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239 Commits

Author SHA1 Message Date
Michael Niedermayer
1014e20492 atrac3: Fix crash in tonal component decoding.
Fixes Ticket780
Bug Found by: cosminamironesei

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9af6abdc17)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-12 22:10:33 +01:00
Michael Niedermayer
431cf16963 h264: check chroma_format_idc range.
Fixes Ticket758
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7fff64e00d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-12 22:09:56 +01:00
Michael Niedermayer
e85296beae Merge remote-tracking branch 'qatar/release/0.6' into release/0.6
* qatar/release/0.6:
  Release notes and changelog for 0.6.5
  Bump version number for 0.6.5 release.
  vorbis: An additional defense in the Vorbis codec.
  vorbisdec: Fix decoding bug with channel handling

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-12 22:06:50 +01:00
Reinhard Tartler
62c4739348 Release notes and changelog for 0.6.5 2012-01-10 21:17:30 +01:00
Reinhard Tartler
7efa13b4b4 Bump version number for 0.6.5 release. 2012-01-10 21:02:32 +01:00
Chris Evans
a5e0afe3c9 vorbis: An additional defense in the Vorbis codec.
Fixes Bug: #190
Chromium Bug: #100543
Related to CVE-2011-3893

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit afb2aa5379)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit b0283ccb9e)

Conflicts:

	libavcodec/vorbis_dec.c
2012-01-08 09:29:16 +01:00
Reinhard Tartler
42f0a66968 vorbisdec: Fix decoding bug with channel handling
Fixes Bug: #191
Chromium Bug: #101458
CVE-2011-3895

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit e6d527ff72)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 97f23c72a3)

Conflicts:

	libavcodec/vorbis_dec.c
2012-01-08 09:24:13 +01:00
Michael Niedermayer
f1c9dbe40b Merge remote-tracking branch 'qatar/release/0.6' into release/0.6
* qatar/release/0.6:
  matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
  vorbis: Avoid some out-of-bounds reads
  vp3: fix streams with non-zero last coefficient
  vp3: fix oob read for negative tokens and memleaks on error. (cherry picked from commit 8370e426e4)

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-08 05:13:49 +01:00
Michael Niedermayer
b945f558c7 vp3: fix regression with mplayer-crash.ogv
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a2a12e3358)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-08 05:13:32 +01:00
Chris Evans
90a4a46747 matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
Fixes bug #190
Chromium bug #100492
related to CVE-2011-3893

Signed-off-by: Reinhard Tartler <siretart@tauware.de>

(cherry-picked from commit faaec4676c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 1f625431e2)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-01-07 22:03:48 +01:00
Chris Evans
6d6254ba9f vorbis: Avoid some out-of-bounds reads
Fixes Bug: #190
Chromium Bug: #100543
Related to CVE-2011-3893

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 57cd6d7095)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 4a94678f1b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-01-07 22:03:34 +01:00
Janne Grunau
ae24b5ce3a vp3: fix streams with non-zero last coefficient
Fixes a regression introduced in 8b94df0f20.
(cherry picked from commit 9b4767e478)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 82a11fcff2)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-01-07 21:33:24 +01:00
Ronald S. Bultje
c9c7db0af2 vp3: fix oob read for negative tokens and memleaks on error.
(cherry picked from commit 8370e426e4)

Fixes: #189
Chromium-Bug: 101172,100465
CVE-2011-3892

Removed the parts that are related to multi-threading, which is not
included before 0.7.

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit c624935554)

Conflicts:

	libavcodec/vp3.c
2012-01-07 09:35:15 +01:00
Michael Niedermayer
e1a2bcbec8 h264: fix init of topleft ref/mv.
Fixes Ticket778

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 680880c98d)
2011-12-28 02:20:09 +01:00
Michael Niedermayer
d32ea79ea2 Merge remote-tracking branch 'qatar/release/0.6' into release/0.6
* qatar/release/0.6:
  Release notes and changelog for 0.6.4

Conflicts:
	Changelog

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-25 20:11:09 +01:00
Reinhard Tartler
6b156c4563 Release notes and changelog for 0.6.4 2011-12-25 10:03:08 +01:00
Michael Niedermayer
57eb787ed3 Merge remote-tracking branch 'qatar/release/0.6' into release/0.6
* qatar/release/0.6: (58 commits)
  Bump version number for 0.6.4 release.
  qdm2: check output buffer size before decoding
  Fix qdm2 decoder packet handling to match the api
  4xm: Add a check in decode_i_frame to prevent buffer overreads
  wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
  swscale: #include "libavutil/mathematics.h"
  vp3dec: Check coefficient index in vp3_dequant()
  svq1dec: call avcodec_set_dimensions() after dimensions changed.
  vp6: Fix illegal read.
  vp6: Fix illegal read.
  vp6: Reset the internal state when aborting key frames header parsing
  vp6: Check for huffman tree build errors
  vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
  Fix out of bound reads in the QDM2 decoder.
  Check for out of bound writes in the QDM2 decoder.
  vmd: fix segfaults on corruped streams
  rv34: Check for invalid slice offsets
  rv34: Fix potential overreads
  rv34: Avoid NULL dereference on corrupted bitstream
  rv10: Reject slices that does not have the same type as the first one
  ...

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-25 01:24:40 +01:00
Reinhard Tartler
dbe7e209df Bump version number for 0.6.4 release. 2011-12-24 15:59:10 +01:00
Justin Ruggles
cfb9b47a1e qdm2: check output buffer size before decoding
(cherry picked from commit 7d49f79f1c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 7347205351)

Conflicts:

	libavcodec/qdm2.c
2011-12-24 15:57:17 +01:00
Baptiste Coudurier
b26c1a8b7e Fix qdm2 decoder packet handling to match the api
Originally committed as revision 25767 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-12-24 15:54:51 +01:00
Shitiz Garg
ccd2ca0246 4xm: Add a check in decode_i_frame to prevent buffer overreads
Fixes bugzilla #135

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit 355d917c0b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit d912a30c7d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Justin Ruggles
92b964969b wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
The initial values are not checked against the number of block sizes.
Initializing them to frame_len_bits will result in a block size index of 0
in these cases instead of something that might be out-of-range.

Fixes Bug 81.
(cherry picked from commit 05d1e45d1f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 8dba5608dc)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Reinhard Tartler
ca87ec53e9 swscale: #include "libavutil/mathematics.h"
this file uses the M_PI macro since
4e74187db2, so include the correct header
directly.

Signed-off-by: Reinhard Tartler <siretart@tauware.de>

(cherry picked from commit 5089ce1b5a)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 851098c9e0)

Conflicts:

	libswscale/utils.c
2011-12-24 15:47:57 +01:00
Reinhard Tartler
bd071de29a vp3dec: Check coefficient index in vp3_dequant()
Based on a patch by Michael Niedermayer <michaelni@gmx.at>

Fixes NGS00145, CVE-2011-4352

Found-by: Phillip Langlois
Signed-off-by: Reinhard Tartler <siretart@tauware.de>

(cherry picked from commit 8b94df0f20)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit bba709214a)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Michael Niedermayer
8ddc0b491d svq1dec: call avcodec_set_dimensions() after dimensions changed.
Fixes NGS00148, CVE-2011-4579

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

(cherry picked from commit 6e24b9488e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 0eca0da06e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Thierry Foucu
94aacaf508 vp6: Fix illegal read.
Found with Address Sanitizer

Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit e0966eb140)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit ba4b08b789)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Alex Converse
8d68083298 vp6: Fix illegal read.
(cherry picked from commit 2a6eb06254)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 67a7ed623b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Laurent Aimar
e28bb18fdc vp6: Reset the internal state when aborting key frames header parsing
It prevents leaving the state only half initialized.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit a72cad0a6c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit c76505e0de)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Laurent Aimar
a62779d986 vp6: Check for huffman tree build errors
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit 066fff755a)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 30c08e2261)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:57 +01:00
Dustin Brody
201fcfb894 vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit f913eeea43)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 7367cbec1b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:47:56 +01:00
Laurent Aimar
8856c4c5c9 Fix out of bound reads in the QDM2 decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit 5a19acb17c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 0d93d5c461)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:17:00 +01:00
Laurent Aimar
0f7bf1786e Check for out of bound writes in the QDM2 decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit 291d74a46d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit a31ccacb1a)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:16:51 +01:00
Laurent Aimar
b99366faef vmd: fix segfaults on corruped streams
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 494cfacdb9)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 15:16:36 +01:00
Laurent Aimar
da0900e8bb rv34: Check for invalid slice offsets
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 4cc7732386)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 2bbb142a14)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
d5551d7884 rv34: Fix potential overreads
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit b4ed3d78cb)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit b4a1bf0bbf)

Conflicts:

	libavcodec/rv34.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
35f1888585 rv34: Avoid NULL dereference on corrupted bitstream
rv34_decode_slice() can return without allocating any pictures.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit d0f6ab0298)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
7cd7461ec8 rv10: Reject slices that does not have the same type as the first one
This prevents crashes with some corrupted bitstreams.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 4a29b47186)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 28d948ac44)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
23f622de60 oggdec: fix out of bound write in the ogg demuxer
Between ogg_save() and ogg_restore() calls, the number of streams
could have been reduced.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 0e7efb9d23)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit a3d471e500)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Michael Niedermayer
19a99b6e6b smacker: fix a few off by 1 errors
stereo & 16bit is untested due to lack of samples

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 5166376f24)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 78cd2e18a4)

Conflicts:

	libavcodec/smacker.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
117e04cdfa Check for invalid VLC value in smacker decoder.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 6489455495)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
5d6fe49ac9 Check and propagate errors when VLC trees cannot be built in smacker decoder.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 9676ffba83)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
9f28eede5e Fixed off by one packet size allocation in the smacker demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit a92d0fa5d2)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
6f70111e81 Check for invalid packet size in the smacker demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit e055932f56)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
4492523938 ape demuxer: fix segfault on memory allocation failure.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 273aab99bf)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 4ee014309c)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
a97e82c487 Fixed size given to init_get_bits() in xan decoder.
(cherry picked from commit 393d5031c6)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Kostya Shishkov
f79f3a946f smacker demuxer: handle possible av_realloc() failure.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 47a8589f7b)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 0b9b3570a3)

Conflicts:

	libavformat/smacker.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
5394cdf775 Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 8bfea4ab4e)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Alex Converse
7f163e5a87 indeo2: fail if input buffer too small
(cherry picked from commit b7ce4f1d1c)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Alex Converse
23999c45bc indeo2: init_get_bits size in bits instead of bytes
(cherry picked from commit 68ca330cbd)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Alex Converse
14fae6eab0 wavpack: Check error codes rather than working around error conditions.
(cherry picked from commit dba2b63a98)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 5d4c065476)

Conflicts:

	libavcodec/wavpack.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:33 +01:00
Laurent Aimar
f5a8c4242e Fixed invalid writes in wavpack decoder on corrupted bitstreams.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 0aedab0340)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 685940da4c)

Conflicts:

	libavcodec/wavpack.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Laurent Aimar
1edc513bcf Fixed invalid access in wavpack decoder on corrupted bitstream.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 55354b7de2)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 4b84e995ad)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Laurent Aimar
9864518544 Fixed invalid access in wavpack decoder on corrupted extra bits sub-blocks.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit beefafda63)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Sean McGovern
4ccb8f5b7b cpu detection: avoid a signed overflow
1<<31 overflows because 1 is signed, so force it to unsigned.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 5938e02185)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Jeff Downs
01b9a6e447 h264: correct implicit weight table computation for long ref pics
Correct computation of implicit weight tables when referencing pictures
that are marked for long reference.

Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 87cf70eb23)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Jeff Downs
34d2fe6860 h264: correct the check for invalid long term frame index in MMCO decode
The current check on MMCO parameters prohibits a "max long term frame index
plus 1" of 16 (frame idx of 15) for the "set max long term frame index" MMCO.
Fix this off-by-one error to allow the full range of legal values.

Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 29a09eae9a)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Kostya Shishkov
e7746f834a rv10/20: tell decoder to use edge emulation
This removes out-of-edge motion compensation artifacts (easily spotted green
blocks in avplay, gray blocks in transcoding), for example here:
http://samples.libav.org/samples/real/tv_watching_t1.rm

Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 331971116d)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Luca Barbato
973bdafe05 flvenc: use int64_t to store offsets
Metadata currently is written only at the start of the file in normal
cases, when transcoding from a rtmp source metadata could be
written later and the offset recorded can exceed 32bit.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 7f5bf4fbaf)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit fe3e7297fe)

Conflicts:

	libavformat/flvenc.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Reimar Döffinger
f4a5a730d8 VC-1: fix reading of custom PAR.
Custom PAR num/denum are in 1-256 range.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 0e86965514)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 12:20:32 +01:00
Michael Niedermayer
603a282f8f Merge remote-tracking branch 'qatar/release/0.6' into release/0.6
* qatar/release/0.6:
  update version
  Release notes and changelog for 0.6.3
  Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
  cavs: fix some crashes with invalid bitstreams
  libvo-aacenc: Sync up with 0.7.2
  Fix MMX rgb24 to yuv conversion with gcc 4.6
  oggdec: prevent heap corruption.
  Fix ff_imdct_calc_sse() on gcc-4.6
  backport libvo-aacenc wrapper for aac encoding

Conflicts:
	Changelog
	VERSION

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 01:50:29 +01:00
Dustin Brody
36c196bca4 h264: notice memory allocation failure
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit bac3ab13ea)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 59a22afa0b)

Conflicts:

	libavcodec/h264.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Baptiste Coudurier
7b733e4b7f libx264: do not set pic quality if no frame is output
Avoids uninitialized reads.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 5caa2de19e)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Alex Converse
e07086d8ae mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
This prevents out of bounds reads when extradata is being decoded.
(cherry picked from commit 1f6f58d585)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Diego Biurrun
d2411412f1 rv30: return AVERROR(EINVAL) instead of EINVAL
On some platforms EINVAL could be positive, ensure we return negative values.
(cherry picked from commit e5985185d2)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Rafaël Carré
0facc63ff6 Do not decode RV30 files if the extradata is too small
Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 289c60001f)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Mans Rullgard
5ed9457260 aacps: skip some memcpy() if src and dst would be equal
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e5902d60ce)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Jindrich Makovicka
002e6d185c mpegts: fix Continuity Counter error detection
According to MPEG-TS specs, the continuity_counter shall not be
incremented when the adaptation_field_control of the packet
equals '00' or '10'.

Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 8923cfa328)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Justin Ruggles
81d5ceff04 alsa: fallback to buffer_size/4 for period_size.
buffer_size/4 is the value used by aplay. This fixes output to null
devices, e.g. writing ALSA output to a file.
(cherry picked from commit 8bfd7f6a47)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Clément Bœsch
86849f0976 mxfenc: fix ignored drop flag in binary timecode representation.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 4d5e7ab5c4)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
John Stebbins
84fce203bd dca: set AVCodecContext frame_size for DTS audio
Set the frame size when decoding DTS audio.

This has the side effect of fixing the computation of timestamps for DTS-HD in compute_pkt_fields.  Since frame_size is
not currently set, the duration of a frame is being guessed based on the streams bitrate.  But for DTS-HD, the bitrate
currently used is the rate of the DTS core which is much different than the whole DTS-HD stream and leads to a wildly
inaccurate frame duration estimate.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 49c7006c7e)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 0631896885)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Jason Garrett-Glaser
572e94bc51 H.264: fix overreads of qscale_table
filter_mb_fast assumed that qscale_table was padded like many of the other tables.
(cherry picked from commit 5029a40633)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:32 +01:00
Piotr Kaczuba
7487d53d01 postprocess.c: filter name needs to be double 0 terminated
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit f4f3300c09)
(cherry picked from commit 20ca827019)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:31 +01:00
Alex Converse
21ad6e08e4 Replace strncpy() with av_strlcpy() in libpostproc.
(cherry picked from commit 1a5e4fd8c5)

Conflicts:

	ffmpeg.c
	libavcodec/ac3enc.c
	libavcodec/ass.c
	libavformat/movenc.c
	libavformat/mp3enc.c
	libavutil/log.c
2011-11-05 13:18:31 +01:00
Michael Niedermayer
feca20021c jpegdec: actually search for and parse RSTn
Fixes decoding of MJPEG files produced by some UVC Logitec web cameras,
such as "Notebook Pro" and "HD C910".

References:
http://trac.videolan.org/vlc/ticket/4215
http://ffmpeg.org/trac/ffmpeg/ticket/267

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Reviewed-by: Kostya <kostya.shishkov@gmail.com>
(cherry picked from commit 8c0fa61a97)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:31 +01:00
Alex Converse
394c3e78d5 riff: Add mpgv MPEG-2 fourcc
Supported by mplayer and seen in the wild.
(cherry picked from commit 505345ed5d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:31 +01:00
Tomas Härdin
2fc39af1de Added M701 codec_tag for mpeg2video
Originally committed as revision 23513 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 77d3f1f792)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 13:18:31 +01:00
Reinhard Tartler
973d752dfb update version 2011-11-05 13:01:01 +01:00
Reinhard Tartler
3cc3f351a6 Release notes and changelog for 0.6.3 2011-11-05 12:46:18 +01:00
Michael Niedermayer
222d18ab20 Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
Whitespace of the patch cleaned up by Aurel
Some of the issues have been reported by Steve Manzuik / Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

(cherry picked from commit 956c901c68)

Further suggestions from Kostya <kostya.shishkov@gmail.com> have been
implemented by Reinhard Tartler <siretart@tauware.de>

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 77d2ef13a8)

NB: MSVR-11-0080 doesn't seem to exist. This issue seems to be known
as MSVR11-011 instead.

Fixes: CVE-2011-3504

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 12:31:43 +01:00
Mans Rullgard
7a6bba627d cavs: fix some crashes with invalid bitstreams
This removes all valgrind-reported invalid writes with one
specific test file.

Fixes http://www.ocert.org/advisories/ocert-2011-002.html

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 4a71da0f3a)

Fixes CVE-2011-3362, CVE-2011-3973, CVE-2011-3974

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 12:29:37 +01:00
Reinhard Tartler
9f95d7aab3 libvo-aacenc: Sync up with 0.7.2
This version was checked out from the 0.7.2 release and has some bits
reverted to compile in the release/0.6 branch

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 12:27:42 +01:00
Mans Rullgard
fe4a355805 Fix MMX rgb24 to yuv conversion with gcc 4.6
When built with gcc 4.6, the MMX rgb24 to yuv conversion gives
wrong output.  The compiler produces this warning:

libswscale/swscale_template.c:1885:5: warning: use of memory input without lvalue in asm operand 4 is deprecated

Changing the memory operand to a register makes it work.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f344903ca5)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 08:59:14 +01:00
Chris Evans
88ea7ca627 oggdec: prevent heap corruption.
Specifically crafted samples can reinit ogg->streams[] while
reading samples, and thus we should not cache old pointers since
these may no longer be valid.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry-picked from commit 4cc3467e7a)
2011-08-04 22:01:25 +02:00
Reinhard Tartler
03ef9828b5 Fix ff_imdct_calc_sse() on gcc-4.6
Gcc 4.6 only preserves the first value when using an array with an "m"
constraint.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 770c410fbb)

References:
http://bugs.debian.org/635324

Conflicts:

	libavcodec/x86/fft_sse.c
2011-07-25 09:38:48 +02:00
Reinhard Tartler
2ad6e14457 backport libvo-aacenc wrapper for aac encoding 2011-05-13 12:43:40 +02:00
Michael Niedermayer
39c8434bce Update VERSION for 0.6.4
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-02 03:22:11 +02:00
Michael Niedermayer
701dd6eab0 Merge remote branch 'qatar/release/0.6' into release/0.6
* qatar/release/0.6:
  Release notes for 0.6.3
  update Changelog
Duplicate  AMV: disable DR1 and don't override EMU_EDGE
  lavf: inspect more frames for fps when container time base is coarse
Duplicate  Fix races in default av_log handler
  aac: add headers needed for log2f()
  mjpeg: Detect overreads in mjpeg_decode_scan() and error out.

Conflicts:
	Changelog

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-02 03:21:55 +02:00
Reinhard Tartler
f5d14a92cb Release notes for 0.6.3 2011-04-30 09:01:09 +02:00
Reinhard Tartler
b5c3b18bfc update Changelog 2011-04-30 08:44:07 +02:00
Michael Niedermayer
8539ea3265 AMV: disable DR1 and don't override EMU_EDGE
This works around a possibly exploitable crash.
Appearently, vlc can be exploited with a malicous file. This should get
reverted as soon as a proper fix is found.

Reported-at: Thu, 21 Apr 2011 14:38:25 +0000
Reported-by: Dominic Chell <Dominic.Chell@ngssecure.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 89f903b3d5)
(cherry picked from commit 9b919571e5)
(cherry picked from commit 7089265756)

Signed-off-by: Reinhard Tartler <siretart@sandy.tauware.de>
2011-04-30 08:19:29 +02:00
Reinhard Tartler
9848e1be2d lavf: inspect more frames for fps when container time base is coarse
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.

Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 78431098f9)

Tested with mplayer based on this report
http://thread.gmane.org/gmane.comp.video.mplayer.user/66043/focus=66063

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 7c152a458d)
2011-04-30 08:18:32 +02:00
Reinhard Tartler
66ebf37c73 Fix races in default av_log handler
Prevent competing threads from overwriting (shared) buffers.

Original patch by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0247bdee25)
2011-04-30 08:16:18 +02:00
Michael Niedermayer
036fefebb9 aac: add headers needed for log2f()
Allows compilation of aaccoder.c under cygwin

Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>

(cherry picked from commit 30fe971934)
2011-04-30 08:16:07 +02:00
Michael Niedermayer
1916656641 mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Ronald S. Bultje <rbultje@google.com>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>

(cherry picked from commit 0d9cba562b88899f0769e686d19b7953f589069b)
2011-04-30 08:12:10 +02:00
Michael Niedermayer
ace432f62c Update VERSION for 0.6.3
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-26 12:44:19 +02:00
Michael Niedermayer
f8969f8ea9 update changelog
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-26 12:29:34 +02:00
Michael Niedermayer
36f3244ec8 Fix apparently exploitable race condition.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-26 12:28:46 +02:00
Michael Niedermayer
553ae88b2e uodate changelog with AMV fix
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-26 11:59:25 +02:00
Michael Niedermayer
9b919571e5 AMV: Fix possibly exploitable crash.
Reported-at: Thu, 21 Apr 2011 14:38:25 +0000
Reported-by: Dominic Chell <Dominic.Chell@ngssecure.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 89f903b3d5)
2011-04-26 11:57:22 +02:00
Michael Niedermayer
99d3772ef4 Update changelog to match our (late) 0.6.2 release 2011-03-24 02:22:39 +01:00
Michael Niedermayer
15cfc6c97d Merge remote-tracking branch 'newdev/release/0.6' into release/0.6
* newdev/release/0.6:
  Support writing 2d float arrays.
  Add missing dependencies for the AAC-HE backport for hardcoded tables
  configure: Add the -D parameter to the dlltool command
  Set the correct target for mingw64 dlltool
  configure: use dlltools instead of lib.exe
  release notes and changelog for 0.6.2

Conflicts:
	Changelog

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-03-24 02:22:00 +01:00
Michael Kostylev
b4eafa8b04 Support writing 2d float arrays.
Patch by Michael Kostylev, michael D kostylev A gmail

Originally committed as revision 23105 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 56b37cd1c7)

Unbreaks compilation since the HE-AAC v2 backport
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-03-22 21:15:20 +01:00
Reinhard Tartler
2b2856bac2 Add missing dependencies for the AAC-HE backport for hardcoded tables
Unbreaks compilation with --enable-hardcoded-tables
2011-03-22 21:08:48 +01:00
Martin Storsjö
52f9b632b5 configure: Add the -D parameter to the dlltool command
This is required for the generated .lib file to actually
be usable by MSVC.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 77e41e047d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-03-22 12:42:56 +01:00
Luca Barbato
79e0cbae33 Set the correct target for mingw64 dlltool
That fixes .lib creation for the win64 target.
(cherry picked from commit 417516f63f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-03-22 12:42:55 +01:00
Luca Barbato
7fc5bffb54 configure: use dlltools instead of lib.exe
This way building ffmpeg on mingw won't require windows specific tools
(cherry picked from commit ec10a9ab46)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-03-22 12:42:55 +01:00
Reinhard Tartler
4940d1c0ae release notes and changelog for 0.6.2
(cherry picked from commit b0f8fdc411)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-03-19 17:47:14 +01:00
Reinhard Tartler
b0f8fdc411 release notes and changelog for 0.6.2 2011-03-18 18:34:49 +01:00
Reinhard Tartler
ba1927dda9 partially rename FFmpeg to Libav
update mailing list references and irc channels in configure
(cherry picked from commit 070c5d0f35)
2011-03-18 18:34:49 +01:00
Janne Grunau
dda2062c31 consolidate .gitignore patters into a single file
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 2c3589bfda)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-03-18 17:47:01 +01:00
Janne Grunau
5d01cd8429 convert svn:ignore properties to .gitignore files
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 348b8218f7)
2011-03-18 17:46:29 +01:00
Kostya
f9bf2a4ab1 Do not attempt to decode APE file with no frames
This fixes invalid reads/writes with this sample:
http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt
(cherry picked from commit 8312e3fc90)
2011-03-18 17:02:34 +01:00
Reimar Döffinger
cf69619141 Fix invalid reads in VC1 decoder
Patch discussed and taken from https://roundup.ffmpeg.org/issue2584
(cherry picked from commit 2bbec1eda4)

Change related to CVE-2011-0723
2011-03-18 17:01:08 +01:00
Reinhard Tartler
2d782b9646 Bump version number for 0.6.2 release. 2011-03-18 16:59:07 +01:00
Frank Barchard
5e3d023702 Check rangebits to avoid a possible crash.
Fixes issue 2548 (and Chrome issue 68115 and unknown CERT issues).

Patch by Frank Barchard, fbarchard at google

Originally committed as revision 26365 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 13184036a6)

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 23:26:15 +01:00
Jason Garrett-Glaser
4ac56bf7dc Fix crashes in vorbis decoding found by zzuf
Fixes issue 2322.

Originally committed as revision 25591 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 3dde66752d)

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 23:25:45 +01:00
Diego Biurrun
a38489c0d8 slight rewording for the release focus description
Originally committed as revision 25524 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-18 19:46:51 +00:00
Diego Biurrun
a5200fac2c Rewording for the 0.6.1 release notes.
Originally committed as revision 25514 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-18 11:28:24 +00:00
Diego Biurrun
32b0876fb5 small spelling fixes
Originally committed as revision 25513 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-18 10:52:44 +00:00
Reinhard Tartler
e33f58af5a update version for 0.6.1
Originally committed as revision 25469 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-13 12:44:47 +00:00
Reinhard Tartler
540c6cca6e add note about flic fix (CVE-2010-3429) to RELEASE notes
Originally committed as revision 25468 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-13 12:43:18 +00:00
Reinhard Tartler
08f5b2464d add note about flic fix (CVE-2010-3429)
Originally committed as revision 25467 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-13 12:42:38 +00:00
Reinhard Tartler
79a16f0cac Fix several security issues in flicvideo.c
This fixes CVE-2010-3429


backport r25223 by michael


Originally committed as revision 25307 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-02 07:42:42 +00:00
Reinhard Tartler
9ad774ff33 property changes from the AAC backport
Originally committed as revision 25306 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-10-02 07:35:56 +00:00
Reinhard Tartler
84149812a6 clarify addition of VP80 fourcc code
Originally committed as revision 24636 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-07-31 21:56:15 +00:00
Reinhard Tartler
d41187cb9e update release notes for the upcoming 0.6.1 point release
Originally committed as revision 24635 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-07-31 21:55:38 +00:00
Reinhard Tartler
094e849f32 update changelog for upcoming 0.6.1 point release
Originally committed as revision 24633 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-07-31 21:13:20 +00:00
Reinhard Tartler
ba90f93852 Backport AAC-HE v2 from trunk
This patch has seen testing for a couple of weeks in ubuntu maverick and debian/experimental w/o negative feedback so far.



Originally committed as revision 24576 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-07-28 20:14:28 +00:00
Reinhard Tartler
93883b0c39 aviobuf: Do short seeks forward by reading and skipping data instead of a proper seek
This improves performance on e.g. seekable http.


backport r24280 by mstorsjo


Originally committed as revision 24335 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-07-19 19:51:29 +00:00
Reinhard Tartler
61eb165514 ac3: make the value of codec_id during (E-)AC-3 parsing stay CODEC_ID_EAC3
if any E-AC-3 frames have been detected instead of switching back and forth for
AC-3 core + dependent E-AC-3 substream(s).
Fixes Issue 2022.

backport r24103 by jbr


Originally committed as revision 24136 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-07-09 01:16:42 +00:00
9d1213912f Add VP80 fourcc
Patch by Google

backport r23193 by conrad


Originally committed as revision 23863 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-28 20:38:36 +00:00
Reinhard Tartler
edf5072fea Doxygen av_codec_get_id() and av_codec_get_tag().
backport r23102 by stefano


Originally committed as revision 23801 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-26 15:32:27 +00:00
Jean-Daniel Dupas
32fa7cecd7 Some fields were incorrectly reset (to NULL) when calling avcodec_copy_context().
Patch by Jean-Daniel Dupas, devlists shadowlab org 


backport r23344 by cehoyos


Originally committed as revision 23793 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-26 14:24:14 +00:00
Reinhard Tartler
b9ebd7dc7d 10l: aacsbr: Fix f_master[2] calculation when k2diff == -1.
backport r23660 by alexc


Originally committed as revision 23747 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-24 05:46:58 +00:00
Diego Biurrun
55b6e09e5e Fix two small typos.
Originally committed as revision 23678 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-21 09:17:19 +00:00
Reinhard Tartler
aa48d9e64c remove note that libvpx was considered (L)GPL incompatible
Originally committed as revision 23623 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-16 06:10:30 +00:00
Reinhard Tartler
d7e98ea6ac configure: fix symver detection with icc 10
Apparently the compiler sometimes bypasses the normal assembler, in
which case the asm label variant works.  If the file contains any
inline asm, this bypassing is not done, and the assembler chokes on
invalid syntax.  Inserting a dummy asm() statement in the test when
the compiler supports it makes sure no shortcuts are taken and the
test fails as it should.

backport r23616 by mru


Originally committed as revision 23618 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-15 19:44:30 +00:00
Reinhard Tartler
8182bcae90 Fix symbol version compat wrappers on systems with export prefixes
backport r23615 by mru


Originally committed as revision 23617 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-15 19:44:09 +00:00
Reinhard Tartler
522f1861ca Add compatibility wrappers for functions moved from lavf to lavc
When symbol versioning is enabled, moving symbols from one library to
another breaks binary compatibility.  This adds wrappers with the old
version tag for the av_*packet functions recently moved to lavc.

backport r23611 by mru


Originally committed as revision 23613 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-15 13:31:27 +00:00
Reinhard Tartler
44f91788a5 revert r23610, better patch is pending
Originally committed as revision 23612 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-15 13:29:20 +00:00
Reinhard Tartler
03b47bb8d7 add symver hacks for gnu linkers
this patch restores binary compatibility for the av_*_packet symbols that have
been moved from libavformat to libavcodec.  This patch works for gnu toolchains
only; support for ARM RCVT will be handed in for a later point release as soon
as the patch is ready and approved by the ARM maintainer(s).


