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260 Commits

Author SHA1 Message Date
Michael Niedermayer
9eaec5b8f0 update for 0.5.10
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-09 22:18:07 +02:00
Michael Niedermayer
48ef116631 wmv1: check that the input buffer is large enough
Fixes null ptr deref
Fixes Ticket1367

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f23a2418fb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-09 21:40:36 +02:00
Michael Niedermayer
cc511b36f3 truemotion1: Check index, fix out of array read
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fd4c1c0b70)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-09 21:40:13 +02:00
Michael Niedermayer
85b22a8924 Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Release notes for 0.5.9
  Update changelog for 0.5.9 release

Conflicts:
	RELEASE

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-09 19:02:31 +02:00
Reinhard Tartler
6ec1d3b3ba Release notes for 0.5.9 2012-06-09 12:13:27 +02:00
Derek Buitenhuis
d843e7dc94 Update changelog for 0.5.9 release
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2012-06-08 15:41:31 -04:00
Michael Niedermayer
acc665f22c Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Bump version number for 0.5.9 release.
  png: check bit depth for PAL8/Y400A pixel formats.
  tqi: Pass errors from the MB decoder
  eatqi: move "block" variable into context to ensure sufficient alignment for idct_put for compilers/architectures that can not align stack variables that much. This is also consistent with similar code in eatgq.c
  ea: check chunk_size for validity.
  vfwcap: Include windows.h before vfw.h since the latter requires defines from the former. Patch by kemuri <kemuri9 at gmail dot com>
  mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
  mingw32: properly check if vfw capture is supported by the system headers
  Replace every usage of -lvfw32 with what is particularly necessary for that case: Avisynth -> -lavifil32 VFW Cap -> -lavicap32 Patch by kemuri <kemuri9 at gmail dot com>
  configure: properly check for mingw-w64 through installed headers. mingw-w64 can also target 32-bit code.
  qdm2: clip array indices returned by qdm2_get_vlc().
  kmvc: Check palsize.
  adpcm: ADPCM Electronic Arts has always two channels
  h264: Add check for invalid chroma_format_idc
  dpcm: ignore extra unpaired bytes in stereo streams.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-04 12:29:25 +02:00
Reinhard Tartler
4c223fe519 Bump version number for 0.5.9 release. 2012-06-03 22:42:30 +02:00
Reinhard Tartler
5a9588b088 png: check bit depth for PAL8/Y400A pixel formats.
Wrong bit depth can lead to invalid rowsize values, which crashes the
decoder further down.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit d2205d6543)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit b8d6ba9d50e80fdce2ed74cdaffd4960df8a21c5)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 33f93005f1a86c108302b4c5978aa1a3d8e092cc)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 4c8c2660bd9252775c9a1dc2e2f36cb34718595a)

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

Conflicts:

	libavcodec/pngdec.c
2012-06-03 19:35:50 +02:00
Michael Niedermayer
02cd93f4ad tqi: Pass errors from the MB decoder
This silences some valgrind warnings.
CC: libav-stable@libav.org

Fixes second half of http://ffmpeg.org/trac/ffmpeg/ticket/794
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit f85334f58e)
(cherry picked from commit 90290a5150)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 5872580e65)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 2f2fd8c6d1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit c3edce42704142f4c66954e9f24d7fbf0e5ae423)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-06-03 19:35:13 +02:00
Reimar Döffinger
f8a31e2113 eatqi: move "block" variable into context to ensure sufficient alignment for
idct_put for compilers/architectures that can not align stack variables that much.
This is also consistent with similar code in eatgq.c

Originally committed as revision 18927 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 1eda87ce63)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-06-03 19:35:13 +02:00
Ronald S. Bultje
8ba939fcda ea: check chunk_size for validity.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
(cherry picked from commit 273e6af47b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 6a86b705e1d4b72f0dddfbe23ad3eed9947001d5)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit e74bc64dd376c4691a610ba62a66ed30affc97ec)

Conflicts:

	libavformat/electronicarts.c
(cherry picked from commit 38c45adfca299e3d96c07a700032695ec7ff2aeb)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-06-03 19:35:13 +02:00
kemuri
79674d27d9 vfwcap: Include windows.h before vfw.h since the latter requires defines from the former. Patch by kemuri <kemuri9 at gmail dot com>
Originally committed as revision 21411 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 420755dd28)
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2012-06-02 19:25:58 -04:00
Ramiro Polla
b061ee9a5d mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
Originally committed as revision 24204 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit e26011d0f4)
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2012-06-02 19:25:58 -04:00
Ramiro Polla
baba561c0b mingw32: properly check if vfw capture is supported by the system headers
Remove check for an specific w32api version, checking instead if vfw.h
supports vfw capture. The defines in w32api 3.12 were wrong, so this must be
accounted for in the check.

Originally committed as revision 24203 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit ec1ee802a2)
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Conflicts:

	configure
2012-06-02 19:25:58 -04:00
kemuri
f6147effd2 Replace every usage of -lvfw32 with what is particularly necessary for that case: Avisynth -> -lavifil32 VFW Cap -> -lavicap32 Patch by kemuri <kemuri9 at gmail dot com>
Originally committed as revision 21410 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit a1b3c5a377)
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Conflicts:

	configure
2012-06-02 19:25:57 -04:00
Ramiro Polla
b990187f99 configure: properly check for mingw-w64 through installed headers. mingw-w64 can also target 32-bit code.
Originally committed as revision 24156 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 0a4307d630)
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2012-06-02 19:25:57 -04:00
Ronald S. Bultje
ae6c57859c qdm2: clip array indices returned by qdm2_get_vlc().
Prevents subsequent overreads when these numbers are used as indices
in arrays.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org

Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit 64953f67f9)
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Conflicts:

	libavcodec/qdm2.c
2012-06-02 19:25:57 -04:00
Alex Converse
5629c39101 kmvc: Check palsize.
Fixes: CVE-2011-3952

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Based on fix by Michael Niedermayer
(cherry picked from commit 386741f887)
(cherry picked from commit 416849f2e0)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit e7392dc349291eb94379d8cfb7ef73d32a768858)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-05-28 23:46:08 +02:00
Janne Grunau
7867cbaf6c adpcm: ADPCM Electronic Arts has always two channels
Fixes half of http://ffmpeg.org/trac/ffmpeg/ticket/794
Adresses CVE-2012-0852

(cherry picked from commit bb5b3940b0)

Conflicts:

	libavcodec/adpcm.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit b581580bd1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit a0f58c3a605b8123039628d1598cb36f1da0e815)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-05-28 23:46:08 +02:00
Alexander Strange
0bf8e22cdb h264: Add check for invalid chroma_format_idc
Fixes a crash when FF_DEBUG_PICT_INFO is used.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 6ef4063957)

Fixes: CVE-2012-0851

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 4713234518)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit c5f7c755cf)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 00d2c432581cf61326973a1a48f2e63690b65515)
2012-05-28 23:46:08 +02:00
Alex Converse
7944a87ba8 dpcm: ignore extra unpaired bytes in stereo streams.
Fixes: CVE-2011-3951

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit ce7aee9b73)
(cherry picked from commit eaeaeb265f)

Conflicts:

	libavcodec/dpcm.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 1ce9c93198fc997e8f23934a78e2937af670e4e9)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 41f1f146c9e29dde63e293078819474c9b8111a1)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-05-28 23:46:08 +02:00
Michael Niedermayer
a55db1fc49 dsp: fix diff_bytes_mmx() with small width
Fixes Ticket1068

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 73089eccd3e48539555349b36d8aabbf1cea416e)
2012-05-11 22:39:50 +02:00
Michael Niedermayer
123e925956 mmdemux: dont set pkt->size to an invalid value.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0c97fd336e)
2012-05-11 22:39:37 +02:00
Michael Niedermayer
96c6b3a11c h261: check mtype.
Fixes out of array read

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ec3cd74f2d)
2012-05-11 22:39:31 +02:00
Michael Niedermayer
5e3cd42b6d 4xmdemux: Check chunk size
Fixes over reading the header array

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 474e31c904)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-11 22:38:28 +02:00
Michael Niedermayer
f73106256d fix release number
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-11 22:37:20 +02:00
Michael Niedermayer
782c3ab777 Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Bump version number for 0.5.8 release.
  Release notes and changelog for 0.5.7
  vqavideo: return error if image size is not a multiple of block size
  motionpixels: Clip YUV values after applying a gradient.
  mjpegbdec: Fix overflow in SOS.
  atrac3: Fix crash in tonal component decoding.
  dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
  dv: Fix null pointer dereference due to ach=0
  dv: check stype
  nsvdec: Propagate errors
  nsvdec: Be more careful with av_malloc().
  nsvdec: Fix use of uninitialized streams.

Conflicts:
	libavcodec/atrac3.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-11 22:02:11 +02:00
Reinhard Tartler
4b2e02a4c4 Bump version number for 0.5.8 release. 2012-05-10 20:40:38 +02:00
Reinhard Tartler
79fbcd9f03 Release notes and changelog for 0.5.7 2012-05-10 20:15:51 +02:00
Mans Rullgard
468cc41d6d vqavideo: return error if image size is not a multiple of block size
The decoder assumes in various places that the image size
is a multiple of the block size, and there is no obvious
way to support odd sizes.  Bailing out early if the header
specifies a bad size avoids various errors later on.

Fixes CVE-2012-0947.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 58b2e0f0f2)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit d5207e2af8)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit c71c77e56f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit c90da45d5a7a4045dbf22fba52c63ef55d207269)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-05-09 22:34:07 +02:00
Alex Converse
6c9b404dba motionpixels: Clip YUV values after applying a gradient.
Prevents illegal reads on truncated and malformed input.

CC: libav-stable@libav.org
(cherry picked from commit b5da848fac)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit aaa6a66677)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 50073e2395)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 2134e7f6e88959513ba1713ad6fd7a7c8d5a0f41)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-05-09 22:33:49 +02:00
Alex Converse
b2ac7e585e mjpegbdec: Fix overflow in SOS.
Based in part by a fix from Michael Niedermayer <michaelni@gmx.at>

Fixes CVE-2011-3947

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

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 083a8a0037)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 6ae95a0b93)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 6ca010f209)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-04-21 15:42:28 +02:00
Michael Niedermayer
8bb3ba5541 atrac3: Fix crash in tonal component decoding.
Add a check to avoid writing past the end of the channel_unit.components[]
array.

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

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

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit f43b6e2b1e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit f728ad26f0)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 224025d852)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-04-21 15:42:10 +02:00
Alex Converse
5a92aa378d dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
Found with asan.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 2d1c0dea5f)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 00fa6ffe1a)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit a8f4db0acd)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-04-21 15:41:57 +02:00
Michael Niedermayer
c4e8c99507 dv: Fix null pointer dereference due to ach=0
dv: Fix null pointer dereference due to ach=0

Fixes part2 of CVE-2011-3929

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 5a396bb3a6)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 44e182d41e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit b46141b0d1)

Conflicts:

	libavformat/dv.c
2012-04-21 15:41:30 +02:00
Michael Niedermayer
479869c499 dv: check stype
dv: check stype

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

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 635bcfccd4)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit bb737d381f)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 38421f27b3)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-04-21 15:39:00 +02:00
Alex Converse
ec4979e16e nsvdec: Propagate errors
Related to CVE-2011-3940.

