07:55 +01:00
Ronald S. Bultje
9def2f200e error_resilience: initialize s->block_index[].
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 6193ff6854)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-08 22:07:55 +01:00
Ronald S. Bultje
7b676935ee svq3: protect against negative quantizers.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 11b940a1a8)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-08 22:07:55 +01:00
Reinhard Tartler
9550c63196 Prepare for 0.8.1 Release 2012-03-08 22:07:54 +01:00
Justin Ruggles
4a15240a27 mov: set channel layout for AC-3 streams based on the 'dac3' atom info
fixes Bug 225
(cherry picked from commit 3798205a77)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-08 22:07:54 +01:00
Janne Grunau
a47b96bdd3 rv34: handle size changes during frame multithreading
Factors all context dynamic memory handling to its own functions.
Fixes bug 220.
(cherry picked from commit 2bd730010d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-08 22:07:54 +01:00
Alex Converse
fb049da952 mov: Add more HDV and XDCAM FourCCs.
Reference: VLC
(cherry picked from commit b142496c56)
2012-03-06 15:31:49 -08:00
Alex Converse
4a325ddeae mov: Add support for MPEG2 HDV 720p24 (hdv4)
(cherry picked from commit 0ad522afb3)
2012-03-06 15:31:41 -08:00
Alex Converse
48ac765efe rv10/20: Fix slice overflow with checked bitstream reader.
(cherry picked from commit 9243ec4a50)
2012-03-06 15:31:23 -08:00
Michael Niedermayer
522645e38f h263dec: Disallow width/height changing with frame threads.
Fixes CVE-2011-3937

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 71db86d53b)

Conflicts:

	libavcodec/h263dec.c

Signed-off-by: Alex Converse <alex.converse@gmail.com>
2012-03-06 15:28:01 -08:00
Alex Converse
e891ee4bf6 adpcm: Clip step_index values read from the bitstream at the beginning of each frame.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit bbeb29133b)
2012-03-06 15:28:01 -08:00
Alex Converse
ef673211e7 tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned.
TIFF v6.0 (unimplemented) adds signed equivalents.
(cherry picked from commit e32548d133)
2012-03-06 15:28:01 -08:00
Alex Converse
eaeaeb265f dpcm: ignore extra unpaired bytes in stereo streams.
Fixes: CVE-2011-3951

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit ce7aee9b73)
2012-03-06 15:28:01 -08:00
Alex Converse
db315c796d svq3: Prevent illegal reads while parsing extradata.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit 9e1db721c4)
2012-03-06 15:28:01 -08:00
Alex Converse
035dd77cbb dv: Fix small overread in audio frequency table.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit 0ab3687924)
2012-03-06 15:28:01 -08:00
Michael Niedermayer
e3743869e9 ac3dec: Move center and surround mix level tables to the parser.
That way all mix levels as exported by avpriv_ac3_parse_header()
will have the same meaning.

Previously the 3-bit center mix level for E-AC-3 was used to index in a
4-entry table, leading to out-of-array reads.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit e6d9fa66f1)
2012-03-06 15:28:01 -08:00
Alex Converse
ce14f00dea movdec: Avoid av_malloc(0) in stss
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit 29a20ac4a1)
2012-03-06 15:28:01 -08:00
Mans Rullgard
627f4621f5 ac3: Do not read past the end of ff_ac3_band_start_tab.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 034b03e7a0)
2012-03-06 15:28:01 -08:00
Alex Converse
3e8434bcea dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
Found with asan.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 2d1c0dea5f)
2012-03-06 15:28:01 -08:00
Michael Niedermayer
efd30c4d95 dv: Fix null pointer dereference due to ach=0
dv: Fix null pointer dereference due to ach=0

Fixes part2 of CVE-2011-3929

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 5a396bb3a6)
2012-03-06 15:28:00 -08:00
Michael Niedermayer
d7fddc97d4 dv: check stype
dv: check stype

Fixes part1 of CVE-2011-3929
Possibly fixes part of CVE-2011-3936

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 635bcfccd4)
2012-03-06 15:28:00 -08:00
Dale Curtis
feed0c6b6a mpegaudiodec: Prevent premature clipping of mp3 input buffer.
Instead of clipping extrasize based on EXTRABYTES, clip based on the
amount of buffer actually left. Without this fix, there are warbles
and other distortions in the test case below.

http://kevincennis.com/mix/assets/sounds/1901_voxfx.mp3
(cherry picked from commit b716542691)

Signed-off-by: Alex Converse <alex.converse@gmail.com>
2012-03-06 15:28:00 -08:00
Alex Converse
d0e53ecff7 mp3dec: Fix a heap-buffer-overflow
In some cases, what is left to read from ptr is smaller than EXTRABYTES.

Based on a patch by Thierry Foucu <tfoucu@gmail.com>.

Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit f372ce119b)
2012-03-06 15:28:00 -08:00
Alex Converse
1ca84aa162 mpeg12: Pad framerate tab to 16 entries.
There are many places where we read an unchecked 4-bit index into it.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit dfa37fe8a3)
2012-03-06 15:28:00 -08:00
Michael Niedermayer
d5f2382d03 kgv1dec: Increase offsets array size so it is large enough.
Fixes CVE-2011-3945

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 807a045ab7)

Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit a02e8df973)
2012-03-06 15:28:00 -08:00
Alex Converse
416849f2e0 kmvc: Check palsize.
Fixes: CVE-2011-3952

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Based on fix by Michael Niedermayer
(cherry picked from commit 386741f887)
2012-03-06 15:28:00 -08:00
Alex Converse
dd37038ac7 nsvdec: Propagate errors
Related to CVE-2011-3940.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit c898431ca5)

Conflicts:

	libavformat/nsvdec.c
2012-03-06 15:28:00 -08:00
Alex Converse
e410dd1792 nsvdec: Be more careful with av_malloc().
Check results for av_malloc() and fix an overflow in one call.

Related to CVE-2011-3940.

Based in part on work from Michael Niedermayer.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit 8fd8a48263)
2012-03-06 15:28:00 -08:00
Michael Niedermayer
ffdc41f039 nsvdec: Fix use of uninitialized streams.
Fixes CVE-2011-3940 (Out of bounds read resulting in out of bounds write)

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5c011706bc)

Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 6a89b41d97)
2012-03-06 15:28:00 -08:00
Martin Storsjö
ca7e97bdcf g722: Fix the QMF scaling
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.

This makes the decoder output have double the magnitude
compared to before.

The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.

(cherry picked from commit b087ce2bee)

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-06 15:45:30 +02:00
Justin Ruggles
4ae138cb12 ac3dsp: do not use pshufb in ac3_extract_exponents_ssse3()
We need to do unsigned saturation in order to cover the corner case when the
absolute coefficient value is 16777215 (the maximum value).

Fixes Bug #216
(cherry picked from commit d483bb58c3)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-06 13:55:35 +01:00
Fabian Greffrath
003f7e3dd0 Fix format string vulnerability detected by -Wformat-security.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit c9dbac36ad)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-05 18:01:37 +01:00
Ronald S. Bultje
85eb76a23f h264: fix mmxext chroma deblock to use correct TC values.
(cherry picked from commit b0c4f04338)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-05 18:00:43 +01:00
Ronald S. Bultje
5186984ee9 h264: change underread for 10bit QPEL to overread.
This prevents us from reading before the start of the buffer, and thus
prevents crashes resulting from this behaviour. Fixes bug 237.
(cherry picked from commit 291c9b6285)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-05 18:00:31 +01:00
Ronald S. Bultje
b5331b979b cscd: use negative error values to indicate decode_init() failures.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 8a9faf33f2)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-05 14:48:35 +01:00
Vitor Sessak
11f3173e1b amrnbdec: check frame size before decoding.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 882abda5a2)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-05 14:48:35 +01:00
Ronald S. Bultje
cd17195d1c h264: prevent overreads in intra PCM decoding.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit d1604b3de9)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-05 14:48:35 +01:00
Justin Ruggles
1128b10247 wmaenc: fix m/s stereo encoding for the first frame
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.

CC:libav-stable@libav.org
(cherry picked from commit 51ddf35c90)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:29 +01:00
Justin Ruggles
6a073aa7a7 wmaenc: limit allowed sample rate to 48kHz
ff_wma_init() allows up to 50kHz, but this generates an exponent band
size table that requires 65 bands. The code assumes 25 bands in many
places, and using sample rates higher than 48kHz will lead to buffer
overwrites.

CC:libav-stable@libav.org
(cherry picked from commit 1ec075cfec)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:29 +01:00
Justin Ruggles
073891e875 wmaenc: limit block_align to MAX_CODED_SUPERFRAME_SIZE
This is near the theoretical limit for wma frame size and is the most that
our decoder can handle. Allowing higher bit rates will just end up padding
each frame with empty bytes.

Fixes invalid writes for avconv when using very high bit rates.

CC:libav-stable@libav.org
(cherry picked from commit c2b8dea182)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:29 +01:00
Justin Ruggles
2e341bc99a wmaenc: require a large enough output buffer to prevent overwrites
The maximum theoretical frame size is around 17000 bytes. Although in
practice it will generally be much smaller, we require a larger buffer
just to be safe.

CC: libav-stable@libav.org
(cherry picked from commit dfc4fdedf8)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:29 +01:00
Alex Converse
b7c8fff803 mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 4df369692e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:29 +01:00
Alex Converse
3f7e90cf0c mpegts: Pad the packet buffer in handle_packet().
This allows it to be used with get_bits without the thread of overreads.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 1aa708988a)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:28 +01:00
Ronald S. Bultje
78d4f8cc56 amrwb: remove duplicate arguments from extrapolate_isf().
Prevents warnings because the dst and src overlap (are the same) in the
memcpy() inside the function.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 9d87374ec0)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:28 +01:00
Ronald S. Bultje
de2656ec25 amrwb: error out early if mode is invalid.
Prevents using the invalid mode as an index in a static array, which
would generate invalid reads.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 154b8bb800)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:28 +01:00
Ronald S. Bultje
9686a2c2cf matroska: check buffer size for RM-style byte reordering.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 9c239f6026)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 21:26:28 +01:00
Ronald S. Bultje
b863979c0f wma: fix invalid buffer size assumptions causing random overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 349b7977e4)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Alex Converse
fecd7468fc wmadec: Verify bitstream size makes sense before calling init_get_bits.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit 48f1e5212c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Alex Converse
19da1a39e8 rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 2f6528537f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Ronald S. Bultje
7e88df99e1 lcl: return negative error codes on decode_init() errors.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit bd17a40a7e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Justin Ruggles
7f3f85544c avutil: add AVERROR_UNKNOWN
Useful to return instead of -1 when the cause of the error is unknown,
typically from an external library.
(cherry picked from commit c9bca80132)

Conflicts:

	doc/APIchanges
	libavutil/avutil.h

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Ronald S. Bultje
750f5baf30 h264: error out on invalid bitdepth.
Fixes invalid reads while initializing the dequant tables, which uses
the bit depth to determine the QP table size.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 0ce4fe482c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Ronald S. Bultje
a63f3f714c huffyuv: do not abort on unknown pix_fmt; instead, return an error.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 63c9de6469)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-03-04 12:26:06 +01:00
Ronald S. Bultje
1dd1ee00d5 vmnc: return error on decode_init() failure.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 07a180972f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 21:43:20 +01:00
Ronald S. Bultje
4493af756b rpza: error out on buffer overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 78e9852a2e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 21:43:20 +01:00
Ronald S. Bultje
e904e9b720 qtrle: return error on decode_init() failure.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit e54ae60e46)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 21:43:20 +01:00
Ronald S. Bultje
5f896773e0 swscale: fix another integer overflow.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 791de61bbb)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 21:43:20 +01:00
Ronald S. Bultje
b2dcac7141 vp56: error out on invalid stream dimensions.
Prevents crashes when playing corrupt vp5/6 streams.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 8bc396fc0e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 14:23:11 +01:00
Ronald S. Bultje
40ccc81146 asf: don't seek back on EOF.
Seeking back on EOF will reset the EOF flag, causing us to re-enter
the loop to find the next marker in the ASF file, thus potentially
causing an infinite loop.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit bb6d5411e1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 14:22:35 +01:00
Ronald S. Bultje
1c63d61372 asf: error out on ridiculously large minpktsize values.
They cause various issues further down in demuxing.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 6e57a02b9f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-29 14:21:57 +01:00
Anton Khirnov
2ad77c60ef lavf: add functions for accessing the fourcc<->CodecID mapping tables.
Fixes bug 212.
(cherry picked from commit dd6d3b0e02)

Conflicts:

	doc/APIchanges

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-29 10:44:37 +01:00
Paul B Mahol
a1556d37b8 avutil: make intfloat api public
The functions are already av_ prefixed and intfloat header is already provided.
Install libavutil/intfloat.h

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 8b933129b9)

Conflicts:

	doc/APIchanges

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-29 10:44:37 +01:00
Alex Converse
083a8a0037 mjpegbdec: Fix overflow in SOS.
Based in part by a fix from Michael Niedermayer <michaelni@gmx.at>

Fixes CVE-2011-3947

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit b57d262412)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-28 22:20:45 +01:00
Ronald S. Bultje
71a939fee4 oma: don't read beyond end of leaf_table.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 934cd18a43)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-28 22:10:55 +01:00
Ronald S. Bultje
9dbd437da2 Indeo3: fix crashes on corrupt bitstreams.
Splits at borders of cells are invalid, since it leaves one of the
cells with a width/height of zero. Also, propagate errors on buffer
allocation failures, so we don't continue decoding (which crashes).

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit fc9bc08dca)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-28 20:57:44 +01:00
Ronald S. Bultje
2510e1476e vorbis: fix overflows in floor1[] vector and inverse db table index.
(cherry picked from commit 24947d4988)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 18:11:15 +01:00
Reinhard Tartler
0f839cff6b Fix parser not to clobber has_b_frames when extradata is set.
Because in contrast to the decoder, the parser does not setup low_delay.
The code in parse_nal_units would always end up setting has_b_frames
to "1", except when stream is explicitly marked as low delay.
Since the parser itself would create 'extradata', simply reopening
the parser would cause this.

This happens for instance in estimate_timings_from_pts(), which causes the
parser to be reopened on the same stream.

This fixes Libav #22 and FFmpeg (trac) #360

CC: libav-stable@libav.org

Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(commit 31ac0ac29b)

Comments and description adapted by Reinhard Tartler.

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 790a367d9e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 15:56:55 +01:00
Ronald S. Bultje
abe3572878 rm: prevent infinite loops for index parsing.
Specifically, prevent jumping back in the file for the next index, since
this can lead to infinite loops where we jump between indexes referring
to each other, and don't read indexes that don't fit in the file.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit aac07a7a4c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:04:04 +01:00
Ronald S. Bultje
0d30e2c6f2 fraps: release reference buffer on pix_fmt change.
Prevents crash when trying to copy from a non-existing plane in e.g.
a RGB32 reference image to a YUV420P target image

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 830f70442a)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
a0473085f3 kgv1: release reference picture on size change.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 6c4c27adb6)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
e537dc230b kgv1: use avctx->get/release_buffer().
Also fixes crashes on corrupt bitstreams.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 33cd32b389)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
19f4943d12 lcl: error out if uncompressed input buffer is smaller than framesize.
This prevents crashes when trying to read beyond the end of the buffer
while decoding frame data.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit be129271ea)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
bf6d1a1ca7 mjpeg: abort decoding if packet is too large.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit ab492ca2ab)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Alex Converse
424b6edd19 tiff: Prevent overreads in the type_sizes array.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 447363870f)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
4f48417fe7 swf: check return values for av_get/new_packet().
Prevents crashers when using the packet if allocation failed.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 31632e73f4)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
8e3dc37bc0 truemotion2: error out if the huffman tree has no nodes.
This prevents crashers and errors further down when reading nodes in the
empty tree.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 2b83e8b700)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
0312969b9e rmdec: when using INT4 deinterleaving, error out if sub_packet_h <= 1.
We read sub_packet_h / 2 packets per line of data (during deinterleaving),
which equals zero if sub_packet_h <= 1, thus causing us to not read any
data, leading to an infinite loop.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit e30b3e59a4)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Janne Grunau
62beae313a avplay: fix -threads option
The AVOptions based default to threads auto in 2473a45c8
works only if avplay does not use custom option handling
for -threads.

CC: <libav-stable@libav.org>
(cherry picked from commit e48a70e6da)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
8011a29fa8 vc1parse: call vc1_init_common().
The parser uses VLC tables initialized in vc1_common_init(), therefore
we should call this function on parser init also.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit c742ab4e81)

Conflicts:

	libavcodec/vc1.h

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
fe710f2074 wma: don't return 0 on invalid packets.
Return 0 means "please return the same data again", i.e. it causes an
infinite loop. Instead, return an error.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 9d3050d3e9)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
bba43a1ea0 mjpegb: don't return 0 at the end of frame decoding.
Return 0 indicates "please return the same data again", i.e. it causes
an infinite loop. Instead, return that we consumed the buffer if we
finished decoding succesfully, or return an error if an error occurred.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 74699ac8c8)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
f947e965be asf: prevent packet_size_left from going negative if hdrlen > pktlen.
This prevents failed assertions further down in the packet processing
where we require non-negative values for packet_size_left.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 41afac7f7a)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:16 +01:00
Ronald S. Bultje
5c365dc979 aiff: don't skip block_align==0 check on COMM-after-SSND files.
This prevents SIGFPEs when using block_align for divisions.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 32a659c758)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
95a9d44dc3 mp3on4: require a minimum framesize.
If bufsize < headersize, init_get_bits() will be called with a negative
number, causing it to fail and any subsequent call to get_bits() will
crash because it reads from a NULL pointer.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 3e13005cac)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
27558bd87e huffyuv: error out on bit overrun.
On EOF, get_bits() will continuously return 0, causing an infinite
loop.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 84c202cc37)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
5ab9294a8d als: prevent infinite loop in zero_remaining().
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit af468015d9)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
cfd7d166e2 cook: prevent div-by-zero if channels is zero.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 941fc1ea1e)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
5bcd47cf63 vc1: prevent using last_frame as a reference for I/P first frame.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit ae591aeea5)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
0c60d5c59f swscale: take first/lastline over/underflows into account for MMX.
Fixes crashes for extremely large resizes (several 100-fold).

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 1d8c4af396)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
cd9bdc6395 swscale: fix overflows in filterPos[] calculation for large sizes.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 19a65b5be4)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
b68470707b swscale: enforce a minimum filtersize.
At very small dimensions, this calculation could lead to zero-sized
filters, which leads to uninitialized output, zero-sized allocations,
loop overflows in SIMD that uses do{..}while(i++<filtersize); instead
of for(i=0;i<filtersize;i++){..} and several other similar failures.
Therefore, require a minimum filtersize of 1.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit dae2ce361a)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
7046ae5593 tta: error out if samplerate is zero.
Prevents a division by zero later on.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 7416d61036)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Janne Grunau
d19e3e19d6 vc1: prevent null pointer dereference on broken files
CC: libav-stable@libav.org
(cherry picked from commit 510ef04a46)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Alex Converse
04597e2595 smacker: Sanity check huffman tables found in the headers.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

CC: libav-stable@libav.org
(cherry picked from commit 9adf25c1cf)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Janne Grunau
d16653c3d4 lavf: prevent infinite loops while flushing in avformat_find_stream_info
If no data was seen for a stream decoder are returning 0 when fed with
empty packets for flushing. We can stop flushing when the decoder does
not return delayed delayed frames anymore. Changes try_decode_frame()
return value to got_picture or negative error.

CC: libav-stable@libav.org
(cherry picked from commit b3461c29c1)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Ronald S. Bultje
183e0eb5b9 matroska: don't overwrite string values until read/alloc was succesful.
This prevents certain tags with a default value assigned to them (as per
the EBML syntax elements) from ever being assigned a NULL value. Other
parts of the code rely on these being non-NULL (i.e. they don't check for
NULL before e.g. using the string in strcmp() or similar), and thus in
effect this prevents crashes when reading of such specific tags fails,
either because of low memory or because of targeted file corruption.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit cd40c31ee9)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Alex Converse
be0b3137d0 matroskadec: Pad AAC extradata.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

CC: libav-stable@libav.org
(cherry picked from commit d2ee8c1779)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Alex Converse
683213230e aac: fix infinite loop on end-of-frame with sequence of 1-bits.
Based-on-work-by: Ronald S. Bultje <rsbultje@gmail.com>
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 1cd9a6154b)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Alex Converse
ad0ee682b3 wma: Clip WMA1 and WMA2 frame length to 11 bits.
The MDCT buffers in the decoder are only sized for up to 11 bits. The
reverse engineered documentation for WMA1/2 headers say that that for
all samplerates above 32kHz 11 bits are used. 12 and 13 bit support
were added for WMAPro. I was unable to make any Microsoft tools generate
a test file at a samplerate above 48kHz.

Discovered by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

CC: libav-stable@libav.org
(cherry picked from commit d78bb1a4b2)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:15 +01:00
Janne Grunau
ba418ad400 rv20: prevent calling ff_h263_decode_mba() with unset height/width
Prevents a crash of VLC during playback of a invalid matroska file,
found by John Villamil <johnv@matasano.com>.

CC: libav-stable@libav.org
(cherry picked from commit c3e10ae412)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-02-26 10:03:14 +01:00
Ronald S. Bultje
6dcbbdc011 flac: fix infinite loops on all-zero input or end-of-stream.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 52e4018be4)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:14 +01:00
Ronald S. Bultje
e43bd4fa58 golomb: use HAVE_BITS_REMAINING() macro to prevent infloop on EOF.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 46b3fbc30b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:14 +01:00
Ronald S. Bultje
25b4ed053f get_bits: add HAVE_BITS_REMAINING macro.
(cherry picked from commit b44b41633f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:14 +01:00
Ronald S. Bultje
e1f2a6a32b golomb: avoid infinite loop on all-zero input (or end of buffer).
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit c6643fddba)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:14 +01:00
Michael Niedermayer
6fc3287b9c shorten: Use separate pointers for the allocated memory for decoded samples.
Fixes invalid free() if any of the buffers are not allocated due to either
not decoding a header or an error prior to allocating all buffers.

Fixes CVE-2012-0858
CC: libav-stable@libav.org

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit 204cb29b3c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:14 +01:00
Michael Niedermayer
f43b6e2b1e atrac3: Fix crash in tonal component decoding.
Add a check to avoid writing past the end of the channel_unit.components[]
array.

Bug Found by: cosminamironesei
Fixes CVE-2012-0853
CC: libav-stable@libav.org

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit c509f4f747)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:14 +01:00
Michael Niedermayer
697a45d861 ws_snd1: Fix wrong samples count and crash.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9fb7a5af97)

Addresses CVE-2012-0848

Reviewed-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 10:03:05 +01:00
Ronald S. Bultje
4c7879775e h264: disallow constrained intra prediction modes for luma.
Conversion of the luma intra prediction mode to one of the constrained
("alzheimer") ones can happen by crafting special bitstreams, causing
a crash because we'll call a NULL function pointer for 16x16 block intra
prediction, since constrained intra prediction functions are only
implemented for chroma (8x8 blocks).

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 45b7bd7c53)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 09:09:27 +01:00
Ronald S. Bultje
a2c8db1b79 swscale: fix V plane memory location in bilinear/unscaled RGB/YUYV case.
Fixes bug 221.

CC: libav-stable@libav.org
(cherry picked from commit b7542dd3d7)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 09:09:26 +01:00
Martin Storsjö
fc89f15497 libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set
Earlier, calling avcodec_encode_audio worked fine even if time_base
wasn't set. Now it crashes due to trying to scale the output pts to
the codec context time base. This affects e.g. VLC.

If no time_base is set for audio codecs, set it to the sample
rate.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 9a7dc618c5)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 09:09:26 +01:00
Alex Converse
e364f50718 qdm2: Check data block size for bytes to bits overflow.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

CC: libav-stable@libav.org
(cherry picked from commit dac56d9ce0)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 09:09:26 +01:00
Anton Khirnov
571a4cf273 lavc: set AVCodecContext.codec in avcodec_get_context_defaults3().
This way, if the AVCodecContext is allocated for a specific codec, the
caller doesn't need to store this codec separately and then pass it
again to avcodec_open2().

It also allows to set codec private options using av_opt_set_* before
opening the codec.
(cherry picked from commit bc90199848)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-02-26 09:09:26 +01:00
Anton Khirnov
bafd38a352 lavc: make avcodec_close() work properly on unopened codecs.
I.e. free the priv_data and other stuff allocated in
avcodec_alloc_context3() and not segfault.

(cherry picked from commit 0e72ad95f9)
2012-02-26 09:09:26 +01:00
Anton Khirnov
350d06d63f lavc: add avcodec_is_open().
It allows to check whether an AVCodecContext is open in a documented
way. Right now the undocumented way this check is done in lavf/lavc is
by checking whether AVCodecContext.codec is NULL. However it's desirable
to be able to set AVCodecContext.codec before avcodec_open2().