Originally committed as revision 23610 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-15 12:29:19 +00:00
Reinhard Tartler
05b4e4fe49 remove redundant changelog entry
Originally committed as revision 23560 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-10 08:10:32 +00:00
Reinhard Tartler
3229bf8573 librtmp may link to different ssl implementations
Make sure we link using the right libs by using pkg-config



backport r23511 by lu_zero


Originally committed as revision 23550 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-09 15:55:02 +00:00
James Zern
db06ddf8c6 Add WebM muxer
Patch by James Zern <jzern at google>

backport r23474 by conrad


Originally committed as revision 23549 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-09 15:54:27 +00:00
Reinhard Tartler
eb4b8b96fc libvpx now has an (L)GPL-compatible license.
backport r23482 by diego


Originally committed as revision 23487 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-05 13:58:13 +00:00
Reinhard Tartler
373010971e update RELEASE notes
VP8 license issues seem to have solved, plus include wording suggestion
from discussion on -cvslog


Originally committed as revision 23483 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-05 08:30:55 +00:00
James Zern
c3cd8890fe VP8 encoding via libvpx.
Patch by James Zern, Google, Inc.


backport r23449 by cehoyos


Originally committed as revision 23455 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-03 21:36:38 +00:00
Reinhard Tartler
20d4a1f020 Add an entry to APIchanges for the addition of sws_convertPalette8ToPacked32 -24
backport r23415 by siretart


Originally committed as revision 23446 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-03 20:16:42 +00:00
Reinhard Tartler
e5880b3f24 updates: introductionary and more general notes, add missing items from changelog
Originally committed as revision 23420 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-02 09:25:37 +00:00
Reinhard Tartler
4b51239df3 disable (native) vorbis encoder for the 0.6 branch
Originally committed as revision 23416 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-01 20:14:00 +00:00
Reinhard Tartler
4979362ae4 ffmpeg: offer alternatives for experimental codecs if they exist
backport r23398 by janne


Originally committed as revision 23400 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-01 06:29:50 +00:00
Reinhard Tartler
0546386dff ffmpeg: fail if user selected codec is experimental and strict_std_compliance > experimental
backport r23397 by janne


Originally committed as revision 23399 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-06-01 06:28:19 +00:00
Reinhard Tartler
70ea5176a5 Add a short note about libvpx.
backport r23394 by diego


Originally committed as revision 23395 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-31 21:49:53 +00:00
Diego Biurrun
661092004f Require --enable-nonfree flag for libvpx.
The license of libvpx is incompatible with the (L)GPL.  As long as this is
the case, the only way to use it is by marking the result as nonfree.

backport r23371 by diego


Originally committed as revision 23390 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-30 22:25:40 +00:00
Reinhard Tartler
f7f9f18f83 Treat SIPP like xvid, fixed issue1966
backport r23336 by michael


Originally committed as revision 23385 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 15:07:08 +00:00
Reinhard Tartler
55291e37f5 Samsung uses SIPP as FourCC for MPEG-4 ASP.
backport r23309 by cehoyos


Originally committed as revision 23384 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:45:13 +00:00
Reinhard Tartler
1f1d2fa75c ffserver: Send a Content-Base header in the reply to RTSP DESCRIBE requests
This is needed for QuickTime Player to be able to connect properly.


backport r23325 by mstorsjo


Originally committed as revision 23383 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:41:26 +00:00
Reinhard Tartler
2a546cecbf backport latest ffserver fixes like memory leaks and invalid reads
Patches by Howard Chu, hyc at highlandsun dot com


backport r23290-23295 by mstorsjo


Originally committed as revision 23382 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:40:31 +00:00
Reinhard Tartler
8f3504c3ca Mark vorbis encoder as experimental.
backport r23339 by cehoyos


Originally committed as revision 23381 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:34:37 +00:00
Reinhard Tartler
bf20d5e4bb Mark AAC encoder as experimental.
backport r23350 by alexc


Originally committed as revision 23380 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:33:50 +00:00
Janne Grunau
3fa19f6828 Add CODEC_CAP_EXPERIMENTAL and prefer encoders without it.
Patch by Janne Grunau, janne-ffmpeg jannau net 


backport r23334,23337-23338 by cehoyos and stefano


Originally committed as revision 23379 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:33:02 +00:00
Reinhard Tartler
4e62a67389 Check for VUI overeading and reset num_reoder_frames.
This helps the video from issue1831


backport r23328 by michael


Originally committed as revision 23378 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:25:18 +00:00
Reinhard Tartler
46ac6315fd api-example: Try to avoid decoding incomplete frames
Use a larger input audio buffer, refill it when it has less than 4 KB data
left.


backport r23323 by mstorsjo


Originally committed as revision 23377 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:20:05 +00:00
9052b5b73b matroska: Add V_VP8
Patch by Google

backport r23192 by conrad


Originally committed as revision 23376 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:11:03 +00:00
Reinhard Tartler
9181adff9c Fix VP8 listing in general.texi
backport r23306 by mstorsjo


Originally committed as revision 23375 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:05:05 +00:00
James Zern
8261cce1f0 VP8 decoding via libvpx
Patch by James Zern for Google, Inc., jzern google com

backportd r23191,23303,23307-23308 by conrad, cehoyos and mstorsjo

Originally committed as revision 23374 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 14:03:03 +00:00
Reinhard Tartler
df62b0d9f5 Display a more descriptive log message when probe buffer limit is
reached.



backport r23288 by jai_menon


Originally committed as revision 23373 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 13:11:56 +00:00
Cyril Russo
e63d03e7e2 Fix documentation of av_audio_convert.
Patch by Cyril Russo, stage D nexvision A laposte net


backport r23285 by cehoyos


Originally committed as revision 23372 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-29 13:11:21 +00:00
Reinhard Tartler
7920dc8802 matroskaenc: Write codec time base as default duration for video tracks.
This isn't exactly semantically equivalent, but the field has already been
long abused to mean this, and writing it helps in determining a decent cfr
time base when transcoding from a mkv where the video codec stores none (VP8).

backport r23284 by conrad


Originally committed as revision 23365 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-28 20:35:13 +00:00
Reinhard Tartler
057287c9cd Reduce the score for the aea demuxer probing function.
backport r23273 by banan


Originally committed as revision 23356 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-27 20:48:19 +00:00
Reinhard Tartler
1746657479 Fix detection of some stereo atrac files by not comparing the
block size mode and info byte.


backport r23272 by banan


Originally committed as revision 23355 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-27 20:47:48 +00:00
Reinhard Tartler
0ff244dea2 write 3gp perf tag for artist metadata
backport r23268 by bcoudurier


Originally committed as revision 23354 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-27 20:47:03 +00:00
Reinhard Tartler
0078216f07 change author metadata to artist in mov de/muxer
backport r23266 by bcoudurier


Originally committed as revision 23353 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-27 20:40:09 +00:00
Reinhard Tartler
12ca60ec15 Cosmetics : re-indent after last commit.
backport r23265 by jai_menon


Originally committed as revision 23327 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 20:56:34 +00:00
Reinhard Tartler
2c3439b008 FFplay : Avoid manipulating NULL data pointers so that future checks
remain valid. This fixes segfaults for those cases where data copy to 
this invalid pointer is attempted.



backport r23264 by jai_menon


Originally committed as revision 23326 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 20:55:57 +00:00
Reinhard Tartler
cdfe333493 Document CODEC_FLAG_EMU_EDGE and avcodec_align_dimensions interaction.
backport r23258 by reimar


Originally committed as revision 23321 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 18:34:19 +00:00
Reinhard Tartler
5f1a8c5e80 Update regression tests after removing track timecode scale from mkvenc
backport r23248 by conrad


Originally committed as revision 23320 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 18:33:38 +00:00
Reinhard Tartler
b5afece607 matroskaenc: Don't write track timecode scale
It's not required for mkv and unsupported in webm

backport r23247 by conrad


Originally committed as revision 23315 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 18:22:30 +00:00
Reinhard Tartler
6ae05f541a matroskadec: Allow unknown EBML doctype
backport r23246 by conrad


Originally committed as revision 23314 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 18:21:59 +00:00
James Zern
9d3bb3501d matroskadec: Support webm doctype
Patch by James Zern <jzern at google>

backport r23245 by conrad


Originally committed as revision 23313 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-25 18:20:59 +00:00
Diego Biurrun
0b374e40b1 Convert NaN to 0/0 in av_d2q
This fixes aspect ratio calculation for encoding from files with 0/0 stored,
common with ogg/theora

backport r23280 by conrad


Originally committed as revision 23283 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-24 02:27:43 +00:00
Diego Biurrun
a2a26d108b Enable AVFMT_GENERIC_INDEX for Ogg demuxer. This avoids the many
seeks needed for binary search when seeking to a previously seen
location.

backport r23279 by reimar


Originally committed as revision 23282 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-24 02:09:36 +00:00
Diego Biurrun
49aa53f2d7 Ignore generated files in the libswscale directory.
Originally committed as revision 23281 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-24 01:33:39 +00:00
Diego Biurrun
fb2fc3a4f0 Merge last round of Changelog updates for HTML 5 features.
Originally committed as revision 23278 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-23 23:28:24 +00:00
Diego Biurrun
36edfb558b Add VERSION file for 0.6 release.
Originally committed as revision 23274 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-23 22:07:22 +00:00
Diego Biurrun
4a7c99d6f3 small spelling fixes
Originally committed as revision 23271 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-23 21:30:21 +00:00
Reinhard Tartler
327afd42b4 In ogg muxer, pack multiple frames into one page, much lower overhead
backport r23231 by bcoudurier


Originally committed as revision 23263 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-23 19:12:27 +00:00
Reinhard Tartler
a22222dadb Open 2-pass logfile in binary mode for both reading and writing.
This fixes a regression on Windows introduced by r22769 in which the data read
from the file was not properly zero terminated. The file was read as text,
which caused the \r characters to be suppressed. Since the zero termination
happens at the end of the buffer, and there was one byte less read per line,
this caused the remaining space on the buffer to contain random data.


backport r23251 by ramiro


Originally committed as revision 23261 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-23 18:50:15 +00:00
Martin Storsjö
28631e9393 Mention RTP hinting in the 0.6 release notes
Originally committed as revision 23242 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:50:07 +00:00
Martin Storsjö
f8a6c707a5 Add an APIchanges note regarding the new rtp hinting flag
backport r23180 by mstorsjo


Originally committed as revision 23241 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:49:28 +00:00
Martin Storsjö
c9b841c2b1 Late bump of the minor version, for the addition of AVFMT_FLAG_RTP_HINT
backport r23179 by mstorsjo


Originally committed as revision 23240 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:48:17 +00:00
Martin Storsjö
15b20ef6f9 Add a changelog entry for the RTP hinting in the mov muxer
backport r23178 by mstorsjo


Originally committed as revision 23239 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:47:48 +00:00
Martin Storsjö
1dea5cfbb0 Use a heuristic for describing the RTP packets using sample data
backport r23165 by mstorsjo


Originally committed as revision 23238 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:46:52 +00:00
Martin Storsjö
d48539fb85 Add initial support for RTP hinting in the mov muxer
backport r23164 by mstorsjo


Originally committed as revision 23237 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:46:23 +00:00
Martin Storsjö
a14f86aac7 Make mov_write_packet non-static, add ff_ prefix
backport r23163 by mstorsjo


Originally committed as revision 23236 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:45:54 +00:00
Martin Storsjö
6438cb834c Move the mov muxer structures to a separate header
backport r23162 by mstorsjo


Originally committed as revision 23235 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:45:21 +00:00
Martin Storsjö
ed2b1f80fa Add a flag for enabling RTP hinting
backport r23161 by mstorsjo


Originally committed as revision 23234 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:44:43 +00:00
Martin Storsjö
3d24c2ce02 Make ff_sdp_write_media a lavf-internal function
This is in preparation for RTP hinting in the MOV muxer, where
it needs to be able to create SDP fragments for each media stream.


backport r23160 by mstorsjo


Originally committed as revision 23233 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 21:43:40 +00:00
Reinhard Tartler
7d8b893b7e Disable non interleaved avi code when there is no index available.
Fixes issue1956.


backport r23202 by michael


Originally committed as revision 23230 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 20:24:08 +00:00
Reinhard Tartler
e6ef35bc22 Make aac_decode_frame() consume zero padding at the end of a packet.
backport r23195 by alexc


Originally committed as revision 23229 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 20:23:22 +00:00
Reinhard Tartler
3588e6b7c5 Fix a crash when opening WMS RTSP streams
Fixes issue 1948


backport r23181 by mstorsjo


Originally committed as revision 23228 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 20:22:18 +00:00
Maxim Poliakovski
34a0dee17e Fix signedness of q_delta field of the IVIMbInfo.
Patch by Maxim max_pole () gmx * de


backport r23170 by benoit


Originally committed as revision 23227 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 20:20:01 +00:00
Janne Grunau
b9a35435a5 In ts demuxer, output pes packet as soon as they are complete.
This is needed for subtitles where packets are infrequent.
Patch by Janne Grunau, janne-ffmpeg at jannau dot net.


backport r23157 by bcoudurier


Originally committed as revision 23226 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 20:15:53 +00:00
Reinhard Tartler
85ed859e6b set avg frame rate in mkv demuxer
backport r23152 by bcoudurier


Originally committed as revision 23225 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 20:13:34 +00:00
Reinhard Tartler
fcaba83909 ffserver: Make sure a destination URL is set when creating the SDP
Debugged by Howard Chu, hyc at highlandsun dot com.


backport r23151 by mstorsjo


Originally committed as revision 23224 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:49:32 +00:00
Reinhard Tartler
daf864016d Clarify description for the MONOWHITE and MONOBLACK pixel formats.
backport r23144 by stefano


Originally committed as revision 23223 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:47:19 +00:00
Reinhard Tartler
7454142424 Clarify descriptions for RGB4, BGR4, NV12, NV21,
RGB48BE, and RGB48LE pixel formats.

backport r23142 by stefano


Originally committed as revision 23222 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:46:59 +00:00
Reinhard Tartler
a7216204eb Change MAX_READ_SIZE message during av_find_stream_info to DEBUG level.
It is not harmful and it scares too many users.


backport r23139 by bcoudurier


Originally committed as revision 23221 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:46:03 +00:00
Reinhard Tartler
cff1f85c3f Fix compilation of AC3 decoder if E-AC3 decoder was disabled.
backport r23131 by cehoyos


Originally committed as revision 23220 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:38:39 +00:00
Reinhard Tartler
dd92ff49c9 Make lsf_sf_expand() 4 times faster.
backport r23096 by michael


Originally committed as revision 23219 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:32:58 +00:00
Reinhard Tartler
b12045c8c7 Set cur_channel in the AAC encoder context where needed.
Most coder functions read it. Carting this around in the context may be
suboptimal; a refactor should be considered.


backport r23094 by alexc


Originally committed as revision 23218 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:27:16 +00:00
Janne Grunau
a70e7cc928 Add -ldl to libfaadbin_extralibs instead of libfaadbin_decoder_extralibs.
The latter does not exist and thus compilation fails.
patch by Janne Grunau, janne-ffmpeg jannau.net


backport r23087 by diego


Originally committed as revision 23217 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:25:46 +00:00
Nick Brereton
6f8fbbf8d7 mlpdec: Comment channel_params field in struct SubStream.
Patch by Nick Brereton <nick at nbrereton dot net>

backport r23085 by ramiro


Originally committed as revision 23216 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:22:49 +00:00
Nick Brereton
b68d9df427 mlpdec: Allocate channel decoding parameters for each substream. Some file
was encountered with a channel range that overlapped the previous substreams,
and the code assumed no such overlap was possible.
Patch by Nick Brereton <nick at nbrereton dot net>


backport r23084 by ramiro


Originally committed as revision 23215 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:22:32 +00:00
Reinhard Tartler
26fe0ec7d6 Adding missing () to mathops.h.
backport r23083 by michael


Originally committed as revision 23214 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:19:49 +00:00
Reinhard Tartler
7590718414 Replace more "m" constraints with MANGLE to fix compilation issues
with x86_32 gcc 4.4.4 and -fPIC.


backport r23082 by reimar


Originally committed as revision 23213 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:19:13 +00:00
Ronaldo Moura
37a0852009 blackfin: fix yuv422 to yuv420 conversion
The old code is correct only when stride = 2*width.

Patch by Ronaldo Moura <ronaldo d moura monity com br>


backport r31142 by mru


Originally committed as revision 23212 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:05:38 +00:00
Thomas Devanneaux
b50601e882 Check NAL unit size to avoid reading past the buffer.
This fixes issue1907

Patch by Thomas Devanneaux gmail(thomdev)


backport r23078 by benoit


Originally committed as revision 23211 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-21 18:03:57 +00:00
Reinhard Tartler
d1026ab506 Make the nut demuxer issue a more meaningful error message if it
cannot recognize the provided codec tag.

backport r23071 by stefano


Originally committed as revision 23205 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-20 22:46:11 +00:00
Reinhard Tartler
08a441d4ad bswap: 10L add missing parens around macro args
backport r23070 by mru


Originally committed as revision 23204 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-20 22:45:00 +00:00
Reinhard Tartler
369e37e900 qt-faststart: Avoid leaking memory if encountering a file with double ftyp atoms
backport r23065 by mstorsjo


Originally committed as revision 23190 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 13:51:54 +00:00
Reinhard Tartler
40630a90ed Fix build with swscale disabled
backport r23062 by mru


Originally committed as revision 23189 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 13:43:34 +00:00
Reinhard Tartler
dc04fb36d7 Remove hardcoded-tables hack for IA-64: with latest binutils that now actually
causes linking errors instead of avoiding them.


backport r23058 by reimar


Originally committed as revision 23188 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 13:30:30 +00:00
Reinhard Tartler
8740ed496c Another try for fixing/improving decode_video documentation.
backport r23057 by reimar


Originally committed as revision 23187 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 13:29:36 +00:00
Reinhard Tartler
77a023f422 Document cmdutils.c:print_error().
backport r23051 by stefano


Originally committed as revision 23186 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 11:46:39 +00:00
Reinhard Tartler
ba9cecf5e5 Favor chunk size over hitting the correct position after reading the chunk size in asf.
Fixes issue1923


backport r23040 by michael


Originally committed as revision 23185 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 11:44:45 +00:00
Reinhard Tartler
8ab1409a59 10l: store the result of clipping added in r23035
backport r23037 by alexc


Originally committed as revision 23184 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 11:39:02 +00:00
Reinhard Tartler
3589ac6334 Error out when too many bits per frame are requested.
backport r23036 by alexc


Originally committed as revision 23183 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 11:38:30 +00:00
Reinhard Tartler
dd68bb3cab Make the faac inspired quantizer search make sense for a slightly narrower definition of "make sense."
backport r23035 by alexc


Originally committed as revision 23182 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 11:38:07 +00:00
Reinhard Tartler
50dab63690 Reindent after the last commit.
backport r23034 by stefano


Originally committed as revision 23177 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 08:04:41 +00:00
Reinhard Tartler
da8e4f5f79 Simplify print_error(), directly use av_strerror()/strerror() for
printing the error code associated to FF_NETERROR(EPROTONOSUPPORT).

backport r23033 by stefano


Originally committed as revision 23176 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 08:04:25 +00:00
Reinhard Tartler
e0e0283778 Make print_error() use strerror() in case av_strerror() fails.
Should provide a meaningful error message for systems which do not
support strerror_r().

Fix roundup issue #1894.

backport r23032 by stefano


Originally committed as revision 23175 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 08:03:50 +00:00
Reinhard Tartler
1673bf86d6 Make av_strerror() return -1 even in the case when av_strerror_r() is
not defined.

This allows applications to check if av_strerror() cannot provide a
meaningful representation for the provided error code, without having
to actually check the filled string.

backport r23031 by stefano


Originally committed as revision 23174 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 08:03:22 +00:00
Reinhard Tartler
0db6f6cfed movenc: Write QuickTime chapters
backport r23022 by conrad


Originally committed as revision 23173 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 08:02:33 +00:00
Reinhard Tartler
9d08b20fea movenc: Swap positions of mov_write_header and mov_write_packet
backport r23021 by conrad


Originally committed as revision 23172 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 08:02:05 +00:00
Reinhard Tartler
a03dcb9682 mov: Read nero chapters
backport r23020 by conrad


Originally committed as revision 23171 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-19 07:58:17 +00:00
Martin Storsjö
26899578f1 Mention that both packetizers and depacketizers of H.263 and AMR were added
Originally committed as revision 23130 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-13 19:42:32 +00:00
Reinhard Tartler
eb9df4ac33 first draft of the release notes
Originally committed as revision 23117 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-13 16:37:21 +00:00
Reinhard Tartler
64c2076bfc update version string in changelog
Originally committed as revision 23114 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6
2010-05-13 16:26:55 +00:00
2341 changed files with 120188 additions and 326640 deletions

15
.gitignore vendored
View File

@@ -7,25 +7,15 @@
*-example
*-test
*_g
*.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*
@@ -34,23 +24,22 @@ libavformat/libavformat*
libavutil/avconfig.h
libavutil/libavutil*
libpostproc/libpostproc*
libswresample/libswresample*
libswscale/libswscale*
tests/audiogen
tests/base64
tests/data
tests/rotozoom
tests/seek_test
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
tools/trasher*.d
version.h

View File

@@ -1,7 +1,5 @@
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.
This file contains the name of the people who have contributed to
FFmpeg. The names are sorted alphabetically by last name.
Dénes Balatoni
Michel Bardiaux

538
Changelog
View File

@@ -1,435 +1,114 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version next:
version 0.6.5:
- vorbis: An additional defense in the Vorbis codec. (CVE-2011-3895)
- vorbisdec: Fix decoding bug with channel handling.
- matroskadec: Fix a bug where a pointer was cached to an array that might
later move due to a realloc(). (CVE-2011-3893)
- vorbis: Avoid some out-of-bounds reads. (CVE-2011-3893)
- vp3: fix oob read for negative tokens and memleaks on error, (CVE-2011-3892)
- vp3: fix streams with non-zero last coefficient.
version 0.10.8
- kmvc: Clip pixel position to valid range
- kmvc: use fixed sized arrays in the context
- indeo: use a typedef for the mc function pointer
- lavc: check for overflow in init_get_bits
- mjpegdec: properly report unsupported disabled features
- jpegls: return meaningful errors
- jpegls: factorize return paths
- jpegls: check the scan offset
- wavpack: validate samples size parsed in wavpack_decode_block
- ljpeg: use the correct number of components in yuv
- mjpeg: Validate sampling factors
- mjpegdec: validate parameters in mjpeg_decode_scan_progressive_ac
- wavpack: check packet size early
- wavpack: return meaningful errors
- apetag: use int64_t for filesize
- tiff: do not overread the source buffer
- Prepare for 0.8.8 Release
- smacker: fix an off by one in huff.length computation
- smacker: check the return value of smacker_decode_tree
- smacker: pad the extradata allocation
- smacker: check frame size validity
- vmdav: convert to bytestream2
- 4xm: don't rely on get_buffer() initializing the frame.
- 4xm: check the return value of read_huffman_tables().
- 4xm: use the correct logging context
- 4xm: reject frames not compatible with the declared version
- 4xm: check bitstream_size boundary before using it
- 4xm: do not overread the source buffer in decode_p_block
- avfiltergraph: check for sws opts being non-NULL before using them
- bmv: check for len being valid in bmv_decode_frame()
- dfa: check for invalid access in decode_wdlt()
- indeo3: check motion vectors
- indeo3: fix data size check
- indeo3: switch parsing the header to bytestream2
- lavf: make sure stream probe data gets freed.
- oggdec: fix faulty cleanup prototype
- oma: Validate sample rates
- qdm2: check that the FFT size is a power of 2
- rv10: check that extradata is large enough
- xmv: check audio track parameters validity
- xmv: do not leak memory in the error paths in xmv_read_header()
- aac: check the maximum number of channels
- indeo3: fix off by one in MV validity check, Bug #503
- id3v2: check for end of file while unescaping tags
- wav: Always seek to an even offset, Bug #500, LP: #1174737
- proresdec: support mixed interlaced/non-interlaced content
version 0.6.4:
- 4xm: Add a check in decode_i_frame to prevent buffer overreads
- wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
- swscale: #include "libavutil/mathematics.h"
- vp3dec: Check coefficient index in vp3_dequant() (CVE-2011-4352)
- svq1dec: call avcodec_set_dimensions() after dimensions changed. (CVE-2011-4579)
- commits related to CVE-2011-4353:
- vp6: Fix illegal read.
- vp6: Reset the internal state when aborting key frames header parsing
- vp6: Check for huffman tree build errors
- vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
- commits related to CVE-2011-4351:
- qdm2: check output buffer size before decoding
- Fix qdm2 decoder packet handling to match the api
- Fix out of bound reads in the QDM2 decoder.
- Check for out of bound writes in the QDM2 decoder.
- vmd: fix segfaults on corruped streams (CVE-2011-4364)
- rv34: Check for invalid slice offsets
- rv34: Fix potential overreads
- rv34: Avoid NULL dereference on corrupted bitstream
- rv10: Reject slices that does not have the same type as the first one
- oggdec: fix out of bound write in the ogg demuxer
- smacker: fix a few off by 1 errors
- Check for invalid VLC value in smacker decoder.
- Check and propagate errors when VLC trees cannot be built in smacker decoder.
- Fixed off by one packet size allocation in the smacker demuxer.
- Check for invalid packet size in the smacker demuxer.
- ape demuxer: fix segfault on memory allocation failure.
- Fixed size given to init_get_bits() in xan decoder.
- smacker demuxer: handle possible av_realloc() failure.
- Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
- indeo2: fail if input buffer too small
- indeo2: init_get_bits size in bits instead of bytes
- wavpack: Check error codes rather than working around error conditions.
- Fixed invalid writes and reads in wavpack decoder on corrupted bitstreams.
- cpu detection: avoid a signed overflow
- h264: correct implicit weight table computation for long ref pics
- h264: correct the check for invalid long term frame index in MMCO decode
- rv10/20: tell decoder to use edge emulation
- flvenc: use int64_t to store offsets
- VC-1: fix reading of custom PAR.
- h264: notice memory allocation failure
- libx264: do not set pic quality if no frame is output
- mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
- rv30: return AVERROR(EINVAL) instead of EINVAL
- Do not decode RV30 files if the extradata is too small
- aacps: skip some memcpy() if src and dst would be equal
- mpegts: fix Continuity Counter error detection
- alsa: fallback to buffer_size/4 for period_size.
- mxfenc: fix ignored drop flag in binary timecode representation.
- dca: set AVCodecContext frame_size for DTS audio
- H.264: fix overreads of qscale_table
- postprocess.c: filter name needs to be double 0 terminated
- Replace strncpy() with av_strlcpy() in libpostproc.
- jpegdec: actually search for and parse RSTn
- riff: Add mpgv MPEG-2 fourcc
- Added M701 codec_tag for mpeg2video
version 0.10.6:
version 0.6.3:
- many bug fixes that where found with Coverity
- The following CVE fixes where backported:
CVE-2012-2796, CVE-2012-2775, CVE-2012-2772, CVE-2012-2776,
CVE-2012-2779, CVE-2012-2787, CVE-2012-2794, CVE-2012-2800,
CVE-2012-2802, CVE-2012-2801, CVE-2012-2786, CVE-2012-2798,
CVE-2012-2793, CVE-2012-2789, CVE-2012-2788, CVE-2012-2790,
CVE-2012-2777, CVE-2012-2784
- hundreads of other bug fixes, some possibly security relevant,
see the git log for details.
- fix compilation with --enable-hardcoded-tables
- mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
- aac: add headers needed for log2f()
- lavf: inspect more frames for fps when container time base is coarse
- AMV: disable DR1 and don't override EMU_EDGE
(addresses http://seclists.org/bugtraq/2011/Apr/257)
- Fix memory (re)allocation in matroskadec.c (MSVR11-011/CVE-2011-3504)
- Fix some crashes with invalid bitstreams in the CAVS decoder
(CVE-2011-3362, CVE-2011-3973, CVE-2011-3974)
- Compilation fixes for gcc-4.6, testsuite now passes again
- Fix a heap corruption issue in the OGG decoder
- Backported the Android VisualOn AAC encoder wrapper from 0.7.2
version 0.10.5:
version 0.6.3:
- Several bugs and crashes have been fixed as well as build problems
with recent mingw64
- AMV: Fix possibly exploitable crash.
- Fix apparently exploitable race condition.
(addresses http://seclists.org/bugtraq/2011/Apr/257)
version 0.6.2:
- fix compilation with --enable-hardcoded-tables
- Fix invalid reads in VC-1 decoding (related to CVE-2011-0723)
- Do not attempt to decode APE file with no frames
(adresses http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
version 0.10.4:
version 0.6.1:
- Several bugs and crashes have been fixed
Note, CVE-2012-0851 and CVE-2011-3937 have been fixed in previous releases
version 0.10.3:
- Security fixes in the 4xm demuxer, avi demuxer, cook decoder,
mm demuxer, mpegvideo decoder, vqavideo decoder (CVE-2012-0947) and
xmv demuxer.
- Several bugs and crashes have been fixed in the following codecs: AAC,
APE, H.263, H.264, Indeo 4, Mimic, MJPEG, Motion Pixels Video, RAW,
TTA, VC1, VQA, WMA Voice, vqavideo.
- Several bugs and crashes have been fixed in the following formats:
ASF, ID3v2, MOV, xWMA
- This release additionally updates the following codecs to the
bytestream2 API, and therefore benefit from additional overflow
checks: truemotion2, utvideo, vqavideo
version 0.10.1
- Several security fixes, many bugfixes affecting many formats and
codecs, the list below is not complete.
- swapuv filter
- Several bugs and crashes have been fixed in the following codecs: AAC,
AC-3, ADPCM, AMR (both NB and WB), ATRAC3, CAVC, Cook, camstudio, DCA,
DPCM, DSI CIN, DV, EA TGQ, FLAC, fraps, G.722 (both encoder and
decoder), H.264, huvffyuv, BB JV decoder, Indeo 3, KGV1, LCL, the
libx264 wrapper, MJPEG, mp3on4, Musepack, MPEG1/2, PNG, QDM2, Qt RLE,
ROQ, RV10, RV30/RV34/RV40, shorten, smacker, subrip, SVQ3, TIFF,
Truemotion2, TTA, VC1, VMware Screen codec, Vorbis, VP5, VP6, WMA,
Westwood SNDx, XXAN.
- This release additionally updates the following codecs to the
bytestream2 API, and therefore benefit from additional overflow
checks: XXAN, ALG MM, TQG, SMC, Qt SMC, ROQ, PNG
- Several bugs and crashes have been fixed in the following formats:
AIFF, ASF, DV, Matroska, NSV, MOV, MPEG-TS, Smacker, Sony OpenMG, RM,
SWF.
- Libswscale has an potential overflow for large image size fixed.
- The following APIs have been added:
avcodec_is_open()
avformat_get_riff_video_tags()
avformat_get_riff_audio_tags()
Please see the file doc/APIchanges and the Doxygen documentation for
further information.
version 0.10:
- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
- SBaGen (SBG) binaural beats script demuxer
- OpenMG Audio muxer
- Timecode extraction in DV and MOV
- thumbnail video filter
- XML output in ffprobe
- asplit audio filter
- tinterlace video filter
- astreamsync audio filter
- amerge audio filter
- ISMV (Smooth Streaming) muxer
- GSM audio parser
- SMJPEG muxer
- XWD encoder and decoder
- Automatic thread count based on detection number of (available) CPU cores
- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
- ffprobe -show_error option
- Avid 1:1 10-bit RGB Packer codec
- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
- yuv4 libquicktime packed 4:2:0 encoder and decoder
- ffprobe -show_frames option
- silencedetect audio filter
- ffprobe -show_program_version, -show_library_versions, -show_versions options
- rv34: frame-level multi-threading
- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
- Improved PGS subtitle decoder
- dumpgraph option to lavfi device
- r210 and r10k encoders
- ffwavesynth decoder
- aviocat tool
- ffeval tool
version 0.9:
- openal input device added
- boxblur filter added
- BWF muxer
- Flash Screen Video 2 decoder
- lavfi input device added
- added avconv, which is almost the same for now, except
for a few incompatible changes in the options, which will hopefully make them
easier to use. The changes are:
* The options placement is now strictly enforced! While in theory the
options for ffmpeg should be given in [input options] -i INPUT [output
options] OUTPUT order, in practice it was possible to give output options
before the -i and it mostly worked. Except when it didn't - the behavior was
a bit inconsistent. In avconv, it is not possible to mix input and output
options. All non-global options are reset after an input or output filename.
* All per-file options are now truly per-file - they apply only to the next
input or output file and specifying different values for different files
will now work properly (notably -ss and -t options).
* All per-stream options are now truly per-stream - it is possible to
specify which stream(s) should a given option apply to. See the Stream
specifiers section in the avconv manual for details.
* In ffmpeg some options (like -newvideo/-newaudio/...) are irregular in the
sense that they're specified after the output filename instead of before,
like all other options. In avconv this irregularity is removed, all options
apply to the next input or output file.
* -newvideo/-newaudio/-newsubtitle options were removed. Not only were they
irregular and highly confusing, they were also redundant. In avconv the -map
option will create new streams in the output file and map input streams to
them. E.g. avconv -i INPUT -map 0 OUTPUT will create an output stream for
each stream in the first input file.
* The -map option now has slightly different and more powerful syntax:
+ Colons (':') are used to separate file index/stream type/stream index
instead of dots. Comma (',') is used to separate the sync stream instead
of colon.. This is done for consistency with other options.
+ It's possible to specify stream type. E.g. -map 0:a:2 creates an
output stream from the third input audio stream.
+ Omitting the stream index now maps all the streams of the given type,
not just the first. E.g. -map 0:s creates output streams for all the
subtitle streams in the first input file.
+ Since -map can now match multiple streams, negative mappings were
introduced. Negative mappings disable some streams from an already
defined map. E.g. '-map 0 -map -0:a:1' means 'create output streams for
all the stream in the first input file, except for the second audio
stream'.
* There is a new option -c (or -codec) for choosing the decoder/encoder to
use, which allows to precisely specify target stream(s) consistently with
other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
libvorbis sets the codec for the first audio stream and -c copy copies all
the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
aliases to -c:v/a/s
* It is now possible to precisely specify which stream should an AVOption
apply to. E.g. -b:v:0 2M sets the bitrate for the first video stream, while
-b:a 128k sets the bitrate for all audio streams. Note that the old -ab 128k
syntax is deprecated and will stop working soon.
* -map_chapters now takes only an input file index and applies to the next
output file. This is consistent with how all the other options work.
* -map_metadata now takes only an input metadata specifier and applies to
the next output file. Output metadata specifier is now part of the option
name, similarly to the AVOptions/map/codec feature above.
* -metadata can now be used to set metadata on streams and chapters, e.g.
-metadata:s:1 language=eng sets the language of the first stream to 'eng'.
This made -vlang/-alang/-slang options redundant, so they were removed.
* -qscale option now uses stream specifiers and applies to all streams, not
just video. I.e. plain -qscale number would now apply to all streams. To get
the old behavior, use -qscale:v. Also there is now a shortcut -q for -qscale
and -aq is now an alias for -q:a.
* -vbsf/-absf/-sbsf options were removed and replaced by a -bsf option which
uses stream specifiers. Use -bsf:v/a/s instead of the old options.
* -itsscale option now uses stream specifiers, so its argument is only the
scale parameter.
* -intra option was removed, use -g 0 for the same effect.
* -psnr option was removed, use -flags +psnr for the same effect.
* -vf option is now an alias to the new -filter option, which uses stream specifiers.
* -vframes/-aframes/-dframes options are now aliases to the new -frames option.
* -vtag/-atag/-stag options are now aliases to the new -tag option.
- XMV demuxer
- LOAS demuxer
- ashowinfo filter added
- Windows Media Image decoder
- amovie source added
- LATM muxer/demuxer
- Speex encoder via libspeex
- JSON output in ffprobe
- WTV muxer
- Optional C++ Support (needed for libstagefright)
- H.264 Decoding on Android via Stagefright
- Prores decoder
- BIN/XBIN/ADF/IDF text file decoder
- aconvert audio filter added
- audio support to lavfi input device added
- libcdio-paranoia input device for audio CD grabbing
- Apple ProRes decoder
- CELT in Ogg demuxing
- G.723.1 demuxer and decoder
- libmodplug support (--enable-libmodplug)
- VC-1 interlaced decoding
- libutvideo wrapper (--enable-libutvideo)
- aevalsrc audio source added
- Ut Video decoder
- Speex encoding via libspeex
- 4:2:2 H.264 decoding support
- 4:2:2 and 4:4:4 H.264 encoding with libx264
- Pulseaudio input device
- Prores encoder
- Video Decoder Acceleration (VDA) HWAccel module.
- replacement Indeo 3 decoder
- new ffmpeg option: -map_channel
- volume audio filter added
- earwax audio filter added
- libv4l2 support (--enable-libv4l2)
- TLS/SSL and HTTPS protocol support
- AVOptions API rewritten and documented
- most of CODEC_FLAG2_*, some CODEC_FLAG_* and many codec-specific fields in
AVCodecContext deprecated. Codec private options should be used instead.
- Properly working defaults in libx264 wrapper, support for native presets.
- Encrypted OMA files support
- Discworld II BMV decoding support
- VBLE Decoder
- OS X Video Decoder Acceleration (VDA) support
- compact and csv output in ffprobe
- pan audio filter
- IFF Amiga Continuous Bitmap (ACBM) decoder
- ass filter
- CRI ADX audio format muxer and demuxer
- Playstation Portable PMP format demuxer
- Microsoft Windows ICO demuxer
- life source
- PCM format support in OMA demuxer
- CLJR encoder
- new option: -report
- Dxtory capture format decoder
- cellauto source
- Simple segmenting muxer
- Indeo 4 decoder
- SMJPEG demuxer
version 0.8:
- many many things we forgot because we rather write code than changelogs
- WebM support in Matroska de/muxer
- low overhead Ogg muxing
- MMS-TCP support
- VP8 de/encoding via libvpx
- Demuxer for On2's IVF format
- Pictor/PC Paint decoder
- HE-AAC v2 decoder
- HE-AAC v2 encoding with libaacplus
- libfaad2 wrapper removed
- DTS-ES extension (XCh) decoding support
- native VP8 decoder
- RTSP tunneling over HTTP
- RTP depacketization of SVQ3
- -strict inofficial replaced by -strict unofficial
- ffplay -exitonkeydown and -exitonmousedown options added
- native GSM / GSM MS decoder
- RTP depacketization of QDM2
- ANSI/ASCII art playback system
- Lego Mindstorms RSO de/muxer
- libavcore added (and subsequently removed)
- SubRip subtitle file muxer and demuxer
- Chinese AVS encoding via libxavs
- ffprobe -show_packets option added
- RTP packetization of Theora and Vorbis
- RTP depacketization of MP4A-LATM
- RTP packetization and depacketization of VP8
- hflip filter
- Apple HTTP Live Streaming demuxer
- a64 codec
- MMS-HTTP support
- G.722 ADPCM audio encoder/decoder
- R10k video decoder
- ocv_smooth filter
- frei0r wrapper filter
- change crop filter syntax to width:height:x:y
- make the crop filter accept parametric expressions
- make ffprobe accept AVFormatContext options
- yadif filter
- blackframe filter
- Demuxer for Leitch/Harris' VR native stream format (LXF)
- RTP depacketization of the X-QT QuickTime format
- SAP (Session Announcement Protocol, RFC 2974) muxer and demuxer
- cropdetect filter
- ffmpeg -crop* options removed
- transpose filter added
- ffmpeg -force_key_frames option added
- demuxer for receiving raw rtp:// URLs without an SDP description
- single stream LATM/LOAS decoder
- setpts filter added
- Win64 support for optimized x86 assembly functions
- MJPEG/AVI1 to JPEG/JFIF bitstream filter
- ASS subtitle encoder and decoder
- IEC 61937 encapsulation for E-AC-3, TrueHD, DTS-HD (for HDMI passthrough)
- overlay filter added
- rename aspect filter to setdar, and pixelaspect to setsar
- IEC 61937 demuxer
- Mobotix .mxg demuxer
- frei0r source added
- hqdn3d filter added
- RTP depacketization of QCELP
- FLAC parser added
- gradfun filter added
- AMR-WB decoder
- replace the ocv_smooth filter with a more generic ocv filter
- Windows Televison (WTV) demuxer
- FFmpeg metadata format muxer and demuxer
- SubRip (srt) subtitle encoder and decoder
- floating-point AC-3 encoder added
- Lagarith decoder
- ffmpeg -copytb option added
- IVF muxer added
- Wing Commander IV movies decoder added
- movie source added
- Bink version 'b' audio and video decoder
- Bitmap Brothers JV playback system
- Apple HTTP Live Streaming protocol handler
- sndio support for playback and record
- Linux framebuffer input device added
- Chronomaster DFA decoder
- DPX image encoder
- MicroDVD subtitle file muxer and demuxer
- Playstation Portable PMP format demuxer
- fieldorder video filter added
- AAC encoding via libvo-aacenc
- AMR-WB encoding via libvo-amrwbenc
- xWMA demuxer
- Mobotix MxPEG decoder
- VP8 frame-multithreading
- NEON optimizations for VP8
- Lots of deprecated API cruft removed
- fft and imdct optimizations for AVX (Sandy Bridge) processors
- showinfo filter added
- SMPTE 302M AES3 audio decoder
- Apple Core Audio Format muxer
- 9bit and 10bit per sample support in the H.264 decoder
- 9bit and 10bit FFV1 encoding / decoding
- split filter added
- select filter added
- sdl output device added
- libmpcodecs video filter support (3 times as many filters than before)
- mpeg2 aspect ratio dection fixed
- libxvid aspect pickiness fixed
- Frame multithreaded decoding
- E-AC-3 audio encoder
- ac3enc: add channel coupling support
- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
- H264/MPEG frame-level multi-threading
- All av_metadata_* functions renamed to av_dict_* and moved to libavutil
- 4:4:4 H.264 decoding support
- 10-bit H.264 optimizations for x86
- lut, lutrgb, and lutyuv filters added
- buffersink libavfilter sink added
- Bump libswscale for recently reported ABI break
- New J2K encoder (via OpenJPEG)
version 0.7:
- all the changes for 0.8, but keeping API/ABI compatibility with the 0.6 release
- fix autodetection of E-AC-3 substream samples
- performance fix for seekable HTTP
- backport AAC-HE v2 from trunk
- add missing VP80 fourcc code for the VP8 codec
- small documentation fixes
- fix several potentially exploitable issues in the FLIC decoder
(addresses CVE-2010-3429)
version 0.6:
@@ -438,7 +117,7 @@ version 0.6:
- deprecated vhook subsystem removed
- deprecated old scaler removed
- VQF demuxer
- Alpha channel scaler
- alpha channel scaler
- PCX encoder
- RTP packetization of H.263
- RTP packetization of AMR
@@ -507,7 +186,11 @@ version 0.6:
- 35% faster VP3/Theora decoding
- faster AAC decoding
- faster H.264 decoding
- RealAudio 1.0 (14.4K) encoder
- WebM support in Matroska de/muxer
- low overhead Ogg muxing
- VP8 de/encoding via libvpx
- CODEC_CAP_EXPERIMENTAL added
version 0.5:
@@ -670,7 +353,7 @@ version 0.5:
- Gopher client support
- MXF D-10 muxer
- generic metadata API
- flash ScreenVideo2 encoder
version 0.4.9-pre1:
@@ -757,6 +440,7 @@ version 0.4.9-pre1:
- FunCom ISS audio file demuxer and according ADPCM decoding
version 0.4.8:
- MPEG-2 video encoding (Michael)
@@ -768,6 +452,7 @@ version 0.4.8:
- Duck DK3 and DK4 ADPCM audio decoders (Mike Melanson)
version 0.4.7:
- RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from mplayerhq
@@ -809,6 +494,7 @@ version 0.4.7:
.. And lots more new enhancements and fixes.
version 0.4.6:
- completely new integer only MPEG audio layer 1/2/3 decoder rewritten
@@ -846,6 +532,7 @@ version 0.4.6:
- 16-bit and 15-bit RGB/BGR/GBR support (Bisqwit)
version 0.4.5:
- some header fixes (Zdenek Kabelac <kabi at informatics.muni.cz>)
@@ -877,6 +564,7 @@ version 0.4.5:
- added JPEG image format support (input/output)
version 0.4.4:
- fixed some std header definitions (Bjorn Lindgren
@@ -902,6 +590,7 @@ version 0.4.4:
segfault)
version 0.4.3:
- BGR24 patch (initial patch by Jeroen Vreeken <pe1rxq at amsat.org>)
@@ -923,6 +612,7 @@ version 0.4.3:
instead of a protocol)
version 0.4.2:
- added H.263/MPEG-4/MSMPEG4 decoding support. MPEG-4 decoding support
@@ -953,6 +643,7 @@ version 0.4.1:
- factorized some motion estimation code
version 0.4.0:
- removing grab code from ffserver and moved it to ffmpeg. Added
@@ -981,11 +672,13 @@ version 0.4.0:
codecs
version 0.3.4:
- added stereo in MPEG audio encoder
version 0.3.3:
- added 'high quality' mode which use motion vectors. It can be used in
@@ -994,12 +687,14 @@ version 0.3.3:
bitrates and large GOP size
version 0.3.2: small fixes
- ASF fixes
- put_seek bug fix
version 0.3.1: added avi/divx support
- added AVI support
@@ -1007,4 +702,5 @@ version 0.3.1: added avi/divx support
- added sound for flash format (not tested)
version 0.3: initial public release

1068
Doxyfile

File diff suppressed because it is too large Load Diff

View File

@@ -9,7 +9,3 @@ path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
3) Type 'make install' to install all binaries and libraries you built.
NOTICE
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.