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

Conflicts:

	libavformat/nsvdec.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 0100c4b1b0)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 3253dd2b42)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-04-21 15:38:44 +02:00
Alex Converse
056c909d9d nsvdec: Be more careful with av_malloc().
Check results for av_malloc() and fix an overflow in one call.

Related to CVE-2011-3940.

Based in part on work from Michael Niedermayer.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit 8fd8a48263)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit be524c186b)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 87007519c8)

Conflicts:

	libavformat/nsvdec.c
2012-04-21 15:38:10 +02:00
Michael Niedermayer
bde4b66063 nsvdec: Fix use of uninitialized streams.
Fixes CVE-2011-3940 (Out of bounds read resulting in out of bounds write)

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

Signed-off-by: Alex Converse <alex.converse@gmail.com>
(cherry picked from commit 6a89b41d97)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 65beb8c117)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
(cherry picked from commit 1edf848a81)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-04-21 15:36:40 +02:00
Michael Niedermayer
0fda37cff9 Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  id3v2: fix skipping extended header in id3v2.4

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-04-02 02:25:49 +02:00
Anton Khirnov
2e693be7e9 id3v2: fix skipping extended header in id3v2.4
In v2.4, the length includes the length field itself.
(cherry picked from commit ddb4431208)

Conflicts:

	libavformat/id3v2.c

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-04-01 19:49:37 +02:00
Michael Niedermayer
8e101086eb update for 0.5.8
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-12 22:19:09 +01:00
Michael Niedermayer
f13de3c653 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 22:15:00 +01:00
Michael Niedermayer
1eb7872238 Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Release notes and changelog for 0.5.7
  Bump version number for 0.5.7 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 22:14:10 +01:00
Reinhard Tartler
15df4428d2 Release notes and changelog for 0.5.7 2012-01-10 22:22:05 +01:00
Reinhard Tartler
ec0124203c Bump version number for 0.5.7 release. 2012-01-10 21:23:27 +01:00
Chris Evans
6b01bcebb9 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
(cherry picked from commit a5e0afe3c9)

Conflicts:

	libavcodec/vorbis_dec.c
2012-01-08 09:49:19 +01:00
Reinhard Tartler
efd453d82d 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
(cherry picked from commit 42f0a66968)

Conflicts:

	libavcodec/vorbis_dec.c
2012-01-08 09:40:38 +01:00
Michael Niedermayer
7209c2b13f Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  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 oob read for negative tokens and memleaks on error.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-08 05:03:39 +01:00
Chris Evans
7ee536e87a 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>
(cherry picked from commit 90a4a46747)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-01-07 22:16:29 +01:00
Chris Evans
665421f3b1 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>
(cherry picked from commit 6d6254ba9f)

Conflicts:

	libavcodec/vorbis.c
2012-01-07 22:15:53 +01:00
Ronald S. Bultje
3eb6983dbc 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
(cherry picked from commit c9c7db0af2)

Conflicts:

	libavcodec/vp3.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2012-01-07 21:39:50 +01:00
Michael Niedermayer
e75056bc54 Update for 0.5.7
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-25 21:43:56 +01:00
Michael Niedermayer
8755a7890e Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Release notes and changelog for 0.5.6

Conflicts:
	RELEASE

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-25 20:19:13 +01:00
Reinhard Tartler
d0688fdd31 Release notes and changelog for 0.5.6 2011-12-25 10:18:18 +01:00
Michael Niedermayer
0e05292a6c Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Bump version number for 0.5.6 release.
  svq1dec: call avcodec_set_dimensions() after dimensions changed.
  vmd: fix segfaults on corruped streams
  vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
  Plug some memory leaks in the VP6 decoder
  vp6: Reset the internal state when aborting key frames header parsing
  vp6: Fix illegal read.
  vp6: Fix illegal read.
  Fix out of bound reads in the QDM2 decoder.
  Check for out of bound writes in the QDM2 decoder.
  qdm2: check output buffer size before decoding
  Fix qdm2 decoder packet handling to match the api

Conflicts:
	libavcodec/qdm2.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-25 00:53:49 +01:00
Reinhard Tartler
23f228a0d0 Bump version number for 0.5.6 release. 2011-12-24 16:32:06 +01:00
Michael Niedermayer
110aff4b24 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>
(cherry picked from commit 8ddc0b491d)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:28:20 +01:00
Laurent Aimar
4a1c3df592 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>
(cherry picked from commit b99366faef)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:28:20 +01:00
Dustin Brody
185abfb218 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>
(cherry picked from commit 201fcfb894)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:28:20 +01:00
Vitor Sessak
280590e338 Plug some memory leaks in the VP6 decoder
Originally committed as revision 22172 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 0a41faa9a7)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:28:19 +01:00
Laurent Aimar
9767ea7aa7 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>
(cherry picked from commit e28bb18fdc)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:17:42 +01:00
Thierry Foucu
771ceb19f2 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>
(cherry picked from commit 94aacaf508)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:14:55 +01:00
Alex Converse
7739947671 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>
(cherry picked from commit 8d68083298)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:14:41 +01:00
Laurent Aimar
8abf1d882e 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 16:07:49 +01:00
Laurent Aimar
1a53095406 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 16:07:36 +01:00
Justin Ruggles
60eebf5c12 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
(cherry picked from commit cfb9b47a1e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:06:10 +01:00
Baptiste Coudurier
30ee6c1995 Fix qdm2 decoder packet handling to match the api
Originally committed as revision 25767 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit b26c1a8b7e)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-12-24 16:05:52 +01:00
Michael Niedermayer
b769df8ff2 update for 0.5.6
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 22:22:04 +01:00
Michael Niedermayer
cc2580e6e9 svq1dec: call avcodec_set_dimensions() after dimensions changed.
Fixes NGS00148

Found-by: Phillip Langlois
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4931c8f0f1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 661ee45f88)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 22:19:22 +01:00
Michael Niedermayer
4b0f8aed13 qdm2dec: fix buffer overflow.
Fixes NGS00144

This also adds a few lines of code from master that are needed for this fix.