(cherry picked from commit af08d9aeea)

Conflicts:

	doc/APIchanges
2012-02-26 09:03:33 +01:00
Derek Buitenhuis
9f82cbf7c1 wavpack: Don't shift minclip/maxclip
Since we are clipping before we shift the values to
16 or 32 bits, we should not shift the min/max clip
values to compensate.

Fixes 8 and 24 bit lossy decoding.

Fixes ticket #871.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 480b133e6f)
2012-02-25 20:50:27 +01:00
Michael Niedermayer
dcde8e1c90 Revert "Improve decoding quality for lossy wavpack."
This has been implemented more correctly.

This reverts commit a915618a29.
(cherry picked from commit 32e74395a8)
2012-02-25 20:50:19 +01:00
Carl Eugen Hoyos
569cb94869 Fix ffmpeg -codecs output.
(cherry picked from commit f6492476a6)
2012-02-18 00:00:06 +01:00
Justin Ruggles
0df7d7482c wavpack: add needed braces for 2 statements inside an if block
(cherry picked from commit 9d7cee50aa)
2012-02-12 01:48:07 +01:00
Carl Eugen Hoyos
b2f27d2926 Improve decoding quality for lossy wavpack.
This reverts e6e7bfc1 and 365e1ec2.
The code may be incorrect both before and after the revert, but we
do not have any samples that were fixed by the original commits.

Fixes ticket #871.
(cherry picked from commit a915618a29)
2012-01-29 18:02:12 +01:00
Michael Niedermayer
7e16636995 doc: remove doc/ffmpeg-mt-authorship.txt for release/0.10
we dont carry the whole git history in releases so theres no
point in having this in them either.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-26 22:44:59 +01:00
Michael Niedermayer
83d78fece0 Update for 0.10
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-26 22:43:32 +01:00
4164 changed files with 219552 additions and 474219 deletions

123
.gitignore vendored
View File

@@ -1,79 +1,56 @@
*.a
.config
.version
*.o
*.d
*.def
*.dll
*.dylib
*.exe
*.exp
*.gcda
*.gcno
*.h.c
*.ilk
*.lib
*.pc
*.pdb
*.so
*.so.*
*.ver
*.ho
*-example
*-test
*_g
/.config
/.version
/ffmpeg
/ffplay
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/version.h
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/doc/*.html
/doc/*.pod
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/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/pc-uninstalled
/doc/examples/resampling_audio
/doc/examples/scaling_video
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/doc/doxy/html/
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/libavcodec/*_tablegen
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/tests/audiogen
/tests/base64
/tests/data/
/tests/rotozoom
/tests/tiny_psnr
/tests/tiny_ssim
/tests/videogen
/tests/vsynth1/
/tools/aviocat
/tools/ffbisect
/tools/bisect.need
/tools/crypto_bench
/tools/cws2fws
/tools/fourcc2pixfmt
/tools/ffescape
/tools/ffeval
/tools/ffhash
/tools/graph2dot
/tools/ismindex
/tools/pktdumper
/tools/probetest
/tools/qt-faststart
/tools/trasher
/tools/seek_print
/tools/zmqsend
*.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

View File

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

59
CREDITS
View File

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

354
Changelog
View File

@@ -1,326 +1,44 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>
version next:
version 0.10.1
- Several security fixes, many bugfixes affecting many formats and
codecs, the list below is not complete.
version 2.1:
- aecho filter
- perspective filter ported from libmpcodecs
- ffprobe -show_programs option
- compand filter
- RTMP seek support
- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
even when used as an input option. Previous behavior can be restored with
the -noaccurate_seek option.
- ffmpeg -t option can now be used for inputs, to limit the duration of
data read from an input file
- incomplete Voxware MetaSound decoder
- read EXIF metadata from JPEG
- DVB teletext decoder
- phase filter ported from libmpcodecs
- w3fdif filter
- Opus support in Matroska
- FFV1 version 1.3 is stable and no longer experimental
- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
more consistent with other muxers.
- adelay filter
- pullup filter ported from libmpcodecs
- ffprobe -read_intervals option
- Lossless and alpha support for WebP decoder
- Error Resilient AAC syntax (ER AAC LC) decoding
- Low Delay AAC (ER AAC LD) decoding
- mux chapters in ASF files
- SFTP protocol (via libssh)
- libx264: add ability to encode in YUVJ422P and YUVJ444P
- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
- make decoding alpha optional for prores, ffv1 and vp6 by setting
the skip_alpha flag.
- ladspa wrapper filter
- native VP9 decoder
- dpx parser
- max_error_rate parameter in ffmpeg
- PulseAudio output device
- ReplayGain scanner
- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
- Linux framebuffer output device
- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
- mergeplanes filter
version 2.0:
- curves filter
- reference-counting for AVFrame and AVPacket data
- ffmpeg now fails when input options are used for output file
or vice versa
- support for Monkey's Audio versions from 3.93
- perms and aperms filters
- audio filtering support in ffplay
- 10% faster aac encoding on x86 and MIPS
- sine audio filter source
- WebP demuxing and decoding support
- new ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
- OpenCL support
- audio phaser filter
- separatefields filter
- libquvi demuxer
- uniform options syntax across all filters
- telecine filter
- new interlace filter
- smptehdbars source
- inverse telecine filters (fieldmatch and decimate)
- colorbalance filter
- colorchannelmixer filter
- The matroska demuxer can now output proper verbatim ASS packets. It will
become the default at the next libavformat major bump.
- decent native animated GIF encoding
- asetrate filter
- interleave filter
- timeline editing with filters
- vidstabdetect and vidstabtransform filters for video stabilization using
the vid.stab library
- astats filter
- trim and atrim filters
- ffmpeg -t and -ss (output-only) options are now sample-accurate when
transcoding audio
- Matroska muxer can now put the index at the beginning of the file.
- extractplanes filter
- avectorscope filter
- ADPCM DTK decoder
- ADP demuxer
- RSD demuxer
- RedSpark demuxer
- ADPCM IMA Radical decoder
- zmq filters
- DCT denoiser filter (dctdnoiz)
- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
- Apple Intermediate Codec decoder
- Escape 130 video decoder
- FTP protocol support
- V4L2 output device
- 3D LUT filter (lut3d)
- SMPTE 302M audio encoder
- support for slice multithreading in libavfilter
- Hald CLUT support (generation and filtering)
- VC-1 interlaced B-frame support
- support for WavPack muxing (raw and in Matroska)
- XVideo output device
- vignette filter
- True Audio (TTA) encoder
- Go2Webinar decoder
- mcdeint filter ported from libmpcodecs
- sab filter ported from libmpcodecs
- ffprobe -show_chapters option
- WavPack encoding through libwavpack
- rotate filter
- spp filter ported from libmpcodecs
- libgme support
- psnr filter
version 1.2:
- VDPAU hardware acceleration through normal hwaccel
- SRTP support
- Error diffusion dither in Swscale
- Chained Ogg support
- Theora Midstream reconfiguration support
- EVRC decoder
- audio fade filter
- filtering audio with unknown channel layout
- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
and treble audio filter
- improved showspectrum filter, with multichannel support and sox-like colors
- histogram filter
- tee muxer
- il filter ported from libmpcodecs
- support ID3v2 tags in ASF files
- encrypted TTA stream decoding support
- RF64 support in WAV muxer
- noise filter ported from libmpcodecs
- Subtitles character encoding conversion
- blend filter
- stereo3d filter ported from libmpcodecs
version 1.1:
- stream disposition information printing in ffprobe
- filter for loudness analysis following EBU R128
- Opus encoder using libopus
- ffprobe -select_streams option
- Pinnacle TARGA CineWave YUV16 decoder
- TAK demuxer, decoder and parser
- DTS-HD demuxer
- remove -same_quant, it hasn't worked for years
- FFM2 support
- X-Face image encoder and decoder
- 24-bit FLAC encoding
- multi-channel ALAC encoding up to 7.1
- metadata (INFO tag) support in WAV muxer
- subtitles raw text decoder
- support for building DLLs using MSVC
- LVF demuxer
- ffescape tool
- metadata (info chunk) support in CAF muxer
- field filter ported from libmpcodecs
- AVR demuxer
- geq filter ported from libmpcodecs
- remove ffserver daemon mode
- AST muxer/demuxer
- new expansion syntax for drawtext
- BRender PIX image decoder
- ffprobe -show_entries option
- ffprobe -sections option
- ADPCM IMA Dialogic decoder
- BRSTM demuxer
- animated GIF decoder and demuxer
- PVF demuxer
- subtitles filter
- IRCAM muxer/demuxer
- Paris Audio File demuxer
- Virtual concatenation demuxer
- VobSub demuxer
- JSON captions for TED talks decoding support
- SOX Resampler support in libswresample
- aselect filter
- SGI RLE 8-bit decoder
- Silicon Graphics Motion Video Compressor 1 & 2 decoder
- Silicon Graphics Movie demuxer
- apad filter
- Resolution & pixel format change support with multithreading for H.264
- documentation split into per-component manuals
- pp (postproc) filter ported from MPlayer
- NIST Sphere demuxer
- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
- Sony Wave64 muxer
- adobe and limelight publisher authentication in RTMP
- data: URI scheme
- support building on the Plan 9 operating system
- kerndeint filter ported from MPlayer
- histeq filter ported from VirtualDub
- Megalux Frame demuxer
- 012v decoder
- Improved AVC Intra decoding support
version 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
- 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
- support for building with MSVC
version 0.11:
- 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
- ffv1: support (draft) version 1.3
- 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,
@@ -656,7 +374,7 @@ version 0.6:
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- ATRAC1 decoder
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -756,7 +474,7 @@ version 0.5:
- MXF demuxer
- VC-1/WMV3/WMV9 video decoder
- MacIntel support
- AviSynth support
- AVISynth support
- VMware video decoder
- VP5 video decoder
- VP6 video decoder
@@ -784,7 +502,7 @@ version 0.5:
- Interplay C93 demuxer and video decoder
- Bethsoft VID demuxer and video decoder
- CRYO APC demuxer
- ATRAC3 decoder
- Atrac3 decoder
- V.Flash PTX decoder
- RoQ muxer, RoQ audio encoder
- Renderware TXD demuxer and decoder
@@ -1061,7 +779,7 @@ 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)

View File

@@ -31,9 +31,9 @@ PROJECT_NAME = FFmpeg
# This could be handy for archiving the generated documentation or
# if some version control system is used.
PROJECT_NUMBER = 2.1.8
PROJECT_NUMBER = 0.10.1
# With the PROJECT_LOGO tag one can specify a logo or icon that is included
# With the PROJECT_LOGO tag one can specify an logo or icon that is included
# in the documentation. The maximum height of the logo should not exceed 55
# pixels and the maximum width should not exceed 200 pixels. Doxygen will
# copy the logo to the output directory.
@@ -44,7 +44,7 @@ PROJECT_LOGO =
# If a relative path is entered, it will be relative to the location
# where doxygen was started. If left blank the current directory will be used.
OUTPUT_DIRECTORY = doc/doxy
OUTPUT_DIRECTORY = doxy
# If the CREATE_SUBDIRS tag is set to YES, then doxygen will create
# 4096 sub-directories (in 2 levels) under the output directory of each output
@@ -277,7 +277,7 @@ SUBGROUPING = YES
# be useful for C code in case the coding convention dictates that all compound
# types are typedef'ed and only the typedef is referenced, never the tag name.
TYPEDEF_HIDES_STRUCT = YES
TYPEDEF_HIDES_STRUCT = NO
# The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
# determine which symbols to keep in memory and which to flush to disk.
@@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT = YES
# causing a significant performance penality.
# If the system has enough physical memory increasing the cache will improve the
# performance by keeping more symbols in memory. Note that the value works on
# a logarithmic scale so increasing the size by one will roughly double the
# a logarithmic scale so increasing the size by one will rougly double the
# memory usage. The cache size is given by this formula:
# 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
# corresponding to a cache size of 2^16 = 65536 symbols
@@ -409,7 +409,7 @@ INLINE_INFO = YES
# alphabetically by member name. If set to NO the members will appear in
# declaration order.
SORT_MEMBER_DOCS = NO
SORT_MEMBER_DOCS = YES
# If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
# brief documentation of file, namespace and class members alphabetically
@@ -489,6 +489,12 @@ MAX_INITIALIZER_LINES = 30
SHOW_USED_FILES = YES
# If the sources in your project are distributed over multiple directories
# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy
# in the documentation. The default is NO.
SHOW_DIRECTORIES = NO
# Set the SHOW_FILES tag to NO to disable the generation of the Files page.
# This will remove the Files entry from the Quick Index and from the
# Folder Tree View (if specified). The default is YES.
@@ -639,14 +645,15 @@ EXCLUDE_SYMBOLS =
# directories that contain example code fragments that are included (see
# the \include command).
EXAMPLE_PATH = doc/examples/
EXAMPLE_PATH = libavcodec/ \
libavformat/
# If the value of the EXAMPLE_PATH tag contains directories, you can use the
# EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
# and *.h) to filter out the source-files in the directories. If left
# blank all files are included.
EXAMPLE_PATTERNS = *.c
EXAMPLE_PATTERNS = *-example.c
# If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
# searched for input files to be used with the \include or \dontinclude
@@ -709,7 +716,7 @@ INLINE_SOURCES = NO
# doxygen to hide any special comment blocks from generated source code
# fragments. Normal C and C++ comments will always remain visible.
STRIP_CODE_COMMENTS = NO
STRIP_CODE_COMMENTS = YES
# If the REFERENCED_BY_RELATION tag is set to YES
# then for each documented function all documented
@@ -793,13 +800,13 @@ HTML_FILE_EXTENSION = .html
# each generated HTML page. If it is left blank doxygen will generate a
# standard header.
#HTML_HEADER = doc/doxy/header.html
HTML_HEADER = doc/doxy/header.html
# The HTML_FOOTER tag can be used to specify a personal HTML footer for
# each generated HTML page. If it is left blank doxygen will generate a
# standard footer.
#HTML_FOOTER = doc/doxy/footer.html
HTML_FOOTER = doc/doxy/footer.html
# The HTML_STYLESHEET tag can be used to specify a user-defined cascading
# style sheet that is used by each HTML page. It can be used to
@@ -808,7 +815,7 @@ HTML_FILE_EXTENSION = .html
# the style sheet file to the HTML output directory, so don't put your own
# stylesheet in the HTML output directory as well, or it will be erased!
#HTML_STYLESHEET = doc/doxy/doxy_stylesheet.css
HTML_STYLESHEET = doc/doxy/doxy_stylesheet.css
# The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
# Doxygen will adjust the colors in the stylesheet and background images
@@ -818,7 +825,7 @@ HTML_FILE_EXTENSION = .html
# 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
# The allowed range is 0 to 359.
#HTML_COLORSTYLE_HUE = 120
HTML_COLORSTYLE_HUE = 120
# The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of
# the colors in the HTML output. For a value of 0 the output will use
@@ -841,6 +848,12 @@ HTML_COLORSTYLE_GAMMA = 80
HTML_TIMESTAMP = YES
# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
# files or namespaces will be aligned in HTML using tables. If set to
# NO a bullet list will be used.
HTML_ALIGN_MEMBERS = YES
# If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML
# documentation will contain sections that can be hidden and shown after the
# page has loaded. For this to work a browser that supports
@@ -1021,6 +1034,11 @@ ENUM_VALUES_PER_LINE = 4
GENERATE_TREEVIEW = NO
# By enabling USE_INLINE_TREES, doxygen will generate the Groups, Directories,
# and Class Hierarchy pages using a tree view instead of an ordered list.
USE_INLINE_TREES = NO
# If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
# used to set the initial width (in pixels) of the frame in which the tree
# is shown.
@@ -1356,9 +1374,14 @@ INCLUDE_FILE_PATTERNS =
# instead of the = operator.
PREDEFINED = "__attribute__(x)=" \
"RENAME(x)=x ## _TMPL" \
"DEF(x)=x ## _TMPL" \
HAVE_AV_CONFIG_H \
HAVE_MMX \
HAVE_MMX2 \
HAVE_AMD3DNOW \
"DECLARE_ALIGNED(a,t,n)=t n" \
"offsetof(x,y)=0x42" \
av_alloc_size \
"offsetof(x,y)=0x42"
# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
# this tag can be used to specify a list of macro names that should be expanded.

93
LICENSE
View File

@@ -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
@@ -13,91 +14,33 @@ 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:
- f_ebur128.c
- vf_blackframe.c
- vf_boxblur.c
- vf_colormatrix.c
- vf_cropdetect.c
- vf_decimate.c
- vf_delogo.c
- vf_geq.c
- vf_histeq.c
- vf_hqdn3d.c
- vf_kerndeint.c
- vf_mcdeint.c
- vf_mp.c
- vf_owdenoise.c
- vf_perspective.c
- vf_phase.c
- vf_pp.c
- vf_pullup.c
- vf_sab.c
- vf_smartblur.c
- vf_spp.c
- vf_stereo3d.c
- vf_super2xsai.c
- vf_tinterlace.c
- vf_yadif.c
- vsrc_mptestsrc.c
There are a handful of files under other licensing terms, namely:
* 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
for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
There are a handful of files under other licensing terms, namely:
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
licensing details. Specifically note that you must credit the IJG in the
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
external libraries:
-------------------
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.
external 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.
FFmpeg can be combined with a number of external libraries, which sometimes
affect the licensing of binaries resulting from the combination.
compatible libraries
--------------------
The following libraries are under GPL:
- frei0r
- libcdio
- libutvideo
- libvidstab
- libx264
- libxavs
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing --enable-gpl to configure.
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing --enable-version3 to configure.
incompatible libraries
----------------------
The Fraunhofer AAC library, FAAC and aacplus are under licenses which
are incompatible with the GPLv2 and v3. We do not know for certain if their
licenses are compatible with the LGPL.
If you wish to enable these libraries, pass --enable-nonfree to configure.
But note that if you enable any of these libraries the resulting binary will
be under a complex license mix that is more restrictive than the LGPL and that
may result in additional obligations. It is possible that these
restrictions cause the resulting binary to be unredistributeable.
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.

View File

@@ -4,11 +4,11 @@ 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 <address>) after the name means that the maintainer prefers to be CC-ed on
patches and related discussions.
A CC after the name means that the maintainer prefers to be CC-ed on patches
and related discussions.
Project Leader
@@ -43,24 +43,16 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu
documentation Mike Melanson
website Robert Swain, Lou Logan
build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
Communication
=============
website Robert Swain, Lou Logan
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan
Launchpad Timothy Gu
libavutil
=========
@@ -70,23 +62,11 @@ Internal Interfaces:
libavutil/common.h Michael Niedermayer
Other:
bprint Nicolas George
bswap.h
des Reimar Doeffinger
eval.c, eval.h Michael Niedermayer
float_dsp Loren Merritt
hash Reimar Doeffinger
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
mathematics.c, mathematics.h Michael Niedermayer
mem.c, mem.h Michael Niedermayer
opencl.c, opencl.h Wei Gao
opt.c, opt.h Michael Niedermayer
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
timecode Clément Bœsch
mathematics.c, mathematics.h Michael Niedermayer
integer.c, integer.h Michael Niedermayer
bswap.h
libavcodec
@@ -97,6 +77,10 @@ Generic Parts:
avcodec.h Michael Niedermayer
utility code:
utils.c Michael Niedermayer
mem.c Michael Niedermayer
opt.c, opt.h Michael Niedermayer
arithmetic expression evaluator:
eval.c Michael Niedermayer
audio and video frame extraction:
parser.c Michael Niedermayer
bitstream reading:
@@ -127,8 +111,6 @@ Generic Parts:
libpostproc/* Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
fixed point FFT:
fft* Zeljko Lukac
Codecs:
4xm.c Michael Niedermayer
@@ -147,22 +129,18 @@ Codecs:
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
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
dnxhd* Baptiste Coudurier
dpcm.c Mike Melanson
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
dv.c Roman Shaposhnik
eacmv*, eaidct*, eat* Peter Ross
exif.c, exif.h Thilo Borgmann
ffv1.c Michael Niedermayer
ffwavesynth.c Nicolas George
flac* Justin Ruggles
@@ -171,9 +149,9 @@ Codecs:
g722.c Martin Storsjo
g726.c Roman Shaposhnik
gifdec.c Baptiste Coudurier
h264* Loren Merritt, Michael Niedermayer
h261* Michael Niedermayer
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
huffyuv.c Michael Niedermayer
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
@@ -181,41 +159,36 @@ Codecs:
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
ivi* Kostya Shishkov
jacosub* Clément Bœsch
jpeg2000* Nicolas Bertrand
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libdirac* David Conrad
libgsm.c Michel Bardiaux
libdirac* David Conrad
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libutvideo* Derek Buitenhuis
libvorbis.c David Conrad
libx264.c Mans Rullgard, Jason Garrett-Glaser
libxavs.c Stefan Gehrer
libx264.c Mans Rullgard, Jason Garrett-Glaser
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
mimic.c Ramiro Polla
mjpeg*.c Michael Niedermayer
mjpeg.c Michael Niedermayer
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpc* Kostya Shishkov
mpeg12.c, mpeg12data.h Michael Niedermayer
mpegvideo.c, mpegvideo.h Michael Niedermayer
mqc* Nicolas Bertrand
msmpeg4.c, msmpeg4data.h Michael Niedermayer
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nellymoserdec.c Benjamin Larsson
nuv.c Reimar Doeffinger
paf.* Paul B Mahol
pcx.c Ivo van Poorten
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
@@ -235,13 +208,11 @@ Codecs:
s3tc* Ivo van Poorten
smacker.c Kostya Shishkov
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow.c Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
targa.c Kostya Shishkov
tiff.c Kostya Shishkov
truemotion1* Mike Melanson
@@ -249,7 +220,6 @@ Codecs:
truespeech.c Kostya Shishkov
tscc.c Kostya Shishkov
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
@@ -257,11 +227,9 @@ Codecs:
vble.c Derek Buitenhuis
vc1* Kostya Shishkov
vcr1.c Michael Niedermayer
vda_h264_dec.c Xidorn Quan
vima.c Paul B Mahol
vmnc.c Kostya Shishkov
vorbis_dec.c Denes Balatoni, David Conrad
vorbis_enc.c Oded Shimon
vorbis_dec.c Denes Balatoni, David Conrad
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
@@ -273,12 +241,8 @@ Codecs:
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
xbm* Paul B Mahol
xface Stefano Sabatini
xl.c Kostya Shishkov
xvmc.c Ivan Kalvachev
xwd* Paul B Mahol
zerocodec.c Derek Buitenhuis
zmbv* Kostya Shishkov
Hardware acceleration:
@@ -296,54 +260,22 @@ libavdevice
libavdevice/avdevice.h
dshow.c Roger Pack
fbdev_enc.c Lukasz Marek
iec61883.c Georg Lippitsch
lavfi Stefano Sabatini
libdc1394.c Roman Shaposhnik
pulse_audio_enc.c Lukasz Marek
sdl Stefano Sabatini
v4l2.c Luca Abeni
vfwcap.c Ramiro Polla
libavfilter
===========
Generic parts:
Video filters:
graphdump.c Nicolas George
Filters:
af_adelay.c Paul B Mahol
af_aecho.c Paul B Mahol
af_afade.c Paul B Mahol
af_amerge.c Nicolas George
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_astreamsync.c Nicolas George
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_compand.c Paul B Mahol
af_ladspa.c Paul B Mahol
af_pan.c Nicolas George
avf_avectorscope.c Paul B Mahol
vf_blend.c Paul B Mahol
vf_colorbalance.c Paul B Mahol
vf_delogo.c Jean Delvare (CC <khali@linux-fr.org>)
vf_drawbox.c/drawgrid Andrey Utkin
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_il.c Paul B Mahol
vf_mergeplanes.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_scale.c Michael Niedermayer
vf_separatefields.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vsrc_mandelbrot.c Michael Niedermayer
vf_yadif.c Michael Niedermayer
Sources:
vsrc_mandelbrot.c Michael Niedermayer
libavformat
===========
@@ -358,27 +290,17 @@ Generic parts:
Muxers/Demuxers:
4xm.c Mike Melanson
adtsenc.c Robert Swain
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
aiff.c Baptiste Coudurier
ape.c Kostya Shishkov
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
avi* Michael Niedermayer
avisynth.c AvxSynth Team (avxsynth.testing at gmail dot com)
avr.c Paul B Mahol
bink.c Peter Ross
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flac* Justin Ruggles
flic.c Mike Melanson
@@ -388,26 +310,22 @@ Muxers/Demuxers:
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
img2*.c Michael Niedermayer
ipmovie.c Mike Melanson
ircam* Paul B Mahol
img2.c Michael Niedermayer
iss.c Stefan Gehrer
jacosub* Clément Bœsch
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Michael Niedermayer, Baptiste Coudurier
movenc.c Baptiste Coudurier, Matthieu Bouron
movenc.c Michael Niedermayer, Baptiste Coudurier
mpc.c Kostya Shishkov
mpeg.c Michael Niedermayer
mpegenc.c Michael Niedermayer
@@ -416,7 +334,6 @@ Muxers/Demuxers:
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
mxfdec.c Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut.c Michael Niedermayer
nuv.c Reimar Doeffinger
@@ -424,10 +341,8 @@ Muxers/Demuxers:
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
paf.c Paul B Mahol
psxstr.c Mike Melanson
pva.c Ivo van Poorten
pvfdec.c Paul B Mahol
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
@@ -443,51 +358,30 @@ Muxers/Demuxers:
segafilm.c Mike Melanson
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
smjpeg* Paul B Mahol
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
takdec.c Paul B Mahol
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
voc.c Aurelien Jacobs
wav.c Michael Niedermayer
wc3movie.c Mike Melanson
webvtt* Matthew J Heaney
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
wvenc.c Paul B Mahol
Protocols:
bluray.c Petri Hintukainen
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
mms*.c Ronald S. Bultje
udp.c Luca Abeni
libswresample
=============
Generic parts:
audioconvert.c Michael Niedermayer
dither.c Michael Niedermayer
rematrix*.c Michael Niedermayer
swresample*.c Michael Niedermayer
Resamplers:
resample*.c Michael Niedermayer
soxr_resample.c Rob Sykes
Operating systems / CPU architectures
=====================================
Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard, Nedeljko Babic
MIPS Mans Rullgard
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Luca Barbato
@@ -501,30 +395,23 @@ x86 Michael Niedermayer
Releases
========
2.1 Michael Niedermayer
2.0 Michael Niedermayer
0.9 Michael Niedermayer
If you want to maintain an older release, please contact us
GnuPG Fingerprints of maintainers and contributors
==================================================
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
Ash Hughes 694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
Ben Littler 3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Bœsch Clément 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
@@ -538,7 +425,5 @@ Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
Stefano Sabatini 9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
Tomas Härdin D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD

View File

@@ -15,24 +15,22 @@ 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 $(EXEOBJS)
OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
OBJS = $(PROGS-yes:%=%.o) cmdutils.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr base64
HOSTPROGS := $(TESTTOOLS:%=tests/%)
TOOLS = qt-faststart trasher
TOOLS-$(CONFIG_ZLIB) += cws2fws
BASENAMES = ffmpeg ffplay ffprobe ffserver
ALLPROGS = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLPROGS_G = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
ALLMANPAGES = $(BASENAMES:%=%.1)
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
@@ -41,9 +39,8 @@ FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
SKIPHEADERS = cmdutils_common_opts.h
include $(SRC_PATH)/common.mak
@@ -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 $(EXEOBJS)
$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
$(TOOLS): %$(EXESUF): %.o
$(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 \
HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ARMV5TE-OBJS ARMV6-OBJS VFP-OBJS NEON-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 cmdutils.o $(EXEOBJS)
$(1)$(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
ffplay.o: CFLAGS += $(SDL_CFLAGS)
ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)
$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
$(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
@@ -152,18 +137,26 @@ uninstall-data:
clean::
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(TOOLS)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) coverage.info
$(RM) -r coverage-html
$(RM) -rf coverage.info lcov
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h
$(RM) config.* .version version.h libavutil/avconfig.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
check: all alltools examples testprogs fate
# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
coverage-html: coverage.info
$(Q)mkdir -p $@
$(Q)genhtml -o $@ $<
$(Q)touch $@
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/tests/Makefile
@@ -179,5 +172,5 @@ $(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 *clean config examples install*
.PHONY: testprogs uninstall*

8
README
View File

@@ -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.