10
LICENSE
View File

@@ -14,7 +14,11 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are
- libpostproc
- optional MMX optimizations for YUV to RGB colorspace conversion in
libswscale/x86/yuv2rgb_template.c
- optional x86 optimizations in the files
libavcodec/x86/h264_deblock_sse2.asm
libavcodec/x86/h264_idct_sse2.asm
libavcodec/x86/idct_mmx.c
- the X11 grabber in libavdevice/x11grab.c
@@ -41,6 +45,6 @@ is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
version needs to be upgraded by passing --enable-version3 to configure.
The 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.
The nonfree external library libfaac 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

@@ -4,12 +4,6 @@ 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 upto date!
Names in () mean that the maintainer currently has no time to maintain the code.
A CC after the name means that the maintainer prefers to be CC-ed on patches
and related discussions.
Project Leader
==============
@@ -25,10 +19,7 @@ ffmpeg:
ffmpeg.c Michael Niedermayer
ffplay:
ffplay.c Marton Balint
ffprobe:
ffprobe.c Stefano Sabatini
ffplay.c Michael Niedermayer
ffserver:
ffserver.c, ffserver.h Baptiste Coudurier
@@ -43,14 +34,14 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Mike Melanson
website Robert Swain, Lou Logan
documentation Mike Melanson, Diego Biurrun
website Robert Swain
build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
project server Diego Biurrun, Mans Rullgard
mailinglists Michael Niedermayer, Baptiste Coudurier
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
release management Diego Biurrun, Reinhard Tartler
libavutil
@@ -87,8 +78,6 @@ Generic Parts:
bitstream.c, bitstream.h Michael Niedermayer
CABAC:
cabac.h, cabac.c Michael Niedermayer
codec names:
codec_names.sh Nicolas George
DSP utilities:
dsputils.c, dsputils.h Michael Niedermayer
entropy coding:
@@ -109,19 +98,17 @@ Generic Parts:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
libpostproc/* Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
Codecs:
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aasc.c Kostya Shishkov
aac*, sbr.h Alex Converse
ac3* Justin Ruggles
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann
apedec.c Kostya Shishkov
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3* Benjamin Larsson
bgmc.c, bgmc.h Thilo Borgmann
@@ -133,7 +120,6 @@ Codecs:
cinepak.c Roberto Togni
cljr Alex Beregszaszi
cook.c, cookdata.h Benjamin Larsson
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
dca.c Kostya Shishkov, Benjamin Larsson
dnxhd* Baptiste Coudurier
@@ -142,11 +128,9 @@ Codecs:
dv.c Roman Shaposhnik
eacmv*, eaidct*, eat* Peter Ross
ffv1.c Michael Niedermayer
ffwavesynth.c Nicolas George
flac* Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
g722.c Martin Storsjo
g726.c Roman Shaposhnik
gifdec.c Baptiste Coudurier
h264* Loren Merritt, Michael Niedermayer
@@ -160,19 +144,14 @@ Codecs:
interplayvideo.c Mike Melanson
ivi* Kostya Shishkov
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
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
libvorbis.c David Conrad
libxavs.c Stefan Gehrer
libx264.c Mans Rullgard, Jason Garrett-Glaser
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
@@ -190,7 +169,6 @@ Codecs:
nellymoserdec.c Benjamin Larsson
nuv.c Reimar Doeffinger
pcx.c Ivo van Poorten
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni, Benjamin Larsson
@@ -210,7 +188,6 @@ Codecs:
smc.c Mike Melanson
snow.c Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
targa.c Kostya Shishkov
@@ -222,18 +199,15 @@ Codecs:
tta.c Alex Beregszaszi, Jaikrishnan Menon
txd.c Ivo van Poorten
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov
vcr1.c Michael Niedermayer
vmnc.c Kostya Shishkov
vorbis_enc.c Oded Shimon
vorbis_dec.c Denes Balatoni, David Conrad
vorbis_dec.c Denes Balatoni
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
@@ -246,11 +220,8 @@ Codecs:
zmbv* Kostya Shishkov
Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Laurent Aimar
libstagefright.cpp Mohamed Naufal
vaapi* Gwenole Beauchesne
vda* Sebastien Zwickert
vdpau* Carl Eugen Hoyos
@@ -265,18 +236,6 @@ libavdevice
vfwcap.c Ramiro Polla
libavfilter
===========
Video filters:
graphdump.c Nicolas George
af_amerge.c Nicolas George
af_astreamsync.c Nicolas George
af_pan.c Nicolas George
vsrc_mandelbrot.c Michael Niedermayer
vf_yadif.c Michael Niedermayer
libavformat
===========
@@ -292,10 +251,8 @@ Muxers/Demuxers:
adtsenc.c Robert Swain
aiff.c Baptiste Coudurier
ape.c Kostya Shishkov
ass* Aurelien Jacobs
avi* Michael Niedermayer
bink.c Peter Ross
caf* Peter Ross
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dv.c Roman Shaposhnik
@@ -308,21 +265,17 @@ Muxers/Demuxers:
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
idcin.c Mike Melanson
idroqdec.c Mike Melanson
idroq.c Mike Melanson
iff.c Jaikrishnan Menon
ipmovie.c Mike Melanson
img2.c Michael Niedermayer
iss.c Stefan Gehrer
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
metadata* Aurelien Jacobs
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Michael Niedermayer, Baptiste Coudurier
movenc.c Michael Niedermayer, Baptiste Coudurier
@@ -333,14 +286,12 @@ Muxers/Demuxers:
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
mxfdec.c Tomas Härdin
nsvdec.c Francois Revol
nut.c Michael Niedermayer
nuv.c Reimar Doeffinger
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
psxstr.c Mike Melanson
pva.c Ivo van Poorten
r3d.c Baptiste Coudurier
@@ -349,16 +300,14 @@ Muxers/Demuxers:
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rtp.c, rtpenc.c Martin Storsjo
rtpdec_asf.* Ronald S. Bultje
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
rtp.c, rtpenc.c Luca Abeni
rtp_asf.* Ronald S. Bultje
rtp_mpv.*, rtp_aac.* Luca Abeni
rtsp.c Luca Barbato
sbgdec.c Nicolas George
sdp.c Martin Storsjo
sdp.c Luca Abeni
segafilm.c Mike Melanson
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
@@ -366,12 +315,10 @@ Muxers/Demuxers:
wav.c Michael Niedermayer
wc3movie.c Mike Melanson
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
Protocols:
http.c Ronald S. Bultje
mms*.c Ronald S. Bultje
udp.c Luca Abeni
@@ -382,6 +329,7 @@ Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard
BeOS Francois Revol
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Luca Barbato
@@ -392,38 +340,20 @@ Sparc Roman Shaposhnik
x86 Michael Niedermayer
Releases
========
GnuPG Fingerprints of maintainers and others who have svn write access
======================================================================
0.9 Michael Niedermayer
GnuPG Fingerprints of maintainers and contributors
==================================================
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
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
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Luca Barbato 6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
Panagiotis Issaris 515C E262 10A8 FDCE 5481 7B9C 3AD7 D9A5 071D B3A9
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Reimar Döffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
Tomas Härdin D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD

342
Makefile
View File

@@ -1,72 +1,62 @@
MAIN_MAKEFILE=1
include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(SRC_PATH)
vpath %.S $(SRC_PATH)
vpath %.asm $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %/fate_config.sh.template $(SRC_PATH)
SRC_DIR = $(SRC_PATH_BARE)
vpath %.texi $(SRC_PATH_BARE)
PROGS-$(CONFIG_FFMPEG) += ffmpeg
PROGS-$(CONFIG_FFPLAY) += ffplay
PROGS-$(CONFIG_FFPROBE) += ffprobe
PROGS-$(CONFIG_FFSERVER) += ffserver
PROGS := $(PROGS-yes:%=%$(EXESUF))
INSTPROGS = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
OBJS = $(PROGS-yes:%=%.o) cmdutils.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr base64
HOSTPROGS := $(TESTTOOLS:%=tests/%)
TOOLS = qt-faststart trasher
TOOLS-$(CONFIG_ZLIB) += cws2fws
PROGS := $(addsuffix $(EXESUF), $(PROGS-yes))
PROGS_G = $(addsuffix _g$(EXESUF), $(PROGS-yes))
OBJS = $(addsuffix .o, $(PROGS-yes)) cmdutils.o
MANPAGES = $(addprefix doc/, $(addsuffix .1, $(PROGS-yes)))
TOOLS = $(addprefix tools/, $(addsuffix $(EXESUF), cws2fws pktdumper probetest qt-faststart trasher))
HOSTPROGS = $(addprefix tests/, audiogen videogen rotozoom tiny_psnr)
BASENAMES = ffmpeg ffplay ffprobe ffserver
ALLPROGS = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLPROGS_G = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
ALLMANPAGES = $(BASENAMES:%=%.1)
ALLPROGS = $(addsuffix $(EXESUF), $(BASENAMES))
ALLPROGS_G = $(addsuffix _g$(EXESUF), $(BASENAMES))
ALLMANPAGES = $(addsuffix .1, $(BASENAMES))
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)
SKIPHEADERS = cmdutils_common_opts.h
include $(SRC_PATH)/common.mak
include common.mak
FF_LDFLAGS := $(FFLDFLAGS)
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
all: $(PROGS)
ALL_TARGETS-$(CONFIG_DOC) += documentation
$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@$(PROGSSUF)
$(STRIP) $@$(PROGSSUF)
ifdef PROGS
INSTALL_TARGETS-yes += install-progs install-data
INSTALL_TARGETS-$(CONFIG_DOC) += install-man
endif
INSTALL_PROGS_TARGETS-$(CONFIG_SHARED) = install-libs
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
tools/cws2fws$(EXESUF): ELIBS = -lz
config.h: .config
.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
$(PROGS): %$(EXESUF): %_g$(EXESUF)
$(CP) $< $@
$(STRIP) $@
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
HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS
define RESET
$(1) :=
@@ -76,28 +66,30 @@ endef
define DOSUBDIR
$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
include $(SRC_PATH)/library.mak
include $(1)/Makefile
endef
$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
ffplay.o: CFLAGS += $(SDL_CFLAGS)
ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)
ffserver_g$(EXESUF): FF_LDFLAGS += $(FFSERVERLDFLAGS)
%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
%_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
$(LD) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
OBJDIRS += tools
tools/%$(EXESUF): tools/%.o
$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
-include $(wildcard tools/*.d)
tools/%.o: tools/%.c
$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_O) $<
VERSION_SH = $(SRC_PATH)/version.sh
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
ffplay.o ffplay.d: CFLAGS += $(SDL_CFLAGS)
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
VERSION_SH = $(SRC_PATH_BARE)/version.sh
GIT_LOG = $(SRC_PATH_BARE)/.git/logs/HEAD
SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
.version: $(wildcard $(GIT_LOG) $(SVN_ENTRIES)) $(VERSION_SH) config.mak
.version: M=@
version.h .version:
@@ -107,26 +99,39 @@ version.h .version:
# force version.sh to run whenever version might have changed
-include .version
ifdef PROGS
install: install-progs install-data
endif
alltools: $(TOOLS)
install: install-libs install-headers
documentation: $(addprefix doc/, developer.html faq.html ffmpeg-doc.html \
ffplay-doc.html ffprobe-doc.html ffserver-doc.html \
general.html libavfilter.html $(ALLMANPAGES))
install-libs: install-libs-yes
doc/%.html: TAG = HTML
doc/%.html: doc/%.texi
$(M)cd doc && texi2html -monolithic -number $(<:doc/%=%)
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
doc/%.pod: TAG = POD
doc/%.pod: doc/%-doc.texi
$(M)doc/texi2pod.pl $< $@
install-progs: install-progs-yes $(PROGS)
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
install: $(INSTALL_TARGETS-yes)
install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
$(INSTALL) -c -m 755 $(PROGS) "$(BINDIR)"
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
uninstall: uninstall-progs uninstall-data uninstall-man
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
@@ -134,44 +139,215 @@ uninstall-progs:
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
testclean:
$(RM) -r tests/vsynth1 tests/vsynth2 tests/data
$(RM) $(addprefix tests/,$(CLEANSUFFIXES))
$(RM) tests/seek_test$(EXESUF) tests/seek_test.o
$(RM) $(addprefix tests/,$(addsuffix $(HOSTEXESUF),audiogen videogen rotozoom tiny_psnr))
clean:: testclean
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) coverage.info
$(RM) -r coverage-html
$(RM) doc/*.html doc/*.pod doc/*.1
$(RM) $(TOOLS)
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .version version.h libavutil/avconfig.h
$(RM) version.h config.* libavutil/avconfig.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
# regression tests
coverage-html: coverage.info
$(Q)mkdir -p $@
$(Q)genhtml -o $@ $<
$(Q)touch $@
check: test checkheaders
check: all alltools checkheaders examples testprogs fate
fulltest test: codectest lavftest seektest
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/tests/Makefile
FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
SEEK_REFFILE = $(SRC_PATH)/tests/seek.regression.ref
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
ENCDEC = $(and $(CONFIG_$(1)_ENCODER),$(CONFIG_$(1)_DECODER))
MUXDEM = $(and $(CONFIG_$(1)_MUXER),$(CONFIG_$(or $(2),$(1))_DEMUXER))
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
VCODEC_TESTS =
VCODEC_TESTS-$(call ENCDEC,ASV1) += asv1
VCODEC_TESTS-$(call ENCDEC,ASV2) += asv2
VCODEC_TESTS-$(call ENCDEC,DNXHD) += dnxhd_1080i dnxhd_720p dnxhd_720p_rd
VCODEC_TESTS-$(call ENCDEC,DVVIDEO) += dv dv50
VCODEC_TESTS-$(call ENCDEC,FFV1) += ffv1
VCODEC_TESTS-$(call ENCDEC,FLASHSV) += flashsv
VCODEC_TESTS-$(call ENCDEC,FLV) += flv
VCODEC_TESTS-$(call ENCDEC,H261) += h261
VCODEC_TESTS-$(call ENCDEC,H263) += h263 h263p
VCODEC_TESTS-$(call ENCDEC,HUFFYUV) += huffyuv
VCODEC_TESTS-$(call ENCDEC,JPEGLS) += jpegls
VCODEC_TESTS-$(call ENCDEC,MJPEG) += mjpeg ljpeg
VCODEC_TESTS-$(call ENCDEC,MPEG1VIDEO) += mpeg mpeg1b
VCODEC_TESTS-$(call ENCDEC,MPEG2VIDEO) += mpeg2 mpeg2thread
VCODEC_TESTS-$(call ENCDEC,MPEG4) += mpeg4 mpeg4adv mpeg4nr mpeg4thread error rc
VCODEC_TESTS-$(call ENCDEC,MSMPEG4V1) += msmpeg4
VCODEC_TESTS-$(call ENCDEC,MSMPEG4V2) += msmpeg4v2
VCODEC_TESTS-$(call ENCDEC,ROQ) += roq
VCODEC_TESTS-$(call ENCDEC,RV10) += rv10
VCODEC_TESTS-$(call ENCDEC,RV20) += rv20
VCODEC_TESTS-$(call ENCDEC,SNOW) += snow snowll
VCODEC_TESTS-$(call ENCDEC,SVQ1) += svq1
VCODEC_TESTS-$(call ENCDEC,WMV1) += wmv1
VCODEC_TESTS-$(call ENCDEC,WMV2) += wmv2
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
ACODEC_TESTS =
ACODEC_TESTS-$(call ENCDEC,AC3) += ac3
ACODEC_TESTS-$(call ENCDEC,ADPCM_G726) += g726
ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_QT) += adpcm_ima_qt
ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_WAV) += adpcm_ima_wav
ACODEC_TESTS-$(call ENCDEC,ADPCM_MS) += adpcm_ms
ACODEC_TESTS-$(call ENCDEC,ADPCM_SWF) += adpcm_swf
ACODEC_TESTS-$(call ENCDEC,ADPCM_YAMAHA) += adpcm_yam
ACODEC_TESTS-$(call ENCDEC,ALAC) += alac
ACODEC_TESTS-$(call ENCDEC,FLAC) += flac
ACODEC_TESTS-$(call ENCDEC,MP2) += mp2
ACODEC_TESTS-$(call ENCDEC,PCM_S16LE) += pcm # fixme
ACODEC_TESTS-$(call ENCDEC,WMAV1) += wmav1
ACODEC_TESTS-$(call ENCDEC,WMAV1) += wmav2
.PHONY: all all-yes alltools check *clean config examples install*
.PHONY: testprogs uninstall*
LAVF_TESTS =
LAVF_TESTS-$(call MUXDEM,AIFF) += aiff
LAVF_TESTS-$(call MUXDEM,PCM_ALAW) += alaw
LAVF_TESTS-$(call MUXDEM,ASF) += asf
LAVF_TESTS-$(call MUXDEM,AU) += au
LAVF_TESTS-$(call MUXDEM,AVI) += avi
LAVF_TESTS-$(call ENCDEC,BMP) += bmp
LAVF_TESTS-$(call MUXDEM,DV) += dv_fmt
LAVF_TESTS-$(call MUXDEM,FFM) += ffm
LAVF_TESTS-$(call MUXDEM,FLV) += flv_fmt
LAVF_TESTS-$(call ENCDEC,GIF) += gif
LAVF_TESTS-$(call MUXDEM,GXF) += gxf
LAVF_TESTS-$(call ENCDEC,MJPEG) += jpg
LAVF_TESTS-$(call MUXDEM,MATROSKA) += mkv
LAVF_TESTS-$(call MUXDEM,MMF) += mmf
LAVF_TESTS-$(call MUXDEM,MOV) += mov
LAVF_TESTS-$(call MUXDEM,MPEG1SYSTEM,MPEGPS) += mpg
LAVF_TESTS-$(call MUXDEM,PCM_MULAW) += mulaw
LAVF_TESTS-$(call MUXDEM,MXF) += mxf
LAVF_TESTS-$(call MUXDEM,NUT) += nut
LAVF_TESTS-$(call MUXDEM,OGG) += ogg
LAVF_TESTS-$(call ENCDEC,PBM) += pbmpipe
LAVF_TESTS-$(call ENCDEC,PCX) += pcx
LAVF_TESTS-$(call ENCDEC,PGM) += pgm pgmpipe
LAVF_TESTS-$(call MUXDEM,RAWVIDEO) += pixfmt
LAVF_TESTS-$(call ENCDEC,PPM) += ppm ppmpipe
LAVF_TESTS-$(call MUXDEM,RM) += rm
LAVF_TESTS-$(call ENCDEC,SGI) += sgi
LAVF_TESTS-$(call MUXDEM,SWF) += swf
LAVF_TESTS-$(call ENCDEC,TARGA) += tga
LAVF_TESTS-$(call ENCDEC,TIFF) += tiff
LAVF_TESTS-$(call MUXDEM,MPEGTS) += ts
LAVF_TESTS-$(call MUXDEM,VOC) += voc
LAVF_TESTS-$(call MUXDEM,WAV) += wav
LAVF_TESTS-$(call MUXDEM,YUV4MPEGPIPE) += yuv4mpeg
LAVFI_TESTS = \
crop \
crop_scale \
crop_scale_vflip \
crop_vflip \
null \
scale200 \
scale500 \
vflip \
vflip_crop \
vflip_vflip \
ACODEC_TESTS := $(addprefix regtest-, $(ACODEC_TESTS) $(ACODEC_TESTS-yes))
VCODEC_TESTS := $(addprefix regtest-, $(VCODEC_TESTS) $(VCODEC_TESTS-yes))
LAVF_TESTS := $(addprefix regtest-, $(LAVF_TESTS) $(LAVF_TESTS-yes))
LAVFI_TESTS := $(addprefix regtest-, $(LAVFI_TESTS) $(LAVFI_TESTS-yes))
CODEC_TESTS = $(VCODEC_TESTS) $(ACODEC_TESTS)
codectest: $(CODEC_TESTS)
lavftest: $(LAVF_TESTS)
lavfitest: $(LAVFI_TESTS)
$(ACODEC_TESTS): regtest-aref
$(VCODEC_TESTS): regtest-vref
$(LAVF_TESTS) $(LAVFI_TESTS): regtest-ref
REFFILE = $(SRC_PATH)/tests/ref/$(1)/$(2:regtest-%=%)
RESFILE = tests/data/$(2:regtest-%=%).$(1).regression
define CODECTEST_CMD
$(SRC_PATH)/tests/codec-regression.sh $@ vsynth1 tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(SRC_PATH)/tests/codec-regression.sh $@ vsynth2 tests/vsynth2 "$(TARGET_EXEC)" "$(TARGET_PATH)"
endef
regtest-ref: regtest-aref regtest-vref
regtest-vref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
$(CODECTEST_CMD)
regtest-aref: ffmpeg$(EXESUF) tests/data/asynth1.sw
@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(VCODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
@echo "TEST VCODEC $(@:regtest-%=%)"
@$(CODECTEST_CMD)
@diff -u -w $(call REFFILE,vsynth1,$@) $(call RESFILE,vsynth1,$@)
@diff -u -w $(call REFFILE,vsynth2,$@) $(call RESFILE,vsynth2,$@)
$(ACODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
@echo "TEST ACODEC $(@:regtest-%=%)"
@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
@diff -u -w $(call REFFILE,acodec,$@) $(call RESFILE,acodec,$@)
$(LAVF_TESTS):
@echo "TEST LAVF $(@:regtest-%=%)"
@$(SRC_PATH)/tests/lavf-regression.sh $@ lavf tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
@diff -u -w $(call REFFILE,lavf,$@) $(call RESFILE,lavf,$@)
$(LAVFI_TESTS):
@echo "TEST LAVFI $(@:regtest-%=%)"
@$(SRC_PATH)/tests/lavfi-regression.sh $@ lavfi tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
@diff -u -w $(call REFFILE,lavfi,$@) $(call RESFILE,lavfi,$@)
seektest: codectest lavftest tests/seek_test$(EXESUF)
$(SRC_PATH)/tests/seek-regression.sh $(SRC_PATH) "$(TARGET_EXEC)" "$(TARGET_PATH)"
ffservertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/data/asynth1.sw
@echo
@echo "Unfortunately ffserver is broken and therefore its regression"
@echo "test fails randomly. Treat the results accordingly."
@echo
$(SRC_PATH)/tests/ffserver-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/ffserver.conf
tests/vsynth1/00.pgm: tests/videogen$(HOSTEXESUF)
mkdir -p tests/vsynth1
$(BUILD_ROOT)/$< 'tests/vsynth1/'
tests/vsynth2/00.pgm: tests/rotozoom$(HOSTEXESUF)
mkdir -p tests/vsynth2
$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF)
mkdir -p tests/data
$(BUILD_ROOT)/$< $@
tests/seek_test$(EXESUF): tests/seek_test.o $(FF_DEP_LIBS)
$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
ifdef SAMPLES
include $(SRC_PATH_BARE)/tests/fate.mak
fate: $(FATE_TESTS)
$(FATE_TESTS): ffmpeg$(EXESUF)
@echo "TEST FATE $(@:fate-%=%)"
@$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)'
else
fate:
@echo "SAMPLES not specified, cannot run FATE"
endif
.PHONY: documentation *test regtest-* zlib-error alltools check config

195
RELEASE
View File

@@ -1 +1,194 @@
0.10.9
Release Notes
=============
* 0.6 "Works with HTML5" June, 2010
General notes
-------------
This release focuses on improvements for the new multimedia elements in HTML5.
The H.264 and Theora decoders are now significantly faster, the Vorbis decoder
has seen important updates and this release supports Google's newly released
libvpx library for the VP8 codec and WebM container.
Other important changes are additions of decoders including, but not limited to,
Intel Indeo 5, WMA Pro, WMA Voice and HE-AAC.
See the Changelog file for a list of significant changes.
Please note that our policy on bug reports has not changed. We still only accept
bug reports against HEAD of the FFmpeg trunk repository. If you are experiencing
any issues with any formally released version of FFmpeg, please try a current
version of the development code to check if the issue still exists. If it does,
make your report against the development code following the usual bug reporting
guidelines.
API and other notable Changes
-----------------------------
Please see the file doc/APIchanges for programmer-centric information.
Notable changes:
- deprecated vhook subsystem removed
- deprecated old scaler removed
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- RTMP support in libavformat
- -formats option split into -formats, -codecs, -bsfs, and -protocols
- ffprobe tool
- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
- CODEC_CAP_EXPERIMENTAL added
Added Codecs:
-------------
- VQF demuxer
- PCX encoder
- CorePNG decoding support
- 8088flex TMV demuxer and decoder
- enable symbol versioning by default for linkers that support it
- V210 decoder and encoder
- QCP demuxer
- SoX native format muxer and demuxer
- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
- DPX image decoder
- Electronic Arts Madcow decoder
- DivX (XSUB) subtitle encoder
- experimental AAC encoder
- Wave64 demuxer
- IEC-61937 compatible Muxer
- TwinVQ decoder
- Bluray (PGS) subtitle decoder
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
- IV8 demuxer
- CDG demuxer and decoder
- R210 decoder
- Auravision Aura 1 and 2 decoders
- Deluxe Paint Animation playback system
- SIPR decoder
- Adobe Filmstrip muxer and demuxer
- RTP packetization and depacketization of H.263 and AMR
- Bink demuxer and audio/video decoders
- IFF PBM/ILBM bitmap decoder
- Indeo 5 decoder
- WMA Voice decoder
- AMR-NB decoder
- RTSP muxer
- HE-AAC v1 decoder
- Kega Game Video (KGV1) decoder
- Psygnosis YOP demuxer and video decoder
- RTP hinting in the mov/3gp/mp4 muxer
- VP8 decoding via libvpx
Notable license related changes
-------------------------------
- remaining GPL parts in AC-3 decoder converted to LGPL
- libswscale can now be compiled in LGPL mode
* 0.6.1
General notes
-------------
This point release includes some updates to make the 0.6 release series usable
for users that need to retain the existing behavior as closely as possible.
The changes follow below:
Bugfixes
--------
- fix autodetection of E-AC-3 substream samples
- performance fix for seekable HTTP
- add missing VP80 fourcc code for the VP8 codec
- small documentation fixes
- fix several potentially exploitable issues in the FLIC decoder
(addresses CVE-2010-3429)
HE-AAC v2 backport
------------------
This release includes a backport of the AAC decoder from trunk, which
enables proper playback of HE-AAC v2 media.
* 0.6.2
General notes
-------------
This is a maintenance-only release that addresses a small number of security
and portability issues. Distributors and system integrators are encouraged
to update and share their patches against this branch.
Security fixes
--------------
Programming errors in container and codec implementations may lead to
denial of service or the execution of arbitrary code if the user is
tricked into opening a malformed media file or stream.
Affected and updated have been the implementations of the following
codecs and container formats:
- VC1 decoder (Change related to CVE-2011-0723)
- APE decoder (cf. http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
* 0.6.3
General notes
-------------
This is a mostly maintenance-only release that addresses a small number
of bugs such as security and compilation issues. Moreover, this release
has been updated to work with gcc-4.6 and the VisualOn AAC encoder has
been backported from the Libav 0.7.2 release. Distributors and system
integrators are encouraged to update and share their patches against
this branch.
For a full list of changes please see the Changelog file.
* 0.6.4
General notes
-------------
This mostly maintenance-only release that addresses a number a number of
bugs such as security and compilation issues that have been brought to
our attention. Among other (rather minor) fixes, this release features
fixes for the QDM2 decoder (CVE-2011-4351), vp3 decoder (CVE-2011-4352),
DoS in the VP5/VP6 decoders (CVE-2011-4353), a buffer overflow in the
Sierra VMD decoder CVE-2011-4364, and a safety fix in the svq1 decoder
(CVE-2011-4579).
Distributors and system integrators are encouraged
to update and share their patches against this branch. For a full list
of changes please see the Changelog file.
* 0.6.5
General notes
-------------
This mostly maintenance-only release that addresses a number a number of
bugs such as security and compilation issues that have been brought to
our attention. Among other (rather minor) fixes, this release features
fixes for the VP3 decoder (CVE-2011-3892), vorbis decoder, and matroska
demuxer (CVE-2011-3893 and CVE-2011-3895).
Distributors and system integrators are encouraged
to update and share their patches against this branch. For a full list
of changes please see the Changelog file.