Thanks to Phillip for suggestions to improve the patch.
Found-by: Phillip Langlois
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a6a61a6d1d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 22:01:29 +01:00
Michael Niedermayer
2c8ac66456 qdm2dec: check remaining input bits in the mainloop of qdm2_fft_decode_tones()
This is neccessary but likely not sufficient to prevent out of array reads.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 14db3af4f2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8120a1d9bd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 21:59:45 +01:00
Michael Niedermayer
4f209fe960 cinepak: check strip_size
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cea0c82d9b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 211a107208)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 21:59:34 +01:00
Michael Niedermayer
037b1142cd wma: Check channel number before init.
Fixes Ticket240

Based on patch by ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 20431a9982)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 21:59:17 +01:00
Alex Converse
37cc48861d vp5: Fix illegal read.
Found with Address Sanitizer
(cherry picked from commit bb4b0ad83b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f62fa1ce9f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 21:57:27 +01:00
Thierry Foucu
0cd61bfa6d 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: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8a63deab15)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-21 21:57:09 +01:00
Michael Niedermayer
16ea6af381 0.5: some updates.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 20:57:55 +01:00
Michael Niedermayer
8bd374858f resample: Fix array size
Found-by: Jim Radford
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3e7db0a9ee)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit edf3c5a3eb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Michael Niedermayer
d39cc3c092 resample2: fix potential overflow
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a39b5e8b32)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Michael Niedermayer
e124c3c298 resample: Fix overflow
Found-by: Jim Radford
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6ae93d0304)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Ronald S. Bultje
8acc0546bb matroskadec: fix out of bounds write
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit 723229c11f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d51c7b4cbe)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Michael Niedermayer
36e4be0a0a mem: fix memalign hack av_realloc()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fc11927890)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5ae87280e2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
c603cf5170 qtrle: check for out of bound writes.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7fb92be7e5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a65045915f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
e1a46eff7a qtrle: check for invalid line offset
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a4ed7c3fe9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 67c46b9b30)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
23aaa82b1d vqa: fix double free on corrupted streams
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e3123856c7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Justin Ruggles
58087a4e64 mpc7: return error if packet is too small.
(cherry picked from commit 8290d1f38b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 490617b6ff)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Justin Ruggles
8d1fa1c97e mpc7: check output buffer size before decoding
(cherry picked from commit c8b5c4d274)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b833859daa)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
2eb5f77bc8 h264: do not let invalid values in h->ref_count after a decoder reset.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0333d234b0)
(cherry picked from commit f74d1c6de7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
ddbbe500b0 h264: fix the check for invalid SPS:num_ref_frames.
This patch set the limit to 16.

For information, thoses previous commits:
41f7e2d11d
5cbb0e70a0
assumed it was either 30 or 32.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bcf881a685)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
d1a5b53ede h264: do not let invalid values in h->ref_count on ff_h264_decode_ref_pic_list_reordering() errors.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2428b53f6d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
8e6173c76a Check for out of bound accesses in the 4xm decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9c661e952f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
1a6f024520 Prevent block size from inreasing in the shorten decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b399cbfba5)
(cherry picked from commit 55a96a984e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
3699a46ed7 Check for out of bound writes in the QDM2 decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4a7876c6e4)
(cherry picked from commit b08df314dc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
62da9203fd Check for out of bound writes in the avs demuxer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5d44c061cf)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
2e1e3c1e41 Check for corrupted data in avs demuxer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1cce7def0a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:12 +01:00
Laurent Aimar
635256a324 Fix out of bound writes in fix_bitshift() of the shorten decoder.
The data pointers s->decoded[*] already take into account s->nwrap.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f42b3195d3)
(cherry picked from commit 107ea3057e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:11 +01:00
Laurent Aimar
240546a185 Check for out of bounds writes in the Delphine Software International CIN decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3035c4034b)
(cherry picked from commit 6e774cf67e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:11 +01:00
Laurent Aimar
07df40db6e Check for invalid update parameters in vmd video decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e7aed1280e)
(cherry picked from commit 1ed90c84f6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 19:49:11 +01:00
Laurent Aimar
b24c2e59fe Release old pictures after a resolution change in vp5/6 decoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dba20b8478)
(cherry picked from commit c9c6e5f4e8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 18:40:47 +01:00
Laurent Aimar
25bc1108c2 Check output buffer size in nellymoser decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 741ec30bd2)
(cherry picked from commit 533dbaa55b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 18:40:19 +01:00
Michael Niedermayer
8ef917c033 check all svq3_get_ue_golomb() returns.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 979bea1300)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 18:34:49 +01:00
Michael Niedermayer
1883249be3 rv34: check for size mismatch
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 35f38b3ab9)
(cherry picked from commit ed9e561490)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 18:32:40 +01:00
Laurent Aimar
648dc68098 Reject audio tracks with invalid interleaver parameters in RM demuxer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4907f81358)
(cherry picked from commit 24e0a9e451)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 18:32:11 +01:00
Michael Niedermayer
10da0edddc Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  update version
  Release notes and changelog for 0.5.5
  Fix ff_imdct_calc_sse() on gcc-4.6
  Make DECLARE_ALIGNED macros work with external array specifiers
  Fix MMX rgb24 to yuv conversion with gcc 4.6