View File

@@ -1 +1 @@
2.1.8
0.10.1

View File

@@ -1 +1 @@
2.1.8
0.10.1

View File

@@ -1,16 +0,0 @@
OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
OBJS-$(HAVE_ARMV6) += $(ARMV6-OBJS) $(ARMV6-OBJS-yes)
OBJS-$(HAVE_VFP) += $(VFP-OBJS) $(VFP-OBJS-yes)
OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_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)

1318
cmdutils.c

File diff suppressed because it is too large Load Diff

View File

@@ -51,18 +51,7 @@ extern const int this_year;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern struct SwsContext *sws_opts;
extern AVDictionary *swr_opts;
extern AVDictionary *format_opts, *codec_opts, *resample_opts;
/**
* Register a program-specific cleanup routine.
*/
void register_exit(void (*cb)(int ret));
/**
* Wraps exit with a program-specific cleanup routine.
*/
void exit_program(int ret);
extern AVDictionary *format_opts, *codec_opts;
/**
* Initialize the cmdutils option system, in particular
@@ -81,34 +70,27 @@ void uninit_opts(void);
*/
void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
/**
* Override the cpuflags.
*/
int opt_cpuflags(void *optctx, const char *opt, const char *arg);
/**
* Fallback for options that are not explicitly handled, these will be
* parsed through AVOptions.
*/
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_codec_debug(void *optctx, const char *opt, const char *arg);
int opt_opencl(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.
@@ -138,7 +120,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
* not zero timestr is interpreted as a duration, otherwise as a
* date
*
* @see av_parse_time()
* @see parse_date()
*/
int64_t parse_time_or_die(const char *context, const char *timestr,
int is_duration);
@@ -154,7 +136,7 @@ typedef struct SpecifierOpt {
} u;
} SpecifierOpt;
typedef struct OptionDef {
typedef struct {
const char *name;
int flags;
#define HAS_ARG 0x0001
@@ -163,42 +145,32 @@ typedef struct OptionDef {
#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
an int containing element count in the array. */
#define OPT_TIME 0x10000
#define OPT_DOUBLE 0x20000
#define OPT_INPUT 0x40000
#define OPT_OUTPUT 0x80000
union {
void *dst_ptr;
int (*func_arg)(void *, const char *, const char *);
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
@@ -206,23 +178,10 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
*/
void show_help_children(const AVClass *class, int flags);
/**
* Per-fftool specific help handler. Implemented in each
* fftool, called by show_help().
*/
void show_help_default(const char *opt, const char *arg);
/**
* Generic -h handler common to all fftools.
*/
int show_help(void *optctx, const char *opt, const char *arg);
/**
* Parse the command line arguments.
*
* @param optctx an opaque options context
* @param argc number of command line arguments
* @param argv values of command line arguments
* @param options Array with the definitions required to interpret every
* option of the form: -option_name [argument]
* @param parse_arg_function Name of the function called to process every
@@ -240,112 +199,11 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
int parse_option(void *optctx, const char *opt, const char *arg,
const OptionDef *options);
/**
* An option extracted from the commandline.
* Cannot use AVDictionary because of options like -map which can be
* used multiple times.
*/
typedef struct Option {
const OptionDef *opt;
const char *key;
const char *val;
} Option;
typedef struct OptionGroupDef {
/**< group name */
const char *name;
/**
* Option to be used as group separator. Can be NULL for groups which
* are terminated by a non-option argument (e.g. ffmpeg output files)
*/
const char *sep;
/**
* Option flags that must be set on each option that is
* applied to this group
*/
int flags;
} OptionGroupDef;
typedef struct OptionGroup {
const OptionGroupDef *group_def;
const char *arg;
Option *opts;
int nb_opts;
AVDictionary *codec_opts;
AVDictionary *format_opts;
AVDictionary *resample_opts;
struct SwsContext *sws_opts;
AVDictionary *swr_opts;
} OptionGroup;
/**
* A list of option groups that all have the same group type
* (e.g. input files or output files)
*/
typedef struct OptionGroupList {
const OptionGroupDef *group_def;
OptionGroup *groups;
int nb_groups;
} OptionGroupList;
typedef struct OptionParseContext {
OptionGroup global_opts;
OptionGroupList *groups;
int nb_groups;
/* parsing state */
OptionGroup cur_group;
} OptionParseContext;
/**
* Parse an options group and write results into optctx.
*
* @param optctx an app-specific options context. NULL for global options group
*/
int parse_optgroup(void *optctx, OptionGroup *g);
/**
* Split the commandline into an intermediate form convenient for further
* processing.
*
* The commandline is assumed to be composed of options which either belong to a
* group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
* (everything else).
*
* A group (defined by an OptionGroupDef struct) is a sequence of options
* terminated by either a group separator option (e.g. -i) or a parameter that
* is not an option (doesn't start with -). A group without a separator option
* must always be first in the supplied groups list.
*
* All options within the same group are stored in one OptionGroup struct in an
* OptionGroupList, all groups with the same group definition are stored in one
* OptionGroupList in OptionParseContext.groups. The order of group lists is the
* same as the order of group definitions.
*/
int split_commandline(OptionParseContext *octx, int argc, char *argv[],
const OptionDef *options,
const OptionGroupDef *groups, int nb_groups);
/**
* Free all allocated memory in an OptionParseContext.
*/
void uninit_parse_context(OptionParseContext *octx);
/**
* Find the '-loglevel' option in the command line args and apply it.
*/
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.
*
@@ -363,16 +221,12 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
* Create a new options dictionary containing only the options from
* opts which apply to the codec with ID codec_id.
*
* @param opts dictionary to place options in
* @param codec_id ID of the codec that should be filtered for
* @param s Corresponding format context.
* @param st A stream from s for which the options should be filtered.
* @param codec The particular codec for which the options should be filtered.
* 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().
@@ -412,87 +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);
/**
* Print a listing containing all the color names and values recognized
* by the program.
*/
void show_colors(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
@@ -504,10 +333,9 @@ int read_yesno(void);
* Read the file with name filename, and put its content in a newly
* allocated 0-terminated buffer.
*
* @param filename file to read from
* @param bufptr location where pointer to buffer is returned
* @param size location where size of buffer is returned
* @return >= 0 in case of success, a negative value corresponding to an
* @return 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
@@ -521,7 +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
@@ -533,39 +361,20 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
FILE *get_preset_file(char *filename, size_t filename_size,
const char *preset_name, int is_path, const char *codec_name);
/**
* Do all the necessary cleanup and abort.
* This function is implemented in the avtools, not cmdutils.
*/
void exit_program(int ret);
/**
* Realloc array to hold new_size elements of elem_size.
* Calls exit() on failure.
* Calls exit_program() on failure.
*
* @param array array to reallocate
* @param elem_size size in bytes of each element
* @param size new element count will be written here
* @param new_size number of elements to place in reallocated array
* @return reallocated array
*/
void *grow_array(void *array, int elem_size, int *size, int new_size);
#define media_type_string av_get_media_type_string
#define GROW_ARRAY(array, nb_elems)\
array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
#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 */

View File

@@ -1,25 +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" },
{ "colors" , OPT_EXIT, {.func_arg = show_colors }, "show available color names" },
{ "loglevel" , HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" },
{ "v", HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" },
{ "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" },
#if CONFIG_OPENCL
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" },
#endif
{ "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" },

View File

@@ -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,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CPPFLAGS) $(CFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS += $(CPPFLAGS) $(CFLAGS)
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
CCFLAGS = $(CFLAGS)
CXXFLAGS := $(CFLAGS) $(CXXFLAGS)
YASMFLAGS += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
HOSTCFLAGS += $(IFLAGS)
LDFLAGS := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
define COMPILE
$(call $(1)DEP,$(1))
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
$($(1)DEP)
$($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
@@ -56,11 +53,8 @@ COMPILE_S = $(call COMPILE,AS)
%.o: %.S
$(COMPILE_S)
%.i: %.c
$(CC) $(CCFLAGS) $(CC_E) $<
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.ho: %.h
$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
@@ -79,14 +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)%))
@@ -97,46 +90,31 @@ HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
HEADERS += $(HEADERS-yes)
PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
SRC_DIR := $(SRC_PATH)/lib$(NAME)
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
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)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
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
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))

View File

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

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@@ -1,879 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
// NOTE: this is a partial update of the Avisynth C interface to recognize
// new color spaces added in Avisynth 2.60. By no means is this document
// completely Avisynth 2.60 compliant.
#ifndef __AVISYNTH_C__
#define __AVISYNTH_C__
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
typedef unsigned char BYTE;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVISYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 4 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
AVS_PLANAR_A=1<<4,
AVS_PLANAR_R=1<<5,
AVS_PLANAR_G=1<<6,
AVS_PLANAR_B=1<<7,
AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 1 << 3,
AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
AVS_CS_UPLANEFIRST = 1 << 4, // I420
AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
AVS_CS_PLANAR_FILTER = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
// AVS_CS_YV12 = 1<<3 Reserved
// AVS_CS_I420 = 1<<4 Reserved
AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
AVS_CS_YV24 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1, // YVU 4:4:4 planar
AVS_CS_YV16 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:2 planar
AVS_CS_YV12 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:0 planar
AVS_CS_I420 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YUV 4:2:0 planar
AVS_CS_IYUV = AVS_CS_I420,
AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:1 planar
AVS_CS_YUV9 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:0 planar
AVS_CS_Y8 = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 // Y 4:0:0 planar
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoInfo
//
// AVS_VideoInfo is layed out identicly to VideoInfo
typedef struct AVS_VideoInfo {
int width, height; // width=0 means no video
unsigned fps_numerator, fps_denominator;
int num_frames;
int pixel_type;
int audio_samples_per_second; // 0 means no audio
int sample_type;
INT64 num_audio_samples;
int nchannels;
// Imagetype properties
int image_type;
} AVS_VideoInfo;
// useful functions of the above
AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
{ return (p->width!=0); }
AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
{ return (p->audio_samples_per_second!=0); }
AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
{
switch (p->sample_type) {
case AVS_SAMPLE_INT8: return sizeof(signed char);
case AVS_SAMPLE_INT16: return sizeof(signed short);
case AVS_SAMPLE_INT24: return 3;
case AVS_SAMPLE_INT32: return sizeof(signed int);
case AVS_SAMPLE_FLOAT: return sizeof(float);
default: return 0;
}
}
AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
{ return p->nchannels*avs_bytes_per_channel_sample(p);}
AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
{ return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
{ return bytes / avs_bytes_per_audio_sample(p); }
AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return samples * avs_bytes_per_audio_sample(p); }
AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
{ return p->nchannels; }
AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
{ return p->sample_type;}
// useful mutator
AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
{ p->image_type|=property; }
AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
{ p->image_type&=~property; }
AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
{ if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
{
unsigned x=numerator, y=denominator;
while (y) { // find gcd
unsigned t = x%y; x = y; y = t;
}
p->fps_numerator = numerator/x;
p->fps_denominator = denominator/x;
}
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
BYTE * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
volatile long sequence_number;
volatile long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
volatile long refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
int row_sizeUV, heightUV;
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return p->pitch;}
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_sizeUV;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_sizeUV;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->heightUV;
return 0;
}
return p->height;}
AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
return p->vfb->data + p->offset;}
AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on an AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, int frame_range);
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
AVS_CPUF_SSE3 = 0x100, // PIV+, K8 Venice
AVS_CPUF_SSSE3 = 0x200, // Core 2
AVS_CPUF_SSE4 = 0x400, // Penryn, Wolfdale, Yorkfield
AVS_CPUF_SSE4_1 = 0x400,
AVS_CPUF_SSE4_2 = 0x800, // Nehalem
};
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#ifdef AVSC_NO_DECLSPEC
// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
/*
The following functions needs to have been declared, probably from windows.h
void* malloc(size_t)
void free(void*);
HMODULE LoadLibrary(const char*);
void* GetProcAddress(HMODULE, const char*);
FreeLibrary(HMODULE);
*/
typedef struct AVS_Library AVS_Library;
#define AVSC_DECLARE_FUNC(name) name##_func name
struct AVS_Library {
HMODULE handle;
AVSC_DECLARE_FUNC(avs_add_function);
AVSC_DECLARE_FUNC(avs_at_exit);
AVSC_DECLARE_FUNC(avs_bit_blt);
AVSC_DECLARE_FUNC(avs_check_version);
AVSC_DECLARE_FUNC(avs_clip_get_error);
AVSC_DECLARE_FUNC(avs_copy_clip);
AVSC_DECLARE_FUNC(avs_copy_value);
AVSC_DECLARE_FUNC(avs_copy_video_frame);
AVSC_DECLARE_FUNC(avs_create_script_environment);
AVSC_DECLARE_FUNC(avs_delete_script_environment);
AVSC_DECLARE_FUNC(avs_function_exists);
AVSC_DECLARE_FUNC(avs_get_audio);
AVSC_DECLARE_FUNC(avs_get_cpu_flags);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_get_frame);
AVSC_DECLARE_FUNC(avs_get_parity);
AVSC_DECLARE_FUNC(avs_get_var);
AVSC_DECLARE_FUNC(avs_get_version);
AVSC_DECLARE_FUNC(avs_get_video_info);
AVSC_DECLARE_FUNC(avs_invoke);
AVSC_DECLARE_FUNC(avs_make_writable);
AVSC_DECLARE_FUNC(avs_new_c_filter);
AVSC_DECLARE_FUNC(avs_new_video_frame_a);
AVSC_DECLARE_FUNC(avs_release_clip);
AVSC_DECLARE_FUNC(avs_release_value);
AVSC_DECLARE_FUNC(avs_release_video_frame);
AVSC_DECLARE_FUNC(avs_save_string);
AVSC_DECLARE_FUNC(avs_set_cache_hints);
AVSC_DECLARE_FUNC(avs_set_global_var);
AVSC_DECLARE_FUNC(avs_set_memory_max);
AVSC_DECLARE_FUNC(avs_set_to_clip);
AVSC_DECLARE_FUNC(avs_set_var);
AVSC_DECLARE_FUNC(avs_set_working_dir);
AVSC_DECLARE_FUNC(avs_sprintf);
AVSC_DECLARE_FUNC(avs_subframe);
AVSC_DECLARE_FUNC(avs_subframe_planar);
AVSC_DECLARE_FUNC(avs_take_clip);
AVSC_DECLARE_FUNC(avs_vsprintf);
};
#undef AVSC_DECLARE_FUNC
AVSC_INLINE AVS_Library * avs_load_library() {
AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
if (library == NULL)
return NULL;
library->handle = LoadLibrary("avisynth");
if (library->handle == NULL)
goto fail;
#define __AVSC_STRINGIFY(x) #x
#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
#define AVSC_LOAD_FUNC(name) {\
library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
if (library->name == NULL)\
goto fail;\
}
AVSC_LOAD_FUNC(avs_add_function);
AVSC_LOAD_FUNC(avs_at_exit);
AVSC_LOAD_FUNC(avs_bit_blt);
AVSC_LOAD_FUNC(avs_check_version);
AVSC_LOAD_FUNC(avs_clip_get_error);
AVSC_LOAD_FUNC(avs_copy_clip);
AVSC_LOAD_FUNC(avs_copy_value);
AVSC_LOAD_FUNC(avs_copy_video_frame);
AVSC_LOAD_FUNC(avs_create_script_environment);
AVSC_LOAD_FUNC(avs_delete_script_environment);
AVSC_LOAD_FUNC(avs_function_exists);
AVSC_LOAD_FUNC(avs_get_audio);
AVSC_LOAD_FUNC(avs_get_cpu_flags);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_get_frame);
AVSC_LOAD_FUNC(avs_get_parity);
AVSC_LOAD_FUNC(avs_get_var);
AVSC_LOAD_FUNC(avs_get_version);
AVSC_LOAD_FUNC(avs_get_video_info);
AVSC_LOAD_FUNC(avs_invoke);
AVSC_LOAD_FUNC(avs_make_writable);
AVSC_LOAD_FUNC(avs_new_c_filter);
AVSC_LOAD_FUNC(avs_new_video_frame_a);
AVSC_LOAD_FUNC(avs_release_clip);
AVSC_LOAD_FUNC(avs_release_value);
AVSC_LOAD_FUNC(avs_release_video_frame);
AVSC_LOAD_FUNC(avs_save_string);
AVSC_LOAD_FUNC(avs_set_cache_hints);
AVSC_LOAD_FUNC(avs_set_global_var);
AVSC_LOAD_FUNC(avs_set_memory_max);
AVSC_LOAD_FUNC(avs_set_to_clip);
AVSC_LOAD_FUNC(avs_set_var);
AVSC_LOAD_FUNC(avs_set_working_dir);
AVSC_LOAD_FUNC(avs_sprintf);
AVSC_LOAD_FUNC(avs_subframe);
AVSC_LOAD_FUNC(avs_subframe_planar);
AVSC_LOAD_FUNC(avs_take_clip);
AVSC_LOAD_FUNC(avs_vsprintf);
#undef __AVSC_STRINGIFY
#undef AVSC_STRINGIFY
#undef AVSC_LOAD_FUNC
return library;
fail:
free(library);
return NULL;
}
AVSC_INLINE void avs_free_library(AVS_Library *library) {
if (library == NULL)
return;
FreeLibrary(library->handle);
free(library);
}
#endif
#endif

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

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@@ -1,727 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef __AVXSYNTH_C__
#define __AVXSYNTH_C__
#include "windowsPorts/windows2linux.h"
#include <stdarg.h>
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVXSYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 3 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar
AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar
AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR // same as above
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoInfo
//
// AVS_VideoInfo is layed out identicly to VideoInfo
typedef struct AVS_VideoInfo {
int width, height; // width=0 means no video
unsigned fps_numerator, fps_denominator;
int num_frames;
int pixel_type;
int audio_samples_per_second; // 0 means no audio
int sample_type;
INT64 num_audio_samples;
int nchannels;
// Imagetype properties
int image_type;
} AVS_VideoInfo;
// useful functions of the above
AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
{ return (p->width!=0); }
AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
{ return (p->audio_samples_per_second!=0); }
AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
{
switch (p->sample_type) {
case AVS_SAMPLE_INT8: return sizeof(signed char);
case AVS_SAMPLE_INT16: return sizeof(signed short);
case AVS_SAMPLE_INT24: return 3;
case AVS_SAMPLE_INT32: return sizeof(signed int);
case AVS_SAMPLE_FLOAT: return sizeof(float);
default: return 0;
}
}
AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
{ return p->nchannels*avs_bytes_per_channel_sample(p);}
AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
{ return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
{ return bytes / avs_bytes_per_audio_sample(p); }
AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return samples * avs_bytes_per_audio_sample(p); }
AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
{ return p->nchannels; }
AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
{ return p->sample_type;}
// useful mutator
AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
{ p->image_type|=property; }
AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
{ p->image_type&=~property; }
AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
{ if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
{
unsigned x=numerator, y=denominator;
while (y) { // find gcd
unsigned t = x%y; x = y; y = t;
}
p->fps_numerator = numerator/x;
p->fps_denominator = denominator/x;
}
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
unsigned char * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
long sequence_number;
long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
int refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return p->pitch;}
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_size>>1;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_size>>1;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->height>>1;
return 0;
}
return p->height;}
AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
return p->vfb->data + p->offset;}
AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
INT64 integer64; // match addition of __int64 to avxplugin.h
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
#if defined __cplusplus
}
#endif // __cplusplus
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on am AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, size_t frame_range);
#if defined __cplusplus
}
#endif // __cplusplus
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
#if defined __cplusplus
}
#endif // __cplusplus
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
};
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
#if defined __cplusplus
}
#endif // __cplusplus
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#if defined __cplusplus
}
#endif // __cplusplus
#endif //__AVXSYNTH_C__

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

View File

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

View File

@@ -1,86 +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;
}

View File

@@ -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;
}

View File

@@ -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 */

View File

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

View File

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

View File

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

View File

@@ -1,93 +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 <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 (av_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;
}

View File

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

3790
configure vendored

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@@ -1,46 +1,24 @@
LIBRARIES-$(CONFIG_AVUTIL) += libavutil
LIBRARIES-$(CONFIG_SWSCALE) += libswscale
LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
LIBRARIES-$(CONFIG_AVCODEC) += libavcodec
LIBRARIES-$(CONFIG_AVFORMAT) += libavformat
LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice
LIBRARIES-$(CONFIG_AVFILTER) += libavfilter
COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils
COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler
COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters
COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols
COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices
COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters
MANPAGES1 = $(PROGS-yes:%=doc/%.1) $(PROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1)
MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3)
MANPAGES = $(MANPAGES1) $(MANPAGES3)
PODPAGES = $(PROGS-yes:%=doc/%.pod) $(PROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
MANPAGES = $(PROGS-yes:%=doc/%.1)
PODPAGES = $(PROGS-yes:%=doc/%.pod)
HTMLPAGES = $(PROGS-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/nut.html \
doc/libavfilter.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS-$(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
all-$(CONFIG_DOC): documentation
doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
@@ -50,88 +28,37 @@ 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 --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: TAG = HTML
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
doc/%.pod: doc/%.texi
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
doc/%-all.pod: TAG = POD
doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
$(DOCS): | doc
OBJDIRS += doc
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
install-doc: install-html install-man
install-html:
install-man:
ifdef CONFIG_HTMLPAGES
install-progs-$(CONFIG_DOC): install-html
install-html: $(HTMLPAGES)
$(Q)mkdir -p "$(DOCDIR)"
$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
endif
ifdef CONFIG_MANPAGES
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
$(Q)mkdir -p "$(MANDIR)/man3"
$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
endif
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
uninstall: uninstall-doc
uninstall-doc: uninstall-html uninstall-man
uninstall-html:
$(RM) -r "$(DOCDIR)"
uninstall: uninstall-man
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
clean:: docclean
distclean:: docclean
$(RM) doc/config.texi
docclean:
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
$(RM) -r doc/doxy/html
clean::
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
-include $(wildcard $(DOCS:%=%.d))
.PHONY: apidoc doc documentation
.PHONY: documentation

View File

@@ -1,11 +1,13 @@
Release Notes
=============
* 2.1 "Fourier" October, 2013
* 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.