View File

@@ -1 +1 @@
0.10.9
0.6.5

1152
cmdutils.c

File diff suppressed because it is too large Load Diff

View File

@@ -22,17 +22,11 @@
#ifndef FFMPEG_CMDUTILS_H
#define FFMPEG_CMDUTILS_H
#include <stdint.h>
#include <inttypes.h>
#include "libavcodec/avcodec.h"
#include "libavfilter/avfilter.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#ifdef __MINGW32__
#undef main /* We don't want SDL to override our main() */
#endif
/**
* program name, defined by the program for show_version().
*/
@@ -43,32 +37,12 @@ extern const char program_name[];
*/
extern const int program_birth_year;
/**
* this year, defined by the program for show_banner()
*/
extern const int this_year;
extern const char **opt_names;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern struct SwsContext *sws_opts;
extern AVDictionary *format_opts, *codec_opts;
/**
* Initialize the cmdutils option system, in particular
* allocate the *_opts contexts.
*/
void init_opts(void);
/**
* Uninitialize the cmdutils option system, in particular
* free the *_opts contexts and their contents.
*/
void uninit_opts(void);
/**
* Trivial log callback.
* Only suitable for opt_help and similar since it lacks prefix handling.
*/
void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
/**
* Fallback for options that are not explicitly handled, these will be
@@ -77,44 +51,37 @@ void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
int opt_default(const char *opt, const char *arg);
/**
* Set the libav* libraries log level.
* Sets the libav* libraries log level.
*/
int opt_loglevel(const char *opt, const char *arg);
int opt_report(const char *opt);
int opt_max_alloc(const char *opt, const char *arg);
int opt_codec_debug(const char *opt, const char *arg);
/**
* Limit the execution time.
*/
int opt_timelimit(const char *opt, const char *arg);
/**
* Parse a string and return its corresponding value as a double.
* Exit from the application if the string cannot be correctly
* Parses a string and returns its corresponding value as a double.
* Exits from the application if the string cannot be correctly
* parsed or the corresponding value is invalid.
*
* @param context the context of the value to be set (e.g. the
* corresponding command line option name)
* corresponding commandline option name)
* @param numstr the string to be parsed
* @param type the type (OPT_INT64 or OPT_FLOAT) as which the
* string should be parsed
* @param min the minimum valid accepted value
* @param max the maximum valid accepted value
*/
double parse_number_or_die(const char *context, const char *numstr, int type,
double min, double max);
double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);
/**
* Parse a string specifying a time and return its corresponding
* value as a number of microseconds. Exit from the application if
* Parses a string specifying a time and returns its corresponding
* value as a number of microseconds. Exits from the application if
* the string cannot be correctly parsed.
*
* @param context the context of the value to be set (e.g. the
* corresponding command line option name)
* corresponding commandline option name)
* @param timestr the string to be parsed
* @param is_duration a flag which tells how to interpret timestr, if
* not zero timestr is interpreted as a duration, otherwise as a
@@ -122,19 +89,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
*
* @see parse_date()
*/
int64_t parse_time_or_die(const char *context, const char *timestr,
int is_duration);
typedef struct SpecifierOpt {
char *specifier; /**< stream/chapter/program/... specifier */
union {
uint8_t *str;
int i;
int64_t i64;
float f;
double dbl;
} u;
} SpecifierOpt;
int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);
typedef struct {
const char *name;
@@ -149,101 +104,38 @@ typedef struct {
#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_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_FUNC2 0x0400
#define OPT_INT64 0x0800
#define OPT_EXIT 0x1000
union {
void *dst_ptr;
int (*func_arg)(const char *, const char *);
int (*func2_arg)(void *, const char *, const char *);
size_t off;
void (*func_arg)(const char *); //FIXME passing error code as int return would be nicer then exit() in the func
int *int_arg;
char **str_arg;
float *float_arg;
int (*func2_arg)(const char *, const char *);
int64_t *int64_arg;
} u;
const char *help;
const char *argname;
} OptionDef;
void show_help_options(const OptionDef *options, const char *msg, int mask,
int value);
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
* children.
*/
void show_help_children(const AVClass *class, int flags);
/**
* Parse the command line arguments.
*
* @param optctx an opaque options context
* Parses the command line arguments.
* @param options Array with the definitions required to interpret every
* option of the form: -option_name [argument]
* option of the form: -<option_name> [<argument>]
* @param parse_arg_function Name of the function called to process every
* argument without a leading option name flag. NULL if such arguments do
* not have to be processed.
*/
void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
void (* parse_arg_function)(void *optctx, const char*));
void parse_options(int argc, char **argv, const OptionDef *options,
void (* parse_arg_function)(const char*));
void set_context_opts(void *ctx, void *opts_ctx, int flags);
/**
* Parse one given option.
*
* @return on success 1 if arg was consumed, 0 otherwise; negative number on error
*/
int parse_option(void *optctx, const char *opt, const char *arg,
const OptionDef *options);
/**
* Find the '-loglevel' option in the command line args and apply it.
*/
void parse_loglevel(int argc, char **argv, const OptionDef *options);
/**
* Check if the given stream matches a stream specifier.
*
* @param s Corresponding format context.
* @param st Stream from s to be checked.
* @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
*
* @return 1 if the stream matches, 0 if it doesn't, <0 on error
*/
int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
/**
* Filter out options for given codec.
*
* Create a new options dictionary containing only the options from
* opts which apply to the codec with ID codec_id.
*
* @param s Corresponding format context.
* @param st A stream from s for which the options should be filtered.
* @return a pointer to the created dictionary
*/
AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
AVFormatContext *s, AVStream *st);
/**
* Setup AVCodecContext options for avformat_find_stream_info().
*
* Create an array of dictionaries, one dictionary for each stream
* contained in s.
* Each dictionary will contain the options from codec_opts which can
* be applied to the corresponding stream codec context.
*
* @return pointer to the created array of dictionaries, NULL if it
* cannot be created
*/
AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
AVDictionary *codec_opts);
/**
* Print an error message to stderr, indicating filename and a human
* Prints an error message to stderr, indicating filename and a human
* readable description of the error code err.
*
* If strerror_r() is not available the use of this function in a
@@ -253,128 +145,79 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
*/
void print_error(const char *filename, int err);
void list_fmts(void (*get_fmt_string)(char *buf, int buf_size, int fmt), int nb_fmts);
/**
* Print the program banner to stderr. The banner contents depend on the
* Prints the program banner to stderr. The banner contents depend on the
* current version of the repository and of the libav* libraries used by
* the program.
*/
void show_banner(int argc, char **argv, const OptionDef *options);
void show_banner(void);
/**
* Print the version of the program to stdout. The version message
* Prints the version of the program to stdout. The version message
* depends on the current versions of the repository and of the libav*
* libraries.
* This option processing function does not utilize the arguments.
*/
int opt_version(const char *opt, const char *arg);
void show_version(void);
/**
* Print the license of the program to stdout. The license depends on
* Prints 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 opt_license(const char *opt, const char *arg);
void show_license(void);
/**
* Print a listing containing all the formats supported by the
* program.
* This option processing function does not utilize the arguments.
*/
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 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 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 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 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 opt_pix_fmts(const char *opt, const char *arg);
/**
* Print a listing containing all the sample formats supported by the
* Prints a listing containing all the formats supported by the
* program.
*/
int show_sample_fmts(const char *opt, const char *arg);
void show_formats(void);
/**
* Return a positive value if a line read from standard input
* starts with [yY], otherwise return 0.
* Prints a listing containing all the codecs supported by the
* program.
*/
void show_codecs(void);
/**
* Prints a listing containing all the filters supported by the
* program.
*/
void show_filters(void);
/**
* Prints a listing containing all the bit stream filters supported by the
* program.
*/
void show_bsfs(void);
/**
* Prints a listing containing all the protocols supported by the
* program.
*/
void show_protocols(void);
/**
* Prints a listing containing all the pixel formats supported by the
* program.
*/
void show_pix_fmts(void);
/**
* Returns a positive value if reads from standard input a line
* starting with [yY], otherwise returns 0.
*/
int read_yesno(void);
/**
* Read the file with name filename, and put its content in a newly
* Reads the file with name filename, and puts its content in a newly
* allocated 0-terminated buffer.
*
* @param bufptr location where pointer to buffer is returned
* @param size location where size of buffer is returned
* @param bufptr puts here the pointer to the newly allocated buffer
* @param size puts here the size of the newly allocated buffer
* @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);
int read_file(const char *filename, char **bufptr, size_t *size);
/**
* Get a file corresponding to a preset file.
*
* If is_path is non-zero, look for the file in the path preset_name.
* Otherwise search for a file named arg.ffpreset in the directories
* $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined
* 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.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
* @param preset_name name of the preset to search
* @param is_path tell if preset_name is a filename path
* @param codec_name name of the codec for which to look for the
* preset, may be NULL
*/
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_program() on failure.
*
* @param elem_size size in bytes of each element
* @param size new element count will be written here
* @return reallocated array
*/
void *grow_array(void *array, int elem_size, int *size, int new_size);
#endif /* CMDUTILS_H */
#endif /* FFMPEG_CMDUTILS_H */

View File

@@ -1,18 +1,13 @@
{ "L", OPT_EXIT, {(void*)opt_license}, "show license" },
{ "h", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "?", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "help", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "-help", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "version", OPT_EXIT, {(void*)opt_version}, "show version" },
{ "formats" , OPT_EXIT, {(void*)opt_formats }, "show available formats" },
{ "codecs" , OPT_EXIT, {(void*)opt_codecs }, "show available codecs" },
{ "bsfs" , OPT_EXIT, {(void*)opt_bsfs }, "show available bit stream filters" },
{ "protocols", OPT_EXIT, {(void*)opt_protocols}, "show available protocols" },
{ "filters", OPT_EXIT, {(void*)opt_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {(void*)opt_pix_fmts }, "show available pixel formats" },
{ "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
{ "loglevel", HAS_ARG, {(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" },
{ "L", OPT_EXIT, {(void*)show_license}, "show license" },
{ "h", OPT_EXIT, {(void*)show_help}, "show help" },
{ "?", OPT_EXIT, {(void*)show_help}, "show help" },
{ "help", OPT_EXIT, {(void*)show_help}, "show help" },
{ "-help", OPT_EXIT, {(void*)show_help}, "show help" },
{ "version", OPT_EXIT, {(void*)show_version}, "show version" },
{ "formats" , OPT_EXIT, {(void*)show_formats }, "show available formats" },
{ "codecs" , OPT_EXIT, {(void*)show_codecs }, "show available codecs" },
{ "bsfs" , OPT_EXIT, {(void*)show_bsfs }, "show available bit stream filters" },
{ "protocols", OPT_EXIT, {(void*)show_protocols}, "show available protocols" },
{ "filters", OPT_EXIT, {(void*)show_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {(void*)show_pix_fmts }, "show available pixel formats" },
{ "loglevel", HAS_ARG | OPT_FUNC2, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },

View File

@@ -6,64 +6,60 @@
all: all-yes
ifndef SUBDIR
vpath %.c $(SRC_DIR)
vpath %.h $(SRC_DIR)
vpath %.S $(SRC_DIR)
vpath %.asm $(SRC_DIR)
vpath %.v $(SRC_DIR)
ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
BUILD_ROOT_REL = .
else
BUILD_ROOT_REL = ..
endif
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX AS YASM AR LD HOSTCC STRIP CP
BRIEF = CC AS YASM AR LD HOSTCC STRIP CP
SILENT = DEPCC YASMDEP RM RANLIB
MSG = $@
M = @$(call ECHO,$(TAG),$@);
$(foreach VAR,$(BRIEF), \
$(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
$(foreach VAR,$(SILENT),$(eval $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CFLAGS)
CXXFLAGS := $(CFLAGS) $(CXXFLAGS)
YASMFLAGS += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
HOSTCFLAGS += $(IFLAGS)
LDFLAGS := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
define COMPILE
$($(1)DEP)
$($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
CPPFLAGS := -I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
%.o: %.c
$(COMPILE_C)
%.o: %.cpp
$(COMPILE_CXX)
%.s: %.c
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
$(CCDEP)
$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_DEPFLAGS) -c $(CC_O) $<
%.o: %.S
$(COMPILE_S)
$(ASDEP)
$(AS) $(CPPFLAGS) $(ASFLAGS) $(AS_DEPFLAGS) -c -o $@ $<
%.ho: %.h
$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
%$(EXESUF): %.c
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
install: install-libs install-headers
install-libs: install-libs-yes
uninstall: uninstall-libs uninstall-headers
.PHONY: all depend dep *clean install* uninstall* examples testprogs
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
@@ -75,55 +71,39 @@ endif
OBJS-$(HAVE_MMX) += $(MMX-OBJS-yes)
CFLAGS += $(CFLAGS-yes)
OBJS += $(OBJS-yes)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(EXTRALIBS)
FFEXTRALIBS := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
FFLDFLAGS := $(addprefix -L$(BUILD_ROOT)/lib,$(ALLFFLIBS)) $(LDFLAGS)
EXAMPLES := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
EXAMPLES := $(addprefix $(SUBDIR),$(addsuffix -example$(EXESUF),$(EXAMPLES)))
OBJS := $(addprefix $(SUBDIR),$(sort $(OBJS)))
TESTOBJS := $(addprefix $(SUBDIR),$(TESTOBJS))
TESTPROGS := $(addprefix $(SUBDIR),$(addsuffix -test$(EXESUF),$(TESTPROGS)))
HOSTOBJS := $(addprefix $(SUBDIR),$(addsuffix .o,$(HOSTPROGS)))
HOSTPROGS := $(addprefix $(SUBDIR),$(addsuffix $(HOSTEXESUF),$(HOSTPROGS)))
DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
DEP_LIBS := $(foreach NAME,$(FFLIBS),$(BUILD_ROOT_REL)/lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
SKIPHEADERS += $(addprefix $(ARCH)/,$(ARCH_HEADERS))
SKIPHEADERS := $(addprefix $(SUBDIR),$(SKIPHEADERS-) $(SKIPHEADERS))
checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
alltools: $(TOOLS)
$(HOSTOBJS): %.o: %.c
$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
DEPS := $(OBJS:.o=.d)
depend dep: $(DEPS)
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver
DISTCLEANSUFFIXES = *.pc
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) $(TESTOBJS:.o=.d))
-include $(wildcard $(DEPS))

1832
configure vendored

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File diff suppressed because it is too large Load Diff

View File

@@ -1,64 +0,0 @@
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/libavfilter.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
ifdef HAVE_MAKEINFO
DOCS += $(TXTPAGES)
endif
all-$(CONFIG_DOC): documentation
documentation: $(DOCS)
TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
doc/%.html: TAG = HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
$(Q)$(TEXIDEP)
$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi
$(Q)$(TEXIDEP)
$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
$(DOCS): | doc
OBJDIRS += doc
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
uninstall: uninstall-man
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
clean::
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
-include $(wildcard $(DOCS:%=%.d))
.PHONY: documentation

View File

@@ -1,49 +0,0 @@
Release Notes
=============
* 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.
Bugreports against FFmpeg git master or the most recent FFmpeg release are
accepted. If you are experiencing issues with any formally released version of
FFmpeg, please try git master to check if the issue still exists. If it does,
make your report against the development code following the usual bug reporting
guidelines.
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.

90
doc/TODO Normal file
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@@ -0,0 +1,90 @@
ffmpeg TODO list:
----------------
Fabrice's TODO list: (unordered)
-------------------
Short term:
- use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work
- add RTSP regression test (both client and server)
- make ffserver allocate AVFormatContext
- clean up (incompatible change, for 0.5.0):
* AVStream -> AVComponent
* AVFormatContext -> AVInputStream/AVOutputStream
* suppress rate_emu from AVCodecContext
- add new float/integer audio filterting and conversion : suppress
CODEC_ID_PCM_xxc and use CODEC_ID_RAWAUDIO.
- fix telecine and frame rate conversion
Long term (ask me if you want to help):
- commit new imgconvert API and new PIX_FMT_xxx alpha formats
- commit new LGPL'ed float and integer-only AC3 decoder
- add WMA integer-only decoder
- add new MPEG4-AAC audio decoder (both integer-only and float version)
Michael's TODO list: (unordered) (if anyone wanna help with sth, just ask)
-------------------
- optimize H264 CABAC
- more optimizations
- simper rate control
Francois' TODO list: (unordered, without any timeframe)
-------------------
- test MACE decoder against the openquicktime one as suggested by A'rpi
- BeOS audio input grabbing backend
- BeOS video input grabbing backend
- publish my BeOS libposix on BeBits so I can officially support ffserver :)
- check the whole code for thread-safety (global and init stuff)
Philip'a TODO list: (alphabetically ordered) (please help)
------------------
- Add a multi-ffm filetype so that feeds can be recorded into multiple files rather
than one big file.
- Authenticated users support -- where the authentication is in the URL
- Change ASF files so that the embedded timestamp in the frames is right rather
than being an offset from the start of the stream
- Make ffm files more resilient to changes in the codec structures so that you
can play old ffm files.
Baptiste's TODO list:
-----------------
- mov edit list support (AVEditList)
- YUV 10 bit per component support "2vuy"
- mxf muxer
- mpeg2 non linear quantizer
unassigned TODO: (unordered)
---------------
- use AVFrame for audio codecs too
- rework aviobuf.c buffering strategy and fix url_fskip
- generate optimal huffman tables for mjpeg encoding
- fix ffserver regression tests
- support xvids motion estimation
- support x264s motion estimation
- support x264s rate control
- SNOW: non translational motion compensation
- SNOW: more optimal quantization
- SNOW: 4x4 block support
- SNOW: 1/8 pel motion compensation support
- SNOW: iterative motion estimation based on subsampled images
- SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves
- SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder
- SNOW: think about/analyize how to make snow use multiple cpus/threads
- SNOW: finish spec
- FLAC: lossy encoding (viterbi and naive scalar quantization)
- libavfilter
- JPEG2000 decoder & encoder
- MPEG4 GMC encoding support
- macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res)
- regression tests for codecs which do not have an encoder (I+P-frame bitstream in svn)
- add support for using mplayers video filters to ffmpeg
- H264 encoder
- per MB ratecontrol (so VCD and such do work better)
- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
- generic audio mixing API
- extract PES packetizer from PS muxer and use it for new TS muxer
- implement automatic AVBistreamFilter activation
- make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651)
- merge imdct and windowing, the current code does considerable amounts of redundant work

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@@ -1,168 +0,0 @@
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.

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@@ -19,6 +19,7 @@ integer.c 128bit integer math
lls.c
mathematics.c greatest common divisor, integer sqrt, integer log2, ...
mem.c memory allocation routines with guaranteed alignment
softfloat.c
Headers:
bswap.h big/little/native-endian conversion code

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@@ -1,91 +0,0 @@
@chapter Bitstream Filters
@c man begin BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option @code{--list-bsfs}.
You can disable all the bitstream filters using the configure option
@code{--disable-bsfs}, and selectively enable any bitstream filter using
the option @code{--enable-bsf=BSF}, or you can disable a particular
bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
@section chomp
@section dump_extradata
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format ("mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
@section imx_dump_header
@section mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
@example
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
@end example
Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG
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 -vbsf mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpega_dump_header
@section movsub
@section mp3_header_compress
@section mp3_header_decompress
@section noise
@section remove_extradata
@c man end BITSTREAM FILTERS

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@@ -1,50 +0,0 @@
FFmpeg currently uses a custom build system, this text attempts to document
some of its obscure features and options.
Makefile variables:
V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
Makefile targets:
all
Default target, builds all the libraries and the executables.
fate
Run the fate test suite, note you must have installed it
fate-list
Will list all fate/regression test targets
install
Install headers, libraries and programs.
libavformat/output-example
Build the libavformat basic example.
libavcodec/api-example
Build the libavcodec basic example.
libswscale/swscale-test
Build the swscale self-test (useful also as example).
Useful standard make commands:
make -t <target>
Touch all files that otherwise would be build, this is useful to reduce
unneeded rebuilding when changing headers, but note you must force rebuilds
of files that actually need it by hand then.
make -j<num>
rebuild with multiple jobs at the same time. Faster on multi processor systems
make -k
continue build in case of errors, this is useful for the regression tests
sometimes but note it will still not run all reg tests.

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@@ -1,63 +0,0 @@
@chapter Decoders
@c man begin DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available decoders using the configure option @code{--list-decoders}.
You can disable all the decoders with the configure option
@code{--disable-decoders} and selectively enable / disable single decoders
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-codecs} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@chapter Video Decoders
@c man begin VIDEO DECODERS
A description of some of the currently available video decoders
follows.
@section rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
@subsection Options
@table @option
@item top @var{top_field_first}
Specify the assumed field type of the input video.
@table @option
@item -1
the video is assumed to be progressive (default)
@item 0
bottom-field-first is assumed
@item 1
top-field-first is assumed
@end table
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
@section ffwavesynth
Internal wave synthetizer.
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.
@c man end AUDIO DECODERS

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@@ -1,108 +0,0 @@
@chapter Demuxers
@c man begin DEMUXERS
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 "--list-demuxers".
You can disable all the demuxers using the configure option
"--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 "-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.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen
@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
script looks like that:
@example
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
@end example
A SBG script can mix absolute and relative timestamps. If the script uses
either only absolute timestamps (including the script start time) or only
relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of
timestamps, then the @var{NOW} reference for relative timestamps will be
taken from the current time of day at the time the script is read, and the
script layout will be frozen according to that reference. That means that if
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.
@c man end INPUT DEVICES

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@@ -2,12 +2,11 @@
@settitle Developer Documentation
@titlepage
@sp 7
@center @titlefont{Developer Documentation}
@sp 3
@end titlepage
@top
@contents
@chapter Developers Guide
@@ -34,57 +33,59 @@ 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
There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
@item Submitting Patches to the main developer mailing list
see @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the @ref{Coding Rules}.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@section Coding Rules
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
Indent size is 4.
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
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also
compiles with several other compilers, such as the Compaq ccc compiler
or Sun Studio 9, and we would like to keep it that way unless it would
be exceedingly involved. To ensure compatibility, please do not use any
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
Indent size is 4.
The presentation is one inspired by 'indent -i4 -kr -nut'.
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.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
rejected by the Subversion repository.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@subsection Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
Comments: Use the JavaDoc/Doxygen
format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
@code{//!} with @code{///} and similar. Also @@ syntax should be employed
for markup commands, i.e. use @code{@@param} and not @code{\param}.
@example
/**
* @@file
* @@file mpeg.c
* MPEG codec.
* @@author ...
*/
@@ -112,97 +113,11 @@ int myfunc(int my_parameter)
...
@end example
@subsection C language features
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
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
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 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 following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For variables and functions declared as @code{static} no prefixes are required.
@item
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 used internally across multiple libraries, use
@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
@item
For exported names, each library has its own prefixes. Just check the existing
code and name accordingly.
@end itemize
@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.
@end itemize
@subsection Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
(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
@@ -234,9 +149,9 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
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 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!
Do not change behavior of the program (renaming options etc) 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
@@ -264,13 +179,10 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
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
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
If you apply a patch by someone else, include the name and email address in
the log message. Since the ffmpeg-cvslog mailing list is publicly
archived you should add some SPAM protection to the email address. Send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
@@ -299,7 +211,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav*
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
@@ -326,14 +238,12 @@ 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
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
First, (@pxref{Coding Rules}) above if you did not yet.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-)
When you submit your patch, try to send a unified diff (diff '-up'
option). We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
@@ -345,13 +255,13 @@ for us and greatly increases your chances of getting your patch applied.
Use the patcheck tool of FFmpeg to check your patch.
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.
Run the regression tests before submitting a patch so that you can
verify that there are no big 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}
@url{http://lists.mplayerhq.hu/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
@@ -379,14 +289,14 @@ send a reminder by email. Your patch should eventually be dealt with.
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 in @file{avcodec.h} or
@file{avformat.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the CodecID to @file{avcodec.h}?
@item
If it has a fourCC, did you add it to @file{libavformat/riff.c},
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?
@@ -401,31 +311,22 @@ send a reminder by email. Your patch should eventually be dealt with.
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?
@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 "svn add" the appropriate files before commiting?
@end enumerate
@section patch submission checklist
@enumerate
@item
Does @code{make fate} pass with the patch applied?
Do the regression tests pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
Does @code{make checkheaders} pass with the patch applied?
@item
Did you sign off your patch? (git commit -s)
See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
of sign off.
Is the patch a unified diff?
@item
Did you provide a clear git commit log message?
Is the patch against latest FFmpeg SVN?
@item
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
Are you subscribed to ffmpeg-dev?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
@@ -441,6 +342,9 @@ send a reminder by email. Your patch should eventually be dealt with.
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
Is the patch created from the root of the source tree, so it can be
applied with @code{patch -p0}?
@item
Does the patch not mix functional and cosmetic changes?
@item
@@ -477,15 +381,13 @@ send a reminder by email. Your patch should eventually be dealt with.
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider to add a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm
Did you provide a suggestion for a clear commit log message?
@end enumerate
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
clear note that the patch is not for SVN.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
@@ -498,23 +400,34 @@ After a patch is approved it will be committed to the repository.
We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
be rejected. Instead, submit significant changes or new features as
separate patches.
@anchor{Regression tests}
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
Running 'make fate' accomplishes this, please see @url{fate.html} for details.
The regression tests build a synthetic video stream and a synthetic
audio stream. These are then encoded and decoded with all codecs or
formats. The CRC (or MD5) of each generated file is recorded in a
result file. A 'diff' is launched to compare the reference results and
the result file. The output is checked immediately after each test
has run.
The regression tests then go on to test the FFserver code with a
limited set of streams. It is important that this step runs correctly
as well.
Run 'make test' to test all the codecs and formats. Commands like
'make regtest-mpeg2' can be used to run a single test. By default,
make will abort if any test fails. To run all tests regardless,
use make -k. To get a more verbose output, use 'make V=1 test' or
'make V=2 test'.
Run 'make fulltest' to test all the codecs, formats and FFserver.
[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified

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</div>
<div id="footer">
Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
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<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
<meta http-equiv="X-UA-Compatible" content="IE=9"/>
<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
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<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
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@chapter Encoders
@c man begin ENCODERS
Encoders are configured elements in FFmpeg which allow the encoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available encoders using the configure option @code{--list-encoders}.
You can disable all the encoders with the configure option
@code{--disable-encoders} and selectively enable / disable single encoders
with the options @code{--enable-encoder=@var{ENCODER}} /
@code{--disable-encoder=@var{ENCODER}}.
The option @code{-codecs} of the ff* tools will display the list of
enabled encoders.
@c man end ENCODERS
@chapter Audio Encoders
@c man begin AUDIO ENCODERS
A description of some of the currently available audio encoders
follows.
@section ac3 and ac3_fixed
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed}
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option @code{-acodec ac3_fixed} in order to use it.
@subsection AC-3 Metadata
The AC-3 metadata options are used to set parameters that describe the audio,
but in most cases do not affect the audio encoding itself. Some of the options
do directly affect or influence the decoding and playback of the resulting
bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for
audio data, and will thus affect the quality of the output. Those will be
indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available
documents.
@itemize
@item @uref{http://www.atsc.org/cms/standards/a_52-2010.pdf,A/52:2010 - Digital Audio Compression (AC-3) (E-AC-3) Standard}
@item @uref{http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf,A/54 - Guide to the Use of the ATSC Digital Television Standard}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf,Dolby Metadata Guide}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines}
@end itemize
@subsubsection Metadata Control Options
@table @option
@item -per_frame_metadata @var{boolean}
Allow Per-Frame Metadata. Specifies if the encoder should check for changing
metadata for each frame.
@table @option
@item 0
The metadata values set at initialization will be used for every frame in the
stream. (default)
@item 1
Metadata values can be changed before encoding each frame.
@end table
@end table
@subsubsection Downmix Levels
@table @option
@item -center_mixlev @var{level}
Center Mix Level. The amount of gain the decoder should apply to the center
channel when downmixing to stereo. This field will only be written to the
bitstream if a center channel is present. The value is specified as a scale
factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6dB gain
@end table
@item -surround_mixlev @var{level}
Surround Mix Level. The amount of gain the decoder should apply to the surround
channel(s) when downmixing to stereo. This field will only be written to the
bitstream if one or more surround channels are present. The value is specified
as a scale factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.500
Apply -6dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Audio Production Information
Audio Production Information is optional information describing the mixing
environment. Either none or both of the fields are written to the bitstream.
@table @option
@item -mixing_level @var{number}
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the @code{room_type} option is not the default value, the @code{mixing_level}
option must not be -1.
@item -room_type @var{type}
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the @code{mixing_level}
option and the @code{room_type} option have the default values.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx large
Large Room
@item 2
@itemx small
Small Room
@end table
@end table
@subsubsection Other Metadata Options
@table @option
@item -copyright @var{boolean}
Copyright Indicator. Specifies whether a copyright exists for this audio.
@table @option
@item 0
@itemx off
No Copyright Exists (default)
@item 1
@itemx on
Copyright Exists
@end table
@item -dialnorm @var{value}
Dialogue Normalization. Indicates how far the average dialogue level of the
program is below digital 100% full scale (0 dBFS). This parameter determines a
level shift during audio reproduction that sets the average volume of the
dialogue to a preset level. The goal is to match volume level between program
sources. A value of -31dB will result in no volume level change, relative to
the source volume, during audio reproduction. Valid values are whole numbers in
the range -31 to -1, with -31 being the default.
@item -dsur_mode @var{mode}
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
(Pro Logic). This field will only be written to the bitstream if the audio
stream is stereo. Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx off
Not Dolby Surround Encoded
@item 2
@itemx on
Dolby Surround Encoded
@end table
@item -original @var{boolean}
Original Bit Stream Indicator. Specifies whether this audio is from the
original source and not a copy.
@table @option
@item 0
@itemx off
Not Original Source
@item 1
@itemx on
Original Source (default)
@end table
@end table
@subsection Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the @code{center_mixlev}
and @code{surround_mixlev} options if it supports the Alternate Bit Stream
Syntax.
@subsubsection Extended Bitstream Information - Part 1
@table @option
@item -dmix_mode @var{mode}
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx ltrt
Lt/Rt Downmix Preferred
@item 2
@itemx loro
Lo/Ro Downmix Preferred
@end table
@item -ltrt_cmixlev @var{level}
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lt/Rt mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -ltrt_surmixlev @var{level}
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lt/Rt mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@item -loro_cmixlev @var{level}
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lo/Ro mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -loro_surmixlev @var{level}
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lo/Ro mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Extended Bitstream Information - Part 2
@table @option
@item -dsurex_mode @var{mode}
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround EX processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Surround EX Off
@item 2
@itemx off
Dolby Surround EX On
@end table
@item -dheadphone_mode @var{mode}
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
option does @b{NOT} mean the encoder will actually apply Dolby Headphone
processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Headphone Off
@item 2
@itemx off
Dolby Headphone On
@end table
@item -ad_conv_type @var{type}
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
conversion.
@table @option
@item 0
@itemx standard
Standard A/D Converter (default)
@item 1
@itemx hdcd
HDCD A/D Converter
@end table
@end table
@subsection Other AC-3 Encoding Options
@table @option
@item -stereo_rematrixing @var{boolean}
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This
is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.
@end table
@subsection Floating-Point-Only AC-3 Encoding Options
These options are only valid for the floating-point encoder and do not exist
for the fixed-point encoder due to the corresponding features not being
implemented in fixed-point.
@table @option
@item -channel_coupling @var{boolean}
Enables/Disables use of channel coupling, which is an optional AC-3 feature
that increases quality by combining high frequency information from multiple
channels into a single channel. The per-channel high frequency information is
sent with less accuracy in both the frequency and time domains. This allows
more bits to be used for lower frequencies while preserving enough information
to reconstruct the high frequencies. This option is enabled by default for the
floating-point encoder and should generally be left as enabled except for
testing purposes or to increase encoding speed.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@item 0
@itemx off
Disable Channel Coupling
@item 1
@itemx on
Enable Channel Coupling
@end table
@item -cpl_start_band @var{number}
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a
value higher than the bandwidth is used, it will be reduced to 1 less than the
coupling end band. If @var{auto} is used, the start band will be determined by
the encoder based on the bit rate, sample rate, and channel layout. This option
has no effect if channel coupling is disabled.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@end table
@end table
@c man end AUDIO ENCODERS
@chapter Video Encoders
@c man begin VIDEO ENCODERS
A description of some of the currently available video encoders
follows.
@section libvpx
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libvpx}.
@subsection Options
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
@table @option
@item threads
g_threads
@item profile
g_profile
@item vb
rc_target_bitrate
@item g
kf_max_dist
@item keyint_min
kf_min_dist
@item qmin
rc_min_quantizer
@item qmax
rc_max_quantizer
@item bufsize, vb
rc_buf_sz
@code{(bufsize * 1000 / vb)}
rc_buf_optimal_sz
@code{(bufsize * 1000 / vb * 5 / 6)}
@item rc_init_occupancy, vb
rc_buf_initial_sz
@code{(rc_init_occupancy * 1000 / vb)}
@item rc_buffer_aggressivity
rc_undershoot_pct
@item skip_threshold
rc_dropframe_thresh
@item qcomp
rc_2pass_vbr_bias_pct
@item maxrate, vb
rc_2pass_vbr_maxsection_pct
@code{(maxrate * 100 / vb)}
@item minrate, vb
rc_2pass_vbr_minsection_pct
@code{(minrate * 100 / vb)}
@item minrate, maxrate, vb
@code{VPX_CBR}
@code{(minrate == maxrate == vb)}
@item crf
@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
@item quality
@table @option
@item @var{best}
@code{VPX_DL_BEST_QUALITY}
@item @var{good}
@code{VPX_DL_GOOD_QUALITY}
@item @var{realtime}
@code{VPX_DL_REALTIME}
@end table
@item speed
@code{VP8E_SET_CPUUSED}
@item nr
@code{VP8E_SET_NOISE_SENSITIVITY}
@item mb_threshold
@code{VP8E_SET_STATIC_THRESHOLD}
@item slices
@code{VP8E_SET_TOKEN_PARTITIONS}
@item Alternate reference frame related
@table @option
@item vp8flags altref
@code{VP8E_SET_ENABLEAUTOALTREF}
@item @var{arnr_max_frames}
@code{VP8E_SET_ARNR_MAXFRAMES}
@item @var{arnr_type}
@code{VP8E_SET_ARNR_TYPE}
@item @var{arnr_strength}
@code{VP8E_SET_ARNR_STRENGTH}
@item @var{rc_lookahead}
g_lag_in_frames
@end table
@item vp8flags error_resilient
g_error_resilient
@end table
For more information about libvpx see:
@url{http://www.webmproject.org/}
@section libx264
H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through
libx264.
Requires the presence of the libx264 headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libx264}.
@subsection Options
@table @option
@item preset @var{preset_name}
Set the encoding preset.
@item tune @var{tune_name}
Tune the encoding params.
@item fastfirstpass @var{bool}
Use fast settings when encoding first pass, default value is 1.
@item profile @var{profile_name}
Set profile restrictions.
@item level @var{level}
Specify level (as defined by Annex A).
Deprecated in favor of @var{x264opts}.
@item passlogfile @var{filename}
Specify filename for 2 pass stats.
Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).
@item wpredp @var{wpred_type}
Specify Weighted prediction for P-frames.
Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
@item x264opts @var{options}
Allow to set any x264 option, see x264 --fullhelp for a list.
@var{options} is a list of @var{key}=@var{value} couples separated by
":".
@end table
For example to specify libx264 encoding options with @command{ffmpeg}:
@example
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
For more information about libx264 and the supported options see:
@url{http://www.videolan.org/developers/x264.html}
@c man end VIDEO ENCODERS

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@@ -1,174 +0,0 @@
The following table lists most error codes found in various operating
systems supported by FFmpeg.
OS
Code Std F LBMWwb Text (YMMV)
E2BIG POSIX ++++++ Argument list too long
EACCES POSIX ++++++ Permission denied
EADDRINUSE POSIX +++..+ Address in use
EADDRNOTAVAIL POSIX +++..+ Cannot assign requested address
EADV +..... Advertise error
EAFNOSUPPORT POSIX +++..+ Address family not supported
EAGAIN POSIX + ++++++ Resource temporarily unavailable
EALREADY POSIX +++..+ Operation already in progress
EAUTH .++... Authentication error
EBADARCH ..+... Bad CPU type in executable
EBADE +..... Invalid exchange
EBADEXEC ..+... Bad executable
EBADF POSIX ++++++ Bad file descriptor
EBADFD +..... File descriptor in bad state
EBADMACHO ..+... Malformed Macho file
EBADMSG POSIX ++4... Bad message
EBADR +..... Invalid request descriptor
EBADRPC .++... RPC struct is bad
EBADRQC +..... Invalid request code
EBADSLT +..... Invalid slot
EBFONT +..... Bad font file format
EBUSY POSIX - ++++++ Device or resource busy
ECANCELED POSIX +++... Operation canceled
ECHILD POSIX ++++++ No child processes
ECHRNG +..... Channel number out of range
ECOMM +..... Communication error on send
ECONNABORTED POSIX +++..+ Software caused connection abort
ECONNREFUSED POSIX - +++ss+ Connection refused
ECONNRESET POSIX +++..+ Connection reset
EDEADLK POSIX ++++++ Resource deadlock avoided
EDEADLOCK +..++. File locking deadlock error
EDESTADDRREQ POSIX +++... Destination address required
EDEVERR ..+... Device error
EDOM C89 - ++++++ Numerical argument out of domain
EDOOFUS .F.... Programming error
EDOTDOT +..... RFS specific error
EDQUOT POSIX +++... Disc quota exceeded
EEXIST POSIX ++++++ File exists
EFAULT POSIX - ++++++ Bad address
EFBIG POSIX - ++++++ File too large
EFTYPE .++... Inappropriate file type or format
EHOSTDOWN +++... Host is down
EHOSTUNREACH POSIX +++..+ No route to host
EHWPOISON +..... Memory page has hardware error
EIDRM POSIX +++... Identifier removed
EILSEQ C99 ++++++ Illegal byte sequence
EINPROGRESS POSIX - +++ss+ Operation in progress
EINTR POSIX - ++++++ Interrupted system call
EINVAL POSIX + ++++++ Invalid argument
EIO POSIX + ++++++ I/O error
EISCONN POSIX +++..+ Socket is already connected
EISDIR POSIX ++++++ Is a directory
EISNAM +..... Is a named type file
EKEYEXPIRED +..... Key has expired
EKEYREJECTED +..... Key was rejected by service
EKEYREVOKED +..... Key has been revoked
EL2HLT +..... Level 2 halted
EL2NSYNC +..... Level 2 not synchronized
EL3HLT +..... Level 3 halted
EL3RST +..... Level 3 reset
ELIBACC +..... Can not access a needed shared library
ELIBBAD +..... Accessing a corrupted shared library
ELIBEXEC +..... Cannot exec a shared library directly
ELIBMAX +..... Too many shared libraries
ELIBSCN +..... .lib section in a.out corrupted
ELNRNG +..... Link number out of range
ELOOP POSIX +++..+ Too many levels of symbolic links
EMEDIUMTYPE +..... Wrong medium type
EMFILE POSIX ++++++ Too many open files
EMLINK POSIX ++++++ Too many links
EMSGSIZE POSIX +++..+ Message too long
EMULTIHOP POSIX ++4... Multihop attempted
ENAMETOOLONG POSIX - ++++++ Filen ame too long
ENAVAIL +..... No XENIX semaphores available
ENEEDAUTH .++... Need authenticator
ENETDOWN POSIX +++..+ Network is down
ENETRESET SUSv3 +++..+ Network dropped connection on reset
ENETUNREACH POSIX +++..+ Network unreachable
ENFILE POSIX ++++++ Too many open files in system
ENOANO +..... No anode
ENOATTR .++... Attribute not found
ENOBUFS POSIX - +++..+ No buffer space available
ENOCSI +..... No CSI structure available
ENODATA XSR +N4... No message available
ENODEV POSIX - ++++++ No such device
ENOENT POSIX - ++++++ No such file or directory
ENOEXEC POSIX ++++++ Exec format error
ENOFILE ...++. No such file or directory
ENOKEY +..... Required key not available
ENOLCK POSIX ++++++ No locks available
ENOLINK POSIX ++4... Link has been severed
ENOMEDIUM +..... No medium found
ENOMEM POSIX ++++++ Not enough space
ENOMSG POSIX +++..+ No message of desired type
ENONET +..... Machine is not on the network
ENOPKG +..... Package not installed
ENOPROTOOPT POSIX +++..+ Protocol not available
ENOSPC POSIX ++++++ No space left on device
ENOSR XSR +N4... No STREAM resources
ENOSTR XSR +N4... Not a STREAM
ENOSYS POSIX + ++++++ Function not implemented
ENOTBLK +++... Block device required
ENOTCONN POSIX +++..+ Socket is not connected
ENOTDIR POSIX ++++++ Not a directory
ENOTEMPTY POSIX ++++++ Directory not empty
ENOTNAM +..... Not a XENIX named type file
ENOTRECOVERABLE SUSv4 - +..... State not recoverable
ENOTSOCK POSIX +++..+ Socket operation on non-socket
ENOTSUP POSIX +++... Operation not supported
ENOTTY POSIX ++++++ Inappropriate I/O control operation
ENOTUNIQ +..... Name not unique on network
ENXIO POSIX ++++++ No such device or address
EOPNOTSUPP POSIX +++..+ Operation not supported (on socket)
EOVERFLOW POSIX +++..+ Value too large to be stored in data type
EOWNERDEAD SUSv4 +..... Owner died
EPERM POSIX - ++++++ Operation not permitted
EPFNOSUPPORT +++..+ Protocol family not supported
EPIPE POSIX - ++++++ Broken pipe
EPROCLIM .++... Too many processes
EPROCUNAVAIL .++... Bad procedure for program
EPROGMISMATCH .++... Program version wrong
EPROGUNAVAIL .++... RPC prog. not avail
EPROTO POSIX ++4... Protocol error
EPROTONOSUPPORT POSIX - +++ss+ Protocol not supported
EPROTOTYPE POSIX +++..+ Protocol wrong type for socket
EPWROFF ..+... Device power is off
ERANGE C89 - ++++++ Result too large
EREMCHG +..... Remote address changed
EREMOTE +++... Object is remote
EREMOTEIO +..... Remote I/O error
ERESTART +..... Interrupted system call should be restarted
ERFKILL +..... Operation not possible due to RF-kill
EROFS POSIX ++++++ Read-only file system
ERPCMISMATCH .++... RPC version wrong
ESHLIBVERS ..+... Shared library version mismatch
ESHUTDOWN +++..+ Cannot send after socket shutdown
ESOCKTNOSUPPORT +++... Socket type not supported
ESPIPE POSIX ++++++ Illegal seek
ESRCH POSIX ++++++ No such process
ESRMNT +..... Srmount error
ESTALE POSIX +++..+ Stale NFS file handle
ESTRPIPE +..... Streams pipe error
ETIME XSR +N4... Stream ioctl timeout
ETIMEDOUT POSIX - +++ss+ Connection timed out
ETOOMANYREFS +++... Too many references: cannot splice
ETXTBSY POSIX +++... Text file busy
EUCLEAN +..... Structure needs cleaning
EUNATCH +..... Protocol driver not attached
EUSERS +++... Too many users
EWOULDBLOCK POSIX +++..+ Operation would block
EXDEV POSIX ++++++ Cross-device link
EXFULL +..... Exchange full
Notations:
F: used in FFmpeg (-: a few times, +: a lot)
SUSv3: Single Unix Specification, version 3
SUSv4: Single Unix Specification, version 4
XSR: XSI STREAMS (obsolete)
OS: availability on some supported operating systems
L: GNU/Linux
B: BSD (F: FreeBSD, N: NetBSD)
M: MacOS X
W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h)
w: Mingw32 (3.17) and Mingw64 (2.0.1)
b: BeOS