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-06 01:35:41 +01:00
Reinhard Tartler
d6bf79993f update version 2011-11-05 12:57:22 +01:00
Reinhard Tartler
0f2735e839 Release notes and changelog for 0.5.5 2011-11-05 12:53:16 +01:00
Alex Converse
9463a28792 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)

Conflicts:

	libavcodec/x86/fft_sse.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 11:29:12 +01:00
Måns Rullgård
1330a8a1cb Make DECLARE_ALIGNED macros work with external array specifiers
The macro implementation might need the name of the variable being
declared for compiler-specific syntax.  Moving array specifiers outside
the macro invocation allows this to work.

Originally committed as revision 21363 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 8a24e98d50)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 11:29:12 +01:00
Mans Rullgard
8135c35528 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)

Conflicts:

	libswscale/swscale_template.c

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-11-05 09:04:11 +01:00
Michael Niedermayer
38423fe0b7 smacker: add forgotten *
found by fenrir

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f98edc73c5)
2011-11-03 03:52:55 +01:00
Laurent Aimar
71132596ae segafilm: Fix potential division by 0 on corrupted segafilm streams in the demuxer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-03 03:52:55 +01:00
Laurent Aimar
d6f8b65417 segafilm: Check for memory allocation failures in segafilm demuxer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7cbe025758)
2011-11-03 03:52:55 +01:00
Kostya Shishkov
d8439f0452 rv34: check that subsequent slices have the same type as first one.
This prevents some crashes when corrupted bitstream reports e.g. P-type
slice in I-frame. Official RealVideo decoder demands all slices to be
of the same type too.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 23a1f0c592)
2011-11-03 03:52:55 +01:00
Laurent Aimar
eb6b0ed8be Fixed invalid read access on extra data in cinepak decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dc255275f6)
2011-11-03 03:33:47 +01:00
Laurent Aimar
6108f04d4f Fixed segfault on corrupted smacker streams in the demuxer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d0121e8d96)
2011-11-03 03:33:41 +01:00
Laurent Aimar
b261ebfd22 Fixed segfaults on corruped smacker streams in the decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d07ac1853d)
2011-11-03 03:33:35 +01:00
Laurent Aimar
03db051b43 Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 8bfea4ab4e)
2011-11-03 03:32:50 +01:00
Laurent Aimar
cc885682e3 Fixed deference of NULL pointer in motionpixels decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 824f98f442)
2011-11-03 03:32:23 +01:00
Chris Rankin
1415ebf031 qcelpdec: fix the return value of qcelp_decode_frame().
(cherry picked from commit 04c13dca88)
2011-11-03 03:31:25 +01:00
Reimar Döffinger
4e0315b30e Check extradata size on resolution change.
Ignore resolution change if resolution not defined in extradata.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit 09c5f990bc)
2011-11-03 03:30:35 +01:00
Laurent Aimar
a9ded3d272 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>
2011-11-03 03:29:32 +01:00
Laurent Aimar
ef93642aac 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>
2011-11-03 03:29:21 +01:00
Laurent Aimar
9cda3d7915 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>
2011-11-03 03:29:12 +01:00
Laurent Aimar
52b8edc94c 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>
2011-11-03 03:28:27 +01:00
Laurent Aimar
f936799f0b 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>
2011-11-03 03:27:53 +01:00
Laurent Aimar
f6d3dfe78b 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>
2011-11-03 03:27:46 +01:00
Laurent Aimar
2e17744a90 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>
2011-11-03 03:27:39 +01:00
Laurent Aimar
19431d4d4e 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>
2011-11-03 03:27:24 +01:00
Laurent Aimar
ecd6fa11c2 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>
2011-11-03 03:27:16 +01:00
Alex Converse
7cb35d4954 cljr: init_get_bits size in bits instead of bytes
(cherry picked from commit 0c1f5b93d9)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2011-11-03 03:26:33 +01:00
Alex Converse
457f869b73 indeo2: fail if input buffer too small
(cherry picked from commit b7ce4f1d1c)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2011-11-03 03:26:19 +01:00
Alex Converse
70f01f1262 indeo2: init_get_bits size in bits instead of bytes
(cherry picked from commit 68ca330cbd)

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2011-11-03 03:26:11 +01:00
Michael Niedermayer
80fb9f2c57 cavsdec: avoid possible crash with crafted input
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9f06c1c61e)
2011-11-03 03:22:37 +01:00
Carl Eugen Hoyos
46f9a6203a Fix possible double free when encoding using xvid.
(cherry picked from commit 315f0e3fd8)
2011-11-03 03:17:19 +01:00
Michael Niedermayer
e9210b4e17 Merge remote-tracking branch 'qatar/release/0.5' into release/0.5
* qatar/release/0.5:
  Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
  cavs: fix some crashes with invalid bitstreams
  mjpeg: Detect overreads in mjpeg_decode_scan() and error out.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-03 02:23:42 +01:00
Michael Niedermayer
4f07a3aa2c 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>
2011-11-02 21:49:55 +01:00
Mans Rullgard
04888edef3 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>
2011-11-02 21:14:57 +01:00
Michael Niedermayer
24cd7c5df7 Fix apparently exploitable race condition.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-26 12:30:57 +02:00
Michael Niedermayer
8210ee22e2 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>
2011-04-26 12:12:11 +02:00
Michael Niedermayer
eed5697f99 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>
2011-04-26 09:40:06 +02:00
Reinhard Tartler
52ee20f2aa update release date 2011-03-17 13:10:27 +01:00
Reinhard Tartler
f17b892787 document APE patch 2011-03-17 13:09:40 +01:00
Kostya
18c5fe919f 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)
2011-03-16 13:27:01 +01:00
Diego Biurrun
c5898d7c1d Fix a bunch of typos in the release documentation. 2011-03-06 11:02:36 +01:00
Reinhard Tartler
2c4d6aeabc Bump version number for 0.5.4 release. 2011-02-20 22:13:05 +01:00
Reinhard Tartler
31c8dcedb2 release notes for 0.5.4 2011-02-20 09:44:18 +01:00
Reinhard Tartler
2adad90ae7 Amend Changelog for 0.5.4 2011-02-20 09:38:11 +01:00
Kostya Shishkov
808f9ce727 Call avcodec_set_dimensions() instead of simply setting avctx->width/height
when frame dimensions change in RV3/4.