View File

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

View File

@@ -0,0 +1,168 @@
All the numerical options, if not specified otherwise, accept in input
a string representing a number, which may contain one of the
International System number postfixes, for example 'K', 'M', 'G'.
If 'i' is appended after the postfix, powers of 2 are used instead of
powers of 10. The 'B' postfix multiplies the value for 8, and can be
appended after another postfix or used alone. This allows using for
example 'KB', 'MiB', 'G' and 'B' as postfix.
Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
with "no" the option name, for example using "-nofoo" in the
command line will set to false the boolean option with name "foo".
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
@code{a:1} stream 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 stream, the option is then applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams, for example @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data and 't' for attachments. If @var{stream_index} is given, then
matches stream number @var{stream_index} of this type. Otherwise matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then matches stream number @var{stream_index} in
program with id @var{program_id}. Otherwise matches all streams in this program.
@end table
@section Generic options
These options are shared amongst the av* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help
Show help.
@item -version
Show version.
@item -formats
Show available formats.
The fields preceding the format names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@end table
@item -codecs
Show available codecs.
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.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -loglevel @var{loglevel} | -v @var{loglevel}
Set the logging level used by the library.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
@item panic
@item fatal
@item error
@item warning
@item info
@item verbose
@item debug
@end table
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Note: setting the environment variable @code{FFREPORT} to any value has the
same effect.
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are obviously per-stream, so the chapter on stream
specifiers applies to them
Note @option{-nooption} syntax cannot be used for boolean AVOptions,
use @option{-option 0}/@option{-option 1}.
Note2 old undocumented way of specifying per-stream AVOptions by prepending
v/a/s to the options name is now obsolete and will be removed soon.

View File

@@ -17,46 +17,9 @@ Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
bitstream filter.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
ADTS header and removes the ADTS header.
This is required for example when copying an AAC stream from a raw
ADTS AAC container to a FLV or a MOV/MP4 file.
@section chomp
Remove zero padding at the end of a packet.
@section dump_extra
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item a
add extradata to all key packets, but only if @var{local_header} is
set in the @option{flags2} codec context field
@item k
add extradata to all key packets
@item e
add extradata to all packets
@end table
If not specified it is assumed @samp{k}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the @code{libx264} encoder, but corrects them by adding
the header stored in extradata to the key packets:
@example
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dump_extradata
@section h264_mp4toannexb
@@ -108,7 +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
@@ -123,6 +86,6 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@section noise
@section remove_extra
@section remove_extradata
@c man end BITSTREAM FILTERS

File diff suppressed because it is too large Load Diff

View File

@@ -60,143 +60,4 @@ This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
documented.
@section libcelt
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libcelt}.
@section libgsm
libgsm decoder wrapper.
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
the presence of the libgsm headers and library during configuration. You need
to explicitly configure the build with @code{--enable-libgsm}.
This decoder supports both the ordinary GSM and the Microsoft variant.
@section libilbc
libilbc decoder wrapper.
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
audio codec. Requires the presence of the libilbc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libilbc}.
@subsection Options
The following option is supported by the libilbc wrapper.
@table @option
@item enhance
Enable the enhancement of the decoded audio when set to 1. The default
value is 0 (disabled).
@end table
@section libopencore-amrnb
libopencore-amrnb decoder wrapper.
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrnb}.
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
without this library.
@section libopencore-amrwb
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrwb}.
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
without this library.
@section libopus
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
also be found in VobSub file pairs and in some Matroska files.
@subsection Options
@table @option
@item palette
Specify the global palette used by the bitmaps. When stored in VobSub, the
palette is normally specified in the index file; in Matroska, the palette is
stored in the codec extra-data in the same format as in VobSub. In DVDs, the
palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@end table
@section libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libzvbi}.
@subsection Options
@table @option
@item txt_page
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
Y offset of generated bitmaps, default is 0.
@item txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext charactes.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
miliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque (black) background.
@end table
@c man end SUBTILES DECODERS

View File

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

View File

@@ -1,23 +1,69 @@
@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg that can read the
Demuxers are configured elements in FFmpeg which allow to read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option @code{--list-demuxers}.
configure option "--list-demuxers".
You can disable all the demuxers using the configure option
@code{--disable-demuxers}, and selectively enable a single demuxer with
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
with the option @code{--disable-demuxer=@var{DEMUXER}}.
"--disable-demuxers", and selectively enable a single demuxer with
the option "--enable-demuxer=@var{DEMUXER}", or disable it
with the option "--disable-demuxer=@var{DEMUXER}".
The option @code{-formats} of the ff* tools will display the list of
The option "-formats" of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
@section image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
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
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 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 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}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
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:
@example
ffmpeg -i img.jpeg img.png
@end example
@section applehttp
Apple HTTP Live Streaming demuxer.
@@ -29,298 +75,6 @@ the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
@table @option
@item -no_resync_search @var{bool}
Do not try to resynchronize by looking for a certain optional start code.
@end table
@anchor{concat}
@section concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packet had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
and each next file starts where the previous one finishes. Note that it is
done globally and may cause gaps if all streams do not have exactly the same
length.
All files must have the same streams (same codecs, same time base, etc.).
The duration of each file is used to adjust the timestamps of the next file:
if the duration is incorrect (because it was computed using the bit-rate or
because the file is truncated, for example), it can cause artifacts. The
@code{duration} directive can be used to override the duration stored in
each file.
@subsection Syntax
The script is a text file in extended-ASCII, with one directive per line.
Empty lines, leading spaces and lines starting with '#' are ignored. The
following directive is recognized:
@table @option
@item @code{file @var{path}}
Path to a file to read; special characters and spaces must be escaped with
backslash or single quotes.
All subsequent directives apply to that file.
@item @code{ffconcat version 1.0}
Identify the script type and version. It also sets the @option{safe} option
to 1 if it was to its default -1.
To make FFmpeg recognize the format automatically, this directive must
appears exactly as is (no extra space or byte-order-mark) on the very first
line of the script.
@item @code{duration @var{dur}}
Duration of the file. This information can be specified from the file;
specifying it here may be more efficient or help if the information from the
file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the
whole concatenated video.
@end table
@subsection Options
This demuxer accepts the following option:
@table @option
@item safe
If set to 1, reject unsafe file paths. A file path is considered safe if it
does not contain a protocol specification and is relative and all components
only contain characters from the portable character set (letters, digits,
period, underscore and hyphen) and have no period at the beginning of a
component.
If set to 0, any file name is accepted.
The default is -1, it is equivalent to 1 if the format was automatically
probed and 0 otherwise.
@end table
@section flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams.
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@end table
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{http://code.google.com/p/game-music-emu/} for more information.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
For very large files, the @option{max_size} option may have to be adjusted.
@section libquvi
Play media from Internet services using the quvi project.
The demuxer accepts a @option{format} option to request a specific quality. It
is by default set to @var{best}.
See @url{http://quvi.sourceforge.net/} for more information.
FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
enabled.
@section image2
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 frame rate 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
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
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
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.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
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.
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:
@example
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 ts_from_file
If set to 1, will set frame timestamp to modification time of image file. Note
that monotonity of timestamps is not provided: images go in the same order as
without this option. Default value is 0.
@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 -framerate 10 -i 'img-%03d.jpeg' out.mkv
@end example
@item
As above, but start by reading from a file with index 100 in the sequence:
@example
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
@end example
@item
Read images matching the "*.png" glob pattern , that is all the files
terminating with the ".png" suffix:
@example
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section mpegts
MPEG-2 transport stream demuxer.
@table @option
@item fix_teletext_pts
Overrides teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
not discarded. Default value is 1, set this option to 0 if you want your
teletext packet PTS and DTS values untouched.
@end table
@section rawvideo
Raw video demuxer.
This demuxer allows to read raw video data. Since there is no header
specifying the assumed video parameters, the user must specify them
in order to be able to decode the data correctly.
This demuxer accepts the following options:
@table @option
@item framerate
Set input video frame rate. Default value is 25.
@item pixel_format
Set the input video pixel format. Default value is @code{yuv420p}.
@item video_size
Set the input video size. This value must be specified explicitly.
@end table
For example to read a rawvideo file @file{input.raw} with
@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
size of @code{320x240}, and a frame rate of 10 images per second, use
the command:
@example
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
@section sbg
SBaGen script demuxer.
@@ -351,25 +105,4 @@ the script is directly played, the actual times will match the absolute
timestamps up to the sound controller's clock accuracy, but if the user
somehow pauses the playback or seeks, all times will be shifted accordingly.
@section tedcaptions
JSON captions used for @url{http://www.ted.com/, TED Talks}.
TED does not provide links to the captions, but they can be guessed from the
page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
contains a bookmarklet to expose them.
This demuxer accepts the following option:
@table @option
@item start_time
Set the start time of the TED talk, in milliseconds. The default is 15000
(15s). It is used to sync the captions with the downloadable videos, because
they include a 15s intro.
@end table
Example: convert the captions to a format most players understand:
@example
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@c man end DEMUXERS
@c man end INPUT DEVICES

View File

@@ -11,23 +11,28 @@
@chapter Developers Guide
@section Notes for external developers
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in @file{doc/examples} and in the source code to
see how the public API is employed.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{libavformat/output-example.c} to use it to generate
audio or video streams.
You can use the FFmpeg libraries in your commercial program, but you
are encouraged to @emph{publish any patch you make}. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
@end itemize
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{http://ffmpeg.org/legal.html}.
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@section Contributing
@@ -51,16 +56,13 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@@ -114,17 +116,13 @@ int myfunc(int my_parameter)
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
@@ -136,72 +134,46 @@ clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@subsection Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in the CamelCase
There are the following conventions for naming variables and functions:
There are following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For file-scope variables and functions declared as @code{static}, no prefix
is required.
For variables and functions declared as @code{static} no prefixes are required.
@item
For variables and functions visible outside of file scope, but only used
internally by a library, an @code{ff_} prefix should be used,
e.g. @samp{ff_w64_demuxer}.
For variables and functions used internally by the library, @code{ff_} prefix
should be used.
For example, @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_aac_parse_header}.
For variables and functions used internally across multiple libraries, use
@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
@item
Each library has its own prefix for public symbols, in addition to the
commonly used @code{av_} (@code{avformat_} for libavformat,
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
Check the existing code and choose names accordingly.
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
@code{lib<name>/lib<name>.v} files.
For exported names, each library has its own prefixes. Just check the existing
code and name accordingly.
@end itemize
Furthermore, name space reserved for the system should not be invaded.
Identifiers ending in @code{_t} are reserved by
@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
Also avoid names starting with @code{__} or @code{_} followed by an uppercase
letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@subsection Miscellaneous conventions
@subsection Miscellanous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@@ -215,10 +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,automake set noexpandtab shiftwidth=8 softtabstop=8
" allow tabs in Makefiles
autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
@@ -228,174 +198,134 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
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
@enumerate
@item
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
a gift-style license, the
@uref{http://www.isc.org/software/license/, ISC} or
@uref{http://mit-license.org/, MIT} license.
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder).
@item
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help
finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@anchor{Submitting patches}
@section Submitting patches
@@ -418,6 +348,11 @@ The tool is located in the tools directory.
Run the @ref{Regression tests} before submitting a patch in order to verify
it does not cause unexpected problems.
Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch will not be trashed during
transmission) to the ffmpeg-devel mailing list, see
@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
@@ -425,13 +360,6 @@ and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to the
@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
@@ -446,51 +374,38 @@ send a reminder by email. Your patch should eventually be dealt with.
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
Did you add the 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?
If it has a fourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you @code{git add} the appropriate files before committing?
Did you @code{git add} the appropriate files before committing?
@item
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate
@@ -498,109 +413,73 @@ Did you make sure it compiles standalone, i.e. with
@enumerate
@item
Does @code{make fate} pass with the patch applied?
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
Was the patch generated with git format-patch or send-email?
@item
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
Did you sign off your patch? (git commit -s)
See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Does the patch not mix functional and cosmetic changes?
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider to add a regression test for your code.
Consider to add a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
If you added YASM code please check that things still work with --disable-yasm
@end enumerate
@section Patch review process
@@ -641,157 +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.
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
the following steps:
@enumerate
@item
Configure to compile with instrumentation enabled:
@code{configure --toolchain=gcov}.
@item
Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.
@item
Run @code{make lcov} to generate coverage data in HTML format.
@item
View @code{lcov/index.html} in your preferred HTML viewer.
@end enumerate
You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@subsection Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
to your configure line, and reasonable defaults will be set for running
FATE under the supervision of either the @strong{memcheck} or the
@strong{massif} tool of the valgrind suite.
In case you need finer control over how valgrind is invoked, use the
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
your configure line instead.
@anchor{Release process}
@section Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a @strong{release
manager} prepares, tests and publishes tarballs on the
@url{http://ffmpeg.org} website.
There are two kinds of releases:
@enumerate
@item
@strong{Major releases} always include the latest and greatest
features and functionality.
@item
@strong{Point releases} are cut from @strong{release} branches,
which are named @code{release/X}, with @code{X} being the release
version number.
@end enumerate
Note that we promise to our users that shared libraries from any FFmpeg
release never break programs that have been @strong{compiled} against
previous versions of @strong{the same release series} in any case!
However, from time to time, we do make API changes that require adaptations
in applications. Such changes are only allowed in (new) major releases and
require further steps such as bumping library version numbers and/or
adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@subsection Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@enumerate
@item
Fixes a security issue, preferably identified by a @strong{CVE
number} issued by @url{http://cve.mitre.org/}.
@item
Fixes a documented bug in @url{https://trac.ffmpeg.org}.
@item
Improves the included documentation.
@item
Retains both source code and binary compatibility with previous
point releases of the same release branch.
@end enumerate
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@subsection Release Checklist
The release process involves the following steps:
@enumerate
@item
Ensure that the @file{RELEASE} file contains the version number for
the upcoming release.
@item
Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
@item
Announce the intent to do a release to the mailing list.
@item
Make sure all relevant security fixes have been backported. See
@url{https://ffmpeg.org/security.html}.
@item
Ensure that the FATE regression suite still passes in the release
branch on at least @strong{i386} and @strong{amd64}
(cf. @ref{Regression tests}).
@item
Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
supplementing files that contain @code{gpg} signatures
@item
Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
push an annotated tag in the form @code{nX}, with @code{X}
containing the version number.
@item
Propose and send a patch to the @strong{ffmpeg-devel} mailing list
with a news entry for the website.
@item
Publish the news entry.
@item
Send announcement to the mailing list.
@end enumerate
@bye

View File

@@ -1,21 +0,0 @@
@chapter Device Options
@c man begin DEVICE OPTIONS
The libavdevice library provides the same interface as
libavformat. Namely, an input device is considered like a demuxer, and
an output device like a muxer, and the interface and generic device
options are the same provided by libavformat (see the ffmpeg-formats
manual).
In addition each input or output device may support so-called private
options, which are specific for that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the device
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
@c man end DEVICE OPTIONS
@include indevs.texi
@include outdevs.texi

View File

@@ -1,14 +0,0 @@
#!/bin/sh
SRC_PATH="${1}"
DOXYFILE="${2}"
shift 2
doxygen - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
HTML_HEADER = ${SRC_PATH}/doc/doxy/header.html
HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html
HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
EOF

File diff suppressed because it is too large Load Diff

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@@ -1,9 +1,10 @@
</div>
<div id="footer">
Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
</div>
<footer class="footer pagination-right">
<span class="label label-info">
Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
</span>
</footer>
</div>
</body>
</html>

View File

@@ -1,16 +1,14 @@
<!DOCTYPE html>
<html>
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
<meta http-equiv="X-UA-Compatible" content="IE=9"/>
<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
<!--Header replace -->
</head>
<div class="container">
<div id="container">
<!--Header replace -->
<div class="menu">
<div id="body">
<div>

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192
doc/eval.texi Normal file
View File

@@ -0,0 +1,192 @@
@chapter Expression Evaluation
@c man begin EXPRESSION EVALUATION
When evaluating an arithmetic expression, FFmpeg uses an internal
formula evaluator, implemented through the @file{libavutil/eval.h}
interface.
An expression may contain unary, binary operators, constants, and
functions.
Two expressions @var{expr1} and @var{expr2} can be combined to form
another expression "@var{expr1};@var{expr2}".
@var{expr1} and @var{expr2} are evaluated in turn, and the new
expression evaluates to the value of @var{expr2}.
The following binary operators are available: @code{+}, @code{-},
@code{*}, @code{/}, @code{^}.
The following unary operators are available: @code{+}, @code{-}.
The following functions are available:
@table @option
@item sinh(x)
@item cosh(x)
@item tanh(x)
@item sin(x)
@item cos(x)
@item tan(x)
@item atan(x)
@item asin(x)
@item acos(x)
@item exp(x)
@item log(x)
@item abs(x)
@item squish(x)
@item gauss(x)
@item isnan(x)
Return 1.0 if @var{x} is NAN, 0.0 otherwise.
@item mod(x, y)
@item max(x, y)
@item min(x, y)
@item eq(x, y)
@item gte(x, y)
@item gt(x, y)
@item lte(x, y)
@item lt(x, y)
@item st(var, expr)
Allow to store the value of the expression @var{expr} in an internal
variable. @var{var} specifies the number of the variable where to
store the value, and it is a value ranging from 0 to 9. The function
returns the value stored in the internal variable.
Note, Variables are currently not shared between expressions.
@item ld(var)
Allow to load the value of the internal variable with number
@var{var}, which was previously stored with st(@var{var}, @var{expr}).
The function returns the loaded value.
@item while(cond, expr)
Evaluate expression @var{expr} while the expression @var{cond} is
non-zero, and returns the value of the last @var{expr} evaluation, or
NAN if @var{cond} was always false.
@item ceil(expr)
Round the value of expression @var{expr} upwards to the nearest
integer. For example, "ceil(1.5)" is "2.0".
@item floor(expr)
Round the value of expression @var{expr} downwards to the nearest
integer. For example, "floor(-1.5)" is "-2.0".
@item trunc(expr)
Round the value of expression @var{expr} towards zero to the nearest
integer. For example, "trunc(-1.5)" is "-1.0".
@item sqrt(expr)
Compute the square root of @var{expr}. This is equivalent to
"(@var{expr})^.5".
@item not(expr)
Return 1.0 if @var{expr} is zero, 0.0 otherwise.
@item pow(x, y)
Compute the power of @var{x} elevated @var{y}, it is equivalent to
"(@var{x})^(@var{y})".
@item random(x)
Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
internal variable which will be used to save the seed/state.
@item hypot(x, y)
This function is similar to the C function with the same name; it returns
"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
right triangle with sides of length @var{x} and @var{y}, or the distance of the
point (@var{x}, @var{y}) from the origin.
@item gcd(x, y)
Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
@var{y} are 0 or either or both are less than zero then behavior is undefined.
@item if(x, y)
Evaluate @var{x}, and if the result is non-zero return the result of
the evaluation of @var{y}, return 0 otherwise.
@item ifnot(x, y)
Evaluate @var{x}, and if the result is zero return the result of the
evaluation of @var{y}, return 0 otherwise.
@end table
The following constants are available:
@table @option
@item PI
area of the unit disc, approximately 3.14
@item E
exp(1) (Euler's number), approximately 2.718
@item PHI
golden ratio (1+sqrt(5))/2, approximately 1.618
@end table
Assuming that an expression is considered "true" if it has a non-zero
value, note that:
@code{*} works like AND
@code{+} works like OR
and the construct:
@example
if A then B else C
@end example
is equivalent to
@example
if(A,B) + ifnot(A,C)
@end example
In your C code, you can extend the list of unary and binary functions,
and define recognized constants, so that they are available for your
expressions.
The evaluator also recognizes the International System number
postfixes. If 'i' is appended after the postfix, powers of 2 are used
instead of powers of 10. The 'B' postfix multiplies the value for 8,
and can be appended after another postfix or used alone. This allows
using for example 'KB', 'MiB', 'G' and 'B' as postfix.
Follows the list of available International System postfixes, with
indication of the corresponding powers of 10 and of 2.
@table @option
@item y
-24 / -80
@item z
-21 / -70
@item a
-18 / -60
@item f
-15 / -50
@item p
-12 / -40
@item n
-9 / -30
@item u
-6 / -20
@item m
-3 / -10
@item c
-2
@item d
-1
@item h
2
@item k
3 / 10
@item K
3 / 10
@item M
6 / 20
@item G
9 / 30
@item T
12 / 40
@item P
15 / 40
@item E
18 / 50
@item Z
21 / 60
@item Y
24 / 70
@end table
@c man end

View File

@@ -1,38 +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 -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= decoding_encoding \
demuxing \
filtering_video \
filtering_audio \
metadata \
muxing \
resampling_audio \
scaling_video \
EXAMPLES=decoding_encoding filtering metadata muxing
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: 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)

View File

@@ -1,18 +0,0 @@
FFmpeg examples README
----------------------
Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
1) Build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
2) Build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then you can go into
doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

View File

@@ -27,76 +27,18 @@
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
* @example doc/examples/decoding_encoding.c
*/
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.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_channels = 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_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
@@ -104,83 +46,44 @@ 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);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* 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);
}
@@ -188,46 +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);
}
@@ -247,30 +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);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* 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");
@@ -288,7 +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
@@ -329,7 +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);
}
/*
@@ -339,26 +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);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
picture= avcodec_alloc_frame();
/* put sample parameters */
c->bit_rate = 400000;
@@ -369,105 +237,79 @@ static void video_encode_example(const char *filename, int codec_id)
c->time_base= (AVRational){1,25};
c->gop_size = 10; /* emit one intra frame every ten frames */
c->max_b_frames=1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
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");
}
@@ -488,42 +330,15 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
pkt->size -= len;
pkt->data += len;
}
return 0;
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_count;
int frame, got_picture, len;
FILE *f;
AVFrame *frame;
AVFrame *picture;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
char buf[1024];
AVPacket avpkt;
av_init_packet(&avpkt);
@@ -531,20 +346,17 @@ 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);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
picture= avcodec_alloc_frame();
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
@@ -554,24 +366,20 @@ 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);
}
/* the codec gives us the frame size, in samples */
f = fopen(filename, "rb");
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_count = 0;
frame = 0;
for(;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
@@ -593,9 +401,26 @@ static void video_decode_example(const char *outfilename, const char *filename)
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0)
if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
while (avpkt.size > 0) {
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", frame);
exit(1);
}
if (got_picture) {
printf("saving frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
avpkt.size -= len;
avpkt.data += len;
}
}
/* some codecs, such as MPEG, transmit the I and P frame with a
@@ -603,48 +428,50 @@ static void video_decode_example(const char *outfilename, const char *filename)
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (got_picture) {
printf("saving last frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
fclose(f);
avcodec_close(c);
av_free(c);
avcodec_free_frame(&frame);
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;
}

View File

@@ -1,341 +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.
* @example doc/examples/demuxing.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
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;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
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 allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
ret = av_image_alloc(video_dst_data, video_dst_linesize,
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
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 audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
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]);
return ret < 0;
}

View File

@@ -24,18 +24,14 @@
/**
* @file
* API example for decoding and filtering
* @example doc/examples/filtering_video.c
*/
#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";
@@ -49,7 +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) {
@@ -57,7 +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;
}
@@ -72,7 +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;
}
@@ -88,18 +84,14 @@ static int init_filters(const char *filters_descr)
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
AVBufferSinkParams *buffersink_params;
enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
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) {
@@ -108,11 +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;
@@ -129,42 +118,41 @@ static int init_filters(const char *filters_descr)
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
return 0;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (picref->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
delay = av_rescale_q(picref->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
last_pts = picref->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
p0 = picref->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
for (y = 0; y < picref->video->h; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
for (x = 0; x < picref->video->w; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
p0 += picref->linesize[0];
}
fflush(stdout);
}
@@ -173,14 +161,9 @@ int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
AVFrame frame;
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
@@ -197,48 +180,43 @@ int main(int argc, char **argv)
/* read all packets */
while (1) {
AVFilterBufferRef *picref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
avcodec_get_frame_defaults(frame);
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
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_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
/* pull filtered pictures from the filtergraph */
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_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_frame_free(&frame);
av_frame_free(&filt_frame);
av_close_input_file(fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];

View File

@@ -1,265 +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
* @example doc/examples/filtering_audio.c
*/
#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 <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
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;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* 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("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const 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. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
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_ptr(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_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t *p = (uint16_t*)frame->data[0];
const 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 = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
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_flags(buffersrc_ctx, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if(ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
}
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

View File

@@ -23,7 +23,6 @@
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example doc/examples/metadata.c
*/
#include <stdio.h>
@@ -51,6 +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;
}