View File

@@ -1,192 +0,0 @@
@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

@@ -1,21 +0,0 @@
# 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 $(FFMPEG_LIBS))
EXAMPLES=decoding_encoding filtering metadata muxing
OBJS=$(addsuffix .o,$(EXAMPLES))
%: %.o
$(CC) $< $(LDFLAGS) -o $@
%.o: %.c
$(CC) $< $(CFLAGS) -c -o $@
.phony: all clean
all: $(OBJS) $(EXAMPLES)
clean:
rm -rf $(EXAMPLES) $(OBJS)

View File

@@ -1,229 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 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
* API example for decoding and filtering
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/vsrc_buffer.h>
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret, i;
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 = av_find_stream_info(fmt_ctx)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
/* init the video decoder */
if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
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), "%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) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
return ret;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, pix_fmts, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
}
static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
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(picref->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = picref->pts;
}
/* Trivial ASCII grayscale display. */
p0 = picref->data[0];
puts("\033c");
for (y = 0; y < picref->video->h; y++) {
p = p0;
for (x = 0; x < picref->video->w; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += picref->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame frame;
int got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
AVFilterBufferRef *picref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
av_free_packet(&packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
if (got_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 */
av_vsrc_buffer_add_frame(buffersrc_ctx, &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);
}
}
}
}
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
av_close_input_file(fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

View File

@@ -1,55 +0,0 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* 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
* Shows how the metadata API can be used in application programs.
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_free_context(fmt_ctx);
return 0;
}

View File

@@ -2,15 +2,29 @@
@settitle FFmpeg FAQ
@titlepage
@sp 7
@center @titlefont{FFmpeg FAQ}
@sp 3
@end titlepage
@top
@contents
@chapter General Questions
@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between?
Like most open source projects FFmpeg suffers from a certain lack of
manpower. For this reason the developers have to prioritize the work
they do and putting out releases is not at the top of the list, fixing
bugs and reviewing patches takes precedence. Please don't complain or
request more timely and/or frequent releases unless you are willing to
help out creating them.
@section I have a problem with an old version of FFmpeg; where should I report it?
Nowhere. Upgrade to the latest release or if there is no recent release upgrade
to Subversion HEAD. You could also try to report it. Maybe you will get lucky and
become the first person in history to get an answer different from "upgrade
to Subversion HEAD".
@section Why doesn't FFmpeg support feature [xyz]?
Because no one has taken on that task yet. FFmpeg development is
@@ -24,6 +38,30 @@ No. Windows DLLs are not portable, bloated and often slow.
Moreover FFmpeg strives to support all codecs natively.
A DLL loader is not conducive to that goal.
@section My bug report/mail to ffmpeg-devel/user has not received any replies.
Likely reasons
@itemize
@item We are busy and haven't had time yet to read your report or
investigate the issue.
@item You didn't follow bugreports.html.
@item You didn't use Subversion HEAD.
@item You reported a segmentation fault without gdb output.
@item You describe a problem but not how to reproduce it.
@item It's unclear if you use ffmpeg as command line tool or use
libav* from another application.
@item You speak about a video having problems on playback but
not what you use to play it.
@item We have no faint clue what you are talking about besides
that it is related to FFmpeg.
@end itemize
@section Is there a forum for FFmpeg? I do not like mailing lists.
You may view our mailing lists with a more forum-alike look here:
@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user},
but, if you post, please remember that our mailing list rules still apply there.
@section I cannot read this file although this format seems to be supported by ffmpeg.
Even if ffmpeg can read the container format, it may not support all its
@@ -47,7 +85,7 @@ The following list of video codecs should work on most Windows systems:
.asf only
@item mpeg4
Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
@item mpeg1video
@item mpeg1
.mpg only
@end table
Note, ASF files often have .wmv or .wma extensions in Windows. It should also
@@ -59,9 +97,9 @@ The following list of audio codecs should work on most Windows systems:
@table @option
@item adpcm_ima_wav
@item adpcm_ms
@item pcm_s16le
@item pcm
always
@item libmp3lame
@item mp3
If some MP3 codec like LAME is installed.
@end table
@@ -83,8 +121,7 @@ problem and an NP-hard problem...
@section ffmpeg does not work; what is wrong?
Try a @code{make distclean} in the ffmpeg source directory before the build.
If this does not help see
Try a @code{make distclean} in the ffmpeg source directory before the build. If this does not help see
(@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
@@ -108,7 +145,7 @@ 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 $x); ln "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@@ -135,15 +172,15 @@ The @file{movie.mpg} used as input will be converted to
Instead of relying on file format self-recognition, you may also use
@table @option
@item -c:v ppm
@item -c:v png
@item -c:v mjpeg
@item -vcodec ppm
@item -vcodec png
@item -vcodec mjpeg
@end table
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 -vcodec mjpeg menu%d.jpg
@end example
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -158,28 +195,79 @@ quite logical that there is a small reduction of quality. This is not a bug.
Use @file{-} as file name.
@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux?
This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at
(@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-420-2.4.26.patch?view=co}). This may also
apply cleanly to other 2.4-series kernels.
@section How do I avoid the ugly aliasing artifacts in bt8x8 captures on Linux?
Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1'
will cause somewhat too strong filtering. A fix is to apply (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co})
or (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co})
and pass 'combfilter=2'.
@section -f jpeg doesn't work.
Try '-f image2 test%d.jpg'.
@section Why can I not change the frame rate?
@section Why can I not change the framerate?
Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
Choose a different codec with the -c:v command line option.
Some codecs, like MPEG-1/2, only allow a small number of fixed framerates.
Choose a different codec with the -vcodec command line option.
@section How do I encode Xvid or DivX video with ffmpeg?
Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
standard (note that there are many other coding formats that use this
same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
same standard). Thus, use '-vcodec mpeg4' to encode in these formats. The
default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
force the fourcc 'xvid' to be stored as the video fourcc rather than the
default.
@section How do I encode videos which play on the iPod?
@table @option
@item needed stuff
-acodec libfaac -vcodec mpeg4 width<=320 height<=240
@item working stuff
4mv, title
@item non-working stuff
B-frames
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4
@end table
@section How do I encode videos which play on the PSP?
@table @option
@item needed stuff
-acodec libfaac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
@item working stuff
4mv, title
@item non-working stuff
B-frames
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4
@item needed stuff for H.264
-acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
@item working stuff for H.264
title, loop filter
@item non-working stuff for H.264
CAVLC
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
@item higher resolution for newer PSP firmwares, width<=480, height<=272
-vcodec libx264 -level 21 -coder 1 -f psp
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640kb -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4
@end table
@section Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
'-mbd rd -flags +4mv+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
@@ -198,18 +286,19 @@ material, and try '-top 0/1' if the result looks really messed-up.
If you have built FFmpeg with @code{./configure --enable-avisynth}
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
(Be aware that this feature has been recently added,
so you will need to help yourself in case of problems.)
Just create an "input.avs" text file with this single line ...
@example
DirectShowSource("C:\path to your file\yourfile.asf")
@end example
... and then feed that text file to ffmpeg:
... and then feed that text file to FFmpeg:
@example
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 @url{http://www.avisynth.org/}.
@section How can I join video files?
@@ -222,13 +311,13 @@ equally humble @code{copy} under Windows), and finally transcoding back to your
format of choice.
@example
ffmpeg -i input1.avi -same_quant intermediate1.mpg
ffmpeg -i input2.avi -same_quant intermediate2.mpg
ffmpeg -i input1.avi -sameq intermediate1.mpg
ffmpeg -i input2.avi -sameq intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -same_quant output.avi
ffmpeg -i intermediate_all.mpg -sameq output.avi
@end example
Notice that you should either use @code{-same_quant} or set a reasonably high
Notice that you should either use @code{-sameq} or set a reasonably high
bitrate for your intermediate and output files, if you want to preserve
video quality.
@@ -238,10 +327,10 @@ of named pipes, should your platform support it:
@example
mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -same_quant -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
ffmpeg -i input1.avi -sameq -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -sameq -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -same_quant -c:v mpeg4 -acodec libmp3lame output.avi
ffmpeg -f mpeg -i - -sameq -vcodec mpeg4 -acodec libmp3lame output.avi
@end example
Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
@@ -268,47 +357,27 @@ 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 \
-same_quant -y output.flv
-sameq -y output.flv
rm temp[12].[av] all.[av]
@end example
@section -profile option fails when encoding H.264 video with AAC audio
@section FFmpeg does not adhere to the -maxrate setting, some frames are bigger than maxrate/fps.
@command{ffmpeg} prints an error like
Read the MPEG spec about video buffer verifier.
@example
Undefined constant or missing '(' in 'baseline'
Unable to parse option value "baseline"
Error setting option profile to value baseline.
@end example
@section I want CBR, but no matter what I do frame sizes differ.
Short answer: write @option{-profile:v} instead of @option{-profile}.
You do not understand what CBR is, please read the MPEG spec.
Read about video buffer verifier and constant bitrate.
The one sentence summary is that there is a buffer and the input rate is
constant, the output can vary as needed.
Long answer: this happens because the @option{-profile} option can apply to both
video and audio. Specifically the AAC encoder also defines some profiles, none
of which are named @var{baseline}.
@section How do I check if a stream is CBR?
The solution is to apply the @option{-profile} option to the video stream only
by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
Appending @code{:v} to it will do exactly that.
To quote the MPEG-2 spec:
"There is no way to tell that a bitstream is constant bitrate without
examining all of the vbv_delay values and making complicated computations."
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
Use @option{-dumpgraph -} to find out exactly where the channel layout is
lost.
Most likely, it is through @code{auto-inserted 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{aconvert} explicitly in the filter graph,
specifying the exact format.
@example
aconvert=s16:stereo:packed
@end example
@chapter Development
@@ -356,26 +425,34 @@ the FFmpeg Windows Help Forum at
No. These tools are too bloated and they complicate the build.
@section Why not rewrite FFmpeg in object-oriented C++?
@section Why not rewrite ffmpeg in object-oriented C++?
FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
read "Programming Religion" at (@url{http://www.tux.org/lkml/#s15}).
@section Why are the ffmpeg programs devoid of debugging symbols?
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.
you need the debug information, used the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
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
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 which seem to be available.
@section I want to compile xyz.c alone but my compiler produced many errors.
Common code is in its own files in libav* and is used by the individual
codecs. They will not work without the common parts, you have to compile
the whole libav*. If you wish, disable some parts with configure switches.
You can also try to hack it and remove more, but if you had problems fixing
the compilation failure then you are probably not qualified for this.
@section I'm using libavcodec 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
you need to explicitly state that you are using a C library. You can do this by
@@ -383,15 +460,23 @@ encompassing your FFmpeg includes using @code{extern "C"}.
See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
FFmpeg is a pure C project using C99 math features, in order to enable C++
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 implement a URLProtocol, see @file{libavformat/file.c} in
FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
You have to implement a URLProtocol, see libavformat/file.c in FFmpeg
and libmpdemux/demux_lavf.c in MPlayer sources.
@section I get "No compatible shell script interpreter found." in MSys.
The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
@section I get "./configure: line <xxx>: pr: command not found" in MSys.
The standard MSys install doesn't come with pr. You need to get it from the coreutils package.
@section I tried to pass RTP packets into a decoder, but it doesn't work.
RTP is a container format like any other, you must first depacketize the
codec frames/samples stored in RTP and then feed to the decoder.
@section Where can I find libav* headers for Pascal/Delphi?
@@ -399,7 +484,7 @@ see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
see @url{http://www.ffmpeg.org/~michael/}
see @url{http://svn.ffmpeg.org/michael/trunk/docs/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
@@ -407,24 +492,12 @@ Even if peculiar since it is network oriented, RTP is a container like any
other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
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.
@section AVStream.r_frame_rate is wrong, it is much larger than the framerate.
r_frame_rate is NOT the average frame rate, it is the smallest frame rate
r_frame_rate is NOT the average framerate, it is the smallest framerate
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 r_frame_rate
will be 150.
@section Why is @code{make fate} not running all tests?
Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
or @code{FATE_SAMPLES} environment variable or the @code{--samples}
@command{configure} option is set to the right path.
@section Why is @code{make fate} not finding the samples?
Do you happen to have a @code{~} character in the samples path to indicate a
home directory? The value is used in ways where the shell cannot expand it,
causing FATE to not find files. Just replace @code{~} by the full path.
@bye

View File

@@ -1,174 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle FATE Automated Testing Environment
@titlepage
@center @titlefont{FATE Automated Testing Environment}
@end titlepage
@node Top
@top
@contents
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
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 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
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@example
make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
`--samples=<path to the samples directory>'. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
@example
./configure --samples=fate-suite/
make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
@example
FATE_SAMPLES=fate-suite/ make fate
@end example
@float NOTE
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
@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 tests/fate.sh from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@example
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{tests/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
@verbatiminclude ../tests/fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The `slot' configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
For your first test runs the `fate_recv' variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@itemize
@item configure.log
@item compile.log
@item test.log
@item report
@item version
@end itemize
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
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:
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.
@chapter FATE makefile targets and variables
@section Makefile targets
@table @option
@item fate-rsync
Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
@item fate
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.
@itemize
@item 0: show just the test arguments
@item 1: show just the command used in the test
@item 2: show everything
@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.
@item THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@end table
Example:
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
@end example

956
doc/ffmpeg-doc.texi Normal file
View File

@@ -0,0 +1,956 @@
\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Documentation
@titlepage
@sp 7
@center @titlefont{FFmpeg Documentation}
@sp 3
@end titlepage
@chapter Introduction
FFmpeg is a very fast video and audio converter. It can also grab from
a live audio/video source.
The command line interface is designed to be intuitive, in the sense
that FFmpeg tries to figure out all parameters that can possibly be
derived automatically. You usually only have to specify the target
bitrate you want.
FFmpeg can also convert from any sample rate to any other, and resize
video on the fly with a high quality polyphase filter.
@chapter Quick Start
@c man begin EXAMPLES
@section Video and Audio grabbing
FFmpeg can grab video and audio from devices given that you specify the input
format and device.
@example
ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
@end example
Note that you must activate the right video source and channel before
launching FFmpeg with any TV viewer such as xawtv
(@url{http://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also
have to set the audio recording levels correctly with a
standard mixer.
@section X11 grabbing
FFmpeg can grab the X11 display.
@example
ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg
@end example
0.0 is display.screen number of your X11 server, same as
the DISPLAY environment variable.
@example
ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg
@end example
0.0 is display.screen number of your X11 server, same as the DISPLAY environment
variable. 10 is the x-offset and 20 the y-offset for the grabbing.
@section Video and Audio file format conversion
* FFmpeg can use any supported file format and protocol as input:
Examples:
* You can use YUV files as input:
@example
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
@end example
It will use the files:
@example
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
@end example
The Y files use twice the resolution of the U and V files. They are
raw files, without header. They can be generated by all decent video
decoders. You must specify the size of the image with the @option{-s} option
if FFmpeg cannot guess it.
* You can input from a raw YUV420P file:
@example
ffmpeg -i /tmp/test.yuv /tmp/out.avi
@end example
test.yuv is a file containing raw YUV planar data. Each frame is composed
of the Y plane followed by the U and V planes at half vertical and
horizontal resolution.
* You can output to a raw YUV420P file:
@example
ffmpeg -i mydivx.avi hugefile.yuv
@end example
* You can set several input files and output files:
@example
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
@end example
Converts the audio file a.wav and the raw YUV video file a.yuv
to MPEG file a.mpg.
* You can also do audio and video conversions at the same time:
@example
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
@end example
Converts a.wav to MPEG audio at 22050 Hz sample rate.
* You can encode to several formats at the same time and define a
mapping from input stream to output streams:
@example
ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
@end example
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map
file:index' specifies which input stream is used for each output
stream, in the order of the definition of output streams.
* You can transcode decrypted VOBs:
@example
ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
@end example
This is a typical DVD ripping example; the input is a VOB file, the
output an AVI file with MPEG-4 video and MP3 audio. Note that in this
command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
input video. Furthermore, the audio stream is MP3-encoded so you need
to enable LAME support by passing @code{--enable-libmp3lame} to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
* You can extract images from a video, or create a video from many images:
For extracting images from a video:
@example
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
@end example
This will extract one video frame per second from the video and will
output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
* You can put many streams of the same type in the output:
@example
ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
@end example
In addition to the first video and audio streams, the resulting
output file @file{test12.avi} will contain the second video
and the second audio stream found in the input streams list.
The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle}
options have to be specified immediately after the name of the output
file to which you want to add them.
@c man end
@chapter Invocation
@section Syntax
The generic syntax is:
@example
@c man begin SYNOPSIS
ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
@c man end
@end example
@c man begin DESCRIPTION
As a general rule, options are applied to the next specified
file. Therefore, order is important, and you can have the same
option on the command line multiple times. Each occurrence is
then applied to the next input or output file.
* To set the video bitrate of the output file to 64kbit/s:
@example
ffmpeg -i input.avi -b 64k output.avi
@end example
* To force the frame rate of the output file to 24 fps:
@example
ffmpeg -i input.avi -r 24 output.avi
@end example
* To force the frame rate of the input file (valid for raw formats only)
to 1 fps and the frame rate of the output file to 24 fps:
@example
ffmpeg -r 1 -i input.m2v -r 24 output.avi
@end example
The format option may be needed for raw input files.
By default, FFmpeg tries to convert as losslessly as possible: It
uses the same audio and video parameters for the outputs as the one
specified for the inputs.
@c man end
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -f @var{fmt}
Force format.
@item -i @var{filename}
input file name
@item -y
Overwrite output files.
@item -t @var{duration}
Restrict the transcoded/captured video sequence
to the duration specified in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
@item -fs @var{limit_size}
Set the file size limit.
@item -ss @var{position}
Seek to given time position in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
@item -itsoffset @var{offset}
Set the input time offset in seconds.
@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
This option affects all the input files that follow it.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by 'offset' seconds.
@item -timestamp @var{time}
Set the timestamp.
@item -metadata @var{key}=@var{value}
Set a metadata key/value pair.
For example, for setting the title in the output file:
@example
ffmpeg -i in.avi -metadata title="my title" out.flv
@end example
@item -v @var{number}
Set the logging verbosity level.
@item -target @var{type}
Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
"ntsc-svcd", ... ). All the format options (bitrate, codecs,
buffer sizes) are then set automatically. You can just type:
@example
ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
@end example
Nevertheless you can specify additional options as long as you know
they do not conflict with the standard, as in:
@example
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dframes @var{number}
Set the number of data frames to record.
@item -scodec @var{codec}
Force subtitle codec ('copy' to copy stream).
@item -newsubtitle
Add a new subtitle stream to the current output stream.
@item -slang @var{code}
Set the ISO 639 language code (3 letters) of the current subtitle stream.
@end table
@section Video Options
@table @option
@item -b @var{bitrate}
Set the video bitrate in bit/s (default = 200 kb/s).
@item -vframes @var{number}
Set the number of video frames to record.
@item -r @var{fps}
Set frame rate (Hz value, fraction or abbreviation), (default = 25).
@item -s @var{size}
Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source).
The following abbreviations are recognized:
@table @samp
@item sqcif
128x96
@item qcif
176x144
@item cif
352x288
@item 4cif
704x576
@item 16cif
1408x1152
@item qqvga
160x120
@item qvga
320x240
@item vga
640x480
@item svga
800x600
@item xga
1024x768
@item uxga
1600x1200
@item qxga
2048x1536
@item sxga
1280x1024
@item qsxga
2560x2048
@item hsxga
5120x4096
@item wvga
852x480
@item wxga
1366x768
@item wsxga
1600x1024
@item wuxga
1920x1200
@item woxga
2560x1600
@item wqsxga
3200x2048
@item wquxga
3840x2400
@item whsxga
6400x4096
@item whuxga
7680x4800
@item cga
320x200
@item ega
640x350
@item hd480
852x480
@item hd720
1280x720
@item hd1080
1920x1080
@end table
@item -aspect @var{aspect}
Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).
@item -croptop @var{size}
Set top crop band size (in pixels).
@item -cropbottom @var{size}
Set bottom crop band size (in pixels).
@item -cropleft @var{size}
Set left crop band size (in pixels).
@item -cropright @var{size}
Set right crop band size (in pixels).
@item -padtop @var{size}
Set top pad band size (in pixels).
@item -padbottom @var{size}
Set bottom pad band size (in pixels).
@item -padleft @var{size}
Set left pad band size (in pixels).
@item -padright @var{size}
Set right pad band size (in pixels).
@item -padcolor @var{hex_color}
Set color of padded bands. The value for padcolor is expressed
as a six digit hexadecimal number where the first two digits
represent red, the middle two digits green and last two digits
blue (default = 000000 (black)).
@item -vn
Disable video recording.
@item -bt @var{tolerance}
Set video bitrate tolerance (in bits, default 4000k).
Has a minimum value of: (target_bitrate/target_framerate).
In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
willing to deviate from the target average bitrate value. This is
not related to min/max bitrate. Lowering tolerance too much has
an adverse effect on quality.
@item -maxrate @var{bitrate}
Set max video bitrate (in bit/s).
Requires -bufsize to be set.
@item -minrate @var{bitrate}
Set min video bitrate (in bit/s).
Most useful in setting up a CBR encode:
@example
ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
@end example
It is of little use elsewise.
@item -bufsize @var{size}
Set video buffer verifier buffer size (in bits).
@item -vcodec @var{codec}
Force video codec to @var{codec}. Use the @code{copy} special value to
tell that the raw codec data must be copied as is.
@item -sameq
Use same video quality as source (implies VBR).
@item -pass @var{n}
Select the pass number (1 or 2). It is used to do two-pass
video encoding. The statistics of the video are recorded in the first
pass into a log file (see also the option -passlogfile),
and in the second pass that log file is used to generate the video
at the exact requested bitrate.
On pass 1, you may just deactivate audio and set output to null,
examples for Windows and Unix:
@example
ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
@end example
@item -passlogfile @var{prefix}
Set two-pass log file name prefix to @var{prefix}, the default file name
prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
stream.
@item -newvideo
Add a new video stream to the current output stream.
@item -vlang @var{code}
Set the ISO 639 language code (3 letters) of the current video stream.
@end table
@section Advanced Video Options
@table @option
@item -pix_fmt @var{format}
Set pixel format. Use 'list' as parameter to show all the supported
pixel formats.
@item -sws_flags @var{flags}
Set SwScaler flags.
@item -g @var{gop_size}
Set the group of pictures size.
@item -intra
Use only intra frames.
@item -vdt @var{n}
Discard threshold.
@item -qscale @var{q}
Use fixed video quantizer scale (VBR).
@item -qmin @var{q}
minimum video quantizer scale (VBR)
@item -qmax @var{q}
maximum video quantizer scale (VBR)
@item -qdiff @var{q}
maximum difference between the quantizer scales (VBR)
@item -qblur @var{blur}
video quantizer scale blur (VBR) (range 0.0 - 1.0)
@item -qcomp @var{compression}
video quantizer scale compression (VBR) (default 0.5).
Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
@item -lmin @var{lambda}
minimum video lagrange factor (VBR)
@item -lmax @var{lambda}
max video lagrange factor (VBR)
@item -mblmin @var{lambda}
minimum macroblock quantizer scale (VBR)
@item -mblmax @var{lambda}
maximum macroblock quantizer scale (VBR)
These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
but you may use the QP2LAMBDA constant to easily convert from 'q' units:
@example
ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end example
@item -rc_init_cplx @var{complexity}
initial complexity for single pass encoding
@item -b_qfactor @var{factor}
qp factor between P- and B-frames
@item -i_qfactor @var{factor}
qp factor between P- and I-frames
@item -b_qoffset @var{offset}
qp offset between P- and B-frames
@item -i_qoffset @var{offset}
qp offset between P- and I-frames
@item -rc_eq @var{equation}
Set rate control equation (@pxref{FFmpeg formula
evaluator}) (default = @code{tex^qComp}).
@item -rc_override @var{override}
rate control override for specific intervals
@item -me_method @var{method}
Set motion estimation method to @var{method}.
Available methods are (from lowest to best quality):
@table @samp
@item zero
Try just the (0, 0) vector.
@item phods
@item log
@item x1
@item hex
@item umh
@item epzs
(default method)
@item full
exhaustive search (slow and marginally better than epzs)
@end table
@item -dct_algo @var{algo}
Set DCT algorithm to @var{algo}. Available values are:
@table @samp
@item 0
FF_DCT_AUTO (default)
@item 1
FF_DCT_FASTINT
@item 2
FF_DCT_INT
@item 3
FF_DCT_MMX
@item 4
FF_DCT_MLIB
@item 5
FF_DCT_ALTIVEC
@end table
@item -idct_algo @var{algo}
Set IDCT algorithm to @var{algo}. Available values are:
@table @samp
@item 0
FF_IDCT_AUTO (default)
@item 1
FF_IDCT_INT
@item 2
FF_IDCT_SIMPLE
@item 3
FF_IDCT_SIMPLEMMX
@item 4
FF_IDCT_LIBMPEG2MMX
@item 5
FF_IDCT_PS2
@item 6
FF_IDCT_MLIB
@item 7
FF_IDCT_ARM
@item 8
FF_IDCT_ALTIVEC
@item 9
FF_IDCT_SH4
@item 10
FF_IDCT_SIMPLEARM
@end table
@item -er @var{n}
Set error resilience to @var{n}.
@table @samp
@item 1
FF_ER_CAREFUL (default)
@item 2
FF_ER_COMPLIANT
@item 3
FF_ER_AGGRESSIVE
@item 4
FF_ER_VERY_AGGRESSIVE
@end table
@item -ec @var{bit_mask}
Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
the following values:
@table @samp
@item 1
FF_EC_GUESS_MVS (default = enabled)
@item 2
FF_EC_DEBLOCK (default = enabled)
@end table
@item -bf @var{frames}
Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
@item -mbd @var{mode}
macroblock decision
@table @samp
@item 0
FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
@item 1
FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
@item 2
FF_MB_DECISION_RD: rate distortion
@end table
@item -4mv
Use four motion vector by macroblock (MPEG-4 only).
@item -part
Use data partitioning (MPEG-4 only).
@item -bug @var{param}
Work around encoder bugs that are not auto-detected.
@item -strict @var{strictness}
How strictly to follow the standards.
@item -aic
Enable Advanced intra coding (h263+).
@item -umv
Enable Unlimited Motion Vector (h263+)
@item -deinterlace
Deinterlace pictures.
@item -ilme
Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
Use this option if your input file is interlaced and you want
to keep the interlaced format for minimum losses.
The alternative is to deinterlace the input stream with
@option{-deinterlace}, but deinterlacing introduces losses.
@item -psnr
Calculate PSNR of compressed frames.
@item -vstats
Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
Dump video coding statistics to @var{file}.
@item -top @var{n}
top=1/bottom=0/auto=-1 field first
@item -dc @var{precision}
Intra_dc_precision.
@item -vtag @var{fourcc/tag}
Force video tag/fourcc.
@item -qphist
Show QP histogram.
@item -vbsf @var{bitstream_filter}
Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump".
@example
ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
@end example
@end table
@section Audio Options
@table @option
@item -aframes @var{number}
Set the number of audio frames to record.
@item -ar @var{freq}
Set the audio sampling frequency (default = 44100 Hz).
@item -ab @var{bitrate}
Set the audio bitrate in bit/s (default = 64k).
@item -aq @var{q}
Set the audio quality (codec-specific, VBR).
@item -ac @var{channels}
Set the number of audio channels (default = 1).
@item -an
Disable audio recording.
@item -acodec @var{codec}
Force audio codec to @var{codec}. Use the @code{copy} special value to
specify that the raw codec data must be copied as is.
@item -newaudio
Add a new audio track to the output file. If you want to specify parameters,
do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..).
Mapping will be done automatically, if the number of output streams is equal to
the number of input streams, else it will pick the first one that matches. You
can override the mapping using @code{-map} as usual.
Example:
@example
ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
@end example
@item -alang @var{code}
Set the ISO 639 language code (3 letters) of the current audio stream.
@end table
@section Advanced Audio options:
@table @option
@item -atag @var{fourcc/tag}
Force audio tag/fourcc.
@item -absf @var{bitstream_filter}
Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
@end table
@section Subtitle options:
@table @option
@item -scodec @var{codec}
Force subtitle codec ('copy' to copy stream).
@item -newsubtitle
Add a new subtitle stream to the current output stream.
@item -slang @var{code}
Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -sn
Disable subtitle recording.
@item -sbsf @var{bitstream_filter}
Bitstream filters available are "mov2textsub", "text2movsub".
@example
ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
@end example
@end table
@section Audio/Video grab options
@table @option
@item -vc @var{channel}
Set video grab channel (DV1394 only).
@item -tvstd @var{standard}
Set television standard (NTSC, PAL (SECAM)).
@item -isync
Synchronize read on input.
@end table
@section Advanced options
@table @option
@item -map @var{input_stream_id}[:@var{sync_stream_id}]
Set stream mapping from input streams to output streams.
Just enumerate the input streams in the order you want them in the output.
@var{sync_stream_id} if specified sets the input stream to sync
against.
@item -map_meta_data @var{outfile}:@var{infile}
Set meta data information of @var{outfile} from @var{infile}.
@item -debug
Print specific debug info.
@item -benchmark
Show benchmarking information at the end of an encode.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
@item -dump
Dump each input packet.
@item -hex
When dumping packets, also dump the payload.
@item -bitexact
Only use bit exact algorithms (for codec testing).
@item -ps @var{size}
Set RTP payload size in bytes.
@item -re
Read input at native frame rate. Mainly used to simulate a grab device.
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
@item -threads @var{count}
Thread count.
@item -vsync @var{parameter}
Video sync method.
0 Each frame is passed with its timestamp from the demuxer to the muxer
1 Frames will be duplicated and dropped to achieve exactly the requested
constant framerate.
2 Frames are passed through with their timestamp or dropped so as to prevent
2 frames from having the same timestamp
-1 Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
With -map you can select from
which stream the timestamps should be taken. You can leave either video or
audio unchanged and sync the remaining stream(s) to the unchanged one.
@item -async @var{samples_per_second}
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
the parameter is the maximum samples per second by which the audio is changed.
-async 1 is a special case where only the start of the audio stream is corrected
without any later correction.
@item -copyts
Copy timestamps from input to output.
@item -shortest
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
Timestamp discontinuity delta threshold.
@item -muxdelay @var{seconds}
Set the maximum demux-decode delay.
@item -muxpreload @var{seconds}
Set the initial demux-decode delay.
@end table
@section Preset files
A preset file contains a sequence of @var{option}=@var{value} pairs,
one for each line, specifying a sequence of options which would be
awkward to specify on the command line. Lines starting with the hash
('#') character are ignored and are used to provide comments. Check
the @file{ffpresets} directory in the FFmpeg source tree for examples.
Preset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the @code{vpre}, @code{apre}, and
@code{spre} options, the options specified in a preset file are
applied to the currently selected codec of the same type as the preset
option.
The argument passed to the @code{vpre}, @code{apre}, and @code{spre}
preset options identifies the preset file to use according to the
following rules:
First ffmpeg searches for a file named @var{arg}.ffpreset in the
directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg})
in that order. For example, if the argument is @code{libx264-max}, it will
search for the file @file{libx264-max.ffpreset}.
If no such file is found, then ffmpeg will search for a file named
@var{codec_name}-@var{arg}.ffpreset in the above-mentioned
directories, where @var{codec_name} is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with @code{-vcodec libx264} and use @code{-vpre max},
then it will search for the file @file{libx264-max.ffpreset}.
@anchor{FFmpeg formula evaluator}
@section FFmpeg formula evaluator
When evaluating a rate control string, FFmpeg uses an internal formula
evaluator.
The following binary operators are available: @code{+}, @code{-},
@code{*}, @code{/}, @code{^}.
The following unary operators are available: @code{+}, @code{-},
@code{(...)}.
The following statements are available: @code{ld}, @code{st},
@code{while}.
The following functions are available:
@table @var
@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 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 bits2qp(bits)
@item qp2bits(qp)
@end table
The following constants are available:
@table @var
@item PI
@item E
@item iTex
@item pTex
@item tex
@item mv
@item fCode
@item iCount
@item mcVar
@item var
@item isI
@item isP
@item isB
@item avgQP
@item qComp
@item avgIITex
@item avgPITex
@item avgPPTex
@item avgBPTex
@item avgTex
@end table
@c man end
@ignore
@setfilename ffmpeg
@settitle FFmpeg video converter
@c man begin SEEALSO
ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}.
@c man end
@c man begin AUTHOR
Fabrice Bellard
@c man end
@end ignore
@section Protocols
The file name can be @file{-} to read from standard input or to write
to standard output.
FFmpeg also handles many protocols specified with an URL syntax.
Use 'ffmpeg -protocols' to see a list of the supported protocols.
The protocol @code{http:} is currently used only to communicate with
FFserver (see the FFserver documentation). When FFmpeg will be a
video player it will also be used for streaming :-)
@chapter Tips
@itemize
@item For streaming at very low bitrate application, use a low frame rate
and a small GOP size. This is especially true for RealVideo where
the Linux player does not seem to be very fast, so it can miss
frames. An example is:
@example
ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
@end example
@item The parameter 'q' which is displayed while encoding is the current
quantizer. The value 1 indicates that a very good quality could
be achieved. The value 31 indicates the worst quality. If q=31 appears
too often, it means that the encoder cannot compress enough to meet
your bitrate. You must either increase the bitrate, decrease the
frame rate or decrease the frame size.
@item If your computer is not fast enough, you can speed up the
compression at the expense of the compression ratio. You can use
'-me zero' to speed up motion estimation, and '-intra' to disable
motion estimation completely (you have only I-frames, which means it
is about as good as JPEG compression).
@item To have very low audio bitrates, reduce the sampling frequency
(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
@item To have a constant quality (but a variable bitrate), use the option
'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
quality).
@item When converting video files, you can use the '-sameq' option which
uses the same quality factor in the encoder as in the decoder.
It allows almost lossless encoding.
@end itemize
@bye