Originally committed as revision 20595 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit d90aeeaf56)
2011-02-19 17:07:58 +01:00
Reimar Döffinger
8069e2f6fb 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
2011-02-19 17:07:57 +01:00
Ronald S. Bultje
f7494394ee Make get_bits_left() available for use in libavcodec (was previously held
private in dv.c for some reason). See "[PATCH] get_bits_left()" thread.

Originally committed as revision 20490 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit c47ca25e74)
2011-02-19 17:07:57 +01:00
Reinhard Tartler
04847a06c2 Update Changelog for 0.5.4 release. 2011-02-13 23:35:38 +01:00
Frank Barchard
329e816ed7 Check rangebits to avoid a possible crash.
Fixes issue 2548 (and Chrome issue 68115 and unknown CERT issues).

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

Addresses: CVE-2011-0480

Conflicts:

	libavcodec/vorbis_dec.c
2011-02-13 21:41:38 +01:00
Jason Garrett-Glaser
d6860fb653 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)

Addresses: CVE-2010-4704
2011-02-13 20:45:18 +01:00
Reinhard Tartler
e332c41670 also ignore *.so for vhook plugins 2011-02-10 14:09:35 +01:00
Janne Grunau
11f6eebdd3 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>
2011-02-10 14:02:23 +01:00
Janne Grunau
9109a58867 convert svn:ignore properties to .gitignore files
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 348b8218f7)

Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-02-10 14:01:36 +01:00
Kostya Shishkov
44511b17cb Update dimensions in AVCodecContext when RV3/4 frame dimensions change
Originally committed as revision 20572 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit ec10d2d539)

Fixes heap corruption crashes

Addresses: CVE-2011-0722
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-02-04 06:42:29 +01:00
Michael Niedermayer
48b086b0ef Update safety check as the maximum pixel size is no longer 4.
New max size is 16bit * 4 samples (RGBA).

Originally committed as revision 18655 to svn://svn.ffmpeg.org/ffmpeg/trunk
(cherry picked from commit 445f0a8b66)

Addresses: CVE-2010-3908
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-02-04 06:41:58 +01:00
Reinhard Tartler
c9864adf34 release notes for 0.5.3
Originally committed as revision 25523 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-10-18 19:43:55 +00:00
Diego Biurrun
7d10059aeb Bump version number for 0.5.3 release.
Originally committed as revision 25522 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-10-18 19:40:09 +00:00
Diego Biurrun
69e8b43812 Update Changelog for 0.5.3 release.
Originally committed as revision 25521 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-10-18 19:38:02 +00:00
Reinhard Tartler
2f504d7a90 Fix several security issues in flicvideo.c
This fixes CVE-2010-3429


backport r25223 by michael


Originally committed as revision 25325 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-10-03 14:51:50 +00:00
Reinhard Tartler
2dea9a1266 unbreak compilation and finish backport r24280 by mstorsjo
Originally committed as revision 25324 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-10-03 14:50:04 +00:00
Reinhard Tartler
84e6629de3 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 24428 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-07-22 11:58:26 +00:00
Reinhard Tartler
fc038df32e configure: improve temp file creation and cleanup
backport r17752 by mru


Originally committed as revision 23393 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-05-31 13:06:51 +00:00
Diego Biurrun
021054a196 release notes for 0.5.2
Originally committed as revision 23300 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-05-24 21:58:47 +00:00
Diego Biurrun
ee20f19b20 Bump version number for 0.5.2 release.
Originally committed as revision 23299 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-05-24 21:55:01 +00:00
Diego Biurrun
2fcb56dab9 Update Changelog for 0.5.2 release.
Originally committed as revision 23298 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-05-24 21:41:51 +00:00
Reinhard Tartler
96ca078b22 Check validity of channels & samplerate.
This may be security relevant.
Based on 2 patches by chrome.

backport r19975 by michael




Originally committed as revision 22658 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-24 19:35:30 +00:00
Reinhard Tartler
7fd4cbb519 fix compilation issue on powerpc
unlike the ARCH_ macros, COMPILE_ALTIVEC needs to be tested more carefully


Originally committed as revision 22488 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-12 20:35:04 +00:00
Reinhard Tartler
557e065d5f Fix compilation on powerpc with --disable-altivec
in case altivec is disabled, even compilation of code using altivec
keywords or asm must be avoided.