View File

@@ -26,7 +26,6 @@
*
* Output a media file in any supported libavformat format.
* The default codecs are used.
* @example doc/examples/muxing.c
*/
#include <stdlib.h>
@@ -34,174 +33,112 @@
#include <string.h>
#include <math.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.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 AV_PIX_FMT_YUV420P /* default pix_fmt */
#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */
static int sws_flags = SWS_BICUBIC;
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id)
/**************************************************************/
/* audio output */
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, enum CodecID codec_id)
{
AVCodecContext *c;
AVStream *st;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
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 = oc->nb_streams-1;
st->id = 1;
c = st->codec;
c->codec_id = codec_id;
c->codec_type = AVMEDIA_TYPE_AUDIO;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
break;
/* put sample parameters */
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* 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;
}
/**************************************************************/
/* audio output */
static float t, tincr, tincr2;
static uint8_t **src_samples_data;
static int src_samples_linesize;
static int src_nb_samples;
static int max_dst_nb_samples;
uint8_t **dst_samples_data;
int dst_samples_linesize;
int dst_samples_size;
struct SwrContext *swr_ctx = NULL;
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
static void open_audio(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
int ret;
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 */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
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;
src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
10000 : c->frame_size;
audio_outbuf_size = 10000;
audio_outbuf = av_malloc(audio_outbuf_size);
ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
src_nb_samples, c->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
exit(1);
}
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
/* 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;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = src_nb_samples;
ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
max_dst_nb_samples, c->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
exit(1);
}
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
c->sample_fmt, 0);
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;
@@ -210,9 +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;
}
}
@@ -220,141 +157,187 @@ 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, ret, dst_nb_samples;
AVPacket pkt;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
get_audio_frame(samples, audio_input_frame_size, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_samples_data[0] = src_samples_data[0];
dst_nb_samples = src_nb_samples;
}
frame->nb_samples = dst_nb_samples;
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
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. */
ret = av_interleaved_write_frame(oc, &pkt);
if (ret != 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
/* 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)
{
avcodec_close(st->codec);
av_free(src_samples_data[0]);
av_free(dst_samples_data[0]);
av_free(samples);
av_free(audio_outbuf);
}
/**************************************************************/
/* video output */
static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;
static AVFrame *picture, *tmp_picture;
static uint8_t *video_outbuf;
static int frame_count, video_outbuf_size;
static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
/* add a video output stream */
static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
{
int ret;
AVCodecContext *c = st->codec;
AVCodecContext *c;
AVStream *st;
AVCodec *codec;
st = avformat_new_stream(oc, NULL);
if (!st) {
fprintf(stderr, "Could not alloc stream\n");
exit(1);
}
c = st->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 */
c->bit_rate = 400000;
/* 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 == CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
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;
}
// some formats want stream headers to be separate
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
{
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 */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* allocate and init a re-usable frame */
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
video_outbuf = NULL;
if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
/* allocate output buffer */
/* XXX: API change will be done */
/* buffers passed into lav* can be allocated any way you prefer,
as long as they're aligned enough for the architecture, and
they're freed appropriately (such as using av_free for buffers
allocated with av_malloc) */
video_outbuf_size = 200000;
video_outbuf = av_malloc(video_outbuf_size);
}
/* allocate 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);
}
/* Allocate the encoded raw picture. */
ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
if (ret < 0) {
fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
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 (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
if (ret < 0) {
fprintf(stderr, "Could not allocate temporary picture: %s\n",
av_err2str(ret));
/* 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) {
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;
}
@@ -363,72 +346,76 @@ 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 != AV_PIX_FMT_YUV420P) {
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, AV_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 - directly store the picture in the packet */
/* 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 {
AVPacket pkt = { 0 };
int got_packet;
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
/* If size is zero, it means the image was buffered. */
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);
if (!ret && got_packet && pkt.size) {
if (c->coded_frame->pts != AV_NOPTS_VALUE)
pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
if(c->coded_frame->key_frame)
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
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;
}
}
if (ret != 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
fprintf(stderr, "Error while writing video frame\n");
exit(1);
}
frame_count++;
@@ -437,9 +424,13 @@ 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);
}
/**************************************************************/
@@ -451,20 +442,17 @@ int main(int argc, char **argv)
AVOutputFormat *fmt;
AVFormatContext *oc;
AVStream *audio_st, *video_st;
AVCodec *audio_codec, *video_codec;
double audio_time, video_time;
int ret;
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) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"Raw images can also be output by using '%%d' in the filename\n"
"\n", argv[0]);
return 1;
}
@@ -482,83 +470,87 @@ 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_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_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)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
fprintf(stderr, "Could not open '%s'\n", filename);
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
/* 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 (frame)
frame->pts = 0;
for (;;) {
/* Compute current audio and video time. */
audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
if (video_st)
video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
else
video_pts = 0.0;
if ((!audio_st || audio_time >= STREAM_DURATION) &&
(!video_st || video_time >= STREAM_DURATION))
if ((!audio_st || audio_pts >= STREAM_DURATION) &&
(!video_st || video_pts >= STREAM_DURATION))
break;
/* write interleaved audio and video frames */
if (!video_st || (video_st && audio_st && audio_time < video_time)) {
if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
write_audio_frame(oc, audio_st);
} else {
write_video_frame(oc, video_st);
frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
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);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
/* free the streams */
for(i = 0; i < oc->nb_streams; i++) {
av_freep(&oc->streams[i]->codec);
av_freep(&oc->streams[i]);
}
if (!(fmt->flags & AVFMT_NOFILE)) {
/* close the output file */
avio_close(oc->pb);
}
/* free the stream */
avformat_free_context(oc);
av_free(oc);
return 0;
}

View File

@@ -1,211 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example doc/examples/resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

View File

@@ -1,141 +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.
* @example doc/examples/scaling_video.c
*/
#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;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
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 alignment */
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;
}

View File

@@ -79,17 +79,6 @@ not a bug they should fix:
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.
To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.
@chapter Usage
@section ffmpeg does not work; what is wrong?
@@ -105,21 +94,12 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
@example
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example
Notice that @samp{%d} is replaced by the image number.
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
Use the @option{-start_number} option to declare a starting number for
the sequence. This is useful if your sequence does not start with
@file{img001.jpg} but is still in a numerical order. The following
example will start with @file{img100.jpg}:
@example
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
@@ -128,7 +108,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@@ -137,23 +117,17 @@ If you want to sequence them by oldest modified first, substitute
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
You can also use @command{cat} to pipe images to ffmpeg:
@example
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example
@section How do I encode movie to single pictures?
Use:
@example
ffmpeg -i movie.mpg movie%d.jpg
ffmpeg -i movie.mpg movie%d.jpg
@end example
The @file{movie.mpg} used as input will be converted to
@@ -169,7 +143,7 @@ to force the encoding.
Applying that to the previous example:
@example
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,79 +201,20 @@ then you may use any file that DirectShow can read as input.
Just create an "input.avs" text file with this single line ...
@example
DirectShowSource("C:\path to your file\yourfile.asf")
DirectShowSource("C:\path to your file\yourfile.asf")
@end example
... and then feed that text file to ffmpeg:
@example
ffmpeg -i input.avs
ffmpeg -i input.avs
@end example
For ANY other help on AviSynth, please visit the
@uref{http://www.avisynth.org/, AviSynth homepage}.
For ANY other help on Avisynth, please visit the
@uref{http://www.avisynth.org/, Avisynth homepage}.
@section How can I join video files?
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-filters.html#amerge, @code{amerge}} filter.
@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.
@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.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 the concat @emph{filter}
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files containing 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
@@ -307,38 +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 - -c:v mpeg4 -acodec libmp3lame 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
@@ -346,8 +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
@@ -364,7 +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 \
-y output.flv
-same_quant -y output.flv
rm temp[12].[av] all.[av]
@end example
@@ -393,61 +297,24 @@ Appending @code{:v} to it will do exactly that.
Use @option{-dumpgraph -} to find out exactly where the channel layout is
lost.
Most likely, it is through @code{auto-inserted aresample}. Try to understand
Most likely, it is through @code{auto-inserted aconvert}. Try to understand
why the converting filter was needed at that place.
Just before the output is a likely place, as @option{-f lavfi} currently
only support packed S16.
Then insert the correct @code{aformat} explicitly in the filtergraph,
Then insert the correct @code{aconvert} explicitly in the filter graph,
specifying the exact format.
@example
aformat=sample_fmts=s16:channel_layouts=stereo
aconvert=s16:stereo:packed
@end example
@section Why does FFmpeg not see the subtitles in my VOB file?
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.
@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
The @option{-sameq} option meant "same quantizer", and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
"same quality" and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.
Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
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}).
@@ -459,8 +326,31 @@ with @code{#ifdef}s related to the compiler.
@section Is Microsoft Visual C++ supported?
Yes. Please see the @uref{platform.html, Microsoft Visual C++}
section in the FFmpeg documentation.
No. Microsoft Visual C++ is not compliant to the C99 standard and does
not - among other things - support the inline assembly used in FFmpeg.
If you wish to use MSVC++ for your
project then you can link the MSVC++ code with libav* as long as
you compile the latter with a working C compiler. For more information, see
the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
documentation.
There have been efforts to make FFmpeg compatible with MSVC++ in the
past. However, they have all been rejected as too intrusive, especially
since MinGW does the job adequately. None of the core developers
work with MSVC++ and thus this item is low priority. Should you find
the silver bullet that solves this problem, feel free to shoot it at us.
We strongly recommend you to move over from MSVC++ to MinGW tools.
@section Can I use FFmpeg or libavcodec under Windows?
Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
Read the @emph{Windows} section in the FFmpeg documentation to find more
information.
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@section Can you add automake, libtool or autoconf support?
@@ -475,10 +365,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?
The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
contain full debug information. Those binaries are stripped to create
@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
the *_g versions.
The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
you need the debug information, use the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
@@ -486,24 +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
@@ -519,8 +390,12 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
You have to create a custom AVIOContext using @code{avio_alloc_context},
see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
You have to implement a URLProtocol, see @file{libavformat/file.c} in
FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
@section Where can I find libav* headers for Pascal/Delphi?
see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
@@ -534,12 +409,11 @@ In this specific case please look at RFC 4629 to see how it should be done.
@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
r_frame_rate is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
For example, if you have mixed 25 and 30 fps content, then r_frame_rate
will be 150.
@section Why is @code{make fate} not running all tests?

View File

@@ -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
@@ -27,7 +27,7 @@ by visiting this website:
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
with there recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
@@ -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
@@ -93,11 +90,11 @@ tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
configuration variables can be found at @file{doc/fate_config.sh.template}.
configuration variables can be found at @file{tests/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@verbatiminclude ../tests/fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
@@ -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
@@ -131,17 +127,7 @@ of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
@table @option
@item RSA
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
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.
b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@@ -153,20 +139,20 @@ the synchronisation of the samples directory.
@table @option
@item fate-rsync
Download/synchronize sample files to the configured samples directory.
Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
Will list all fate/regression test targets.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@table @option
@item V
Verbosity level, can be set to 0, 1 or 2.
Verbosity level, can be set to 0, 1 or 2.
@itemize
@item 0: show just the test arguments
@item 1: show just the command used in the test
@@ -174,32 +160,15 @@ Verbosity level, can be set to 0, 1 or 2.
@end itemize
@item SAMPLES
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
@item THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@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}.
@item GEN
Set to @var{1} to generate the missing or mismatched references.
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

File diff suppressed because it is too large Load Diff

View File

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

View File

@@ -11,7 +11,11 @@
@chapter Synopsis
ffplay [@var{options}] [@file{input_file}]
@example
@c man begin SYNOPSIS
ffplay [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
@@ -24,7 +28,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@@ -73,22 +77,11 @@ Default value is "video", if video is not present or cannot be played
You can interactively cycle through the available show modes by
pressing the key @key{w}.
@item -vf @var{filtergraph}
Create the filtergraph specified by @var{filtergraph} and use it to
filter the video stream.
@var{filtergraph} is a description of the filtergraph to apply to
the stream, and must have a single video input and a single video
output. In the filtergraph, the input is associated to the label
@code{in}, and the output to the label @code{out}. See the
ffmpeg-filters manual for more information about the filtergraph
syntax.
@item -af @var{filtergraph}
@var{filtergraph} is a description of the filtergraph to apply to
the input audio.
@item -vf @var{filter_graph}
@var{filter_graph} is a description of the filter graph to apply to
the input video.
Use the option "-filters" to show all the available filters (including
sources and sinks).
also sources and sinks).
@item -i @var{input_file}
Read @var{input_file}.
@@ -99,13 +92,9 @@ Read @var{input_file}.
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
Show the stream duration, the codec parameters, the current position in
the stream and the audio/video synchronisation drift.
@item -bug
Work around bugs.
@item -fast
@@ -145,20 +134,8 @@ Exit when video is done playing.
Exit if any key is pressed.
@item -exitonmousedown
Exit if any mouse button is pressed.
@item -codec:@var{media_specifier} @var{codec_name}
Force a specific decoder implementation for the stream identified by
@var{media_specifier}, which can assume the values @code{a} (audio),
@code{v} (video), and @code{s} subtitle.
@item -acodec @var{codec_name}
Force a specific audio decoder.
@item -vcodec @var{codec_name}
Force a specific video decoder.
@item -scodec @var{codec_name}
Force a specific subtitle decoder.
@item -codec:@var{stream_type}
Force a specific decoder implementation
@end table
@section While playing
@@ -174,16 +151,13 @@ Toggle full screen.
Pause.
@item a
Cycle audio channel in the curret program.
Cycle audio channel.
@item v
Cycle video channel.
@item t
Cycle subtitle channel in the current program.
@item c
Cycle program.
Cycle subtitle channel.
@item w
Show audio waves.
@@ -204,74 +178,28 @@ Seek to percentage in file corresponding to fraction of width.
@c man end
@include config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include eval.texi
@include decoders.texi
@include demuxers.texi
@include muxers.texi
@include indevs.texi
@include outdevs.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffplay.html,ffplay},
@end ifset
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffplay(1),
@end ifset
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@c man begin SEEALSO
ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye

View File

@@ -11,7 +11,13 @@
@chapter Synopsis
ffprobe [@var{options}] [@file{input_file}]
The generic syntax is:
@example
@c man begin SYNOPSIS
ffprobe [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
@@ -39,20 +45,15 @@ ffprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of a form defined by the selected
writer, which is specified by the @option{print_format} option.
Sections may contain other nested sections, and are identified by a
name (which may be shared by other sections), and an unique
name. See the output of @option{sections}.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
section.
and printed in the corresponding "FORMAT" or "STREAM" section.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@@ -79,7 +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
@@ -93,32 +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 -sections
Print sections structure and section information, and exit. The output
is not meant to be parsed by a machine.
@item -select_streams @var{stream_specifier}
Select only the streams specified by @var{stream_specifier}. This
option affects only the options related to streams
(e.g. @code{show_streams}, @code{show_packets}, etc.).
For example to show only audio streams, you can use the command:
@example
ffprobe -show_streams -select_streams a INPUT
@end example
To show only video packets belonging to the video stream with index 1:
@example
ffprobe -show_packets -select_streams v:1 INPUT
@end example
@item -show_data
Show payload data, as a hexadecimal and ASCII dump. Coupled with
@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.
@@ -131,64 +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.
This option is deprecated, use @code{show_entries} instead.
@item -show_entries @var{section_entries}
Set list of entries to show.
Entries are specified according to the following
syntax. @var{section_entries} contains a list of section entries
separated by @code{:}. Each section entry is composed by a section
name (or unique name), optionally followed by a list of entries local
to that section, separated by @code{,}.
If section name is specified but is followed by no @code{=}, all
entries are printed to output, together with all the contained
sections. Otherwise only the entries specified in the local section
entries list are printed. In particular, if @code{=} is specified but
the list of local entries is empty, then no entries will be shown for
that section.
Note that the order of specification of the local section entries is
not honored in the output, and the usual display order will be
retained.
The formal syntax is given by:
@example
@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
@end example
For example, to show only the index and type of each stream, and the PTS
time, duration time, and stream index of the packets, you can specify
the argument:
@example
packet=pts_time,duration_time,stream_index : stream=index,codec_type
@end example
To show all the entries in the section "format", but only the codec
type in the section "stream", specify the argument:
@example
format : stream=codec_type
@end example
To show all the tags in the stream and format sections:
@example
format_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
sections:
@example
stream_tags=title
@end example
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
@@ -210,90 +127,6 @@ multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -show_programs
Show information about programs and their streams contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "PROGRAM_STREAM".
@item -show_chapters
Show information about chapters stored in the format.
Each chapter is printed within a dedicated section with name "CHAPTER".
@item -count_frames
Count the number of frames per stream and report it in the
corresponding stream section.
@item -count_packets
Count the number of packets per stream and report it in the
corresponding stream section.
@item -read_intervals @var{read_intervals}
Read only the specified intervals. @var{read_intervals} must be a
sequence of interval specifications separated by ",".
@command{ffprobe} will seek to the interval starting point, and will
continue reading from that.
Each interval is specified by two optional parts, separated by "%".
The first part specifies the interval start position. It is
interpreted as an abolute position, or as a relative offset from the
current position if it is preceded by the "+" character. If this first
part is not specified, no seeking will be performed when reading this
interval.
The second part specifies the interval end position. It is interpreted
as an absolute position, or as a relative offset from the current
position if it is preceded by the "+" character. If the offset
specification starts with "#", it is interpreted as the number of
packets to read (not including the flushing packets) from the interval
start. If no second part is specified, the program will read until the
end of the input.
Note that seeking is not accurate, thus the actual interval start
point may be different from the specified position. Also, when an
interval duration is specified, the absolute end time will be computed
by adding the duration to the interval start point found by seeking
the file, rather than to the specified start value.
The formal syntax is given by:
@example
@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
@end example
A few examples follow.
@itemize
@item
Seek to time 10, read packets until 20 seconds after the found seek
point, then seek to position @code{01:30} (1 minute and thirty
seconds) and read packets until position @code{01:45}.
@example
10%+20,01:30%01:45
@end example
@item
Read only 42 packets after seeking to position @code{01:23}:
@example
01:23%+#42
@end example
@item
Read only the first 20 seconds from the start:
@example
%+20
@end example
@item
Read from the start until position @code{02:30}:
@example
%02:30
@end example
@end itemize
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
@@ -317,10 +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}.
@@ -333,9 +162,8 @@ Read @var{input_file}.
A writer defines the output format adopted by @command{ffprobe}, and will be
used for printing all the parts of the output.
A writer may accept one or more arguments, which specify the options
to adopt. The options are specified as a list of @var{key}=@var{value}
pairs, separated by ":".
A writer may accept one or more arguments, which specify the options to
adopt.
A description of the currently available writers follows.
@@ -351,27 +179,11 @@ keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
PROGRAM_STREAM section, and are prefixed by the string "TAG:".
Metadata tags are printed as a line in the corresponding FORMAT or
STREAM section, and are prefixed by the string "TAG:".
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:
@@ -383,29 +195,30 @@ Metadata tags are printed in the corresponding "format" or "stream"
section. A metadata tag key, if printed, is prefixed by the string
"tag:".
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@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}".
@@ -419,83 +232,22 @@ 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).
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.
Each section is printed using JSON notation.
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@@ -513,15 +265,14 @@ 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},
@option{sexagesimal} etc.) are specified.
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@@ -540,98 +291,44 @@ 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 config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@c man end WRITERS
@include decoders.texi
@include demuxers.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffprobe.html,ffprobe},
@end ifset
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffprobe(1),
@end ifset
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@include indevs.texi
@ignore
@setfilename ffprobe
@settitle ffprobe media prober
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye

View File

@@ -11,8 +11,6 @@
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
@@ -41,12 +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">
@@ -58,16 +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" />
<xsd:attribute name="pkt_size" type="xsd:int" />
<!-- 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" />
@@ -80,6 +70,7 @@
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="reference" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
@@ -88,41 +79,14 @@
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="programsType">
<xsd:sequence>
<xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
<xsd:attribute name="original" type="xsd:int" use="required" />
<xsd:attribute name="comment" type="xsd:int" use="required" />
<xsd:attribute name="lyrics" type="xsd:int" use="required" />
<xsd:attribute name="karaoke" type="xsd:int" use="required" />
<xsd:attribute name="forced" type="xsd:int" use="required" />
<xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<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"/>
@@ -138,38 +102,15 @@
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="channel_layout" type="xsd:string"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<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="programType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="end_time" type="xsd:float"/>
<xsd:attribute name="end_pts" type="xsd:long"/>
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="formatType">
@@ -179,14 +120,12 @@
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="nb_programs" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="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"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
<xsd:attribute name="probe_score" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="tagType">
@@ -209,32 +148,12 @@
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="chaptersType">
<xsd:sequence>
<xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="chapterType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="id" type="xsd:int" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="end" type="xsd:int" use="required"/>
<xsd:attribute name="end_time" type="xsd:float" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
<xsd:attribute name="ident" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">

View File

@@ -25,6 +25,10 @@ MaxBandwidth 1000
# '-' is the standard output.
CustomLog -
# Suppress that if you want to launch ffserver as a daemon.
NoDaemon
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
@@ -369,3 +373,5 @@ ACL allow 192.168.0.0 192.168.255.255
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>

View File

@@ -9,35 +9,52 @@
@contents
@chapter Synopsis
@chapter Synopsys
ffserver [@var{options}]
The generic syntax is:
@example
@c man begin SYNOPSIS
ffserver [options]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
@command{ffserver} is a streaming server for both audio and video. It
supports several live feeds, streaming from files and time shifting on
live feeds (you can seek to positions in the past on each live feed,
provided you specify a big enough feed storage in
@file{ffserver.conf}).
ffserver is a streaming server for both audio and video. It supports
@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
RTP/RTSP/HTTP.
several live feeds, streaming from files and time shifting on live feeds
(you can seek to positions in the past on each live feed, provided you
specify a big enough feed storage in ffserver.conf).
An @command{ffserver} instance will listen on some port as specified
in the configuration file. You can launch one or more instances of
@command{ffmpeg} and send one or more FFM streams to the port where
ffserver is expecting to receive them. Alternately, you can make
@command{ffserver} launch such @command{ffmpeg} instances at startup.
ffserver runs in daemon mode by default; that is, it puts itself in
the background and detaches from its TTY, unless it is launched in
debug mode or a NoDaemon option is specified in the configuration
file.
Input streams are called feeds, and each one is specified by a
@code{<Feed>} section in the configuration file.
This documentation covers only the streaming aspects of ffserver /
ffmpeg. All questions about parameters for ffmpeg, codec questions,
etc. are not covered here. Read @file{ffmpeg.html} for more
information.
@section How does it work?
ffserver receives prerecorded files or FFM streams from some ffmpeg
instance as input, then streams them over RTP/RTSP/HTTP.
An ffserver instance will listen on some port as specified in the
configuration file. You can launch one or more instances of ffmpeg and
send one or more FFM streams to the port where ffserver is expecting
to receive them. Alternately, you can make ffserver launch such ffmpeg
instances at startup.
Input streams are called feeds, and each one is specified by a <Feed>
section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a @code{<Stream>} section in the
configuration file.
formats, each one specified by a <Stream> section in the configuration
file.
@section Status stream
@@ -73,6 +90,14 @@ web server can be used to serve up the files just as well.
It can stream prerecorded video from .ffm files, though it is somewhat tricky
to make it work correctly.
@section What do I need?
I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm
using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
I needed some special drivers for my motherboard-based sound card.]
I understand that FreeBSD systems work just fine as well.
@section How do I make it work?
First, build the kit. It *really* helps to have installed LAME first. Then when
@@ -213,23 +238,10 @@ You use this by adding the ?date= to the end of the URL for the stream.
For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end
@section What is FFM, FFM2
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@@ -242,79 +254,26 @@ within the various <Stream> sections. Since ffserver will not launch
any ffmpeg instances, you will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, directs log
messages to stdout.
messages to stdout and causes ffserver to run in the foreground
rather than as a daemon.
@end table
@c man end
@include config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffserver.html,ffserver},
@end ifset
@ifset config-not-all
@url{ffserver-all.html,ffserver-all},
@end ifset
the @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffserver(1),
@end ifset
@ifset config-not-all
ffserver-all(1),
@end ifset
the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffserver
@settitle ffserver video server
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
example and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye

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@@ -1,292 +0,0 @@
All the numerical options, if not specified otherwise, accept a string
representing a number as input, which may be followed by one of the SI
unit prefixes, for example: 'K', 'M', or 'G'.
If 'i' is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiplies, which are based on
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
prefix multiplies the value by 8. This allows using, for example:
'KB', 'MiB', 'G' and 'B' as number suffixes.
Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
the option name with "no". For example using "-nofoo"
will set the boolean option with name "foo" to false.
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) a given option belongs to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
would select the ac3 codec for the second audio stream.
A stream specifier can match several streams, so that the option is applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams. For example, @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
program.
@item #@var{stream_id}
Matches the stream by a format-specific ID.
@end table
@section Generic options
These options are shared amongst the ff* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help [@var{arg}]
Show help. An optional parameter may be specified to print help about a specific
item. If no argument is specified, only basic (non advanced) tool
options are shown.
Possible values of @var{arg} are:
@table @option
@item long
Print advanced tool options in addition to the basic tool options.
@item full
Print complete list of options, including shared and private options
for encoders, decoders, demuxers, muxers, filters, etc.
@item decoder=@var{decoder_name}
Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@item encoder=@var{encoder_name}
Print detailed information about the encoder named @var{encoder_name}. Use the
@option{-encoders} option to get a list of all encoders.
@item demuxer=@var{demuxer_name}
Print detailed information about the demuxer named @var{demuxer_name}. Use the
@option{-formats} option to get a list of all demuxers and muxers.
@item muxer=@var{muxer_name}
Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.
@item filter=@var{filter_name}
Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
@end table
@item -version
Show version.
@item -formats
Show available formats.
@item -codecs
Show all codecs known to libavcodec.
Note that the term 'codec' is used throughout this documentation as a shortcut
for what is more correctly called a media bitstream format.
@item -decoders
Show available decoders.
@item -encoders
Show all available encoders.
@item -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -layouts
Show channel names and standard channel layouts.
@item -colors
Show recognized color names.
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
to the first line and the "Last message repeated n times" line will be
omitted. "repeat" can also be used alone.
If "repeat" is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
'repeat' will not change the loglevel.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
Show nothing at all; be silent.
@item panic
Only show fatal errors which could lead the process to crash, such as
and assert failure. This is not currently used for anything.
@item fatal
Only show fatal errors. These are errors after which the process absolutely
cannot continue after.
@item error
Show all errors, including ones which can be recovered from.
@item warning
Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
@item info
Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
@item verbose
Same as @code{info}, except more verbose.
@item debug
Show everything, including debugging information.
@end table
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Setting the environment variable @code{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
options will affect the report; options values must be escaped if they
contain special characters or the options delimiter ':' (see the
``Quoting and escaping'' section in the ffmpeg-utils manual). The
following option is recognized:
@table @option
@item file
set the file name to use for the report; @code{%p} is expanded to the name
of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
to a plain @code{%}
@end table
Errors in parsing the environment variable are not fatal, and will not
appear in the report.
@item -cpuflags flags (@emph{global})
Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you're doing.
@example
ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
@end example
Possible flags for this option are:
@table @samp
@item x86
@table @samp
@item mmx
@item mmxext
@item sse
@item sse2
@item sse2slow
@item sse3
@item sse3slow
@item ssse3
@item atom
@item sse4.1
@item sse4.2
@item avx
@item xop
@item fma4
@item 3dnow
@item 3dnowext
@item cmov
@end table
@item ARM
@table @samp
@item armv5te
@item armv6
@item armv6t2
@item vfp
@item vfpv3
@item neon
@end table
@item PowerPC
@table @samp
@item altivec
@end table
@item Specific Processors
@table @samp
@item pentium2
@item pentium3
@item pentium4
@item k6
@item k62
@item athlon
@item athlonxp
@item k8
@end table
@end table
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.
@var{options} must be a list of @var{key}=@var{value} option pairs
separated by ':'. See the ``OpenCL Options'' section in the
ffmpeg-utils manual for the list of supported options.
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them.
Note: the @option{-nooption} syntax cannot be used for boolean
AVOptions, use @option{-option 0}/@option{-option 1}.
Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.