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:
ffmpeg.c : libav*
======== : ======
:
:
--------------------------------:---> AVStream...
InputStream input_streams[] / :
/ :
InputFile input_files[] +==========================+ / ^ :
------> 0 | : st ---:-----------:--/ : :
^ +------+-----------+-----+ / +--------------------------+ : :
: | :ist_index--:-----:---------/ 1 | : st : | : :
: +------+-----------+-----+ +==========================+ : :
nb_input_files : | :ist_index--:-----:------------------> 2 | : st : | : :
: +------+-----------+-----+ +--------------------------+ : nb_input_streams :
: | :ist_index : | 3 | ... | : :
v +------+-----------+-----+ +--------------------------+ : :
--> 4 | | : :
| +--------------------------+ : :
| 5 | | : :
| +==========================+ v :
| :
| :
| :
| :
--------- --------------------------------:---> AVStream...
\ / :
OutputStream output_streams[] / :
\ / :
+======\======================/======+ ^ :
------> 0 | : source_index : st-:--- | : :
OuputFile output_files[] / +------------------------------------+ : :
/ 1 | : : : | : :
^ +------+------------+-----+ / +------------------------------------+ : :
: | : ost_index -:-----:------/ 2 | : : : | : :
nb_output_files : +------+------------+-----+ +====================================+ : :
: | : ost_index -:-----|-----------------> 3 | : : : | : :
: +------+------------+-----+ +------------------------------------+ : nb_output_streams :
: | : : | 4 | | : :
: +------+------------+-----+ +------------------------------------+ : :
: | : : | 5 | | : :
v +------+------------+-----+ +------------------------------------+ : :
6 | | : :
+------------------------------------+ : :
7 | | : :
+====================================+ v :
:

View File

@@ -0,0 +1,172 @@
FFmpeg & evaluating performance on the PowerPC Architecture HOWTO
(c) 2003-2004 Romain Dolbeau <romain@dolbeau.org>
I - Introduction
The PowerPC architecture and its SIMD extension AltiVec offer some
interesting tools to evaluate performance and improve the code.
This document tries to explain how to use those tools with FFmpeg.
The architecture itself offers two ways to evaluate the performance of
a given piece of code:
1) The Time Base Registers (TBL)
2) The Performance Monitor Counter Registers (PMC)
The first ones are always available, always active, but they're not very
accurate: the registers increment by one every four *bus* cycles. On
my 667 Mhz tiBook (ppc7450), this means once every twenty *processor*
cycles. So we won't use that.
The PMC are much more useful: not only can they report cycle-accurate
timing, but they can also be used to monitor many other parameters,
such as the number of AltiVec stalls for every kind of instruction,
or instruction cache misses. The downside is that not all processors
support the PMC (all G3, all G4 and the 970 do support them), and
they're inactive by default - you need to activate them with a
dedicated tool. Also, the number of available PMC depends on the
procesor: the various 604 have 2, the various 75x (aka. G3) have 4,
and the various 74xx (aka G4) have 6.
*WARNING*: The PowerPC 970 is not very well documented, and its PMC
registers are 64 bits wide. To properly notify the code, you *must*
tune for the 970 (using --tune=970), or the code will assume 32 bit
registers.
II - Enabling FFmpeg PowerPC performance support
This needs to be done by hand. First, you need to configure FFmpeg as
usual, but add the "--powerpc-perf-enable" option. For instance:
#####
./configure --prefix=/usr/local/ffmpeg-svn --cc=gcc-3.3 --tune=7450 --powerpc-perf-enable
#####
This will configure FFmpeg to install inside /usr/local/ffmpeg-svn,
compiling with gcc-3.3 (you should try to use this one or a newer
gcc), and tuning for the PowerPC 7450 (i.e. the newer G4; as a rule of
thumb, those at 550Mhz and more). It will also enable the PMC.
You may also edit the file "config.h" to enable the following line:
#####
// #define ALTIVEC_USE_REFERENCE_C_CODE 1
#####
If you enable this line, then the code will not make use of AltiVec,
but will use the reference C code instead. This is useful to compare
performance between two versions of the code.
Also, the number of enabled PMC is defined in "libavcodec/ppc/dsputil_ppc.h":
#####
#define POWERPC_NUM_PMC_ENABLED 4
#####
If you have a G4 CPU, you can enable all 6 PMC. DO NOT enable more
PMC than available on your CPU!
Then, simply compile FFmpeg as usual (make && make install).
III - Using FFmpeg PowerPC performance support
This FFmeg can be used exactly as usual. But before exiting, FFmpeg
will dump a per-function report that looks like this:
#####
PowerPC performance report
Values are from the PMC registers, and represent whatever the
registers are set to record.
Function "gmc1_altivec" (pmc1):
min: 231
max: 1339867
avg: 558.25 (255302)
Function "gmc1_altivec" (pmc2):
min: 93
max: 2164
avg: 267.31 (255302)
Function "gmc1_altivec" (pmc3):
min: 72
max: 1987
avg: 276.20 (255302)
(...)
#####
In this example, PMC1 was set to record CPU cycles, PMC2 was set to
record AltiVec Permute Stall Cycles, and PMC3 was set to record AltiVec
Issue Stalls.
The function "gmc1_altivec" was monitored 255302 times, and the
minimum execution time was 231 processor cycles. The max and average
aren't much use, as it's very likely the OS interrupted execution for
reasons of its own :-(
With the exact same settings and source file, but using the reference C
code we get:
#####
PowerPC performance report
Values are from the PMC registers, and represent whatever the
registers are set to record.
Function "gmc1_altivec" (pmc1):
min: 592
max: 2532235
avg: 962.88 (255302)
Function "gmc1_altivec" (pmc2):
min: 0
max: 33
avg: 0.00 (255302)
Function "gmc1_altivec" (pmc3):
min: 0
max: 350
avg: 0.03 (255302)
(...)
#####
592 cycles, so the fastest AltiVec execution is about 2.5x faster than
the fastest C execution in this example. It's not perfect but it's not
bad (well I wrote this function so I can't say otherwise :-).
Once you have that kind of report, you can try to improve things by
finding what goes wrong and fixing it; in the example above, one
should try to diminish the number of AltiVec stalls, as this *may*
improve performance.
IV) Enabling the PMC in Mac OS X
This is easy. Use "Monster" and "monster". Those tools come from
Apple's CHUD package, and can be found hidden in the developer web
site & FTP site. "MONster" is the graphical application, use it to
generate a config file specifying what each register should
monitor. Then use the command-line application "monster" to use that
config file, and enjoy the results.
Note that "MONster" can be used for many other things, but it's
documented by Apple, it's not my subject.
If you are using CHUD 4.4.2 or later, you'll notice that MONster is
no longer available. It's been superseeded by Shark, where
configuration of PMCs is available as a plugin.
V) Enabling the PMC on Linux
On linux you may use oprofile from http://oprofile.sf.net, depending on the
version and the cpu you may need to apply a patch[1] to access a set of the
possibile counters from the userspace application. You can always define them
using the kernel interface /dev/oprofile/* .
[1] http://dev.gentoo.org/~lu_zero/development/oprofile-g4-20060423.patch
--
Romain Dolbeau <romain@dolbeau.org>
Luca Barbato <lu_zero@gentoo.org>

View File

@@ -1,34 +1,33 @@
\input texinfo @c -*- texinfo -*-
@settitle ffplay Documentation
@settitle FFplay Documentation
@titlepage
@center @titlefont{ffplay Documentation}
@sp 7
@center @titlefont{FFplay Documentation}
@sp 3
@end titlepage
@top
@contents
@chapter Introduction
@chapter Synopsis
@example
@c man begin SYNOPSIS
ffplay [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
FFplay is a very simple and portable media player using the FFmpeg
libraries and the SDL library. It is mostly used as a testbed for the
various FFmpeg APIs.
@c man end
@chapter Options
@chapter Invocation
@section Syntax
@example
@c man begin SYNOPSIS
ffplay [options] @file{input_file}
@c man end
@end example
@c man begin OPTIONS
@include avtools-common-opts.texi
@include fftools-common-opts.texi
@section Main options
@@ -38,9 +37,8 @@ Force displayed width.
@item -y @var{height}
Force displayed height.
@item -s @var{size}
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
Set frame size (WxH or abbreviation), needed for videos which don't
contain a header with the frame size like raw YUV.
@item -an
Disable audio.
@item -vn
@@ -59,44 +57,21 @@ Force format.
Set window title (default is the input filename).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
Set the show mode to use.
Available values for @var{mode} are:
@table @samp
@item 0, video
show video
@item 1, waves
show audio waves
@item 2, rdft
show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
@end table
Default value is "video", if video is not present or cannot be played
"rdft" is automatically selected.
You can interactively cycle through the available show modes by
pressing the key @key{w}.
@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
also sources and sinks).
@item -i @var{input_file}
Read @var{input_file}.
@end table
@section Advanced options
@table @option
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Show the stream duration, the codec parameters, the current position in
the stream and the audio/video synchronisation drift.
@item -debug
Print specific debug info.
@item -bug
Work around bugs.
@item -vismv
Visualize motion vectors.
@item -fast
Non-spec-compliant optimizations.
@item -genpts
@@ -128,14 +103,6 @@ Select the desired subtitle stream number, counting from 0. The number
refers to the list of all the input subtitle streams. If it is greater
than the number of subtitle streams minus one, then the last one is
selected, if it is negative the subtitle rendering is disabled.
@item -autoexit
Exit when video is done playing.
@item -exitonkeydown
Exit if any key is pressed.
@item -exitonmousedown
Exit if any mouse button is pressed.
@item -codec:@var{stream_type}
Force a specific decoder implementation
@end table
@section While playing
@@ -168,9 +135,6 @@ Seek backward/forward 10 seconds.
@item down/up
Seek backward/forward 1 minute.
@item page down/page up
Seek backward/forward 10 minutes.
@item mouse click
Seek to percentage in file corresponding to fraction of width.
@@ -178,26 +142,17 @@ Seek to percentage in file corresponding to fraction of width.
@c man end
@include eval.texi
@include decoders.texi
@include demuxers.texi
@include muxers.texi
@include indevs.texi
@include outdevs.texi
@include protocols.texi
@include filters.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@c man begin SEEALSO
ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
ffmpeg(1), ffserver(1) and the HTML documentation of @file{ffmpeg}.
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man begin AUTHOR
Fabrice Bellard
@c man end
@end ignore

121
doc/ffprobe-doc.texi Normal file
View File

@@ -0,0 +1,121 @@
\input texinfo @c -*- texinfo -*-
@settitle FFprobe Documentation
@titlepage
@sp 7
@center @titlefont{FFprobe Documentation}
@sp 3
@end titlepage
@chapter Introduction
@c man begin DESCRIPTION
FFprobe gathers information from multimedia streams and prints it in
human- and machine-readable fashion.
For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
If a filename is specified in input, ffprobe will try to open and
probe the file content. If the file cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
FFprobe may be employed both as a standalone application or in
combination with a textual filter, which may perform more
sophisticated processing, e.g. statistical processing or plotting.
Options are used to list some of the formats supported by ffprobe or
for specifying which information to display, and for setting how
ffprobe will show it.
FFprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of the form:
@example
[SECTION]
key1=val1
...
keyN=valN
[/SECTION]
@end example
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding ``FORMAT'' or ``STREAM'' section, and
are prefixed by the string ``TAG:''.
@c man end
@chapter Invocation
@section Syntax
The generic syntax is:
@example
@c man begin SYNOPSIS
ffprobe [options] [@file{input_file}]
@c man end
@end example
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -convert_tags
Convert the tag names in the format container to the generic FFmpeg tag names.
@item -f @var{format}
Force format to use.
@item -unit
Show the unit of the displayed values.
@item -prefix
Show a SI prefixes of the displayed values.
Unless ``-byte_binary_prefix'' option is used all the prefix
are decimal.
@item -byte_binary_prefix
Force the use of binary prefixes for byte values.
@item -sexagesimal
Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
@item -pretty
Prettify the format of the displayed values, it corresponds to the
options ``-unit -prefix -byte_binary_prefix -sexagesimal''.
@item -show_format
Show information about the container format of the input multimedia
stream.
All the container format information is printed within a section with
name ``FORMAT''.
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name ``STREAM''.
@end table
@c man end
@ignore
@setfilename ffprobe
@settitle FFprobe media prober
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffserver(1)
@c man end
@end ignore
@bye

View File

@@ -1,334 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle ffprobe Documentation
@titlepage
@center @titlefont{ffprobe Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
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
ffprobe gathers information from multimedia streams and prints it in
human- and machine-readable fashion.
For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
If a filename is specified in input, ffprobe will try to open and
probe the file content. If the file cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
ffprobe may be employed both as a standalone application or in
combination with a textual filter, which may perform more
sophisticated processing, e.g. statistical processing or plotting.
Options are used to list some of the formats supported by ffprobe or
for specifying which information to display, and for setting how
ffprobe will show it.
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.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT" or "STREAM" section.
@c man end
@chapter Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@item -f @var{format}
Force format to use.
@item -unit
Show the unit of the displayed values.
@item -prefix
Use SI prefixes for the displayed values.
Unless the "-byte_binary_prefix" option is used all the prefixes
are decimal.
@item -byte_binary_prefix
Force the use of binary prefixes for byte values.
@item -sexagesimal
Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
@item -pretty
Prettify the format of the displayed values, it corresponds to the
options "-unit -prefix -byte_binary_prefix -sexagesimal".
@item -print_format @var{writer_name}[=@var{writer_options}]
Set the output printing format.
@var{writer_name} specifies the name of the writer, and
@var{writer_options} specifies the options to be passed to the writer.
For example for printing the output in JSON format, specify:
@example
-print_format json
@end example
For more details on the available output printing formats, see the
Writers section below.
@item -show_error
Show information about the error found when trying to probe the input.
The error information is printed within a section with name "ERROR".
@item -show_format
Show information about the container format of the input multimedia
stream.
All the container format information is printed within a section with
name "FORMAT".
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
The information for each single packet is printed within a dedicated
section with name "PACKET".
@item -show_frames
Show information about each frame contained in the input multimedia
stream.
The information for each single frame is printed within a dedicated
section with name "FRAME".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
This option is enabled by default, but you may need to disable it
for specific uses, for example when creating XSD-compliant XML output.
@item -show_program_version
Show information related to program version.
Version information is printed within a section with name
"PROGRAM_VERSION".
@item -show_library_versions
Show information related to library versions.
Version information for each library is printed within a section with
name "LIBRARY_VERSION".
@item -show_versions
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 -i @var{input_file}
Read @var{input_file}.
@end table
@c man end
@chapter Writers
@c man begin WRITERS
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.
A description of the currently available writers follows.
@section default
Default format.
Print each section in the form:
@example
[SECTION]
key1=val1
...
keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT or
STREAM section, and are prefixed by the string "TAG:".
@section compact
Compact format.
Each section is printed on a single line.
If no option is specifid, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
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.
@item nokey, nk
If set to 1 specify not to print the key of each field. Its default
value is 0.
@item escape, e
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') 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}".
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline ('\n'), a carriage return ('\r'), a double quote
('"'), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@end table
@end table
@section csv
CSV format.
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
@item compact, c
If set to 1 enable compact output, that is each section will be
printed on a single line. Default value is 0.
@end table
For more information about JSON, see @url{http://www.json.org/}.
@section xml
XML based format.
The XML output is described in the XML schema description file
@file{ffprobe.xsd} installed in the FFmpeg datadir.
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
@item fully_qualified, q
If set to 1 specify if the output should be fully qualified. Default
value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{http://www.w3.org/XML/}.
@chapter Timecode
@command{ffprobe} supports Timecode extraction:
@itemize
@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
stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
@item DV and GXF timecodes are available in format metadata
(@option{-show_format}, see @var{TAG:timecode}).
@end itemize
@c man end WRITERS
@include decoders.texi
@include demuxers.texi
@include protocols.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

@@ -1,164 +0,0 @@
<?xml version="1.0" encoding="UTF-8"?>
<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
<xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<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="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsType">
<xsd:sequence>
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="framesType">
<xsd:sequence>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float" />
<xsd:attribute name="dts" type="xsd:long" />
<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="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_pts" type="xsd:long" />
<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_pos" type="xsd:long" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<xsd:attribute name="height" type="xsd:long" />
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="coded_picture_number" type="xsd:long" />
<xsd:attribute name="display_picture_number" type="xsd:long" />
<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">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamType">
<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="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"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="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_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string" 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:complexType>
<xsd:complexType name="tagType">
<xsd:attribute name="key" type="xsd:string" use="required"/>
<xsd:attribute name="value" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="errorType">
<xsd:attribute name="code" type="xsd:int" use="required"/>
<xsd:attribute name="string" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string" use="required"/>
<xsd:attribute name="build_time" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_type" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="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:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

View File

@@ -1,46 +1,34 @@
\input texinfo @c -*- texinfo -*-
@settitle ffserver Documentation
@settitle FFserver Documentation
@titlepage
@center @titlefont{ffserver Documentation}
@sp 7
@center @titlefont{FFserver Documentation}
@sp 3
@end titlepage
@top
@contents
@chapter Introduction
@chapter Synopsys
The generic syntax is:
@example
@c man begin SYNOPSIS
ffserver [options]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
ffserver is a streaming server for both audio and video. It supports
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 ffserver.conf).
ffserver runs in daemon mode by default; that is, it puts itself in
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.
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
etc. are not covered here. Read @file{ffmpeg-doc.html} for more
information.
@section How does it work?
ffserver receives prerecorded files or FFM streams from some ffmpeg
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
@@ -58,7 +46,7 @@ file.
@section Status stream
ffserver supports an HTTP interface which exposes the current status
FFserver supports an HTTP interface which exposes the current status
of the server.
Simply point your browser to the address of the special status stream
@@ -111,8 +99,8 @@ As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@example
ffserver -f doc/ffserver.conf &
ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
./ffserver -f doc/ffserver.conf &
./ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
@end example
At this point you should be able to go to your Windows machine and fire up
@@ -147,7 +135,7 @@ that only captures in stereo and also requires that one channel be flipped.
If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
starting ffmpeg.
@subsection The audio and video lose sync after a while.
@subsection The audio and video loose sync after a while.
Yes, they do.
@@ -238,10 +226,18 @@ 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
@chapter Options
@chapter Invocation
@section Syntax
@example
@c man begin SYNOPSIS
ffserver [options]
@c man end
@end example
@section Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@include fftools-common-opts.texi
@section Main options
@@ -250,8 +246,8 @@ For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
Use @file{configfile} instead of @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the Launch directives
within the various <Stream> sections. Since ffserver will not launch
any ffmpeg instances, you will have to launch them manually.
within the various <Stream> sections. FFserver will not launch any
ffmpeg instance, so you will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, directs log
messages to stdout and causes ffserver to run in the foreground
@@ -262,16 +258,15 @@ rather than as a daemon.
@ignore
@setfilename ffserver
@settitle ffserver video server
@settitle FFserver video server
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
example and the FFmpeg HTML documentation
ffmpeg(1), ffplay(1), the @file{ffmpeg/doc/ffserver.conf} example and
the HTML documentation of @file{ffmpeg}.
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man begin AUTHOR
Fabrice Bellard
@c man end
@end ignore

View File

@@ -306,27 +306,6 @@ StartSendOnKey
#</Stream>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<Stream live.h264>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</Stream>
##################################################################
# SDP/multicast examples
#

View File

@@ -0,0 +1,72 @@
@section Generic options
These options are shared amongst the ff* 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 -loglevel @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
@end table

File diff suppressed because it is too large Load Diff

View File

@@ -2,99 +2,35 @@
@settitle General Documentation
@titlepage
@sp 7
@center @titlefont{General Documentation}
@sp 3
@end titlepage
@top
@contents
@chapter External libraries
@chapter external libraries
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 @file{./configure}.
@section OpenJPEG
FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to
@url{http://www.openjpeg.org/} to get the libraries and follow the installation
instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
@file{./configure}.
@section OpenCORE and VisualOn libraries
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 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) to use it.
@end float
@subsection OpenCORE AMR
@section OpenCORE AMR
FFmpeg can make use of the OpenCORE libraries for AMR-NB
decoding/encoding and AMR-WB decoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the libraries.
Then pass @code{--enable-libopencore-amrnb} and/or
@code{--enable-libopencore-amrwb} to configure to enable them.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for
installing the libraries. Then pass @code{--enable-libopencore-amrnb} and/or
@code{--enable-libopencore-amrwb} to configure to enable the libraries.
@subsection VisualOn AAC encoder library
FFmpeg can make use of the VisualOn AACenc library for AAC encoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libvo-aacenc} to configure to enable it.
@subsection VisualOn AMR-WB encoder library
FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding.
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.
@section LAME
FFmpeg can make use of the LAME library for MP3 encoding.
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 libvpx
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 x264
FFmpeg can make use of the x264 library for H.264 encoding.
Go to @url{http://www.videolan.org/developers/x264.html} and follow the
instructions for installing the library. Then pass @code{--enable-libx264} to
configure to enable it.
@float NOTE
x264 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
Note that OpenCORE is under the Apache License 2.0 (see
@url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
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) to use it.
@chapter Supported File Formats, Codecs or Features
@chapter Supported File Formats and Codecs
You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
@@ -108,15 +44,11 @@ library:
@item 4xm @tab @tab X
@tab 4X Technologies format, used in some games.
@item 8088flex TMV @tab @tab X
@item ACT Voice @tab @tab X
@tab contains G.729 audio
@item Adobe Filmstrip @tab X @tab X
@item Audio IFF (AIFF) @tab X @tab X
@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 Apple HTTP Live Streaming @tab @tab X
@item Artworx Data Format @tab @tab X
@item ASF @tab X @tab X
@item AVI @tab X @tab X
@item AVISynth @tab @tab X
@@ -126,24 +58,17 @@ library:
@tab Audio and video format used in some games by Beam Software.
@item Bethesda Softworks VID @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Binary text @tab @tab X
@item Bink @tab @tab X
@tab Multimedia format used by many games.
@item Bitmap Brothers JV @tab @tab X
@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 BWF @tab X @tab X
@item CRI ADX @tab X @tab X
@tab Audio-only format used in console video games.
@item Discworld II BMV @tab @tab X
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@item Delphine Software International CIN @tab @tab X
@tab Multimedia format used by Delphine Software games.
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@item Core Audio Format @tab X @tab X
@item Core Audio Format @tab @tab X
@tab Apple Core Audio Format
@item CRC testing format @tab X @tab
@item Creative Voice @tab X @tab X
@@ -152,8 +77,6 @@ library:
@tab Audio format used in some games by CRYO Interactive Entertainment.
@item D-Cinema audio @tab X @tab X
@item Deluxe Paint Animation @tab @tab X
@item DFA @tab @tab X
@tab This format is used in Chronomaster game
@item DV video @tab X @tab X
@item DXA @tab @tab X
@tab This format is used in the non-Windows version of the Feeble Files
@@ -167,48 +90,32 @@ library:
@tab Only embedded audio is decoded.
@item FLI/FLC/FLX animation @tab @tab X
@tab .fli/.flc files
@item Flash Video (FLV) @tab X @tab X
@item Flash Video (FLV) @tab @tab X
@tab Macromedia Flash video files
@item framecrc testing format @tab X @tab
@item FunCom ISS @tab @tab X
@tab Audio format used in various games from FunCom like The Longest Journey.
@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
@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 @tab X
@tab Microsoft Windows ICO
@item id Quake II CIN video @tab @tab X
@item id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@item IEC61937 encapsulation @tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item IEC61937 encapsulation @tab X @tab
@item IFF @tab @tab X
@tab Interchange File Format
@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 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 LXF @tab @tab X
@tab VR native stream format, used by Leitch/Harris' video servers.
@item Matroska @tab X @tab X
@item Matroska audio @tab X @tab
@item FFmpeg metadata @tab X @tab X
@tab Metadata in text format.
@item MAXIS XA @tab @tab X
@tab Used in Sim City 3000; file extension .xa.
@item MD Studio @tab @tab X
@item Mobotix .mxg @tab @tab X
@item Monkey's Audio @tab @tab X
@item Motion Pixels MVI @tab @tab X
@item MOV/QuickTime/MP4 @tab X @tab X
@@ -242,13 +149,12 @@ library:
@item NUT @tab X @tab X
@tab NUT Open Container Format
@item Ogg @tab X @tab X
@item Playstation Portable PMP @tab @tab X
@item TechnoTrend PVA @tab @tab X
@tab Used by TechnoTrend DVB PCI boards.
@item QCP @tab @tab X
@item raw ADTS (AAC) @tab X @tab X
@item raw AC-3 @tab X @tab X
@item raw Chinese AVS video @tab X @tab X
@item raw Chinese AVS video @tab @tab X
@item raw CRI ADX @tab X @tab X
@item raw Dirac @tab X @tab X
@item raw DNxHD @tab X @tab X
@@ -301,13 +207,10 @@ library:
@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 RTMP @tab X @tab X
@tab Output is performed by publishing stream to RTMP server
@item RTP @tab X @tab X
@item RTP @tab @tab X
@item RTSP @tab X @tab X
@item SAP @tab X @tab X
@item SBG @tab @tab X
@item SDP @tab @tab X
@item Sega FILM/CPK @tab @tab X
@tab Used in many Sega Saturn console games.
@@ -317,15 +220,12 @@ library:
@tab Used in Sierra CD-ROM games.
@item Smacker @tab @tab X
@tab Multimedia format used by many games.
@item SMJPEG @tab X @tab X
@tab Used in certain Loki game ports.
@item Sony OpenMG (OMA) @tab X @tab X
@item Sony OpenMG (OMA) @tab @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 @tab X
@item SoX native format @tab X @tab X
@item SUN AU format @tab X @tab X
@item Text files @tab @tab X
@item THP @tab @tab X
@tab Used on the Nintendo GameCube.
@item Tiertex Limited SEQ @tab @tab X
@@ -335,18 +235,12 @@ library:
@item WAV @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
@tab Multimedia format used in Origin's Wing Commander III computer game.
@item Westwood Studios audio @tab @tab X
@tab Multimedia format used in Westwood Studios games.
@item Westwood Studios VQA @tab @tab X
@tab Multimedia format used in Westwood Studios games.
@item XMV @tab @tab X
@tab Microsoft video container used in Xbox games.
@item xWMA @tab @tab X
@tab Microsoft audio container used by XAudio 2.
@item eXtended BINary text (XBIN) @tab @tab X
@item YUV4MPEG pipe @tab X @tab X
@item Psygnosis YOP @tab @tab X
@end multitable
@@ -366,11 +260,12 @@ following image formats are supported:
@tab Only uncompressed GIFs are generated.
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item DPX @tab X @tab X
@item DPX @tab @tab X
@tab Digital Picture Exchange
@item JPEG @tab X @tab X
@tab Progressive JPEG is not supported.
@item JPEG 2000 @tab X @tab X
@item JPEG 2000 @tab @tab E
@tab decoding supported through external library libopenjpeg
@item JPEG-LS @tab X @tab X
@item LJPEG @tab X @tab
@tab Lossless JPEG
@@ -384,9 +279,8 @@ following image formats are supported:
@tab Portable GrayMap image
@item PGMYUV @tab X @tab X
@tab PGM with U and V components in YUV 4:2:0
@item PIC @tab @tab X
@tab Pictor/PC Paint
@item PNG @tab X @tab X
@tab 2/4 bpp not supported yet
@item PPM @tab X @tab X
@tab Portable PixelMap image
@item PTX @tab @tab X
@@ -399,8 +293,6 @@ following image formats are supported:
@tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa @tab X @tab X
@tab Targa (.TGA) image format
@item XWD @tab X @tab X
@tab X Window Dump image format
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@@ -416,15 +308,11 @@ following image formats are supported:
@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 American Laser Games MM @tab @tab X
@tab Used in games like Mad Dog McCree.
@item AMV Video @tab X @tab X
@item AMV Video @tab @tab X
@tab Used in Chinese MP3 players.
@item ANSI/ASCII art @tab @tab X
@item Apple MJPEG-B @tab @tab X
@item Apple ProRes @tab X @tab X
@item Apple QuickDraw @tab @tab X
@tab fourcc: qdrw
@item Asus v1 @tab X @tab X
@@ -440,16 +328,13 @@ following image formats are supported:
@item Autodesk Animator Flic video @tab @tab X
@item Autodesk RLE @tab @tab X
@tab fourcc: AASC
@item Avid 1:1 10-bit RGB Packer @tab X @tab X
@tab fourcc: AVrp
@item AVS (Audio Video Standard) video @tab @tab X
@tab Video encoding used by the Creature Shock game.
@item Beam Software VB @tab @tab X
@item Bethesda VID video @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Bink Video @tab @tab X
@item Bitmap Brothers JV video @tab @tab X
@item y41p Brooktree uncompressed 4:1:1 12-bit @tab X @tab X
@tab Support for version 'b' is missing.
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item C93 video @tab @tab X
@@ -458,18 +343,15 @@ following image formats are supported:
@tab fourcc: CSCD
@item CD+G @tab @tab X
@tab Video codec for CD+G karaoke disks
@item Chinese AVS video @tab E @tab X
@tab AVS1-P2, JiZhun profile, encoding through external library libxavs
@item Chinese AVS video @tab @tab X
@tab AVS1-P2, JiZhun profile
@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 Cinepak @tab @tab X
@item Cirrus Logic AccuPak @tab X @tab X
@item Cirrus Logic AccuPak @tab @tab X
@tab fourcc: CLJR
@item Creative YUV (CYUV) @tab @tab X
@item DFA @tab @tab X
@tab Codec used in Chronomaster game.
@item Dirac @tab E @tab X
@item Dirac @tab E @tab E
@tab supported through external libdirac/libschroedinger libraries
@item Deluxe Paint Animation @tab @tab X
@item DNxHD @tab X @tab X
@@ -479,7 +361,6 @@ following image formats are supported:
@item Duck TrueMotion 2.0 @tab @tab X
@tab fourcc: TM20
@item DV (Digital Video) @tab X @tab X
@item Dxtory capture format @tab @tab X
@item Feeble Files/ScummVM DXA @tab @tab X
@tab Codec originally used in Feeble Files game.
@item Electronic Arts CMV video @tab @tab X
@@ -489,12 +370,10 @@ following image formats are supported:
@item Electronic Arts TGQ video @tab @tab X
@item Electronic Arts TQI video @tab @tab X
@item Escape 124 @tab @tab X
@item Escape 130 @tab @tab X
@item FFmpeg video codec #1 @tab X @tab X
@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 Fraps @tab @tab X
@@ -513,24 +392,21 @@ following image formats are supported:
@item id RoQ video @tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item IFF ILBM @tab @tab X
@tab IFF interleaved bitmap
@tab IFF interlaved bitmap
@item IFF ByteRun1 @tab @tab X
@tab IFF run length encoded bitmap
@item Intel H.263 @tab @tab X
@item Intel Indeo 2 @tab @tab X
@item Intel Indeo 3 @tab @tab X
@item Intel Indeo 4 @tab @tab X
@item Intel Indeo 5 @tab @tab X
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@item Interplay MVE video @tab @tab X
@tab Used in Interplay .MVE files.
@item J2K @tab X @tab X
@item Karl Morton's video codec @tab @tab X
@tab Codec used in Worms games.
@item Kega Game Video (KGV1) @tab @tab X
@tab Kega emulator screen capture codec.
@item Lagarith @tab @tab X
@item LCL (LossLess Codec Library) MSZH @tab @tab X
@item LCL (LossLess Codec Library) ZLIB @tab E @tab E
@item LOCO @tab @tab X
@@ -542,15 +418,14 @@ following image formats are supported:
@item Miro VideoXL @tab @tab X
@tab fourcc: VIXL
@item MJPEG (Motion JPEG) @tab X @tab X
@item Mobotix MxPEG video @tab @tab X
@item Motion Pixels video @tab @tab X
@item MPEG-1 video @tab X @tab X
@item MPEG-1/2 video XvMC (X-Video Motion Compensation) @tab @tab X
@item MPEG-1/2 video (VDPAU acceleration) @tab @tab X
@item MPEG-2 video @tab X @tab X
@item MPEG-4 part 2 @tab X @tab X
@tab libxvidcore can be used alternatively for encoding.
@item MPEG-4 part 2 Microsoft variant version 1 @tab @tab X
@ libxvidcore can be used alternatively for encoding.
@item MPEG-4 part 2 Microsoft variant version 1 @tab X @tab X
@item MPEG-4 part 2 Microsoft variant version 2 @tab X @tab X
@item MPEG-4 part 2 Microsoft variant version 3 @tab X @tab X
@item Nintendo Gamecube THP video @tab @tab X
@@ -562,12 +437,10 @@ following image formats are supported:
@tab fourcc: VP50
@item On2 VP6 @tab @tab X
@tab fourcc: VP60,VP61,VP62
@item VP8 @tab E @tab X
@tab fourcc: VP80, encoding supported through external library libvpx
@item VP8 @tab X @tab X
@tab fourcc: VP80, de/encoding supported through external library libvpx
@item planar RGB @tab @tab X
@tab fourcc: 8BPS
@item Prores @tab @tab X
@tab fourcc: apch,apcn,apcs,apco
@item Q-team QPEG @tab @tab X
@tab fourccs: QPEG, Q1.0, Q1.1
@item QuickTime 8BPS video @tab @tab X
@@ -577,8 +450,7 @@ following image formats are supported:
@tab fourcc: 'smc '
@item QuickTime video (RPZA) @tab @tab X
@tab fourcc: rpza
@item R10K AJA Kona 10-bit RGB Codec @tab X @tab X
@item R210 Quicktime Uncompressed RGB 10-bit @tab X @tab X
@item R210 Quicktime Uncompressed RGB 10-bit @tab @tab X
@item Raw Video @tab X @tab X
@item RealVideo 1.0 @tab X @tab X
@item RealVideo 2.0 @tab X @tab X
@@ -609,29 +481,20 @@ following image formats are supported:
@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 @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 v410 QuickTime uncompressed 4:4:4 10-bit @tab X @tab X
@item VBLE Lossless Codec @tab @tab X
@item V210 Quicktime Uncompressed 4:2:2 10-bit @tab X @tab X
@item VMware Screen Codec / VMware Video @tab @tab X
@tab Codec used in videos captured by VMware.
@item Westwood Studios VQA (Vector Quantized Animation) video @tab @tab X
@item Windows Media Image @tab @tab X
@item Windows Media Video 7 @tab X @tab X
@item Windows Media Video 8 @tab X @tab X
@item Windows Media Video 9 @tab @tab X
@tab not completely working
@item Wing Commander III / Xan @tab @tab X
@tab Used in Wing Commander III .MVE files.
@item Wing Commander IV / Xan @tab @tab X
@tab Used in Wing Commander IV.
@item Winnov WNV1 @tab @tab X
@item WMV7 @tab X @tab X
@item YAMAHA SMAF @tab X @tab X
@item Psygnosis YOP Video @tab @tab X
@item yuv4 @tab X @tab X
@tab libquicktime uncompressed packed 4:2:0
@item ZLIB @tab X @tab X
@tab part of LCL, encoder experimental
@item Zip Motion Blocks Video @tab X @tab X
@@ -647,8 +510,6 @@ following image formats are supported:
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX audio @tab @tab X
@item AAC+ @tab E @tab X
@tab encoding supported through external library libaacplus
@item AAC @tab E @tab X
@tab encoding supported through external library libfaac and libvo-aacenc
@item AC-3 @tab IX @tab X
@@ -664,7 +525,6 @@ following image formats are supported:
@item ADPCM Electronic Arts R2 @tab @tab X
@item ADPCM Electronic Arts R3 @tab @tab X
@item ADPCM Electronic Arts XAS @tab @tab X
@item ADPCM G.722 @tab X @tab X
@item ADPCM G.726 @tab X @tab X
@item ADPCM IMA AMV @tab @tab X
@tab Used in AMV files
@@ -698,24 +558,21 @@ following image formats are supported:
@item ADPCM Yamaha @tab X @tab X
@item AMR-NB @tab E @tab X
@tab encoding supported through external library libopencore-amrnb
@item AMR-WB @tab E @tab X
@tab encoding supported through external library libvo-amrwbenc
@item AMR-WB @tab @tab E
@tab decoding supported through external library libopencore-amrwb
@item Apple lossless audio @tab X @tab X
@tab QuickTime fourcc 'alac'
@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
@tab decoding supported through external library libcelt
@item Delphine Software International CIN audio @tab @tab X
@tab Codec used in Delphine Software International games.
@item Discworld II BMV Audio @tab @tab X
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
@item DCA (DTS Coherent Acoustics) @tab X @tab X
@item DCA (DTS Coherent Acoustics) @tab @tab X
@item DPCM id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item DPCM Interplay @tab @tab X
@tab Used in various Interplay computer games.
@item DPCM Sierra Online @tab @tab X
@@ -725,14 +582,13 @@ following image formats are supported:
@tab Used in Origin's Wing Commander IV AVI files.
@item DSP Group TrueSpeech @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab X @tab X
@item Enhanced AC-3 @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
@item GSM @tab E @tab X
@tab encoding supported through external library libgsm
@item GSM Microsoft variant @tab E @tab X
@tab encoding supported through external library libgsm
@item GSM @tab E @tab E
@tab supported through external library libgsm
@item GSM Microsoft variant @tab E @tab E
@tab supported through external library libgsm
@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
@@ -771,11 +627,11 @@ following image formats are supported:
@item PCM unsigned 24-bit little-endian @tab X @tab X
@item PCM unsigned 32-bit big-endian @tab X @tab X
@item PCM unsigned 32-bit little-endian @tab X @tab X
@item PCM Zork @tab @tab X
@item PCM Zork @tab X @tab X
@item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 2 @tab @tab X
@tab There are still some distortions.
@item RealAudio 1.0 (14.4K) @tab X @tab X
@item RealAudio 1.0 (14.4K) @tab @tab X
@tab Real 14400 bit/s codec
@item RealAudio 2.0 (28.8K) @tab @tab X
@tab Real 28800 bit/s codec
@@ -786,12 +642,11 @@ following image formats are supported:
@item Sierra VMD audio @tab @tab X
@tab Used in Sierra VMD files.
@item Smacker audio @tab @tab X
@item SMPTE 302M AES3 audio @tab @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
@item Speex @tab @tab E
@tab supported through external library libspeex
@item True Audio (TTA) @tab @tab X
@item TrueHD @tab @tab X
@@ -818,12 +673,10 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1 .1 .1 .1
@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
@item SSA/ASS @tab X @tab X @tab X @tab X
@item SSA/ASS @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
@@ -833,11 +686,9 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1
@item Name @tab Support
@item Apple HTTP Live Streaming @tab X
@item file @tab X
@item Gopher @tab X
@item HTTP @tab X
@item MMS @tab X
@item pipe @tab X
@item RTP @tab X
@item TCP @tab X
@@ -852,12 +703,12 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1 .1
@item Name @tab Input @tab Output
@item ALSA @tab X @tab X
@item BEOS audio @tab X @tab X
@item BKTR @tab X @tab
@item DV1394 @tab X @tab
@item JACK @tab X @tab
@item LIBDC1394 @tab X @tab
@item OSS @tab X @tab X
@item Pulseaudio @tab X @tab
@item Video4Linux @tab X @tab
@item Video4Linux2 @tab X @tab
@item VfW capture @tab X @tab
@@ -866,15 +717,345 @@ performance on systems without hardware floating point support).
@code{X} means that input/output is supported.
@section Timecode
@multitable @columnfractions .4 .1 .1
@item Codec/format @tab Read @tab Write
@item DV @tab X @tab X
@item GXF @tab X @tab X
@item MOV @tab X @tab
@item MPEG1/2 @tab X @tab X
@item MXF @tab @tab X
@end multitable
@chapter Platform Specific information
@section BeOS
BeOS support is broken in mysterious ways.
@section DOS
Using a cross-compiler is preferred for various reasons.
@subsection DJGPP
FFmpeg cannot be compiled because of broken system headers, add
@code{--extra-cflags=-U__STRICT_ANSI__} to the configure options as a
workaround.
@section OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@section Unix-like
Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
@example
$(gcc -print-prog-name=as) --version
@end example
If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass @code{--disable-asm}
to configure.
@subsection BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
(@file{gmake}).
@subsubsection FreeBSD
FreeBSD will not compile out-of-the-box due to broken system headers.
Passing @code{--extra-cflags=-D__BSD_VISIBLE} to configure will work
around the problem. This may have unexpected sideeffects, so use it at
your own risk. If you care about FreeBSD, please make an attempt at
getting the system headers fixed.
@subsection (Open)Solaris
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
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
@example
bash ./configure
@end example
@section Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@subsection Native Windows compilation
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}).
Within the MSYS shell, configure and make with:
@example
./configure --enable-memalign-hack
make
make install
@end example
This will install @file{ffmpeg.exe} along with many other development files
to @file{/usr/local}. You may specify another install path using the
@code{--prefix} option in @file{configure}.
Notes:
@itemize
@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 in
@code{make install}).
@item In order to compile FFplay, you must have the MinGW development library
of SDL. Get it from @url{http://www.libsdl.org}.
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 libavutil, libavcodec and libavformat as DLLs.
@end itemize
@subsection Microsoft Visual C++ compatibility
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.
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.
@subsubsection Using static libraries
Assuming you have just built and installed FFmpeg in @file{/usr/local}.
@enumerate
@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.
@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
@subsubsection 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
call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"
@end example
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 --enable-memalign-hack
make
make install
@end example
Your install path (@file{/usr/local/} by default) should now have the
necessary DLL and LIB files under the @file{bin} directory.
@end enumerate
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 @file{libxx.a}
files, you should add @file{avcodec.lib}, @file{avformat.lib}, and
@file{avutil.lib}. 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. The @file{bin} directory contains a bunch
of DLL files, but the ones that are actually used to run your application
are the ones with a major version number in their filenames
(i.e. @file{avcodec-51.dll}).
@subsection Cross compilation for Windows with Linux
You must use the MinGW cross compilation tools available at
@url{http://www.mingw.org/}.
Then configure FFmpeg with the following options:
@example
./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
@end example
(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
Then you can easily test FFmpeg with Wine
(@url{http://www.winehq.com/}).
@subsection Compilation under Cygwin
Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
Install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, subversion, mingw-runtime, texi2html
@end example
And the following "Utils" one:
@example
diffutils
@end example
Then run
@example
./configure --enable-static --disable-shared
@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
"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
@example
libogg-devel, libvorbis-devel
@end example
These library packages are only available from Cygwin Ports
(@url{http://sourceware.org/cygwinports/}) :
@example
yasm, libSDL-devel, libdirac-devel, libfaac-devel, libfaad-devel, libgsm-devel,
libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel,
libxvidcore-devel
@end example
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.
@subsection Crosscompilation for Windows under Cygwin
With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
Just install your Cygwin as explained before, plus these additional
"Devel" packages:
@example
gcc-mingw-core, mingw-runtime, mingw-zlib
@end example
and add some special flags to your configure invocation.
For a static build run
@example
./configure --target-os=mingw32 --enable-memalign-hack --enable-static --disable-shared --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
and for a build with shared libraries
@example
./configure --target-os=mingw32 --enable-memalign-hack --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@bye