backport r30869 from mplayer repo by siretart


Originally committed as revision 22436 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-10 20:55:07 +00:00
Diego Biurrun
461243731d Mention LGPL libswscale in the Changelog.
Originally committed as revision 22253 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-06 19:50:56 +00:00
Diego Biurrun
fe95afe1e2 libswscale is no longer GPL; update help comment accordingly.
Originally committed as revision 22250 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-06 19:40:37 +00:00
Andres Mejia
775aa5f38c Add Hurd to OS list and disable dv1394 in the Hurd case.
patch by Andres Mejia, mcitadel gmail com

backport r18938 by diego


Originally committed as revision 22237 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-06 16:57:43 +00:00
Diego Biurrun
578c32814c Add point release date.
Originally committed as revision 22163 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-03 08:25:10 +00:00
Reinhard Tartler
c46038f6b7 fix 'seektest' again
backport  r19270 by rbultje:

Remove any reference to ASFContext.packet_size and replace it with
AVFormatContext.packet_size. See "[PATCH] asf*.c/h: use
AVFormatContext->packet_size instead of own copy" thread on ML.

and r19361 by reimar:

Check for packet_length 0, it is already treated as invalid by the padding check,
but that resulted in a confusing/wrong error message.



Originally committed as revision 22147 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-02 16:03:06 +00:00
Diego Biurrun
306eefc49f Bump version to 0.5.1.
Originally committed as revision 22146 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-02 14:43:01 +00:00
Diego Biurrun
eade5150e4 Mention licensing-related changes; some whitespace adjustments.
Originally committed as revision 22145 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-02 14:25:48 +00:00
Diego Biurrun
6d767afb7c If we are using partial release names we might as well try to be funny.
Originally committed as revision 22134 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-01 18:03:53 +00:00
Diego Biurrun
015a7d7362 Add release managers, merged from trunk.
Originally committed as revision 22133 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-01 17:58:50 +00:00
Reinhard Tartler
922c55a09b amend release notes for 0.5.1
Originally committed as revision 22129 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-03-01 16:22:27 +00:00
Diego Biurrun
4c83c13bc8 Mention security fixes in the changelog.
Originally committed as revision 22121 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-28 22:22:22 +00:00
Reinhard Tartler
bd7e30ea00 add myself to gpg fingerprint list
backport r22089 by siretart


Originally committed as revision 22090 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-27 10:16:45 +00:00
Reinhard Tartler
4fb58ecea8 bump LIBAVCODEC_VERSION_MICRO for addition of the lock manager API
As discussed with Diego, we'll go for bumping micro in 0.5 and will
consider adding a RELEASEVERSION macro for trunk and 0.6 seperatly


Originally committed as revision 22087 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-27 10:01:45 +00:00
Reinhard Tartler
a317cd5722 Avoid divisions by 0 in the ASF demuxer if packet_size is not valid.
r19330 by reimar


Originally committed as revision 22080 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-26 15:49:52 +00:00
Reinhard Tartler
8e2149d7df fix the remaining ogv segfaults from issue 1240.
First commit:

Make decode_init fail if the huffman tables are invalid and thus init_vlc fails.
Otherwise this will crash during decoding because the vlc tables are NULL.
Partially fixes ogv/smclock.ogv.1.101.ogv from issue 1240.

backport r19355 by reimar

Second commit:

Add extra validation checks to ff_vorbis_len2vlc.
They should not be necessary, but it seems like a reasonable precaution.

r19374 by reimar


Originally committed as revision 22076 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-26 14:32:27 +00:00
Reinhard Tartler
9d9f1ecfaa Make sure we dont read over the end.
Fixes issue1237.

backport r19322 by michael


Originally committed as revision 22074 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-26 10:56:46 +00:00
Reinhard Tartler
53b90bb25e backport libx264.c from trunk
now compiles with x264 API versions 65 up to 85

patch prepared by darkshikari


Originally committed as revision 22042 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-24 22:40:10 +00:00
Reinhard Tartler
a0244ae347 misc. manpage updates, fixes LP: #501729, Debian: #570050
Update ffmpeg documentation regarding metadata setting. -title,
-author, -copyright, -track, -album, and -year options have been
dropped in favor of -metadata.
Add an explanation and complete the metadata usage example.

backported revisions r19285, r19287 and r19320 by stefano.



Originally committed as revision 21858 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-16 23:04:10 +00:00
Reinhard Tartler
26f74e832b cosmetics: K&R coding style, prettyprinting
backported r20083 by diego

This commit does not introduce functional changes.  It was applied in
order to faciliate reviewing the proposed libx264.c backport



Originally committed as revision 21832 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-15 12:45:14 +00:00
Reinhard Tartler
9593c80062 Fix crash in MLP decoder due to integer overflow.
Probably only DoS, init_get_bits sets buffer to NULL, thus causing a
NULL-dereference directly after.

backport r21426 by reimar


Originally committed as revision 21759 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-11 21:03:30 +00:00
Reinhard Tartler
48b98cdc67 Make sure the block array is of the correct size.
This might have been exploitable.

backported r18393 by michael



Originally committed as revision 21758 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-11 20:57:49 +00:00
Reinhard Tartler
9d442d2d7d Fix crash when max_ref_frames was out of range.
This might have been exploitable.
Fixes first crash of issue840.

backport r18388 by michael


Originally committed as revision 21757 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-11 16:22:59 +00:00
Reinhard Tartler
afc97d4735 reverting objected hunks from previous commit
as discussed with diego on irc, the spurious newline deletion and the
LIBAVCODEC_VERSION_MINOR bump are being reverted based on comments on
ffmpeg-cvslog by ramiro, uoti and michael.

See http://comments.gmane.org/gmane.comp.video.ffmpeg.cvs/28112 for the
full context.