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@@ -1,270 +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, sample format (the sample packing is implied by the sample
format) and sample rate.
The lists are not just lists, they are references to shared objects. When
the negotiation mechanism computes the intersection of the formats
supported at each end of a link, all references to both lists are replaced
with a reference to the intersection. And when a single format is
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.
query_formats can leave some formats unset and return AVERROR(EAGAIN) to
cause the negotiation mechanism to try again later. That can be used by
filters with complex requirements to use the format negotiated on one link
to set the formats supported on another.
Buffer references ownership and permissions
===========================================
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_frame 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 filter_frame method (or its start_frame
deprecated version) belongs to the corresponding filter.
Special exception: for video references: the reference may be used
internally for automatic copying and must not be destroyed before
end_frame; it can be given away to ff_start_frame.
* A reference passed to ff_filter_frame (or the deprecated
ff_start_frame) is given away and must no longer be used.
* A reference 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. The
cur_buf and out_buf were used with the deprecated
start_frame/draw_slice/end_frame API and should no longer be used.
src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
and must not be accessed by filters.
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 that implement the deprecated
start_frame/draw_slice/end_frame API, the permissions only apply to the
parts of the buffer that have already been covered by the draw_slice
method.
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 must request a buffer
with the WRITE permission.
* Filters that intend to keep a reference after the filtering process
is finished (after filter_frame returns) must have the PRESERVE
permission on it and remove the WRITE permission if they create a new
reference to give it away.
* 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.
filter_frame
------------
This method is called when a frame is pushed to the filter's input. It
can be called at any time except in a reentrant way.
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: frame rate-doubling filter: filter_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 filter_frame 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 = ff_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 filter_frame method will be called and do the work.
Legacy API
==========
Until libavfilter 3.23, the filter_frame method was split:
- for video filters, it was made of start_frame, draw_slice (that could be
called several times on distinct parts of the frame) and end_frame;
- for audio filters, it was called filter_samples.

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@@ -1,188 +0,0 @@
@chapter Format Options
@c man begin FORMAT OPTIONS
The libavformat library provides some generic global options, which
can be set on all the muxers and demuxers. In addition each muxer or
demuxer may support so-called private options, which are specific for
that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
The list of supported options follows:
@table @option
@item avioflags @var{flags} (@emph{input/output})
Possible values:
@table @samp
@item direct
Reduce buffering.
@end table
@item probesize @var{integer} (@emph{input})
Set probing size in bytes, i.e. the size of the data to analyze to get
stream information. A higher value will allow to detect more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@item packetsize @var{integer} (@emph{output})
Set packet size.
@item fflags @var{flags} (@emph{input/output})
Set format flags.
Possible values:
@table @samp
@item ignidx
Ignore index.
@item genpts
Generate PTS.
@item nofillin
Do not fill in missing values that can be exactly calculated.
@item noparse
Disable AVParsers, this needs @code{+nofillin} too.
@item igndts
Ignore DTS.
@item discardcorrupt
Discard corrupted frames.
@item sortdts
Try to interleave output packets by DTS.
@item keepside
Do not merge side data.
@item latm
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@end table
@item seek2any @var{integer} (@emph{input})
Allow seeking to non-keyframes on demuxer level when supported if set to 1.
Default is 0.
@item analyzeduration @var{integer} (@emph{input})
Specify how many microseconds are analyzed to probe the input. A
higher value will allow to detect more accurate information, but will
increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
@item cryptokey @var{hexadecimal string} (@emph{input})
Set decryption key.
@item indexmem @var{integer} (@emph{input})
Set max memory used for timestamp index (per stream).
@item rtbufsize @var{integer} (@emph{input})
Set max memory used for buffering real-time frames.
@item fdebug @var{flags} (@emph{input/output})
Print specific debug info.
Possible values:
@table @samp
@item ts
@end table
@item max_delay @var{integer} (@emph{input/output})
Set maximum muxing or demuxing delay in microseconds.
@item fpsprobesize @var{integer} (@emph{input})
Set number of frames used to probe fps.
@item audio_preload @var{integer} (@emph{output})
Set microseconds by which audio packets should be interleaved earlier.
@item chunk_duration @var{integer} (@emph{output})
Set microseconds for each chunk.
@item chunk_size @var{integer} (@emph{output})
Set size in bytes for each chunk.
@item err_detect, f_err_detect @var{flags} (@emph{input})
Set error detection flags. @code{f_err_detect} is deprecated and
should be used only via the @command{ffmpeg} tool.
Possible values:
@table @samp
@item crccheck
Verify embedded CRCs.
@item bitstream
Detect bitstream specification deviations.
@item buffer
Detect improper bitstream length.
@item explode
Abort decoding on minor error detection.
@item careful
Consider things that violate the spec and have not been seen in the
wild as errors.
@item compliant
Consider all spec non compliancies as errors.
@item aggressive
Consider things that a sane encoder should not do as an error.
@end table
@item use_wallclock_as_timestamps @var{integer} (@emph{input})
Use wallclock as timestamps.
@item avoid_negative_ts @var{integer} (@emph{output})
Shift timestamps to make them non-negative. A value of 1 enables shifting,
a value of 0 disables it, the default value of -1 enables shifting
when required by the target format.
When shifting is enabled, all output timestamps are shifted by the
same amount. Audio, video, and subtitles desynching and relative
timestamp differences are preserved compared to how they would have
been without shifting.
Also note that this affects only leading negative timestamps, and not
non-monotonic negative timestamps.
@item skip_initial_bytes @var{integer} (@emph{input})
Set number of bytes to skip before reading header and frames if set to 1.
Default is 0.
@item correct_ts_overflow @var{integer} (@emph{input})
Correct single timestamp overflows if set to 1. Default is 1.
@item flush_packets @var{integer} (@emph{output})
Flush the underlying I/O stream after each packet. Default 1 enables it, and
has the effect of reducing the latency; 0 disables it and may slightly
increase performance in some cases.
@end table
@c man end FORMAT OPTIONS
@anchor{Format stream specifiers}
@section Format stream specifiers
Format stream specifiers allow selection of one or more streams that
match specific properties.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' for video, 'a' for audio,
's' for subtitle, 'd' for data, and 't' for attachments. If
@var{stream_index} is given, then it matches the stream number
@var{stream_index} of this type. Otherwise, it matches all streams of
this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number
@var{stream_index} in the program with the id
@var{program_id}. Otherwise, it matches all streams in the program.
@item #@var{stream_id}
Matches the stream by a format-specific ID.
@end table
The exact semantics of stream specifiers is defined by the
@code{avformat_match_stream_specifier()} function declared in the
@file{libavformat/avformat.h} header.
@include demuxers.texi
@include muxers.texi
@include metadata.texi

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@@ -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
@@ -24,22 +23,17 @@ instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
@file{./configure}.
@section OpenCORE, VisualOn, and Fraunhofer libraries
@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
(see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
incompatible to the LGPL version 2.1 and GPL version 2. You have to
incompatible with the LGPL version 2.1 and GPL version 2. You have to
upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
order to use it.
The Fraunhofer AAC library is licensed under a license incompatible to the GPL
and is not known to be compatible to the LGPL. Therefore, you have to pass
@code{--enable-nonfree} to configure to use it.
GPL components, GPL version 3) to use it.
@end float
@subsection OpenCORE AMR
@@ -68,14 +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.
@@ -84,30 +70,14 @@ 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/VP9 encoding.
FFmpeg can make use of the libvpx library for VP8 encoding.
Go to @url{http://www.webmproject.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libvpx} to configure to
enable it.
@section libwavpack
FFmpeg can make use of the libwavpack library for WavPack encoding.
Go to @url{http://www.wavpack.com/} and follow the instructions for
installing the library. Then pass @code{--enable-libwavpack} to configure to
enable it.
@section x264
FFmpeg can make use of the x264 library for H.264 encoding.
@@ -122,31 +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.
@section libzvbi
libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
teletext pages and DVB teletext subtitles.
Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
installing the library. Then pass @code{--enable-libzvbi} to configure to
enable it.
@float NOTE
libzvbi is licensed under the GNU General Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@chapter Supported File Formats, Codecs or Features
@@ -170,21 +115,11 @@ 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 ADP @tab @tab X
@tab Audio format used on the Nintendo Gamecube.
@item AFC @tab @tab X
@tab Audio format used on the Nintendo Gamecube.
@item ASF @tab X @tab X
@item AST @tab X @tab X
@tab Audio format used on the Nintendo Wii.
@item AVI @tab X @tab X
@item AviSynth @tab @tab X
@item AVR @tab @tab X
@tab Audio format used on Mac.
@item AVISynth @tab @tab X
@item AVS @tab @tab X
@tab Multimedia format used by the Creature Shock game.
@item Beam Software SIFF @tab @tab X
@@ -198,8 +133,6 @@ library:
@tab Used in Z and Z95 games.
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item BRSTM @tab @tab X
@tab Audio format used on the Nintendo Wii.
@item BWF @tab X @tab X
@item CRI ADX @tab X @tab X
@tab Audio-only format used in console video games.
@@ -210,8 +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
@@ -230,7 +161,6 @@ library:
@item Electronic Arts cdata @tab @tab X
@item Electronic Arts Multimedia @tab @tab X
@tab Used in various EA games; files have extensions like WVE and UV2.
@item Ensoniq Paris Audio File @tab @tab X
@item FFM (FFserver live feed) @tab X @tab X
@item Flash (SWF) @tab X @tab X
@item Flash 9 (AVM2) @tab X @tab X
@@ -245,12 +175,12 @@ library:
@item G.723.1 @tab X @tab X
@item G.729 BIT @tab X @tab X
@item G.729 raw @tab @tab X
@item GIF Animation @tab X @tab X
@item GIF Animation @tab X @tab
@item GXF @tab X @tab X
@tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
playout servers.
@item 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
@@ -258,20 +188,17 @@ 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
@tab A format generated by IndigoVision 8000 video server.
@item IVF (On2) @tab X @tab X
@tab A format used by libvpx
@item IRCAM @tab X @tab X
@item LATM @tab X @tab X
@item LMLM4 @tab @tab X
@tab Used by Linux Media Labs MPEG-4 PCI boards
@item LOAS @tab @tab X
@tab contains LATM multiplexed AAC audio
@item LVF @tab @tab X
@item LXF @tab @tab X
@tab VR native stream format, used by Leitch/Harris' video servers.
@item Matroska @tab X @tab X
@@ -281,9 +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 Megalux Frame @tab @tab X
@tab Used by Megalux Ultimate Paint
@item Mobotix .mxg @tab @tab X
@item Monkey's Audio @tab @tab X
@item Motion Pixels MVI @tab @tab X
@@ -311,7 +235,6 @@ library:
@tab SMPTE 386M, D-10/IMX Mapping.
@item NC camera feed @tab @tab X
@tab NC (AVIP NC4600) camera streams
@item NIST SPeech HEader REsources @tab @tab X
@item NTT TwinVQ (VQF) @tab @tab X
@tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video @tab @tab X
@@ -320,7 +243,6 @@ library:
@tab NUT Open Container Format
@item Ogg @tab X @tab X
@item Playstation Portable PMP @tab @tab X
@item Portable Voice Format @tab @tab X
@item TechnoTrend PVA @tab @tab X
@tab Used by TechnoTrend DVB PCI boards.
@item QCP @tab @tab X
@@ -331,7 +253,6 @@ library:
@item raw Dirac @tab X @tab X
@item raw DNxHD @tab X @tab X
@item raw DTS @tab X @tab X
@item raw DTS-HD @tab @tab X
@item raw E-AC-3 @tab X @tab X
@item raw FLAC @tab X @tab X
@item raw GSM @tab @tab X
@@ -349,9 +270,8 @@ library:
@item raw video @tab X @tab X
@item raw id RoQ @tab X @tab
@item raw Shorten @tab @tab X
@item raw TAK @tab @tab X
@item raw TrueHD @tab X @tab X
@item raw VC-1 @tab X @tab X
@item raw VC-1 @tab @tab X
@item raw PCM A-law @tab X @tab X
@item raw PCM mu-law @tab X @tab X
@item raw PCM signed 8 bit @tab X @tab X
@@ -377,13 +297,11 @@ library:
@tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
@item RealMedia @tab X @tab X
@item Redirector @tab @tab X
@item RedSpark @tab @tab X
@item Renderware TeXture Dictionary @tab @tab X
@item RL2 @tab @tab X
@tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie @tab @tab X
@item Lego Mindstorms RSO @tab X @tab X
@item RSD @tab @tab X
@item RTMP @tab X @tab X
@tab Output is performed by publishing stream to RTMP server
@item RTP @tab X @tab X
@@ -393,7 +311,6 @@ library:
@item SDP @tab @tab X
@item Sega FILM/CPK @tab @tab X
@tab Used in many Sega Saturn console games.
@item Silicon Graphics Movie @tab @tab X
@item Sierra SOL @tab @tab X
@tab .sol files used in Sierra Online games.
@item Sierra VMD @tab @tab X
@@ -402,12 +319,10 @@ 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
@item Sony Wave64 (W64) @tab X @tab X
@item Sony Wave64 (W64) @tab @tab X
@item SoX native format @tab X @tab X
@item SUN AU format @tab X @tab X
@item Text files @tab @tab X
@@ -417,9 +332,8 @@ library:
@tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio @tab @tab X
@item VC-1 test bitstream @tab X @tab X
@item Vivo @tab @tab X
@item WAV @tab X @tab X
@item WavPack @tab X @tab X
@item 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
@@ -449,14 +363,11 @@ following image formats are supported:
@item .Y.U.V @tab X @tab X
@tab one raw file per component
@item animated GIF @tab X @tab X
@tab Only uncompressed GIFs are generated.
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item PIX @tab @tab X
@tab PIX is an image format used in the Argonaut BRender engine.
@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
@@ -482,18 +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 WebP @tab @tab X
@tab WebP image format
@item XBM @tab X @tab X
@tab X BitMap image format
@item XFace @tab X @tab X
@tab X-Face image format
@item XWD @tab X @tab X
@tab X Window Dump image format
@end multitable
@@ -509,15 +414,15 @@ following image formats are supported:
@item 4X Movie @tab @tab X
@tab Used in certain computer games.
@item 8088flex TMV @tab @tab X
@item 8SVX exponential @tab @tab X
@item 8SVX fibonacci @tab @tab X
@item A64 multicolor @tab X @tab
@tab Creates video suitable to be played on a commodore 64 (multicolor mode).
@item Amazing Studio PAF Video @tab @tab X
@item American Laser Games MM @tab @tab X
@tab Used in games like Mad Dog McCree.
@item AMV Video @tab X @tab X
@tab Used in Chinese MP3 players.
@item ANSI/ASCII art @tab @tab X
@item Apple Intermediate Codec @tab @tab X
@item Apple MJPEG-B @tab @tab X
@item Apple ProRes @tab X @tab X
@item Apple QuickDraw @tab @tab X
@@ -539,8 +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.
@@ -555,23 +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
@@ -592,21 +491,19 @@ 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 Go2Webinar @tab @tab X
@tab fourcc: G2M4
@item H.261 @tab X @tab X
@item H.263 / H.263-1996 @tab X @tab X
@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 @tab E @tab X
@tab encoding supported through external library libx264
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration) @tab E @tab X
@item HuffYUV @tab X @tab X
@item HuffYUV FFmpeg variant @tab X @tab X
@item IBM Ultimotion @tab @tab X
@@ -637,18 +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.
@@ -659,6 +546,7 @@ following image formats are supported:
@item Motion Pixels video @tab @tab X
@item MPEG-1 video @tab X @tab X
@item MPEG-1/2 video XvMC (X-Video Motion Compensation) @tab @tab X
@item MPEG-1/2 video (VDPAU acceleration) @tab @tab X
@item MPEG-2 video @tab X @tab X
@item MPEG-4 part 2 @tab X @tab X
@tab libxvidcore can be used alternatively for encoding.
@@ -676,10 +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 VP9 @tab E @tab X
@tab encoding supported through external library libvpx
@item Pinnacle TARGA CineWave YUV16 @tab @tab X
@tab fourcc: Y216
@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
@@ -703,11 +589,8 @@ following image formats are supported:
@tab Texture dictionaries used by the Renderware Engine.
@item RL2 video @tab @tab X
@tab used in some games by Entertainment Software Partners
@item SGI RLE 8-bit @tab @tab X
@item Sierra VMD video @tab @tab X
@tab Used in Sierra VMD files.
@item Silicon Graphics Motion Video Compressor 1 (MVC1) @tab @tab X
@item Silicon Graphics Motion Video Compressor 2 (MVC2) @tab @tab X
@item Smacker video @tab @tab X
@tab Video encoding used in Smacker.
@item SMPTE VC-1 @tab @tab X
@@ -722,16 +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
@@ -752,7 +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
@@ -767,8 +646,7 @@ following image formats are supported:
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX exponential @tab @tab X
@item 8SVX fibonacci @tab @tab X
@item 8SVX audio @tab @tab X
@item AAC+ @tab E @tab X
@tab encoding supported through external library libaacplus
@item AAC @tab E @tab X
@@ -799,21 +677,19 @@ following image formats are supported:
@item ADPCM IMA Westwood @tab @tab X
@item ADPCM ISS IMA @tab @tab X
@tab Used in FunCom games.
@item ADPCM IMA Dialogic @tab @tab X
@item ADPCM IMA Duck DK3 @tab @tab X
@tab Used in some Sega Saturn console games.
@item ADPCM IMA Duck DK4 @tab @tab X
@tab Used in some Sega Saturn console games.
@item ADPCM IMA Radical @tab @tab X
@item ADPCM Microsoft @tab X @tab X
@item ADPCM MS IMA @tab X @tab X
@item ADPCM Nintendo Gamecube AFC @tab @tab X
@item ADPCM Nintendo Gamecube DTK @tab @tab X
@item ADPCM Nintendo Gamecube THP @tab @tab X
@item ADPCM QT IMA @tab X @tab X
@item ADPCM SEGA CRI ADX @tab X @tab X
@tab Used in Sega Dreamcast games.
@item ADPCM Shockwave Flash @tab X @tab X
@item ADPCM SMJPEG IMA @tab @tab X
@tab Used in certain Loki game ports.
@item ADPCM Sound Blaster Pro 2-bit @tab @tab X
@item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X
@item ADPCM Sound Blaster Pro 4-bit @tab @tab X
@@ -824,11 +700,10 @@ 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 ATRAC1 @tab @tab X
@item ATRAC3 @tab @tab X
@item Atrac 1 @tab @tab X
@item Atrac 3 @tab @tab X
@item Bink Audio @tab @tab X
@tab Used in Bink and Smacker files in many games.
@item CELT @tab @tab E
@@ -851,7 +726,6 @@ following image formats are supported:
@item DSP Group TrueSpeech @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab X @tab X
@item EVRC (Enhanced Variable Rate Codec) @tab @tab X
@item FLAC (Free Lossless Audio Codec) @tab X @tab IX
@item G.723.1 @tab X @tab X
@item G.729 @tab @tab X
@@ -859,33 +733,24 @@ 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
@item MLP (Meridian Lossless Packing) @tab @tab X
@tab Used in DVD-Audio discs.
@item Monkey's Audio @tab @tab X
@tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1) @tab @tab IX
@item MP2 (MPEG audio layer 2) @tab IX @tab IX
@tab libtwolame can be used alternatively for encoding.
@item MP3 (MPEG audio layer 3) @tab E @tab IX
@tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X
@item Musepack SV7 @tab @tab X
@item Musepack SV8 @tab @tab X
@item Nellymoser Asao @tab X @tab X
@item Opus @tab E @tab E
@tab supported through external library libopus
@item PCM A-law @tab X @tab X
@item PCM mu-law @tab X @tab X
@item PCM signed 8-bit planar @tab X @tab X
@item PCM signed 16-bit big-endian planar @tab X @tab X
@item PCM signed 16-bit little-endian planar @tab X @tab X
@item PCM signed 24-bit little-endian planar @tab X @tab X
@item PCM signed 32-bit little-endian planar @tab X @tab X
@item PCM 16-bit little-endian planar @tab @tab X
@item PCM 32-bit floating point big-endian @tab X @tab X
@item PCM 32-bit floating point little-endian @tab X @tab X
@item PCM 64-bit floating point big-endian @tab X @tab X
@@ -916,35 +781,28 @@ 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
@tab Used in Sierra VMD files.
@item Smacker audio @tab @tab X
@item SMPTE 302M AES3 audio @tab X @tab X
@item SMPTE 302M AES3 audio @tab @tab X
@item Sonic @tab X @tab X
@tab experimental codec
@item Sonic lossless @tab X @tab X
@tab experimental codec
@item Speex @tab E @tab E
@tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor) @tab @tab X
@item True Audio (TTA) @tab X @tab X
@item True Audio (TTA) @tab @tab X
@item TrueHD @tab @tab X
@tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor) @tab @tab X
@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 Voxware MetaSound @tab @tab X
@tab imperfect and incomplete support
@item WavPack @tab X @tab X
@item WavPack @tab @tab X
@item Westwood Audio (SND1) @tab @tab X
@item Windows Media Audio 1 @tab X @tab X
@item Windows Media Audio 2 @tab X @tab X
@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
@@ -960,63 +818,34 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1 .1 .1 .1
@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
@item 3GPP Timed Text @tab @tab @tab X @tab X
@item AQTitle @tab @tab X @tab @tab X
@item DVB @tab X @tab X @tab X @tab X
@item DVB teletext @tab @tab X @tab @tab E
@item DVD @tab X @tab X @tab X @tab X
@item JACOsub @tab X @tab X @tab @tab X
@item MicroDVD @tab X @tab X @tab @tab X
@item MPL2 @tab @tab X @tab @tab X
@item MPsub (MPlayer) @tab @tab X @tab @tab X
@item PGS @tab @tab @tab @tab X
@item PJS (Phoenix) @tab @tab X @tab @tab X
@item RealText @tab @tab X @tab @tab X
@item SAMI @tab @tab X @tab @tab X
@item SSA/ASS @tab X @tab X @tab X @tab X
@item SubRip (SRT) @tab X @tab X @tab X @tab X
@item SubViewer v1 @tab @tab X @tab @tab X
@item SubViewer @tab @tab X @tab @tab X
@item TED Talks captions @tab @tab X @tab @tab X
@item VobSub (IDX+SUB) @tab @tab X @tab @tab X
@item VPlayer @tab @tab X @tab @tab X
@item WebVTT @tab X @tab X @tab @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.
@code{E} means that support is provided through an external library.
@section Network Protocols
@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
@@ -1024,18 +853,13 @@ 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 Video4Linux2 @tab X @tab X
@item Video4Linux @tab X @tab
@item Video4Linux2 @tab X @tab
@item VfW capture @tab X @tab
@item X11 grabbing @tab X @tab
@end multitable
@@ -1046,12 +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