View File

@@ -1,344 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Using git to develop FFmpeg
@titlepage
@center @titlefont{Using git to develop FFmpeg}
@end titlepage
@top
@contents
@chapter Introduction
This document aims in giving some quick references on a set of useful git
commands. You should always use the extensive and detailed documentation
provided directly by git:
@example
git --help
man git
@end example
shows you the available subcommands,
@example
git <command> --help
man git-<command>
@end example
shows information about the subcommand <command>.
Additional information could be found on the
@url{http://gitref.org, Git Reference} website
For more information about the Git project, visit the
@url{http://git-scm.com/, Git website}
Consult these resources whenever you have problems, they are quite exhaustive.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project
@chapter Basics Usage
@section Get GIT
You can get git from @url{http://git-scm.com/}
Most distribution and operating system provide a package for it.
@section Cloning the source tree
@example
git clone git://source.ffmpeg.org/ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>}.
@example
git clone git@@source.ffmpeg.org:ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>} and let
you push back your changes to the remote repository.
@section Updating the source tree to the latest revision
@example
git pull (--rebase)
@end example
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
@float IMPORTANT
@command{--rebase} (see below) is recommended.
@end float
@section Rebasing your local branches
@example
git pull --rebase
@end example
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
@section Adding/removing files/directories
@example
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
@end example
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
@section Showing modifications
@example
git diff <filename(s)>
@end example
will show all local modifications in your working directory as unified diff.
@section Inspecting the changelog
@example
git log <filename(s)>
@end example
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org/
@section Checking source tree status
@example
git status
@end example
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
@section Committing
@example
git diff --check
@end example
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
@example
git diff -w -b <filename(s)>
@end example
Also check the output of
@example
git status
@end example
to make sure you don't have untracked files or deletions.
@example
git add [-i|-p|-A] <filenames/dirnames>
@end example
Make sure you have told git your name and email address
@example
git config --global user.name "My Name"
git config --global user.email my@@email.invalid
@end example
Use @var{--global} to set the global configuration for all your git checkouts.
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
@example
git commit
@end example
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
@example
git config --global core.editor
@end example
or set by one of the following environment variables:
@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
@section Preparing a patchset
@example
git format-patch <commit> [-o directory]
@end example
will generate a set of patches for each commit between @var{<commit>} and
current @var{HEAD}. E.g.
@example
git format-patch origin/master
@end example
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
@example
git format-patch -n
@end example
which will generate patches from last @var{n} commits.
By default the patches are created in the current directory.
@section Sending patches for review
@example
git send-email <commit list|directory>
@end example
will send the patches created by @command{git format-patch} or directly
generates them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. @var{git-email}
package on Debian-based distros).
@section Renaming/moving/copying files or contents of files
Git automatically tracks such changes, making those normal commits.
@example
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
@end example
@chapter FFmpeg specific
@section Reverting broken commits
@example
git reset <commit>
@end example
@command{git reset} will uncommit the changes till @var{<commit>} rewriting
the current branch history.
@example
git commit --amend
@end example
allows to amend the last commit details quickly.
@example
git rebase -i origin/master
@end example
will replay local commits over the main repository allowing to edit, merge
or remove some of them in the process.
@float NOTE
@command{git reset}, @command{git commit --amend} and @command{git rebase}
rewrite history, so you should use them ONLY on your local or topic branches.
The main repository will reject those changes.
@end float
@example
git revert <commit>
@end example
@command{git revert} will generate a revert commit. This will not make the
faulty commit disappear from the history.
@section Pushing changes to remote trees
@example
git push
@end example
Will push the changes to the default remote (@var{origin}).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to and to sync the local tree.
@example
git remote add <name> <url>
@end example
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
@example
git push <remote> <refspec>
@end example
Will push the changes to the @var{<remote>} repository.
Omitting @var{<refspec>} makes @command{git push} update all the remote
branches matching the local ones.
@section Finding a specific svn revision
Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
based on a regular expression. see man gitrevisions
@example
git show :/'as revision 23456'
@end example
will show the svn changeset @var{r23456}. With older git versions searching in
the @command{git log} output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
@example
git checkout -b svn_23456 :/'as revision 23456'
@end example
or for git < 1.7.1 with
@example
git checkout -b svn_23456 $SHA1
@end example
where @var{$SHA1} is the commit hash from the @command{git log} output.
@chapter Server Issues
Contact the project admins @email{root@@ffmpeg.org} if you have technical
problems with the GIT server.

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@@ -1,273 +0,0 @@
About Git write access:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Before everything else, you should know how to use GIT properly.
Luckily Git comes with excellent documentation.
git --help
man git
shows you the available subcommands,
git <command> --help
man git-<command>
shows information about the subcommand <command>.
The most comprehensive manual is the website Git Reference
http://gitref.org/
For more information about the Git project, visit
http://git-scm.com/
Consult these resources whenever you have problems, they are quite exhaustive.
You do not need a special username or password.
All you need is to provide a ssh public key to the Git server admin.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines. Read it at least once, if you are granted commit privileges to the
FFmpeg project you are expected to be familiar with these rules.
I. BASICS:
==========
0. Get GIT:
Most distributions have a git package, if not
You can get git from http://git-scm.com/
1. Cloning the source tree:
git clone git://source.ffmpeg.org/ffmpeg <target>
This will put the FFmpeg sources into the directory <target>.
git clone git@source.ffmpeg.org:ffmpeg <target>
This will put the FFmpeg sources into the directory <target> and let
you push back your changes to the remote repository.
2. Updating the source tree to the latest revision:
git pull (--ff-only)
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
Caveat: Since merge commits are forbidden at least for the initial
months of git --ff-only or --rebase (see below) are recommended.
--ff-only will fail and not create merge commits if your branch
has diverged (has a different history) from the tracked branch.
2.a Rebasing your local branches:
git pull --rebase
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
3. Adding/removing files/directories:
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
4. Showing modifications:
git diff <filename(s)>
will show all local modifications in your working directory as unified diff.
5. Inspecting the changelog:
git log <filename(s)>
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org
6. Checking source tree status:
git status
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
7. Committing:
git diff --check
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
git diff -w -b <filename(s)>
Also check the output of
git status
to make sure you don't have untracked files or deletions.
git add [-i|-p|-A] <filenames/dirnames>
Make sure you have told git your name and email address, e.g. by running
git config --global user.name "My Name"
git config --global user.email my@email.invalid
(--global to set the global configuration for all your git checkouts).
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
git commit
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
git config core.editor
or set by one of the following environment variables:
GIT_EDITOR, VISUAL or EDITOR.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
8. Renaming/moving/copying files or contents of files:
Git automatically tracks such changes, making those normal commits.
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
Do not move, rename or copy files of which you are not the maintainer without
discussing it on the mailing list first!
9. Reverting broken commits
git revert <commit>
git revert will generate a revert commit. This will not make the faulty
commit disappear from the history.
git reset <commit>
git reset will uncommit the changes till <commit> rewriting the current
branch history.
git commit --amend
allows to amend the last commit details quickly.
git rebase -i origin/master
will replay local commits over the main repository allowing to edit,
merge or remove some of them in the process.
Note that the reset, commit --amend and rebase rewrite history, so you
should use them ONLY on your local or topic branches.
The main repository will reject those changes.
10. Preparing a patchset.
git format-patch <commit> [-o directory]
will generate a set of patches for each commit between <commit> and
current HEAD. E.g.
git format-patch origin/master
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
git format-patch -n
which will generate patches from last n commits.
By default the patches are created in the current directory.
11. Sending patches for review
git send-email <commit list|directory>
will send the patches created by git format-patch or directly generates
them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. git-email
package on Debian-based distros).
12. Pushing changes to remote trees
git push
Will push the changes to the default remote (origin).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to 2 and 2.a to sync the local tree.
git remote add <name> <url>
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
git push <remote> <refspec>
Will push the changes to the remote repository. Omitting refspec makes git
push update all the remote branches matching the local ones.
13. Finding a specific svn revision
Since version 1.7.1 git supports ':/foo' syntax for specifying commits
based on a regular expression. see man gitrevisions
git show :/'as revision 23456'
will show the svn changeset r23456. With older git versions searching in
the git log output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
git checkout -b svn_23456 :/'as revision 23456'
or for git < 1.7.1 with
git checkout -b svn_23456 $SHA1
where $SHA1 is the commit SHA1 from the 'git log' output.
Contact the project admins <root at ffmpeg dot org> if you have technical
problems with the GIT server.

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@@ -1,632 +0,0 @@
@chapter Input Devices
@c man begin INPUT DEVICES
Input devices are configured elements in FFmpeg which allow to access
the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices
are enabled by default. You can list all available ones using the
configure option "--list-indevs".
You can disable all the input devices using the configure option
"--disable-indevs", and selectively enable an input device using the
option "--enable-indev=@var{INDEV}", or you can disable a particular
input device using the option "--disable-indev=@var{INDEV}".
The option "-formats" of the ff* tools will display the list of
supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound
installed on your system.
This device allows capturing from an ALSA device. The name of the
device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
@example
hw:@var{CARD}[,@var{DEV}[,@var{SUBDEV}]]
@end example
where the @var{DEV} and @var{SUBDEV} components are optional.
The three arguments (in order: @var{CARD},@var{DEV},@var{SUBDEV})
specify card number or identifier, device number and subdevice number
(-1 means any).
To see the list of cards currently recognized by your system check the
files @file{/proc/asound/cards} and @file{/proc/asound/devices}.
For example to capture with @command{ffmpeg} from an ALSA device with
card id 0, you may run the command:
@example
ffmpeg -f alsa -i hw:0 alsaout.wav
@end example
For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
@section bktr
BSD video input device.
@section dshow
Windows DirectShow input device.
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
opened on the same input, which should improve synchronism between them.
The input name should be in the format:
@example
@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}]
@end example
where @var{TYPE} can be either @var{audio} or @var{video},
and @var{NAME} is the device's name.
@subsection Options
If no options are specified, the device's defaults are used.
If the device does not support the requested options, it will
fail to open.
@table @option
@item video_size
Set the video size in the captured video.
@item framerate
Set the framerate in the captured video.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
@item sample_size
Set the sample size (in bits) of the captured audio.
@item channels
Set the number of channels in the captured audio.
@item list_devices
If set to @option{true}, print a list of devices and exit.
@item list_options
If set to @option{true}, print a list of selected device's options
and exit.
@item video_device_number
Set video device number for devices with same name (starts at 0,
defaults to 0).
@item audio_device_number
Set audio device number for devices with same name (starts at 0,
defaults to 0).
@end table
@subsection Examples
@itemize
@item
Print the list of DirectShow supported devices and exit:
@example
$ ffmpeg -list_devices true -f dshow -i dummy
@end example
@item
Open video device @var{Camera}:
@example
$ ffmpeg -f dshow -i video="Camera"
@end example
@item
Open second video device with name @var{Camera}:
@example
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
@end example
@item
Open video device @var{Camera} and audio device @var{Microphone}:
@example
$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
@end example
@item
Print the list of supported options in selected device and exit:
@example
$ ffmpeg -list_options true -f dshow -i video="Camera"
@end example
@end itemize
@section dv1394
Linux DV 1394 input device.
@section fbdev
Linux framebuffer input 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
Documentation/fb/framebuffer.txt included in the Linux source tree.
To record from the framebuffer device @file{/dev/fb0} with
@command{ffmpeg}:
@example
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
@end example
You can take a single screenshot image with the command:
@example
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
@end example
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section jack
JACK input device.
To enable this input device during configuration you need libjack
installed on your system.
A JACK input device creates one or more JACK writable clients, one for
each audio channel, with name @var{client_name}:input_@var{N}, where
@var{client_name} is the name provided by the application, and @var{N}
is a number which identifies the channel.
Each writable client will send the acquired data to the FFmpeg input
device.
Once you have created one or more JACK readable clients, you need to
connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the @command{jack_connect}
and @command{jack_disconnect} programs, or do it through a graphical interface,
for example with @command{qjackctl}.
To list the JACK clients and their properties you can invoke the command
@command{jack_lsp}.
Follows an example which shows how to capture a JACK readable client
with @command{ffmpeg}.
@example
# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav
# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000
# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm
# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1
@end example
For more information read:
@url{http://jackaudio.org/}
@section lavfi
Libavfilter input virtual device.
This input device reads data from the open output pads of a libavfilter
filtergraph.
For each filtergraph open output, the input device will create a
corresponding stream which is mapped to the generated output. Currently
only video data is supported. The filtergraph is specified through the
option @option{graph}.
@subsection Options
@table @option
@item graph
Specify the filtergraph to use as input. Each video open output must be
labelled by a unique string of the form "out@var{N}", where @var{N} is a
number starting from 0 corresponding to the mapped input stream
generated by the device.
The first unlabelled output is automatically assigned to the "out0"
label, but all the others need to be specified explicitly.
If not specified defaults to the filename specified for the input
device.
@end table
@subsection Examples
@itemize
@item
Create a color video stream and play it back with @command{ffplay}:
@example
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=pink
@end example
@item
Create three different video test filtered sources and play them:
@example
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
@end example
@item
Read an audio stream from a file using the amovie source and play it
back with @command{ffplay}:
@example
ffplay -f lavfi "amovie=test.wav"
@end example
@item
Read an audio stream and a video stream and play it back with
@command{ffplay}:
@example
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end example
@end itemize
@section libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.
@section openal
The OpenAL input device provides audio capture on all systems with a
working OpenAL 1.1 implementation.
To enable this input device during configuration, you need OpenAL
headers and libraries installed on your system, and need to configure
FFmpeg with @code{--enable-openal}.
OpenAL headers and libraries should be provided as part of your OpenAL
implementation, or as an additional download (an SDK). Depending on your
installation you may need to specify additional flags via the
@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build
system to locate the OpenAL headers and libraries.
An incomplete list of OpenAL implementations follows:
@table @strong
@item Creative
The official Windows implementation, providing hardware acceleration
with supported devices and software fallback.
See @url{http://openal.org/}.
@item OpenAL Soft
Portable, open source (LGPL) software implementation. Includes
backends for the most common sound APIs on the Windows, Linux,
Solaris, and BSD operating systems.
See @url{http://kcat.strangesoft.net/openal.html}.
@item Apple
OpenAL is part of Core Audio, the official Mac OS X Audio interface.
See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
@end table
This device allows to capture from an audio input device handled
through OpenAL.
You need to specify the name of the device to capture in the provided
filename. If the empty string is provided, the device will
automatically select the default device. You can get the list of the
supported devices by using the option @var{list_devices}.
@subsection Options
@table @option
@item channels
Set the number of channels in the captured audio. Only the values
@option{1} (monaural) and @option{2} (stereo) are currently supported.
Defaults to @option{2}.
@item sample_size
Set the sample size (in bits) of the captured audio. Only the values
@option{8} and @option{16} are currently supported. Defaults to
@option{16}.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
Defaults to @option{44.1k}.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@end table
@subsection Examples
Print the list of OpenAL supported devices and exit:
@example
$ ffmpeg -list_devices true -f openal -i dummy out.ogg
@end example
Capture from the OpenAL device @file{DR-BT101 via PulseAudio}:
@example
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
@end example
Capture from the default device (note the empty string '' as filename):
@example
$ ffmpeg -f openal -i '' out.ogg
@end example
Capture from two devices simultaneously, writing to two different files,
within the same @command{ffmpeg} command:
@example
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
@end example
Note: not all OpenAL implementations support multiple simultaneous capture -
try the latest OpenAL Soft if the above does not work.
@section oss
Open Sound System input device.
The filename to provide to the input device is the device node
representing the OSS input device, and is usually set to
@file{/dev/dsp}.
For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the
command:
@example
ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
@end example
For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}
@section pulse
pulseaudio input device.
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 pulse source devices and their properties you can invoke
the command @command{pactl list sources}.
@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.
To enable this input device during configuration you need libsndio
installed on your system.
The filename to provide to the input device is the device node
representing the sndio input device, and is usually set to
@file{/dev/audio0}.
For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the
command:
@example
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example
@section video4linux and video4linux2
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
(e.g. an USB webcam) is plugged into the system, and has a name of the
kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
the device.
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.
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.
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
# Grab and show the input of a video4linux device, frame rate is set
# to the default of 25/1.
ffplay -s 320x240 -f video4linux /dev/video0
# Grab and show the input of a video4linux2 device, auto-adjust size.
ffplay -f video4linux2 /dev/video0
# Grab and record the input of a video4linux2 device, auto-adjust size,
# frame rate value defaults to 0/0 so it is read from the video4linux2
# driver.
ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
@end example
"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
"video4linux2".
@section vfwcap
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from
0 to 9. You may use "list" as filename to print a list of drivers. Any
other filename will be interpreted as device number 0.
@section x11grab
X11 video input device.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
@example
[@var{hostname}]:@var{display_number}.@var{screen_number}[+@var{x_offset},@var{y_offset}]
@end example
@var{hostname}:@var{display_number}.@var{screen_number} specifies the
X11 display name of the screen to grab from. @var{hostname} can be
omitted, and defaults to "localhost". The environment variable
@env{DISPLAY} contains the default display name.
@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
area with respect to the top-left border of the X11 screen. They
default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the @command{dpyinfo} program for getting basic information about the
properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
# Grab at position 10,20.
ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
@end example
@subsection @var{follow_mouse} AVOption
The syntax is:
@example
-follow_mouse centered|@var{PIXELS}
@end example
When it is specified with "centered", the grabbing region follows the mouse
pointer and keeps the pointer at the center of region; otherwise, the region
follows only when the mouse pointer reaches within @var{PIXELS} (greater than
zero) to the edge of region.
For example:
@example
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
@subsection @var{show_region} AVOption
The syntax is:
@example
-show_region 1
@end example
If @var{show_region} AVOption is specified with @var{1}, then the grabbing
region will be indicated on screen. With this option, it's easy to know what is
being grabbed if only a portion of the screen is grabbed.
For example:
@example
ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
# With follow_mouse
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
@end example
@c man end INPUT DEVICES

View File

@@ -5,42 +5,49 @@ NOTE: This is a draft.
Overview:
---------
FFmpeg uses Trac for tracking issues, new issues and changes to
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, 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.
FFmpeg uses Roundup for tracking issues, new issues and changes to
existing issues can be done through a web interface and through email.
It is possible to subscribe to individual issues by adding yourself to the
Cc list or to subscribe to the ffmpeg-trac mailing list which receives
a mail for every change to every issue.
nosy list or to subscribe to the ffmpeg-issues mailing list which receives
a mail for every change to every issue. Replies to such mails will also
be properly added to the respective issue.
(the above does all work already after light testing)
The subscription URL for the ffmpeg-trac list is:
http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
The subscription URL for the ffmpeg-issues list is:
http://live.polito/mailman/listinfo/ffmpeg-issues
The URL of the webinterface of the tracker is:
http(s)://trac.ffmpeg.org
http(s)://roundup.ffmpeg/roundup/ffmpeg/
Note the URLs in this document are obfuscated, you must append the top level
domain for non-profit organizations to the tracker, and of Italy to the
mailing list.
Email Interface:
----------------
There is a mailing list to which all new issues and changes to existing issues
are sent. You can subscribe through
http://live.polito/mailman/listinfo/ffmpeg-issues
Replies to messages there will have their text added to the specific issues.
Attachments will be added as if they had been uploaded via the web interface.
You can change the status, substatus, topic, ... by changing the subject in
your reply like:
Re: [issue94] register_avcodec and allcodecs.h [type=patch;status=open;substatus=approved]
Roundup will then change things as you requested and remove the [...] from
the subject before forwarding the mail to the mailing list.
NOTE: issue = (bug report || patch || feature request)
Type:
-----
bug / defect
bug
An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
prevents it from behaving as intended.
feature request / enhancement
feature request
Request of support for encoding or decoding of a new codec, container
or variant.
Request of support for more, less or plain different output or behavior
where the current implementation cannot be considered wrong.
license violation
ticket to keep track of (L)GPL violations of ffmpeg by others
patch
A patch as generated by diff which conforms to the patch submission and
development policy.
@@ -61,8 +68,6 @@ important
the separation to normal is somewhat fuzzy.
For feature requests this priority would be used for things many people
want.
Regressions also should be marked as important, regressions are bugs that
don't exist in a past revision or another branch.
normal
@@ -92,17 +97,6 @@ closed
final state
Analyzed flag:
--------------
Bugs which have been analyzed and where it is understood what causes them
and which exact chain of events triggers them. This analysis should be
available as a message in the bug report.
Note, do not change the status to analyzed without also providing a clear
and understandable analysis.
This state implicates that the bug either has been reproduced or that
reproduction is not needed as the bug is already understood.
Type/Status/Substatus:
----------
*/new/new
@@ -130,6 +124,24 @@ Type/Status/Substatus:
Issues for which some information has been requested by the developers,
but which has not been provided by anyone within reasonable time.
bug/open/reproduced
Bugs which have been reproduced.
bug/open/analyzed
Bugs which have been analyzed and where it is understood what causes them
and which exact chain of events triggers them. This analysis should be
available as a message in the bug report.
Note, do not change the status to analyzed without also providing a clear
and understandable analysis.
This state implicates that the bug either has been reproduced or that
reproduction is not needed as the bug is already understood.
bug/open/needs_more_info
Bug reports which are incomplete and or where more information is needed
from the submitter or another person who can provide it.
This state implicates that the bug has not been analyzed or reproduced.
Note, the idea behind needs_more_info is to offload work from the
developers to the users whenever possible.
bug/closed/fixed
Bugs which have to the best of our knowledge been fixed.
@@ -163,6 +175,10 @@ patch/closed/applied
patch/closed/rejected
Patches which have been rejected.
feature_request/open/needs_more_info
Feature requests where it is not clear what exactly is wanted
(these also could be closed as invalid ...).
feature_request/closed/implemented
Feature requests which have been implemented.
@@ -174,10 +190,12 @@ 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 substatus.
needs_more_info substate.
Component:
----------
Topic:
------
A topic is a tag you should add to your issue in order to make grouping them
easier.
avcodec
issues in libavcodec/*
@@ -197,9 +215,6 @@ ffmpeg
ffplay
issues in or related to ffplay.c
ffprobe
issues in or related to ffprobe.c
ffserver
issues in or related to ffserver.c
@@ -207,7 +222,7 @@ build system
issues in or related to configure/Makefile
regression
bugs which were not present in a past revision
bugs which were working in a past revision
trac
roundup
issues related to our issue tracker

View File

@@ -2,20 +2,31 @@
@settitle Libavfilter Documentation
@titlepage
@sp 7
@center @titlefont{Libavfilter Documentation}
@sp 3
@end titlepage
@top
@contents
@chapter Introduction
Libavfilter is the filtering API of FFmpeg. It is the substitute of the
now deprecated 'vhooks' and started as a Google Summer of Code project.
Audio filtering integration into the main FFmpeg repository is a work in
progress, so audio API and ABI should not be considered stable yet.
Integrating libavfilter into the main FFmpeg repository is a work in
progress. If you wish to try the unfinished development code of
libavfilter then check it out from the libavfilter repository into
some directory of your choice by:
@example
svn checkout svn://svn.ffmpeg.org/soc/libavfilter
@end example
And then read the README file in the top directory to learn how to
integrate it into ffmpeg and ffplay.
But note that there may still be serious bugs in the code and its API
and ABI should not be considered stable yet!
@chapter Tutorial
@@ -36,20 +47,21 @@ and the vflip filter before merging it back with the other stream by
overlaying it on top. You can use the following command to achieve this:
@example
ffmpeg -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output
./ffmpeg -i in.avi -s 240x320 -vfilters "[in] split [T1], fifo, [T2] overlay= 0:240 [out]; [T1] fifo, crop=0:0:-1:240, vflip [T2]
@end example
The result will be that in output the top half of the video is mirrored
where input_video.avi has a vertical resolution of 480 pixels. The
result will be that in output the top half of the video is mirrored
onto the bottom half.
Video filters are loaded using the @var{-vf} option passed to
@command{ffmpeg} or to @command{ffplay}. Filters in the same linear
chain are separated by commas. In our example, @var{split, fifo,
overlay} are in one linear chain, and @var{fifo, crop, vflip} are in
another. The points where the linear chains join are labeled by names
enclosed in square brackets. In our example, that is @var{[T1]} and
@var{[T2]}. The magic labels @var{[in]} and @var{[out]} are the points
where video is input and output.
Video filters are loaded using the @var{-vfilters} option passed to
ffmpeg or to 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
@@ -87,6 +99,209 @@ display graph.png
can be used to create and display an image representing the graph
described by the @var{GRAPH_DESCRIPTION} string.
@include filters.texi
@chapter Available video filters
When you configure your FFmpeg build, you can disable any of the
existing video filters.
The configure output will show the video filters included in your
build.
Below is a description of the currently available video filters.
@section crop
Crop the input video to @var{x}:@var{y}:@var{width}:@var{height}.
@example
./ffmpeg -i in.avi -vfilters "crop=0:0:0:240" out.avi
@end example
@var{x} and @var{y} specify the position of the top-left corner of the
output (non-cropped) area.
The default value of @var{x} and @var{y} is 0.
The @var{width} and @var{height} parameters specify the width and height
of the output (non-cropped) area.
A value of 0 is interpreted as the maximum possible size contained in
the area delimited by the top-left corner at position x:y.
For example the parameters:
@example
"crop=100:100:0:0"
@end example
will delimit the rectangle with the top-left corner placed at position
100:100 and the right-bottom corner corresponding to the right-bottom
corner of the input image.
The default value of @var{width} and @var{height} is 0.
@section format
Convert the input video to one of the specified pixel formats.
Libavfilter will try to pick one that is supported for the input to
the next filter.
The filter accepts a list of pixel format names, separated by ``:'',
for example ``yuv420p:monow:rgb24''.
The following command:
@example
./ffmpeg -i in.avi -vfilters "format=yuv420p" out.avi
@end example
will convert the input video to the format ``yuv420p''.
@section noformat
Force libavfilter not to use any of the specified pixel formats for the
input to the next filter.
The filter accepts a list of pixel format names, separated by ``:'',
for example ``yuv420p:monow:rgb24''.
The following command:
@example
./ffmpeg -i in.avi -vfilters "noformat=yuv420p, vflip" out.avi
@end example
will make libavfilter use a format different from ``yuv420p'' for the
input to the vflip filter.
@section null
Pass the source unchanged to the output.
@section scale
Scale the input video to @var{width}:@var{height} and/or convert the image format.
For example the command:
@example
./ffmpeg -i in.avi -vfilters "scale=200:100" out.avi
@end example
will scale the input video to a size of 200x100.
If the input image format is different from the format requested by
the next filter, the scale filter will convert the input to the
requested format.
If the value for @var{width} or @var{height} is 0, the respective input
size is used for the output.
If the value for @var{width} or @var{height} is -1, the scale filter will
use, for the respective output size, a value that maintains the aspect
ratio of the input image.
The default value of @var{width} and @var{height} is 0.
@section slicify
Pass the images of input video on to next video filter as multiple
slices.
@example
./ffmpeg -i in.avi -vfilters "slicify=32" out.avi
@end example
The filter accepts the slice height as parameter. If the parameter is
not specified it will use the default value of 16.
Adding this in the beginning of filter chains should make filtering
faster due to better use of the memory cache.
@section unsharp
Sharpen or blur the input video. It accepts the following parameters:
@multitable @columnfractions .2 .5 .1 .1 .1
@headitem Name @tab Description @tab Min @tab Max @tab Default
@item @var{luma_msize_x}
@tab Luma matrix horizontal size
@tab 3
@tab 13
@tab 5
@item @var{luma_msize_y}
@tab Luma matrix vertical size
@tab 3
@tab 13
@tab 5
@item @var{luma_amount}
@tab Luma effect strength
@tab -2.0
@tab 5.0
@tab 1.0
@item @var{chroma_msize_x}
@tab Chroma matrix horizontal size
@tab 3
@tab 13
@tab 0
@item @var{chroma_msize_y}
@tab Chroma matrix vertical size
@tab 3
@tab 13
@tab 0
@item @var{chroma_amount}
@tab Chroma effect strength
@tab -2.0
@tab 5.0
@tab 0.0
@end multitable
Negative values for the amount will blur the input video, while positive
values will sharpen. All parameters are optional and default to the
equivalent of the string '5:5:1.0:0:0:0.0'.
@example
# Strong luma sharpen effect parameters
unsharp=7:7:2.5
# Strong blur of both luma and chroma parameters
unsharp=7:7:-2:7:7:-2
# Use the default values with @command{ffmpeg}
./ffmpeg -i in.avi -vfilters "unsharp" out.mp4
@end example
@section vflip
Flip the input video vertically.
@example
./ffmpeg -i in.avi -vfilters "vflip" out.avi
@end example
@chapter Available video sources
Below is a description of the currently available video sources.
@section nullsrc
Null video source, never return images. It is mainly useful as a
template and to be employed in analysis / debugging tools.
It accepts as optional parameter a string of the form
@var{width}:@var{height}, where @var{width} and @var{height} specify the size of
the configured source.
The default values of @var{width} and @var{height} are respectively 352
and 288 (corresponding to the CIF size format).
@chapter Available video sinks
Below is a description of the currently available video sinks.
@section nullsink
Null video sink, do absolutely nothing with the input video. It is
mainly useful as a template and to be employed in analysis / debugging
tools.
@bye

View File

@@ -1,68 +0,0 @@
@chapter Metadata
@c man begin METADATA
FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded
INI-like text file and then load it back using the metadata muxer/demuxer.
The file format is as follows:
@enumerate
@item
A file consists of a header and a number of metadata tags divided into sections,
each on its own line.
@item
The header is a ';FFMETADATA' string, followed by a version number (now 1).
@item
Metadata tags are of the form 'key=value'
@item
Immediately after header follows global metadata
@item
After global metadata there may be sections with per-stream/per-chapter
metadata.
@item
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
brackets ('[', ']') and ends with next section or end of file.
@item
At the beginning of a chapter section there may be an optional timebase to be
used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
den are integers. If the timebase is missing then start/end times are assumed to
be in milliseconds.
Next a chapter section must contain chapter start and end times in form
'START=num', 'END=num', where num is a positive integer.
@item
Empty lines and lines starting with ';' or '#' are ignored.
@item
Metadata keys or values containing special characters ('=', ';', '#', '\' and a
newline) must be escaped with a backslash '\'.
@item
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
the tag (in the example above key is 'foo ', value is ' bar').
@end enumerate
A ffmetadata file might look like this:
@example
;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team
[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line
@end example
@c man end METADATA

View File

@@ -1,65 +0,0 @@
FFmpeg multithreading methods
==============================================
FFmpeg provides two methods for multithreading codecs.
Slice threading decodes multiple parts of a frame at the same time, using
AVCodecContext execute() and execute2().
Frame threading decodes multiple frames at the same time.
It accepts N future frames and delays decoded pictures by N-1 frames.
The later frames are decoded in separate threads while the user is
displaying the current one.
Restrictions on clients
==============================================
Slice threading -
* The client's draw_horiz_band() must be thread-safe according to the comment
in avcodec.h.
Frame threading -
* Restrictions with slice threading also apply.
* For best performance, the client should set thread_safe_callbacks if it
provides a thread-safe get_buffer() callback.
* There is one frame of delay added for every thread beyond the first one.
Clients must be able to handle this; the pkt_dts and pkt_pts fields in
AVFrame will work as usual.
Restrictions on codec implementations
==============================================
Slice threading -
None except that there must be something worth executing in parallel.
Frame threading -
* Codecs can only accept entire pictures per packet.
* Codecs similar to ffv1, whose streams don't reset across frames,
will not work because their bitstreams cannot be decoded in parallel.
* The contents of buffers must not be read before ff_thread_await_progress()
has been called on them. reget_buffer() and buffer age optimizations no longer work.
* The contents of buffers must not be written to after ff_thread_report_progress()
has been called on them. This includes draw_edges().
Porting codecs to frame threading
==============================================
Find all context variables that are needed by the next frame. Move all
code changing them, as well as code calling get_buffer(), up to before
the decode process starts. Call ff_thread_finish_setup() afterwards. If
some code can't be moved, have update_thread_context() run it in the next
thread.
If the codec allocates writable tables in its init(), add an init_thread_copy()
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.
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
doing this. Note that draw_edges() needs to be called before reporting progress.
Before accessing a reference frame or its MVs, call ff_thread_await_progress().