View File

@@ -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

View File

@@ -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
@@ -86,7 +86,7 @@ fail to open.
Set the video size in the captured video.
@item framerate
Set the frame rate in the captured video.
Set the framerate in the captured video.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
@@ -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
@@ -192,66 +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.
@item dvguid
Select the capture device by specifying it's GUID. Capturing will only
be performed from the specified device and fails if no device with the
given GUID is found. This is useful to select the input if multiple
devices are connected at the same time.
Look at /sys/bus/firewire/devices to find out the GUIDs.
@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.
@@ -327,12 +254,6 @@ label, but all the others need to be specified explicitly.
If not specified defaults to the filename specified for the input
device.
@item graph_file
Set the filename of the filtergraph to be read and sent to the other
filters. Syntax of the filtergraph is the same as the one specified by
the option @var{graph}.
@end table
@subsection Examples
@@ -341,14 +262,14 @@ the option @var{graph}.
@item
Create a color video stream and play it back with @command{ffplay}:
@example
ffplay -f lavfi -graph "color=c=pink [out0]" dummy
ffplay -f lavfi -graph "color=pink [out0]" dummy
@end example
@item
As the previous example, but use filename for specifying the graph
description, and omit the "out0" label:
@example
ffplay -f lavfi color=c=pink
ffplay -f lavfi color=pink
@end example
@item
@@ -485,52 +406,87 @@ For more information about OSS see:
@section pulse
PulseAudio input device.
pulseaudio input device.
To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
To enable this input device during configuration you need libpulse-simple
installed in your system.
The filename to provide to the input device is a source device or the
string "default"
To list the PulseAudio source devices and their properties you can invoke
To list the pulse source devices and their properties you can invoke
the command @command{pactl list sources}.
More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
@subsection Options
@table @option
@item server
Connect to a specific PulseAudio server, specified by an IP address.
Default server is used when not provided.
@item name
Specify the application name PulseAudio will use when showing active clients,
by default it is the @code{LIBAVFORMAT_IDENT} string.
@item stream_name
Specify the stream name PulseAudio will use when showing active streams,
by default it is "record".
@item sample_rate
Specify the samplerate in Hz, by default 48kHz is used.
@item channels
Specify the channels in use, by default 2 (stereo) is set.
@item frame_size
Specify the number of bytes per frame, by default it is set to 1024.
@item fragment_size
Specify the minimal buffering fragment in PulseAudio, it will affect the
audio latency. By default it is unset.
@end table
@subsection Examples
Record a stream from default device:
@example
ffmpeg -f pulse -i default /tmp/pulse.wav
@end example
@subsection @var{server} AVOption
The syntax is:
@example
-server @var{server name}
@end example
Connects to a specific server.
@subsection @var{name} AVOption
The syntax is:
@example
-name @var{application name}
@end example
Specify the application name pulse will use when showing active clients,
by default it is the LIBAVFORMAT_IDENT string
@subsection @var{stream_name} AVOption
The syntax is:
@example
-stream_name @var{stream name}
@end example
Specify the stream name pulse will use when showing active streams,
by default it is "record"
@subsection @var{sample_rate} AVOption
The syntax is:
@example
-sample_rate @var{samplerate}
@end example
Specify the samplerate in Hz, by default 48kHz is used.
@subsection @var{channels} AVOption
The syntax is:
@example
-channels @var{N}
@end example
Specify the channels in use, by default 2 (stereo) is set.
@subsection @var{frame_size} AVOption
The syntax is:
@example
-frame_size @var{bytes}
@end example
Specify the number of byte per frame, by default it is set to 1024.
@subsection @var{fragment_size} AVOption
The syntax is:
@example
-fragment_size @var{bytes}
@end example
Specify the minimal buffering fragment in pulseaudio, it will affect the
audio latency. By default it is unset.
@section sndio
sndio input device.
@@ -548,15 +504,9 @@ command:
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example
@section video4linux2, v4l2
@section video4linux and video4linux2
Video4Linux2 input video device.
"v4l2" can be used as alias for "video4linux2".
If FFmpeg is built with v4l-utils support (by using the
@code{--enable-libv4l2} configure option), it is possible to use it with the
@code{-use_libv4l2} input device option.
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
@@ -564,107 +514,40 @@ 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
@var{width}x@var{height} sizes and frame rates. You can check which are
supported using @command{-list_formats all} for Video4Linux2 devices.
Some devices, like TV cards, support one or more standards. It is possible
to list all the supported standards using @command{-list_standards all}.
Video4Linux and Video4Linux2 devices only support a limited set of
@var{width}x@var{height} sizes and framerates. You can check which are
supported for example with the command @command{dov4l} for Video4Linux
devices and using @command{-list_formats all} for Video4Linux2 devices.
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.
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.
Some usage examples of the video4linux2 device with @command{ffmpeg}
and @command{ffplay}:
@itemize
@item
Grab and show the input of a video4linux2 device:
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.
Follow some usage examples of the video4linux devices with the ff*
tools.
@example
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
# 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
@item
Grab and record the input of a video4linux2 device, leave the
frame rate and size as previously set:
@example
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@end example
@end itemize
For more information about Video4Linux, check @url{http://linuxtv.org/}.
@subsection Options
@table @option
@item standard
Set the standard. Must be the name of a supported standard. To get a
list of the supported standards, use the @option{list_standards}
option.
@item channel
Set the input channel number. Default to -1, which means using the
previously selected channel.
@item video_size
Set the video frame size. The argument must be a string in the form
@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation.
@item pixel_format
Select the pixel format (only valid for raw video input).
@item input_format
Set the preferred pixel format (for raw video) or a codec name.
This option allows to select the input format, when several are
available.
@item framerate
Set the preferred video frame rate.
@item list_formats
List available formats (supported pixel formats, codecs, and frame
sizes) and exit.
Available values are:
@table @samp
@item all
Show all available (compressed and non-compressed) formats.
@item raw
Show only raw video (non-compressed) formats.
@item compressed
Show only compressed formats.
@end table
@item list_standards
List supported standards and exit.
Available values are:
@table @samp
@item all
Show all supported standards.
@end table
@item timestamps, ts
Set type of timestamps for grabbed frames.
Available values are:
@table @samp
@item default
Use timestamps from the kernel.
@item abs
Use absolute timestamps (wall clock).
@item mono2abs
Force conversion from monotonic to absolute timestamps.
@end table
Default value is @code{default}.
@end table
"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
"video4linux2".
@section vfwcap
@@ -701,25 +584,19 @@ 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 -framerate 25 -video_size cif -i :0.0 out.mpg
ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
# Grab at position 10,20.
ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
@end example
Grab at position @code{10,20}:
@subsection @var{follow_mouse} AVOption
The syntax is:
@example
ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
-follow_mouse centered|@var{PIXELS}
@end example
@subsection Options
@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}.
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
follows only when the mouse pointer reaches within @var{PIXELS} (greater than
@@ -727,37 +604,29 @@ zero) to the edge of region.
For example:
@example
ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
# 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
To follow only when the mouse pointer reaches within 100 pixels to edge:
@subsection @var{show_region} AVOption
The syntax is:
@example
ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
-show_region 1
@end example
@item framerate
Set the grabbing frame rate. Default value is @code{ntsc},
corresponding to a frame rate of @code{30000/1001}.
@item show_region
Show grabbed region on screen.
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 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@end example
ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
With @var{follow_mouse}:
@example
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size 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

View File

@@ -1,4 +1,4 @@
FFmpeg's bug/feature request tracker manual
FFmpeg's bug/patch/feature request tracker manual
=================================================
NOTE: This is a draft.
@@ -11,7 +11,7 @@ existing issues can be done through a web interface.
Issues can be different kinds of things we want to keep track of
but that do not belong into the source tree itself. This includes
bug reports, feature requests and license violations. We
bug reports, patches, feature requests and license violations. We
might add more items to this list in the future, so feel free to
propose a new `type of issue' on the ffmpeg-devel mailing list if
you feel it is worth tracking.
@@ -24,13 +24,10 @@ 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)://trac.ffmpeg.org
http(s)://ffmpeg.org/trac/ffmpeg
Type:
-----
art
Artwork such as photos, music, banners, and logos.
bug / defect
An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
prevents it from behaving as intended.
@@ -44,18 +41,20 @@ feature request / enhancement
license violation
ticket to keep track of (L)GPL violations of ffmpeg by others
sponsoring request
Developer requests for hardware, software, specifications, money,
refunds, etc.
patch
A patch as generated by diff which conforms to the patch submission and
development policy.
Priority:
---------
critical
Bugs about data loss and security issues.
Bugs and patches which deal with data loss and security issues.
No feature request can be critical.
important
Bugs which make FFmpeg unusable for a significant number of users.
Bugs which make FFmpeg unusable for a significant number of users, and
patches fixing them.
Examples here might be completely broken MPEG-4 decoding or a build issue
on Linux.
While broken 4xm decoding or a broken OS/2 build would not be important,
@@ -69,7 +68,7 @@ normal
minor
Bugs about things like spelling errors, "mp2" instead of
Bugs and patches about things like spelling errors, "mp2" instead of
"mp3" being shown and such.
Feature requests about things few people want or which do not make a big
difference.
@@ -104,13 +103,13 @@ This state implicates that the bug either has been reproduced or that
reproduction is not needed as the bug is already understood.
Type/Status:
Type/Status/Substatus:
----------
*/new
Initial state of new bugs and feature requests submitted by
*/new/new
Initial state of new bugs, patches and feature requests submitted by
users.
*/open
*/open/open
Issues which have been briefly looked at and which did not look outright
invalid.
This implicates that no real more detailed state applies yet. Conversely,
@@ -118,7 +117,9 @@ Type/Status:
looked at.
*/closed/duplicate
Bugs or feature requests which are duplicates.
Bugs, patches or feature requests which are duplicates.
Note that patches dealing with the same thing in a different way are not
duplicates.
Note, if you mark something as duplicate, do not forget setting the
superseder so bug reports are properly linked.
@@ -133,7 +134,7 @@ Type/Status:
bug/closed/fixed
Bugs which have to the best of our knowledge been fixed.
bug/closed/wontfix
bug/closed/wont_fix
Bugs which we will not fix. Possible reasons include legality, high
complexity for the sake of supporting obscure corner cases, speed loss
for similarly esoteric purposes, et cetera.
@@ -147,15 +148,33 @@ bug/closed/works_for_me
reproduction failed - that is the code seems to work correctly to the
best of our knowledge.
feature_request/closed/fixed
patch/open/approved
Patches which have been reviewed and approved by a developer.
Such patches can be applied anytime by any other developer after some
reasonable testing (compile + regression tests + does the patch do
what the author claimed).
patch/open/needs_changes
Patches which have been reviewed and need changes to be accepted.
patch/closed/applied
Patches which have been applied.
patch/closed/rejected
Patches which have been rejected.
feature_request/closed/implemented
Feature requests which have been implemented.
feature_request/closed/wontfix
feature_request/closed/wont_implement
Feature requests which will not be implemented. The reasons here could
be legal, philosophical or others.
Note, please do not use type-status-substatus combinations other than the
above without asking on ffmpeg-dev first!
Note2, if you provide the requested info do not forget to remove the
needs_more_info resolution.
needs_more_info substatus.
Component:
----------

View File

@@ -1,48 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libavcodec Documentation
@titlepage
@center @titlefont{Libavcodec Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavcodec library provides a generic encoding/decoding framework
and contains multiple decoders and encoders for audio, video and
subtitle streams, and several bitstream filters.
The shared architecture provides various services ranging from bit
stream I/O to DSP optimizations, and makes it suitable for
implementing robust and fast codecs as well as for experimentation.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavcodec
@settitle media streams decoding and encoding library
@end ignore
@bye

View File

@@ -1,45 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libavdevice Documentation
@titlepage
@center @titlefont{Libavdevice Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavdevice library provides a generic framework for grabbing from
and rendering to many common multimedia input/output devices, and
supports several input and output devices, including Video4Linux2,
VfW, DShow, and ALSA.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-devices(1),
libavutil(3), libavcodec(3), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavdevice
@settitle multimedia device handling library
@end ignore
@bye

View File

@@ -9,36 +9,84 @@
@contents
@chapter Description
@c man begin DESCRIPTION
@chapter Introduction
The libavfilter library provides a generic audio/video filtering
framework containing several filters, sources and sinks.
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.
@c man end DESCRIPTION
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 See Also
@chapter Tutorial
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-filters.html,ffmpeg-filters},
@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
@end ifhtml
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:
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-filters(1),
libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
@end ifnothtml
@example
input --> split --> fifo -----------------------> overlay --> output
| ^
| |
+------> fifo --> crop --> vflip --------+
@end example
@include authors.texi
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:
@ignore
@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
@setfilename libavfilter
@settitle multimedia filtering library
The result will be that in output the top half of the video is mirrored
onto the bottom half.
@end ignore
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

View File

@@ -1,48 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libavformat Documentation
@titlepage
@center @titlefont{Libavformat Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavformat library provides a generic framework for multiplexing
and demultiplexing (muxing and demuxing) audio, video and subtitle
streams. It encompasses multiple muxers and demuxers for multimedia
container formats.
It also supports several input and output protocols to access a media
resource.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-formats(1), ffmpeg-protocols(1),
libavutil(3), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavformat
@settitle multimedia muxing and demuxing library
@end ignore
@bye

View File

@@ -1,44 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libavutil Documentation
@titlepage
@center @titlefont{Libavutil Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavutil library is a utility library to aid portable
multimedia programming. It contains safe portable string functions,
random number generators, data structures, additional mathematics
functions, cryptography and multimedia related functionality (like
enumerations for pixel and sample formats).
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavutil
@settitle multimedia-biased utility library
@end ignore
@bye

View File

@@ -1,70 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libswresample Documentation
@titlepage
@center @titlefont{Libswresample Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libswresample library performs highly optimized audio resampling,
rematrixing and sample format conversion operations.
Specifically, this library performs the following conversions:
@itemize
@item
@emph{Resampling}: is the process of changing the audio rate, for
example from a high sample rate of 44100Hz to 8000Hz. Audio
conversion from high to low sample rate is a lossy process. Several
resampling options and algorithms are available.
@item
@emph{Format conversion}: is the process of converting the type of
samples, for example from 16-bit signed samples to unsigned 8-bit or
float samples. It also handles packing conversion, when passing from
packed layout (all samples belonging to distinct channels interleaved
in the same buffer), to planar layout (all samples belonging to the
same channel stored in a dedicated buffer or "plane").
@item
@emph{Rematrixing}: is the process of changing the channel layout, for
example from stereo to mono. When the input channels cannot be mapped
to the output streams, the process is lossy, since it involves
different gain factors and mixing.
@end itemize
Various other audio conversions (e.g. stretching and padding) are
enabled through dedicated options.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-resampler(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libswresample
@settitle audio resampling library
@end ignore
@bye

View File

@@ -1,63 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libswscale Documentation
@titlepage
@center @titlefont{Libswscale Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libswscale library performs highly optimized image scaling and
colorspace and pixel format conversion operations.
Specifically, this library performs the following conversions:
@itemize
@item
@emph{Rescaling}: is the process of changing the video size. Several
rescaling options and algorithms are available. This is usually a
lossy process.
@item
@emph{Pixel format conversion}: is the process of converting the image
format and colorspace of the image, for example from planar YUV420P to
RGB24 packed. It also handles packing conversion, that is converts
from packed layout (all pixels belonging to distinct planes
interleaved in the same buffer), to planar layout (all samples
belonging to the same plane stored in a dedicated buffer or "plane").
This is usually a lossy process in case the source and destination
colorspaces differ.
@end itemize
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-scaler(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libswscale
@settitle video scaling and pixel format conversion library
@end ignore
@bye

View File

@@ -65,20 +65,4 @@ title=chapter \#1
title=multi\
line
@end example
By using the ffmetadata muxer and demuxer it is possible to extract
metadata from an input file to an ffmetadata file, and then transcode
the file into an output file with the edited ffmetadata file.
Extracting an ffmetadata file with @file{ffmpeg} goes as follows:
@example
ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
@end example
Reinserting edited metadata information from the FFMETADATAFILE file can
be done as:
@example
ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
@end example
@c man end METADATA

View File

@@ -1,75 +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/
float_dsp_mips.c
libm_mips.h
* libavcodec/
fft_fixed_32.c
fft_init_table.c
fft_table.h
mdct_fixed_32.c
* libavcodec/mips/
aaccoder_mips.c
aacpsy_mips.h
ac3dsp_mips.c
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
iirfilter_mips.c
mpegaudiodsp_mips_fixed.c
mpegaudiodsp_mips_float.c

View File

@@ -57,11 +57,6 @@ which re-allocates them for other threads.
Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
speed gain at this point but it should work.
If there are inter-frame dependencies, so the codec calls
ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
frames must then be freed with ff_thread_release_buffer().
Otherwise leave it at zero and decode directly into the user-supplied frames.
Call ff_thread_report_progress() after some part of the current picture has decoded.
A good place to put this is where draw_horiz_band() is called - add this if it isn't
called anywhere, as it's useful too and the implementation is trivial when you're

View File

@@ -18,23 +18,6 @@ enabled muxers.
A description of some of the currently available muxers follows.
@anchor{aiff}
@section aiff
Audio Interchange File Format muxer.
It accepts the following options:
@table @option
@item write_id3v2
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
@item id3v2_version
Select ID3v2 version to write. Currently only version 3 and 4 (aka.
ID3v2.3 and ID3v2.4) are supported. The default is version 4.
@end table
@anchor{crc}
@section crc
@@ -73,37 +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:
@@ -113,98 +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{hls}
@section hls
Apple HTTP Live Streaming muxer that segments MPEG-TS according to
the HTTP Live Streaming specification.
It creates a playlist file and numbered segment files. The output
filename specifies the playlist filename; the segment filenames
receive the same basename as the playlist, a sequential number and
a .ts extension.
@example
ffmpeg -i in.nut out.m3u8
@end example
@table @option
@item -hls_time @var{seconds}
Set the segment length in seconds.
@item -hls_list_size @var{size}
Set the maximum number of playlist entries.
@item -hls_wrap @var{wrap}
Set the number after which index wraps.
@item -start_number @var{number}
Start the sequence from @var{number}.
@end table
@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
@@ -257,24 +142,87 @@ Note also that the pattern must not necessarily contain "%d" or
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example
@table @option
@item start_number @var{number}
Start the sequence from @var{number}. Default value is 1. Must be a
non-negative number.
@item -update @var{number}
If @var{number} is nonzero, the filename will always be interpreted as just a
filename, not a pattern, and this file will be continuously overwritten with new
images.
@end table
The image muxer supports the .Y.U.V image file format. This format is
special in that that each image frame consists of three files, for
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.
@section mov
MOV / MP4 muxer
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.
@end table
@section mpegts
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
@table @option
@item -mpegts_original_network_id @var{number}
Set the original_network_id (default 0x0001). This is unique identifier
of a network in DVB. Its main use is in the unique identification of a
service through the path Original_Network_ID, Transport_Stream_ID.
@item -mpegts_transport_stream_id @var{number}
Set the transport_stream_id (default 0x0001). This identifies a
transponder in DVB.
@item -mpegts_service_id @var{number}
Set the service_id (default 0x0001) also known as program in DVB.
@item -mpegts_pmt_start_pid @var{number}
Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
Set the first PID for data packets (default 0x0100, max 0x0f00).
@end table
The recognized metadata settings in mpegts muxer are @code{service_provider}
and @code{service_name}. If they are not set the default for
@code{service_provider} is "FFmpeg" and the default for
@code{service_name} is "Service01".
@example
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
@end example
@section null
Null muxer.
This muxer does not generate any output file, it is mainly useful for
testing or benchmarking purposes.
For example to benchmark decoding with @command{ffmpeg} you can use the
command:
@example
ffmpeg -benchmark -i INPUT -f null out.null
@end example
Note that the above command does not read or write the @file{out.null}
file, but specifying the output file is required by the @command{ffmpeg}
syntax.
Alternatively you can write the command as:
@example
ffmpeg -benchmark -i INPUT -f null -
@end example
@section matroska
Matroska container muxer.
@@ -338,281 +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
This muxer supports the following options:
@table @option
@item reserve_index_space
By default, this muxer writes the index for seeking (called cues in Matroska
terms) at the end of the file, because it cannot know in advance how much space
to leave for the index at the beginning of the file. However for some use cases
-- e.g. streaming where seeking is possible but slow -- it is useful to put the
index at the beginning of the file.
If this option is set to a non-zero value, the muxer will reserve a given amount
of space in the file header and then try to write the cues there when the muxing
finishes. If the available space does not suffice, muxing will fail. A safe size
for most use cases should be about 50kB per hour of video.
Note that cues are only written if the output is seekable and this option will
have no effect if it is not.
@end table
@anchor{md5}
@section md5
MD5 testing format.
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 by adding @var{faststart} to the @var{movflags}, or
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:
@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.
@item -movflags faststart
Run a second pass moving the index (moov atom) to the beginning of the file.
This operation can take a while, and will not work in various situations such
as fragmented output, thus it is not enabled by default.
@item -movflags rtphint
Add RTP hinting tracks to the output file.
@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 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
@section mpegts
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
@table @option
@item -mpegts_original_network_id @var{number}
Set the original_network_id (default 0x0001). This is unique identifier
of a network in DVB. Its main use is in the unique identification of a
service through the path Original_Network_ID, Transport_Stream_ID.
@item -mpegts_transport_stream_id @var{number}
Set the transport_stream_id (default 0x0001). This identifies a
transponder in DVB.
@item -mpegts_service_id @var{number}
Set the service_id (default 0x0001) also known as program in DVB.
@item -mpegts_pmt_start_pid @var{number}
Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
Set the first PID for data packets (default 0x0100, max 0x0f00).
@item -mpegts_m2ts_mode @var{number}
Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
@item -muxrate @var{number}
Set muxrate.
@item -pes_payload_size @var{number}
Set minimum PES packet payload in bytes.
@item -mpegts_flags @var{flags}
Set flags (see below).
@item -mpegts_copyts @var{number}
Preserve original timestamps, if value is set to 1. Default value is -1, which
results in shifting timestamps so that they start from 0.
@item -tables_version @var{number}
Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
This option allows updating stream structure so that standard consumer may
detect the change. To do so, reopen output AVFormatContext (in case of API
usage) or restart ffmpeg instance, cyclically changing tables_version value:
@example
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...
ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...
@end example
@end table
Option mpegts_flags may take a set of such flags:
@table @option
@item resend_headers
Reemit PAT/PMT before writing the next packet.
@item latm
Use LATM packetization for AAC.
@end table
The recognized metadata settings in mpegts muxer are @code{service_provider}
and @code{service_name}. If they are not set the default for
@code{service_provider} is "FFmpeg" and the default for
@code{service_name} is "Service01".
@example
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
@end example
@section null
Null muxer.
This muxer does not generate any output file, it is mainly useful for
testing or benchmarking purposes.
For example to benchmark decoding with @command{ffmpeg} you can use the
command:
@example
ffmpeg -benchmark -i INPUT -f null out.null
@end example
Note that the above command does not read or write the @file{out.null}
file, but specifying the output file is required by the @command{ffmpeg}
syntax.
Alternatively you can write the command as:
@example
ffmpeg -benchmark -i INPUT -f null -
@end example
@section ogg
Ogg container muxer.
@table @option
@item -page_duration @var{duration}
Preferred page duration, in microseconds. The muxer will attempt to create
pages that are approximately @var{duration} microseconds long. This allows the
user to compromise between seek granularity and container overhead. The default
is 1 second. A value of 0 will fill all segments, making pages as large as
possible. A value of 1 will effectively use 1 packet-per-page in most
situations, giving a small seek granularity at the cost of additional container
overhead.
@end table
@section segment, stream_segment, ssegment
@section segment
Basic stream segmenter.
@@ -620,289 +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 keyframe of the selected reference stream,
which is set through the @option{reference_stream} option.
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.
Every segment starts with a video keyframe, if a video stream is present.
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 reference_stream @var{specifier}
Set the reference stream, as specified by the string @var{specifier}.
If @var{specifier} is set to @code{auto}, the reference is choosen
automatically. Otherwise it must be a stream specifier (see the ``Stream
specifiers'' chapter in the ffmpeg manual) which specifies the
reference stream. The default value is @code{auto}.
@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 @samp
@item cache
Allow caching (only affects M3U8 list files).
@item live
Allow live-friendly file generation.
@end table
Default value is @code{samp}.
Generate also a listfile named @var{name}.
@item segment_list_size @var{size}
Update the list file so that it contains at most the last @var{size}
segments. If 0 the list file will contain all the segments. Default
value is 0.
@item segment_list_type @var{type}
Specify the format for the segment list file.
The following values are recognized:
@table @samp
@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.
@samp{ext} is deprecated in favor or @samp{csv}.
@item ffconcat
Generate an ffconcat file for the created segments. The resulting file
can be read using the FFmpeg @ref{concat} demuxer.
A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will
auto-select this format.
@item m3u8
Generate an extended M3U8 file, version 3, compliant with
@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}.
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}, the value must be a duration
specification. Default value is "2". See also the
@option{segment_times} option.
Note that splitting may not be accurate, unless you force the
reference stream key-frames at the given time. See the introductory
notice and the examples below.
@item segment_time_delta @var{delta}
Specify the accuracy time when selecting the start time for a
segment, expressed as a duration specification. Default value is "0".
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. See also
the @option{segment_time} option.
@item segment_frames @var{frames}
Specify a list of split video frame numbers. @var{frames} contains a
list of comma separated integer numbers, in increasing order.
This option specifies to start a new segment whenever a reference
stream key frame is found and the sequential number (starting from 0)
of the frame is greater or equal to the next value in the list.
@item segment_wrap @var{limit}
Wrap around segment index once it reaches @var{limit}.
@item segment_start_number @var{number}
Set the sequence number of the first segment. Defaults to @code{0}.
@item reset_timestamps @var{1|0}
Reset timestamps at the begin of each segment, so that each segment
will start with near-zero timestamps. It is meant to ease the playback
of the generated segments. May not work with some combinations of
muxers/codecs. It is set to @code{0} by default.
@item initial_offset @var{offset}
Specify timestamp offset to apply to the output packet timestamps. The
argument must be a time duration specification, and defaults to 0.
@end table
@subsection Examples
@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 @command{ffmpeg} @option{force_key_frames}
option to force key frames in the input at the specified location, together
with the segment option @option{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 -codec:v mpeg4 -codec:a 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
Segment the input file by splitting the input file according to the
frame numbers sequence specified with the @option{segment_frames} option:
@example
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
@end example
@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 tee
The tee muxer can be used to write the same data to several files or any
other kind of muxer. It can be used, for example, to both stream a video to
the network and save it to disk at the same time.
It is different from specifying several outputs to the @command{ffmpeg}
command-line tool because the audio and video data will be encoded only once
with the tee muxer; encoding can be a very expensive process. It is not
useful when using the libavformat API directly because it is then possible
to feed the same packets to several muxers directly.
The slave outputs are specified in the file name given to the muxer,
separated by '|'. If any of the slave name contains the '|' separator,
leading or trailing spaces or any special character, it must be
escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
manual).
Muxer options can be specified for each slave by prepending them as a list of
@var{key}=@var{value} pairs separated by ':', between square brackets. If
the options values contain a special character or the ':' separator, they
must be escaped; note that this is a second level escaping.
The following special options are also recognized:
@table @option
@item f
Specify the format name. Useful if it cannot be guessed from the
output name suffix.
@item bsfs[/@var{spec}]
Specify a list of bitstream filters to apply to the specified
output. It is possible to specify to which streams a given bitstream
filter applies, by appending a stream specifier to the option
separated by @code{/}. If the stream specifier is not specified, the
bistream filters will be applied to all streams in the output.
Several bitstream filters can be specified, separated by ",".
@item select
Select the streams that should be mapped to the slave output,
specified by a stream specifier. If not specified, this defaults to
all the input streams.
@end table
Some examples follow.
@itemize
@item
Encode something and both archive it in a WebM file and stream it
as MPEG-TS over UDP (the streams need to be explicitly mapped):
@example
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@end example
@item
Use @command{ffmpeg} to encode the input, and send the output
to three different destinations. The @code{dump_extra} bitstream
filter is used to add extradata information to all the output video
keyframes packets, as requested by the MPEG-TS format. The select
option is applied to @file{out.aac} in order to make it contain only
audio packets.
@example
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
@end example
@end itemize
Note: some codecs may need different options depending on the output format;
the auto-detection of this can not work with the tee muxer. The main example
is the @option{global_header} flag.
@c man end MUXERS

View File

@@ -1,138 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle NUT
@titlepage
@center @titlefont{NUT}
@end titlepage
@top
@contents
@chapter Description
NUT is a low overhead generic container format. It stores audio, video,
subtitle and user-defined streams in a simple, yet efficient, way.
It was created by a group of FFmpeg and MPlayer developers in 2003
and was finalized in 2008.
The official nut specification is at svn://svn.mplayerhq.hu/nut
In case of any differences between this text and the official specification,
the official specification shall prevail.
@chapter Container-specific codec tags
@section Generic raw YUVA formats
Since many exotic planar YUVA pixel formats are not considered by
the AVI/QuickTime FourCC lists, the following scheme is adopted for
representing them.
The first two bytes can contain the values:
Y1 = only Y
Y2 = Y+A
Y3 = YUV
Y4 = YUVA
The third byte represents the width and height chroma subsampling
values for the UV planes, that is the amount to shift the luma
width/height right to find the chroma width/height.
The fourth byte is the number of bits used (8, 16, ...).
If the order of bytes is inverted, that means that each component has
to be read big-endian.
@section Raw Audio
@multitable @columnfractions .4 .4
@item ALAW @tab A-LAW
@item ULAW @tab MU-LAW
@item P<type><interleaving><bits> @tab little-endian PCM
@item <bits><interleaving><type>P @tab big-endian PCM
@end multitable
<type> is S for signed integer, U for unsigned integer, F for IEEE float
<interleaving> is D for default, P is for planar.
<bits> is 8/16/24/32
@example
PFD[32] would for example be signed 32 bit little-endian IEEE float
@end example
@section Subtitles
@multitable @columnfractions .4 .4
@item UTF8 @tab Raw UTF-8
@item SSA[0] @tab SubStation Alpha
@item DVDS @tab DVD subtitles
@item DVBS @tab DVB subtitles
@end multitable
@section Raw Data
@multitable @columnfractions .4 .4
@item UTF8 @tab Raw UTF-8
@end multitable
@section Codecs
@multitable @columnfractions .4 .4
@item 3IV1 @tab non-compliant MPEG-4 generated by old 3ivx
@item ASV1 @tab Asus Video
@item ASV2 @tab Asus Video 2
@item CVID @tab Cinepak
@item CYUV @tab Creative YUV
@item DIVX @tab non-compliant MPEG-4 generated by old DivX
@item DUCK @tab Truemotion 1
@item FFV1 @tab FFmpeg video 1
@item FFVH @tab FFmpeg Huffyuv
@item H261 @tab ITU H.261
@item H262 @tab ITU H.262
@item H263 @tab ITU H.263
@item H264 @tab ITU H.264
@item HFYU @tab Huffyuv
@item I263 @tab Intel H.263
@item IV31 @tab Indeo 3.1
@item IV32 @tab Indeo 3.2
@item IV50 @tab Indeo 5.0
@item LJPG @tab ITU JPEG (lossless)
@item MJLS @tab ITU JPEG-LS
@item MJPG @tab ITU JPEG
@item MPG4 @tab MS MPEG-4v1 (not ISO MPEG-4)
@item MP42 @tab MS MPEG-4v2
@item MP43 @tab MS MPEG-4v3
@item MP4V @tab ISO MPEG-4 Part 2 Video (from old encoders)
@item mpg1 @tab ISO MPEG-1 Video
@item mpg2 @tab ISO MPEG-2 Video
@item MRLE @tab MS RLE
@item MSVC @tab MS Video 1
@item RT21 @tab Indeo 2.1
@item RV10 @tab RealVideo 1.0
@item RV20 @tab RealVideo 2.0
@item RV30 @tab RealVideo 3.0
@item RV40 @tab RealVideo 4.0
@item SNOW @tab FFmpeg Snow
@item SVQ1 @tab Sorenson Video 1
@item SVQ3 @tab Sorenson Video 3
@item theo @tab Xiph Theora
@item TM20 @tab Truemotion 2.0
@item UMP4 @tab non-compliant MPEG-4 generated by UB Video MPEG-4
@item VCR1 @tab ATI VCR1
@item VP30 @tab VP 3.0
@item VP31 @tab VP 3.1
@item VP50 @tab VP 5.0
@item VP60 @tab VP 6.0
@item VP61 @tab VP 6.1
@item VP62 @tab VP 6.2
@item VP70 @tab VP 7.0
@item WMV1 @tab MS WMV7
@item WMV2 @tab MS WMV8
@item WMV3 @tab MS WMV9
@item WV1F @tab non-compliant MPEG-4 generated by ?
@item WVC1 @tab VC-1
@item XVID @tab non-compliant MPEG-4 generated by old Xvid
@item XVIX @tab non-compliant MPEG-4 generated by old Xvid with interlacing bug
@end multitable

View File

@@ -148,7 +148,7 @@ Alignment:
Some instructions on some architectures have strict alignment restrictions,
for example most SSE/SSE2 instructions on x86.
The minimum guaranteed alignment is written in the .h files, for example:
void (*put_pixels_clamped)(const int16_t *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
General Tips:
@@ -253,7 +253,7 @@ Optimization guide for ARM11 (used in Nokia N800 Internet Tablet):
http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf
Optimization guide for Intel XScale (used in Sharp Zaurus PDA):
http://download.intel.com/design/intelxscale/27347302.pdf
Intel Wireless MMX 2 Coprocessor: Programmers Reference Manual
Intel Wireless MMX2 Coprocessor: Programmers Reference Manual
http://download.intel.com/design/intelxscale/31451001.pdf
PowerPC-specific:

View File

@@ -1,7 +1,7 @@
@chapter Output Devices
@c man begin OUTPUT DEVICES
Output devices are configured elements in FFmpeg that can write
Output devices are configured elements in FFmpeg which allow to write
multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices
@@ -22,161 +22,15 @@ A description of the currently available output devices follows.
ALSA (Advanced Linux Sound Architecture) output device.
@section caca
CACA output device.
This output device 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 fbdev
Linux framebuffer output device.
The Linux framebuffer is a graphic hardware-independent abstraction
layer to show graphics on a computer monitor, typically on the
console. It is accessed through a file device node, usually
@file{/dev/fb0}.
For more detailed information read the file
@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
@subsection Options
@table @option
@item xoffset
@item yoffset
Set x/y coordinate of top left corner. Default is 0.
@end table
@subsection Examples
Play a file on framebuffer device @file{/dev/fb0}.
Required pixel format depends on current framebuffer settings.
@example
ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
@end example
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section oss
OSS (Open Sound System) output device.
@section pulse
PulseAudio output device.
To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
@subsection Options
@table @option
@item server
Connect to a specific PulseAudio server, specified by an IP address.
Default server is used when not provided.
@item name
Specify the application name PulseAudio will use when showing active clients,
by default it is the @code{LIBAVFORMAT_IDENT} string.
@item stream_name
Specify the stream name PulseAudio will use when showing active streams,
by default it is set to the specified output name.
@item device
Specify the device to use. Default device is used when not provided.
List of output devices can be obtained with command @command{pactl list sinks}.
@end table
@subsection Examples
Play a file on default device on default server:
@example
ffmpeg -i INPUT -f pulse "stream name"
@end example
@section sdl
SDL (Simple DirectMedia Layer) output device.
This output device allows to show a video stream in an SDL
This output devices allows to show a video stream in an SDL
window. Only one SDL window is allowed per application, so you can
have only one instance of this output device in an application.
@@ -201,12 +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.
@item window_fullscreen
Set fullscreen mode when non-zero value is provided.
Zero is a default.
If not specified it defaults to the size of the input video.
@end table
@subsection Examples
@@ -221,69 +70,4 @@ ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL
sndio audio output device.
@section xv
XV (XVideo) output device.
This output device allows to show a video stream in a X Window System
window.
@subsection Options
@table @option
@item display_name
Specify the hardware display name, which determines the display and
communications domain to be used.
The display name or DISPLAY environment variable can be a string in
the format @var{hostname}[:@var{number}[.@var{screen_number}]].
@var{hostname} specifies the name of the host machine on which the
display is physically attached. @var{number} specifies the number of
the display server on that host machine. @var{screen_number} specifies
the screen to be used on that server.
If unspecified, it defaults to the value of the DISPLAY environment
variable.
For example, @code{dual-headed:0.1} would specify screen 1 of display
0 on the machine named ``dual-headed''.
Check the X11 specification for more detailed information about the
display name format.
@item window_size
Set the created 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 window_x
@item window_y
Set the X and Y window offsets for the created window. They are both
set to 0 by default. The values may be ignored by the window manager.
@item window_title
Set the window title, if not specified default to the filename
specified for the output device.
@end table
For more information about XVideo see @url{http://www.x.org/}.
@subsection Examples
@itemize
@item
Decode, display and encode video input with @command{ffmpeg} at the
same time:
@example
ffmpeg -i INPUT OUTPUT -f xv display
@end example
@item
Decode and display the input video to multiple X11 windows:
@example
ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
@end example
@end itemize
@c man end OUTPUT DEVICES

View File

@@ -1,8 +1,8 @@
\input texinfo @c -*- texinfo -*-
@settitle Platform Specific Information
@settitle Platform Specific information
@titlepage
@center @titlefont{Platform Specific Information}
@center @titlefont{Platform Specific information}
@end titlepage
@top
@@ -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
@@ -77,15 +77,30 @@ For information about compiling FFmpeg on OS/2 see
@chapter Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@section Native Windows compilation using MinGW or MinGW-w64
@section Native Windows compilation
FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}.
You can find detailed installation instructions in the download section and
the FAQ.
FFmpeg can be built to run natively on Windows using the MinGW tools. Install
the latest versions of MSYS and MinGW from @url{http://www.mingw.org/}.
You can find detailed installation instructions in the download
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 version for some packages
are listed below:
@itemize
@item bash 3.1
@item msys-make 3.81-2 (note: not mingw32-make)
@item w32api 3.13
@item mingw-runtime 3.15
@end itemize
FFmpeg automatically passes @code{-fno-common} to the compiler to work around
a GCC bug (see @url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=37216}).
Notes:
@@ -94,11 +109,14 @@ 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,
@@ -106,106 +124,149 @@ libavformat) as DLLs.
@end itemize
@section Microsoft Visual C++ or Intel C++ Compiler for Windows
@section Microsoft Visual C++ compatibility
FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
wrapper. For ICL, only the wrapper is used, since ICL supports C99.
As stated in the FAQ, FFmpeg will not compile under MSVC++. However, if you
want to use the libav* libraries in your own applications, you can still
compile those applications using MSVC++. But the libav* libraries you link
to @emph{must} be built with MinGW. However, you will not be able to debug
inside the libav* libraries, since MSVC++ does not recognize the debug
symbols generated by GCC.
We strongly recommend you to move over from MSVC++ to MinGW tools.
You will need the following prerequisites:
This description of how to use the FFmpeg libraries with MSVC++ is based on
Microsoft Visual C++ 2005 Express Edition. If you have a different version,
you might have to modify the procedures slightly.
@itemize
@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
you want to run @uref{fate.html, FATE}.
@end itemize
@subsection Using static libraries
To set up a proper environment in MSYS, you need to run @code{msys.bat} from
the Visual Studio or Intel Compiler command prompt.
Assuming you have just built and installed FFmpeg in @file{/usr/local}.
Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
somewhere in your @code{PATH}.
@enumerate
Next, make sure @code{inttypes.h} and any other headers and libs you want to use
are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
these directories. Alternatively, you can try and use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.
@item Create a new console application ("File / New / Project") and then
select "Win32 Console Application". On the appropriate page of the
Application Wizard, uncheck the "Precompiled headers" option.
Finally, run:
@item Write the source code for your application, or, for testing, just
copy the code from an existing sample application into the source file
that MSVC++ has already created for you. For example, you can copy
@file{libavformat/output-example.c} from the FFmpeg distribution.
@item Open the "Project / Properties" dialog box. In the "Configuration"
combo box, select "All Configurations" so that the changes you make will
affect both debug and release builds. In the tree view on the left hand
side, select "C/C++ / General", then edit the "Additional Include
Directories" setting to contain the path where the FFmpeg includes were
installed (i.e. @file{c:\msys\1.0\local\include}).
Do not add MinGW's include directory here, or the include files will
conflict with MSVC's.
@item Still in the "Project / Properties" dialog box, select
"Linker / General" from the tree view and edit the
"Additional Library Directories" setting to contain the @file{lib}
directory where FFmpeg was installed (i.e. @file{c:\msys\1.0\local\lib}),
the directory where MinGW libs are installed (i.e. @file{c:\mingw\lib}),
and the directory where MinGW's GCC libs are installed
(i.e. @file{C:\mingw\lib\gcc\mingw32\4.2.1-sjlj}). Then select
"Linker / Input" from the tree view, and add the files @file{libavformat.a},
@file{libavcodec.a}, @file{libavutil.a}, @file{libmingwex.a},
@file{libgcc.a}, and any other libraries you used (i.e. @file{libz.a})
to the end of "Additional Dependencies".
@item Now, select "C/C++ / Code Generation" from the tree view. Select
"Debug" in the "Configuration" combo box. Make sure that "Runtime
Library" is set to "Multi-threaded Debug DLL". Then, select "Release" in
the "Configuration" combo box and make sure that "Runtime Library" is
set to "Multi-threaded DLL".
@item Click "OK" to close the "Project / Properties" dialog box.
@item MSVC++ lacks some C99 header files that are fundamental for FFmpeg.
Get msinttypes from @url{http://code.google.com/p/msinttypes/downloads/list}
and install it in MSVC++'s include directory
(i.e. @file{C:\Program Files\Microsoft Visual Studio 8\VC\include}).
@item MSVC++ also does not understand the @code{inline} keyword used by
FFmpeg, so you must add this line before @code{#include}ing libav*:
@example
#define inline _inline
@end example
@item Build your application, everything should work.
@end enumerate
@subsection Using shared libraries
This is how to create DLL and LIB files that are compatible with MSVC++:
@enumerate
@item Add a call to @file{vcvars32.bat} (which sets up the environment
variables for the Visual C++ tools) as the first line of @file{msys.bat}.
The standard location for @file{vcvars32.bat} is
@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat},
and the standard location for @file{msys.bat} is @file{C:\msys\1.0\msys.bat}.
If this corresponds to your setup, add the following line as the first line
of @file{msys.bat}:
@example
For MSVC:
./configure --toolchain=msvc
call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"
@end example
For ICL:
./configure --toolchain=icl
Alternatively, you may start the @file{Visual Studio 2005 Command Prompt},
and run @file{c:\msys\1.0\msys.bat} from there.
@item Within the MSYS shell, run @code{lib.exe}. If you get a help message
from @file{Microsoft (R) Library Manager}, this means your environment
variables are set up correctly, the @file{Microsoft (R) Library Manager}
is on the path and will be used by FFmpeg to create
MSVC++-compatible import libraries.
@item Build FFmpeg with
@example
./configure --enable-shared
make
make install
@end example
If you wish to compile shared libraries, add @code{--enable-shared} to your
configure options. Note that due to the way MSVC and ICL handle DLL imports and
exports, you cannot compile static and shared libraries at the same time, and
enabling shared libraries will automatically disable the static ones.
Your install path (@file{/usr/local/} by default) should now have the
necessary DLL and LIB files under the @file{bin} directory.
Notes:
@itemize
@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
You can find out by running @code{which link} to see which @code{link.exe} you
are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
@item If you wish to build with zlib support, you will have to grab a compatible
zlib binary from somewhere, with an MSVC import lib, or if you wish to link
statically, you can follow the instructions below to build a compatible
@code{zlib.lib} with MSVC. Regardless of which method you use, you must still
follow step 3, or compilation will fail.
@enumerate
@item Grab the @uref{http://zlib.net/, zlib sources}.
@item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since
this is how FFmpeg is built as well.
@item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets
erroneously included when building FFmpeg.
@item Run @code{nmake -f win32/Makefile.msc}.
@item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC
can see.
@end enumerate
@item FFmpeg has been tested with the following on i686 and x86_64:
@itemize
@item Visual Studio 2010 Pro and Express
@item Visual Studio 2012 Pro and Express
@item Intel Composer XE 2013
@end itemize
Anything else is not officially supported.
Alternatively, build the libraries with a cross compiler, according to
the instructions below in @ref{Cross compilation for Windows with Linux}.
@end itemize
To use those files with MSVC++, do the same as you would do with
the static libraries, as described above. But in Step 4,
you should only need to add the directory where the LIB files are installed
(i.e. @file{c:\msys\usr\local\bin}). This is not a typo, the LIB files are
installed in the @file{bin} directory. And instead of adding the static
libraries (@file{libxxx.a} files) you should add the MSVC import libraries
(@file{avcodec.lib}, @file{avformat.lib}, and
@file{avutil.lib}). Note that you should not use the GCC import
libraries (@file{libxxx.dll.a} files), as these will give you undefined
reference errors. There should be no need for @file{libmingwex.a},
@file{libgcc.a}, and @file{wsock32.lib}, nor any other external library
statically linked into the DLLs.
@subsection Linking to FFmpeg with Microsoft Visual C++
If you plan to link with MSVC-built static libraries, you will need
to make sure you have @code{Runtime Library} set to
@code{Multi-threaded (/MT)} in your project's settings.
You will need to define @code{inline} to something MSVC understands:
FFmpeg headers do not declare global data for Windows DLLs through the usual
dllexport/dllimport interface. Such data will be exported properly while
building, but to use them in your MSVC++ code you will have to edit the
appropriate headers and mark the data as dllimport. For example, in
libavutil/pixdesc.h you should have:
@example
#define inline __inline
extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
@end example
Also note, that as stated in @strong{Microsoft Visual C++}, you will need
an MSVC-compatible @uref{http://code.google.com/p/msinttypes/, inttypes.h}.
If you plan on using import libraries created by dlltool, you must
set @code{References} to @code{No (/OPT:NOREF)} under the linker optimization
settings, otherwise the resulting binaries will fail during runtime.
This is not required when using import libraries generated by @code{lib.exe}.
Note that using import libraries created by dlltool requires
the linker optimization option to be set to
"References: Keep Unreferenced Data (@code{/OPT:NOREF})", otherwise
the resulting binaries will fail during runtime. This isn't
required when using import libraries generated by lib.exe.
This issue is reported upstream at
@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.
@@ -214,24 +275,27 @@ To create import libraries that work with the @code{/OPT:REF} option
@enumerate
@item Open the @emph{Visual Studio 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
(the standard location for this file is something like
@file{C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat}).
(the standard location for this file is
@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat}).
@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}
@@ -260,9 +324,24 @@ 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
@@ -275,12 +354,16 @@ 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
@@ -304,67 +387,4 @@ and for a build with shared libraries
./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@chapter Plan 9
The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler
does not implement all the C99 features needed by FFmpeg so the gcc
port must be used. Furthermore, a few items missing from the C
library and shell environment need to be fixed.
@itemize
@item GNU awk, grep, make, and sed
Working packages of these tools can be found at
@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}.
They can be installed with @uref{http://9front.org/, 9front's} @code{pkg}
utility by setting @code{pkgpath} to
@code{http://ports2plan9.googlecode.com/files/}.
@item Missing/broken @code{head} and @code{printf} commands
Replacements adequate for building FFmpeg can be found in the
@code{compat/plan9} directory. Place these somewhere they will be
found by the shell. These are not full implementations of the
commands and are @emph{not} suitable for general use.
@item Missing C99 @code{stdint.h} and @code{inttypes.h}
Replacement headers are available from
@url{http://code.google.com/p/plan9front/issues/detail?id=152}.
@item Missing or non-standard library functions
Some functions in the C library are missing or incomplete. The
@code{@uref{http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz,
gcc-apelibs-1207}} package from
@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}
includes an updated C library, but installing the full package gives
unusable executables. Instead, keep the files from @code{gccbin.tgz}
under @code{/386/lib/gnu}. From the @code{libc.a} archive in the
@code{gcc-apelibs-1207} package, extract the following object files and
turn them into a library:
@itemize
@item @code{strerror.o}
@item @code{strtoll.o}
@item @code{snprintf.o}
@item @code{vsnprintf.o}
@item @code{vfprintf.o}
@item @code{_IO_getc.o}
@item @code{_IO_putc.o}
@end itemize
Use the @code{--extra-libs} option of @code{configure} to inform the
build system of this library.
@item FPU exceptions enabled by default
Unlike most other systems, Plan 9 enables FPU exceptions by default.
These must be disabled before calling any FFmpeg functions. While the
included tools will do this automatically, other users of the
libraries must do it themselves.
@end itemize
@bye

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