View File

@@ -1,320 +0,0 @@
@chapter Muxers
@c man begin MUXERS
Muxers are configured elements in FFmpeg which allow writing
multimedia streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option @code{--list-muxers}.
You can disable all the muxers with the configure option
@code{--disable-muxers} and selectively enable / disable single muxers
with the options @code{--enable-muxer=@var{MUXER}} /
@code{--disable-muxer=@var{MUXER}}.
The option @code{-formats} of the ff* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
@anchor{crc}
@section crc
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC 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
CRC.
The output of the muxer consists of a single line of the form:
CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file
@file{out.crc}:
@example
ffmpeg -i INPUT -f crc out.crc
@end example
You can print the CRC to stdout with the command:
@example
ffmpeg -i INPUT -f crc -
@end example
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 the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
@example
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example
See also the @ref{framecrc} muxer.
@anchor{framecrc}
@section framecrc
Per-frame CRC (Cyclic Redundancy Check) testing format.
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
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.
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
You can print the CRC of each decoded frame to stdout with the command:
@example
ffmpeg -i INPUT -f framecrc -
@end example
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:
@example
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
@end example
See also the @ref{crc} muxer.
@anchor{image2}
@section image2
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to
produce sequentially numbered series of files.
The pattern may contain the string "%d" or "%0@var{N}d", this string
specifies the position of the characters representing a numbering in
the filenames. If the form "%0@var{N}d" is used, the string
representing the number in each filename is 0-padded to @var{N}
digits. 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 will contain the number 1, all the following
numbers will be sequential.
The pattern may contain a suffix which is used to automatically
determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.
The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
etc.
The following example shows how to use @command{ffmpeg} for creating a
sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
taking one image every second from the input video:
@example
ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg'
@end example
Note that with @command{ffmpeg}, if the format is not specified with the
@code{-f} option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
@example
ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg'
@end example
Note also that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to create a single image file
@file{img.jpeg} from the input video you can employ the command:
@example
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example
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.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
@table @option
@item title=@var{title name}
Name provided to a single track
@end table
@table @option
@item language=@var{language name}
Specifies the language of the track in the Matroska languages form
@end table
@table @option
@item stereo_mode=@var{mode}
Stereo 3D video layout of two views in a single video track
@table @option
@item mono
video is not stereo
@item left_right
Both views are arranged side by side, Left-eye view is on the left
@item bottom_top
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
@item top_bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
@item checkerboard_rl
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
@item checkerboard_lr
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
@item row_interleaved_rl
Each view is constituted by a row based interleaving, Right-eye view is first row
@item row_interleaved_lr
Each view is constituted by a row based interleaving, Left-eye view is first row
@item col_interleaved_rl
Both views are arranged in a column based interleaving manner, Right-eye view is first column
@item col_interleaved_lr
Both views are arranged in a column based interleaving manner, Left-eye view is first column
@item anaglyph_cyan_red
All frames are in anaglyph format viewable through red-cyan filters
@item right_left
Both views are arranged side by side, Right-eye view is on the left
@item anaglyph_green_magenta
All frames are in anaglyph format viewable through green-magenta filters
@item block_lr
Both eyes laced in one Block, Left-eye view is first
@item block_rl
Both eyes laced in one Block, Right-eye view is first
@end table
@end table
For example a 3D WebM clip can be created using the following command line:
@example
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example
@section segment
Basic stream segmenter.
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}.
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 flat list of the created segments, one segment
per line.
@table @option
@item segment_format @var{format}
Override the inner container format, by default it is guessed by the filename
extension.
@item segment_time @var{t}
Set segment duration to @var{t} seconds.
@item segment_list @var{name}
Generate also a listfile named @var{name}.
@item segment_list_size @var{size}
Overwrite the listfile once it reaches @var{size} entries.
@end table
@example
ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
@end example
@c man end MUXERS

View File

@@ -17,15 +17,15 @@ Understanding these overoptimized functions:
As many functions tend to be a bit difficult to understand because
of optimizations, it can be hard to optimize them further, or write
architecture-specific versions. It is recommended to look at older
revisions of the interesting files (web frontends for the various FFmpeg
branches are listed at http://ffmpeg.org/download.html).
revisions of the interesting files (for a web frontend try ViewVC at
http://svn.ffmpeg.org/ffmpeg/trunk/).
Alternatively, look into the other architecture-specific versions in
the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly
comprehend the instructions, it could help understanding the functions
and how they can be optimized.
NOTE: If you still don't understand some function, ask at our mailing list!!!
(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel)
(http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel)
When is an optimization justified?
@@ -157,68 +157,15 @@ Use asm loops like:
__asm__(
"1: ....
...
"jump_instruction ....
"jump_instruciton ....
Do not use C loops:
do{
__asm__(
...
}while()
For x86, mark registers that are clobbered in your asm. This means both
general x86 registers (e.g. eax) as well as XMM registers. This last one is
particularly important on Win64, where xmm6-15 are callee-save, and not
restoring their contents leads to undefined results. In external asm (e.g.
yasm), you do this by using:
cglobal functon_name, num_args, num_regs, num_xmm_regs
In inline asm, you specify clobbered registers at the end of your asm:
__asm__(".." ::: "%eax").
If gcc is not set to support sse (-msse) it will not accept xmm registers
in the clobber list. For that we use two macros to declare the clobbers.
XMM_CLOBBERS should be used when there are other clobbers, for example:
__asm__(".." ::: XMM_CLOBBERS("xmm0",) "eax");
and XMM_CLOBBERS_ONLY should be used when the only clobbers are xmm registers:
__asm__(".." :: XMM_CLOBBERS_ONLY("xmm0"));
Do not expect a compiler to maintain values in your registers between separate
(inline) asm code blocks. It is not required to. For example, this is bad:
__asm__("movdqa %0, %%xmm7" : src);
/* do something */
__asm__("movdqa %%xmm7, %1" : dst);
- first of all, you're assuming that the compiler will not use xmm7 in
between the two asm blocks. It probably won't when you test it, but it's
a poor assumption that will break at some point for some --cpu compiler flag
- secondly, you didn't mark xmm7 as clobbered. If you did, the compiler would
have restored the original value of xmm7 after the first asm block, thus
rendering the combination of the two blocks of code invalid
Code that depends on data in registries being untouched, should be written as
a single __asm__() statement. Ideally, a single function contains only one
__asm__() block.
Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
The latter requires a good optimizing compiler which gcc is not.
Inline asm vs. external asm
---------------------------
Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)
and external asm (.s or .asm files, handled by an assembler such as yasm/nasm)
are accepted in FFmpeg. Which one to use differs per specific case.
- if your code is intended to be inlined in a C function, inline asm is always
better, because external asm cannot be inlined
- if your code calls external functions, yasm is always better
- if your code takes huge and complex structs as function arguments (e.g.
MpegEncContext; note that this is not ideal and is discouraged if there
are alternatives), then inline asm is always better, because predicting
member offsets in complex structs is almost impossible. It's safest to let
the compiler take care of that
- in many cases, both can be used and it just depends on the preference of the
person writing the asm. For new asm, the choice is up to you. For existing
asm, you'll likely want to maintain whatever form it is currently in unless
there is a good reason to change it.
- if, for some reason, you believe that a particular chunk of existing external
asm could be improved upon further if written in inline asm (or the other
way around), then please make the move from external asm <-> inline asm a
separate patch before your patches that actually improve the asm.
Use __asm__() instead of intrinsics. The latter requires a good optimizing compiler
which gcc is not.
Links:

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@@ -1,73 +0,0 @@
@chapter Output Devices
@c man begin OUTPUT DEVICES
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
are enabled by default. You can list all available ones using the
configure option "--list-outdevs".
You can disable all the output devices using the configure option
"--disable-outdevs", and selectively enable an output device using the
option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
input device using the option "--disable-outdev=@var{OUTDEV}".
The option "-formats" of the ff* tools will display the list of
enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) output device.
@section oss
OSS (Open Sound System) output device.
@section sdl
SDL (Simple DirectMedia Layer) output device.
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.
To enable this output device you need libsdl installed on your system
when configuring your build.
For more information about SDL, check:
@url{http://www.libsdl.org/}
@subsection Options
@table @option
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item icon_title
Set the name of the iconified SDL window, if not specified it is set
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.
@end table
@subsection Examples
The following command shows the @command{ffmpeg} output is an
SDL window, forcing its size to the qcif format:
@example
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
@end example
@section sndio
sndio audio output device.
@c man end OUTPUT DEVICES

View File

@@ -1,390 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Platform Specific information
@titlepage
@center @titlefont{Platform Specific information}
@end titlepage
@top
@contents
@chapter Unix-like
Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
@example
$(gcc -print-prog-name=as) --version
@end example
If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass @code{--disable-asm}
to configure.
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
(@file{gmake}).
@section (Open)Solaris
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
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
@example
bash ./configure
@end example
@anchor{Darwin}
@section Darwin (Mac OS X, iPhone)
The toolchain provided with Xcode is sufficient to build the basic
unacelerated code.
Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
assembler functions. Just download the Perl script and put it somewhere
in your PATH, FFmpeg's configure will pick it up automatically.
Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
@uref{http://mxcl.github.com/homebrew/, Homebrew}
or @uref{http://www.macports.org, MacPorts} can easily provide it.
@chapter DOS
Using a cross-compiler is preferred for various reasons.
@url{http://www.delorie.com/howto/djgpp/linux-x-djgpp.html}
@chapter OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@chapter Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@section Native Windows compilation
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:
@itemize
@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 in
@code{make install}).
@item In order to compile FFplay, you must have the MinGW development library
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,
libavformat) as DLLs.
@end itemize
@section Microsoft Visual C++ compatibility
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.
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.
@subsection Using static libraries
Assuming you have just built and installed FFmpeg in @file{/usr/local}.
@enumerate
@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.
@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
call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"
@end example
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
Your install path (@file{/usr/local/} by default) should now have the
necessary DLL and LIB files under the @file{bin} directory.
@end enumerate
Alternatively, build the libraries with a cross compiler, according to
the instructions below in @ref{Cross compilation for Windows with Linux}.
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.
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
extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
@end example
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}.
To create import libraries that work with the @code{/OPT:REF} option
(which is enabled by default in Release mode), follow these steps:
@enumerate
@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
@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 @file{lib.exe}:
@example
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
@end enumerate
@anchor{Cross compilation for Windows with Linux}
@section Cross compilation for Windows with Linux
You must use the MinGW cross compilation tools available at
@url{http://www.mingw.org/}.
Then configure FFmpeg with the following options:
@example
./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
@end example
(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}.
@section Compilation under Cygwin
Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
Install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, git, mingw-runtime, texi2html
@end example
And the following "Utils" one:
@example
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
"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
@example
libogg-devel, libvorbis-devel
@end example
These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
@example
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 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
With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
Just install your Cygwin as explained before, plus these additional
"Devel" packages:
@example
gcc-mingw-core, mingw-runtime, mingw-zlib
@end example
and add some special flags to your configure invocation.
For a static build run
@example
./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
and for a build with shared libraries
@example
./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@bye

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@@ -1,492 +0,0 @@
@chapter Protocols
@c man begin PROTOCOLS
Protocols are configured elements in FFmpeg which allow to access
resources which require the use of a particular protocol.
When you configure your FFmpeg build, all the supported protocols are
enabled by default. You can list all available ones using the
configure option "--list-protocols".
You can disable all the protocols using the configure option
"--disable-protocols", and selectively enable a protocol using the
option "--enable-protocol=@var{PROTOCOL}", or you can disable a
particular protocol using the option
"--disable-protocol=@var{PROTOCOL}".
The option "-protocols" of the ff* tools will display the list of
supported protocols.
A description of the currently available protocols follows.
@section applehttp
Read Apple HTTP Live Streaming compliant segmented stream as
a uniform one. The M3U8 playlists describing the segments can be
remote HTTP resources or local files, accessed using the standard
file protocol.
HTTP is default, specific protocol can be declared by specifying
"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
is either "file" or "http".
@example
applehttp://host/path/to/remote/resource.m3u8
applehttp+http://host/path/to/remote/resource.m3u8
applehttp+file://path/to/local/resource.m3u8
@end example
@section concat
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were
a unique resource.
A URL accepted by this protocol has the syntax:
@example
concat:@var{URL1}|@var{URL2}|...|@var{URLN}
@end example
where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
resource to be concatenated, each one possibly specifying a distinct
protocol.
For example to read a sequence of files @file{split1.mpeg},
@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
command:
@example
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@end example
Note that you may need to escape the character "|" which is special for
many shells.
@section file
File access protocol.
Allow to read from or read to a file.
For example to read from a file @file{input.mpeg} with @command{ffmpeg}
use the command:
@example
ffmpeg -i file:input.mpeg output.mpeg
@end example
The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".
@section gopher
Gopher protocol.
@section http
HTTP (Hyper Text Transfer Protocol).
@section mmst
MMS (Microsoft Media Server) protocol over TCP.
@section mmsh
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
@example
mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example
@section md5
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes
this to the designated output or stdout if none is specified. It can
be used to test muxers without writing an actual file.
Some examples follow.
@example
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5
# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:
@end example
Note that some formats (typically MOV) require the output protocol to
be seekable, so they will fail with the MD5 output protocol.
@section pipe
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
@example
pipe:[@var{number}]
@end example
@var{number} is the number corresponding to the file descriptor of the
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
is not specified, by default the stdout file descriptor will be used
for writing, stdin for reading.
For example to read from stdin with @command{ffmpeg}:
@example
cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:
@end example
For writing to stdout with @command{ffmpeg}:
@example
ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi
@end example
Note that some formats (typically MOV), require the output protocol to
be seekable, so they will fail with the pipe output protocol.
@section rtmp
Real-Time Messaging Protocol.
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
content across a TCP/IP network.
The required syntax is:
@example
rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example
The accepted parameters are:
@table @option
@item server
The address of the RTMP server.
@item port
The number of the TCP port to use (by default is 1935).
@item app
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
@item playpath
It is the path or name of the resource to play with reference to the
application specified in @var{app}, may be prefixed by "mp4:".
@end table
For example to read with @command{ffplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
ffplay rtmp://myserver/vod/sample
@end example
@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
librtmp.
Requires the presence of the librtmp headers and library during
configuration. You need to explicitly configure the build with
"--enable-librtmp". If enabled this will replace the native RTMP
protocol.
This protocol provides most client functions and a few server
functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
@example
@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
@end example
where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
@var{server}, @var{port}, @var{app} and @var{playpath} have the same
meaning as specified for the RTMP native protocol.
@var{options} contains a list of space-separated options of the form
@var{key}=@var{val}.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
@command{ffmpeg}:
@example
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example
To play the same stream using @command{ffplay}:
@example
ffplay "rtmp://myserver/live/mystream live=1"
@end example
@section rtp
Real-Time Protocol.
@section rtsp
RTSP is not technically a protocol handler in libavformat, it is a demuxer
and muxer. The demuxer supports both normal RTSP (with data transferred
over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
The required syntax for a RTSP url is:
@example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
The following options (set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:
Flags for @code{rtsp_transport}:
@table @option
@item udp
Use UDP as lower transport protocol.
@item tcp
Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.
@item udp_multicast
Use UDP multicast as lower transport protocol.
@item http
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
@end table
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the @code{tcp} and @code{udp} options are supported.
Flags for @code{rtsp_flags}:
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
(since they may arrive out of order, or packets may get lost totally). In
order for this to be enabled, a maximum delay must be specified in the
@code{max_delay} field of AVFormatContext.
When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
on the fly by pressing @code{v} and @code{a}.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example
To watch a stream tunneled over HTTP:
@example
ffplay -rtsp_transport http rtsp://server/video.mp4
@end example
To send a stream in realtime to a RTSP server, for others to watch:
@example
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
@section sap
Session Announcement Protocol (RFC 2974). This is not technically a
protocol handler in libavformat, it is a muxer and demuxer.
It is used for signalling of RTP streams, by announcing the SDP for the
streams regularly on a separate port.
@subsection Muxer
The syntax for a SAP url given to the muxer is:
@example
sap://@var{destination}[:@var{port}][?@var{options}]
@end example
The RTP packets are sent to @var{destination} on port @var{port},
or to port 5004 if no port is specified.
@var{options} is a @code{&}-separated list. The following options
are supported:
@table @option
@item announce_addr=@var{address}
Specify the destination IP address for sending the announcements to.
If omitted, the announcements are sent to the commonly used SAP
announcement multicast address 224.2.127.254 (sap.mcast.net), or
ff0e::2:7ffe if @var{destination} is an IPv6 address.
@item announce_port=@var{port}
Specify the port to send the announcements on, defaults to
9875 if not specified.
@item ttl=@var{ttl}
Specify the time to live value for the announcements and RTP packets,
defaults to 255.
@item same_port=@var{0|1}
If set to 1, send all RTP streams on the same port pair. If zero (the
default), all streams are sent on unique ports, with each stream on a
port 2 numbers higher than the previous.
VLC/Live555 requires this to be set to 1, to be able to receive the stream.
The RTP stack in libavformat for receiving requires all streams to be sent
on unique ports.
@end table
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example
Similarly, for watching in @command{ffplay}:
@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
@end example
And for watching in @command{ffplay}, over IPv6:
@example
ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
@end example
@subsection Demuxer
The syntax for a SAP url given to the demuxer is:
@example
sap://[@var{address}][:@var{port}]
@end example
@var{address} is the multicast address to listen for announcements on,
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port.
Once an announcement is received, it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
@example
ffplay sap://
@end example
To play back the first stream announced on one the default IPv6 SAP multicast address:
@example
ffplay sap://[ff0e::2:7ffe]
@end example
@section tcp
Trasmission Control Protocol.
The required syntax for a TCP url is:
@example
tcp://@var{hostname}:@var{port}[?@var{options}]
@end example
@table @option
@item listen
Listen for an incoming connection
@example
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
ffplay tcp://@var{hostname}:@var{port}
@end example
@end table
@section udp
User Datagram Protocol.
The required syntax for a UDP url is:
@example
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
Follow the list of supported options.
@table @option
@item buffer_size=@var{size}
set the UDP buffer size in bytes
@item localport=@var{port}
override the local UDP port to bind with
@item localaddr=@var{addr}
Choose the local IP address. This is useful e.g. if sending multicast
and the host has multiple interfaces, where the user can choose
which interface to send on by specifying the IP address of that interface.
@item pkt_size=@var{size}
set the size in bytes of UDP packets
@item reuse=@var{1|0}
explicitly allow or disallow reusing UDP sockets
@item ttl=@var{ttl}
set the time to live value (for multicast only)
@item connect=@var{1|0}
Initialize the UDP socket with @code{connect()}. In this case, the
destination address can't be changed with ff_udp_set_remote_url later.
If the destination address isn't known at the start, this option can
be specified in ff_udp_set_remote_url, too.
This allows finding out the source address for the packets with getsockname,
and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
@end table
Some usage examples of the udp protocol with @command{ffmpeg} follow.
To stream over UDP to a remote endpoint:
@example
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
@example
ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example
To receive over UDP from a remote endpoint:
@example
ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example
@c man end PROTOCOLS

View File

@@ -9,16 +9,16 @@ it's a little late for this year's soc (2006).
The Goal:
Our goal in respect to soc is and must be of course exactly one thing and
that is to improve FFmpeg, to reach this goal, code must
* conform to the development policy and patch submission guidelines
* conform to the svn policy and patch submission guidelines
* must improve FFmpeg somehow (faster, smaller, "better",
more codecs supported, fewer bugs, cleaner, ...)
for mentors and other developers to help students to reach that goal it is
essential that changes to their codebase are publicly visible, clean and
easy reviewable that again leads us to:
* use of a revision control system like git
* use of a revision control system like svn
* separation of cosmetic from non-cosmetic changes (this is almost entirely
ignored by mentors and students in soc 2006 which might lead to a surprise
ignored by mentors and students in soc 2006 which might lead to a suprise
when the code will be reviewed at the end before a possible inclusion in
FFmpeg, individual changes were generally not reviewable due to cosmetics).
* frequent commits, so that comments can be provided early

View File

@@ -1,46 +0,0 @@
The official guide to swresample for confused developers.
=========================================================
Current (simplified) Architecture:
---------------------------------
Input
v
__________________/|\___________
/ | \
/ input sample format convert v
/ | ___________/
| |/
| v
| ___________/|\___________ _____________
| / | \ | |
| Rematrix | resample <---->| Buffers |
| \___________ | ___________/ |_____________|
v \|/
Special Converter v
v ___________/|\___________ _____________
| / | \ | |
| Rematrix | resample <---->| Buffers |
| \___________ | ___________/ |_____________|
| \|/
| v
| |\___________
\ | \
\ output sample format convert v
\_________________ | ___________/
\|/
v
Output
Planar/Packed convertion is done when needed during sample format convertion
Every step can be skiped without memcpy when its not needed.
Either Resampling and Rematrixing can be performed first depending on which
way its faster.
The Buffers are needed for resampling due to resamplng being a process that
requires future and past data, it thus also introduces inevitably a delay when
used.
Internally 32bit float and 16bit int is supported currently, other formats can
easily be added
Externally all sample formats in packed and planar configuration are supported
Its also trivial to add special converters for common cases
If only sample format and or packed/planar convertion is needed it
is performed from input to output directly in a single pass with no intermediates.

View File

@@ -1,170 +0,0 @@
# no horiz rules between sections
$end_section = \&FFmpeg_end_section;
sub FFmpeg_end_section($$)
{
}
$EXTRA_HEAD =
'<link rel="icon" href="favicon.png" type="image/png" />
<link rel="stylesheet" type="text/css" href="default.css" />
';
$CSS_LINES = <<EOT;
<style type="text/css">
<!--
a.summary-letter { text-decoration: none }
a { color: #2D6198; }
a:visited { color: #884488; }
h1 a, h2 a, h3 a { text-decoration: inherit; color: inherit; }
p { margin-left: 1em; margin-right: 1em; }
table { margin-left: 2em; }
pre { margin-left: 2em; }
#footer { text-align: center; }
#body { margin-left: 1em; margin-right: 1em; }
body { background-color: #313131; margin: 0; }
#container {
background-color: white;
color: #202020;
margin-left: 1em;
margin-right: 1em;
}
h1 {
background-color: #7BB37B;
border: 1px solid #6A996A;
color: #151515;
font-size: 1.2em;
padding-bottom: 0.2em;
padding-left: 0.4em;
padding-top: 0.2em;
}
h2 {
color: #313131;
font-size: 1.2em;
}
h3 {
color: #313131;
font-size: 0.8em;
margin-bottom: -8px;
}
.note {
margin: 1em;
border: 1px solid #bbc9d8;
background-color: #dde1e1;
}
.important {
margin: 1em;
border: 1px solid #d26767;
background-color: #f8e1e1;
}
-->
</style>
EOT
my $FFMPEG_NAVBAR = $ENV{"FFMPEG_NAVBAR"} || '';
$AFTER_BODY_OPEN =
'<div id="container">' .
"\n$FFMPEG_NAVBAR\n" .
'<div id="body">';
$PRE_BODY_CLOSE = '</div></div>';
$SMALL_RULE = '';
$BODYTEXT = '';
$print_page_foot = \&FFmpeg_print_page_foot;
sub FFmpeg_print_page_foot($$)
{
my $fh = shift;
print $fh '<div id="footer">' . "\n";
T2H_DEFAULT_print_page_foot($fh);
print $fh "</div>\n";
}
$float = \&FFmpeg_float;
sub FFmpeg_float($$$$)
{
my $text = shift;
my $float = shift;
my $caption = shift;
my $shortcaption = shift;
my $label = '';
if (exists($float->{'id'}))
{
$label = &$anchor($float->{'id'});
}
my $class = '';
my $subject = '';
if ($caption =~ /NOTE/)
{
$class = "note";
}
elsif ($caption =~ /IMPORTANT/)
{
$class = "important";
}
return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>';
}
$print_page_head = \&FFmpeg_print_page_head;
sub FFmpeg_print_page_head($$)
{
my $fh = shift;
my $longtitle = "$Texi2HTML::THISDOC{'title_no_texi'}";
$longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'};
my $description = $DOCUMENT_DESCRIPTION;
$description = $longtitle if (!defined($description));
$description = "<meta name=\"description\" content=\"$description\">" if
($description ne '');
$description = $Texi2HTML::THISDOC{'documentdescription'} if (defined($Texi2HTML::THISDOC{'documentdescription'}));
my $encoding = '';
$encoding = "<meta http-equiv=\"Content-Type\" content=\"text/html; charset=$ENCODING\">" if (defined($ENCODING) and ($ENCODING ne ''));
$longtitle =~ s/Documentation.*//g;
$longtitle = "FFmpeg documentation : " . $longtitle;
print $fh <<EOT;
$DOCTYPE
<html>
$Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
<!--
$Texi2HTML::THISDOC{program_authors}
-->
<head>
<title>$longtitle</title>
$description
<meta name="keywords" content="$longtitle">
<meta name="resource-type" content="document">
<meta name="distribution" content="global">
<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
$encoding
$CSS_LINES
$EXTRA_HEAD
</head>
<body $BODYTEXT>
$AFTER_BODY_OPEN
EOT
}
# no navigation elements
$SECTION_NAVIGATION = 0;
# the same for texi2html 5.0
$HEADERS = 0;
# TOC and Chapter headings link
$TOC_LINKS = 1;
# print the TOC where @contents is used
$INLINE_CONTENTS = 1;

View File

@@ -12,7 +12,6 @@ the tables as a header file using the tableprint.h helpers.
Both of these files will be compiled for the host system, so to avoid
breakage with cross-compilation neither of them may include, directly
or indirectly, config.h or avconfig.h.
This means that e.g. libavutil/mathematics.h is ok but libavutil/libm.h is not.
Due to this, the .c file or Makefile may have to provide additional defines
or stubs, though if possible this should be avoided.
In particular, CONFIG_HARDCODED_TABLES should always be defined to 0.
@@ -28,14 +27,9 @@ them.
The printing code typically looks like this:
write_fileheader();
printf("static const uint8_t my_array[100] = {\n");
write_uint8_t_array(my_array, 100);
write_uint8_array(my_array, 100);
printf("};\n");
This is the more generic form, in case you need to do something special.
Usually you should instead use the short form:
write_fileheader();
WRITE_ARRAY("static const", uint8_t, my_array);
write_fileheader() adds some minor things like a "this is a generated file"
comment and some standard includes.
tablegen.h defines some write functions for one- and two-dimensional arrays

View File

@@ -26,7 +26,6 @@
$output = 0;
$skipping = 0;
%sects = ();
@sects_sequence = ();
$section = "";
@icstack = ();
@endwstack = ();
@@ -100,21 +99,15 @@ while(<$inf>) {
next;
};
/^\@include\s+(.+)$/ and do {
push @instack, $inf;
$inf = gensym();
# Try cwd and $ibase.
open($inf, "<" . $1)
or open($inf, "<" . $ibase . "/" . $1)
or die "cannot open $1 or $ibase/$1: $!\n";
next;
};
# Look for blocks surrounded by @c man begin SECTION ... @c man end.
# This really oughta be @ifman ... @end ifman and the like, but such
# would require rev'ing all other Texinfo translators.
/^\@c\s+man\s+begin\s+([A-Za-z ]+)/ and $sect = $1, push (@sects_sequence, $sect), $output = 1, next;
/^\@c\s+man\s+begin\s+([A-Z]+)\s+([A-Za-z0-9-]+)/ and do {
$output = 1 if exists $defs{$2};
$sect = $1;
next;
};
/^\@c\s+man\s+begin\s+([A-Z]+)/ and $sect = $1, $output = 1, next;
/^\@c\s+man\s+end/ and do {
$sects{$sect} = "" unless exists $sects{$sect};
$sects{$sect} .= postprocess($section);
@@ -231,18 +224,27 @@ while(<$inf>) {
# Single line command handlers.
/^\@(?:section|unnumbered|unnumberedsec|center|heading)\s+(.+)$/
/^\@include\s+(.+)$/ and do {
push @instack, $inf;
$inf = gensym();
# Try cwd and $ibase.
open($inf, "<" . $1)
or open($inf, "<" . $ibase . "/" . $1)
or die "cannot open $1 or $ibase/$1: $!\n";
next;
};
/^\@(?:section|unnumbered|unnumberedsec|center)\s+(.+)$/
and $_ = "\n=head2 $1\n";
/^\@(?:subsection|subheading)\s+(.+)$/
/^\@subsection\s+(.+)$/
and $_ = "\n=head3 $1\n";
/^\@(?:subsubsection|subsubheading)\s+(.+)$/
and $_ = "\n=head4 $1\n";
# Block command handlers:
/^\@itemize\s*(\@[a-z]+|\*|-)?/ and do {
/^\@itemize\s+(\@[a-z]+|\*|-)/ and do {
push @endwstack, $endw;
push @icstack, $ic;
$ic = $1 ? $1 : "*";
$ic = $1;
$_ = "\n=over 4\n";
$endw = "itemize";
};
@@ -301,8 +303,8 @@ die "No filename or title\n" unless defined $fn && defined $tl;
$sects{NAME} = "$fn \- $tl\n";
$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES};
unshift @sects_sequence, "NAME";
for $sect (@sects_sequence) {
for $sect (qw(NAME SYNOPSIS DESCRIPTION OPTIONS EXAMPLES ENVIRONMENT FILES
BUGS NOTES FOOTNOTES SEEALSO AUTHOR COPYRIGHT)) {
if(exists $sects{$sect}) {
$head = $sect;
$head =~ s/SEEALSO/SEE ALSO/;
@@ -348,11 +350,9 @@ sub postprocess
# @* is also impossible in .pod; we discard it and any newline that
# follows it. Similarly, our macro @gol must be discarded.
s/\@anchor{(?:[^\}]*)\}//g;
s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
s/;\s+\@pxref\{(?:[^\}]*)\}//g;
s/\@ref\{([^\}]*)\}/$1/g;
s/\@noindent\s*//g;
s/\@refill//g;
s/\@gol//g;

7399
ffmpeg.c

File diff suppressed because it is too large Load Diff

2417
ffplay.c

File diff suppressed because it is too large Load Diff

View File

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

View File

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

View File

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

View File

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

View File

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

View File

@@ -0,0 +1,4 @@
coder=0
bf=0
flags2=-wpred-dct8x8
wpredp=0

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=7
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=3
directpred=1
trellis=1
flags2=+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=6
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=2
directpred=1
trellis=1
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=30

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2
rc_lookahead=30

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=4
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=2
directpred=1
trellis=1
flags2=+bpyramid-mixed_refs+wpred+dct8x8+fastpskip
wpredp=1
rc_lookahead=20

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=1
rc_lookahead=20

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=3
trellis=0
flags2=-bpyramid-wpred-mixed_refs-dct8x8+fastpskip
wpredp=2

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=umh
subq=8
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=4
directpred=3
trellis=1
flags2=+wpred+mixed_refs+dct8x8+fastpskip
wpredp=2

View File

@@ -0,0 +1,7 @@
coder=0
bf=0
flags2=-wpred-dct8x8
level=13
maxrate=768000
bufsize=3000000
wpredp=0

View File

@@ -0,0 +1,8 @@
coder=0
bf=0
refs=1
flags2=-wpred-dct8x8
level=30
maxrate=10000000
bufsize=10000000
wpredp=0

View File

@@ -0,0 +1,20 @@
coder=0
flags=+loop
cmp=+chroma
partitions=-parti8x8+parti4x4+partp8x8-partp4x4-partb8x8
me_method=hex
subq=3
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
directpred=1
flags2=+fastpskip
cqp=0
wpredp=0

View File

@@ -0,0 +1,21 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
me_method=esa
subq=8
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
refs=16
directpred=1
flags2=+mixed_refs+dct8x8+fastpskip
cqp=0
wpredp=2

View File

@@ -0,0 +1,20 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
me_method=hex
subq=5
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
directpred=1
flags2=+fastpskip
cqp=0
wpredp=2

View File

@@ -0,0 +1,21 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
me_method=umh
subq=6
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
refs=2
directpred=1
flags2=+dct8x8+fastpskip
cqp=0
wpredp=2

View File

@@ -0,0 +1,21 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
me_method=umh
subq=8
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
refs=4
directpred=1
flags2=+mixed_refs+dct8x8+fastpskip
cqp=0
wpredp=2

View File

@@ -0,0 +1,19 @@
coder=0
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
me_method=dia
subq=0
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
directpred=1
flags2=+fastpskip
cqp=0

View File

@@ -0,0 +1 @@
flags2=-dct8x8

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=tesa
subq=10
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=16
directpred=3
trellis=2
flags2=+wpred+mixed_refs+dct8x8-fastpskip
wpredp=2

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=7
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=3
directpred=1
trellis=1
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2

View File

@@ -0,0 +1,22 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=6
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=2
directpred=3
trellis=0
flags2=+wpred+dct8x8+fastpskip
wpredp=2

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=tesa
subq=10
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=16
refs=16
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8-fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=tesa
subq=10
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=16
refs=16
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8-fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=umh
subq=8
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=5
directpred=3
trellis=1
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=50

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=3
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2
rc_lookahead=50

View File

@@ -0,0 +1,23 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=umh
subq=9
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=8
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=60

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