Originally committed as revision 21755 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-11 11:52:59 +00:00
Reinhard Tartler
e5bea45df7 Add a lock manager API to libavcodec.
Allows an application to register a callback that manages mutexes
on behalf of FFmpeg.
With this callback registered FFmpeg is fully thread safe.

backport r19025 by andoma

NB: This is a feature backport with little regression potential. It was
requested at FOSDEM 2010 by ben@geexbox.org for use by geexbox and the
enna mediacenter in the upcoming debian/squeeze and ubuntu/lucid
release.

Approved by DonDiego on #ffmpeg-devel


Originally committed as revision 21731 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 20:28:42 +00:00
9e3935dfd8 Check submap indexes.
10_vorbis_submap_indexes.patch by chrome.
I am applying this even though Reimar had some comments to improve it as it fixes
a serious security issue and I do not want to leave such things unfixed.

backport r20001 by michael


Originally committed as revision 21730 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:59:11 +00:00
4f5ee3f87b Check begin/end/partition_size.
23_vorbis_sane_partition.patch by chrome.
Also this should be better documented but i prefer not to leave potential
security issues open due to missing documentation.

r19996 by michael


Originally committed as revision 21729 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:54:42 +00:00
736d36b792 Check res_setup->books.
15_more_residue_book_indexes.patch by chrome.

r19992 by michael


Originally committed as revision 21728 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:52:27 +00:00
dc5cc27d5a Check masterbook index and subclass book index.
14_floor_masterbook_index.patch by chrome

r19991 by michael


Originally committed as revision 21727 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:50:47 +00:00
eb70d77e1e Add checks for per-packet mode indexes and per-header mode mapping indexes.
12_vorbis_mode_indexes.patch by chrome
maybe exploitable

r19990 by michael


Originally committed as revision 21726 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:49:28 +00:00
b8ec4c49bd Check classbook value.
11_vorbis_residue_book_index.patch by chrome.

r19989 by michael


Originally committed as revision 21725 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:47:42 +00:00
49487dfd78 Fix book_idx check.
25_vorbis_floor0_index.patch by chrome.

backport r19984 by michael


Originally committed as revision 21724 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:46:22 +00:00
0d3f25eb48 Sanity checks for magnitude and angle.
26_vorbis_mag_angle_index.patch by chrome

backport r19983 by michael


Originally committed as revision 21723 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:45:16 +00:00
9ef13f70f4 = -> == typo.
27_vorbis_residue_loop_error.patch by chrome

backport r19982 by michael


Originally committed as revision 21722 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:44:06 +00:00
Reinhard Tartler
beb0dc5f36 Check dimensions against 0 too.
39_vorbis_zero_dims.patch from chrome

backport r19976 by michael


Originally committed as revision 21721 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:42:30 +00:00
Reinhard Tartler
f3fd428697 Make sure that all memory allocations succeed.
Based on 28_theora_malloc_checks.patch from the Google Chrome team.

backport r20008 by melanson


Originally committed as revision 21720 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:31:04 +00:00
8811fe6987 Fix init_get_bits() buffer size.
18_fix_theora_header_bit_len.patch by chrome

backport r19993 by michael


Originally committed as revision 21719 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:26:47 +00:00
Reinhard Tartler
3835603e7e Set data_size to 0 to avoid having it uninitialized.
based on 31_mp3_outlen.patch by chrome.

backported r19988 by michael


Originally committed as revision 21718 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:22:19 +00:00
Reinhard Tartler
4b360ee2ca Check data_size in decode_frame_mp3on4().
backported r19987 by michael


Originally committed as revision 21717 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:20:25 +00:00
Reinhard Tartler
ab546a7463 check data_size in decode_frame()
backported r19986 by michael




Originally committed as revision 21716 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:09:12 +00:00
Reinhard Tartler
acac7858bd Check num_units_in_tick/time_scale to be valid and within the range we support.
based on a patch by chrome

backported r19979 by michael




Originally committed as revision 21715 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 19:02:39 +00:00
Reinhard Tartler
c42640b200 Fix a possibly exploitable buffer overflow.
backported r18640 by michael



Originally committed as revision 21712 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 18:55:41 +00:00
Reinhard Tartler
ef84190a1a Fix possible buffer over-read in vorbis_comment, fix it double to be sure.
First, make s signed, so that comparisons against end - p will not be made as
unsigned, making the check incorrectly pass if p is beyond end.
Also ensure that p will never be > end, so the code is correct also if
buf is not padded.

backported r20014 by reimar


Originally committed as revision 21711 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 18:51:11 +00:00
Reinhard Tartler
7db16a8173 check stream existence before assignment, fix #1222
backported r19259 by bcoudurier



Originally committed as revision 21710 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 18:44:49 +00:00
Reinhard Tartler
e91ba7dc9d add one missing check for stream existence in read_elst, fix #1364
backported patch r19792 by bcoudurier




Originally committed as revision 21709 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 18:41:17 +00:00
Reinhard Tartler
95f90d27d2 Disable parsing for ogg streams where no ogg header was found,
if no header was found the parser was not initialized and thus will
crash when trying to use it.



Originally committed as revision 21708 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 18:24:30 +00:00
Reinhard Tartler
1e9ac36f66 Make arguments of av_set_pts_info() unsigned.
Fixes issue1240/mpeg1/smclockmpeg1.avi.3.1



Originally committed as revision 21707 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 18:10:07 +00:00
Reinhard Tartler
7d061cfe17 fix aac playback regression
Discussed at http://comments.gmane.org/gmane.comp.video.ffmpeg.devel/103768

related reports:
 - http://bugs.debian.org/540729
 - https://roundup.ffmpeg.org/roundup/ffmpeg/issue800



Originally committed as revision 21706 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-09 17:36:18 +00:00
Reinhard Tartler
e21e76a914 build PIC code on powerpc
this avoids failing 24bit relocations as seen on
http://bugs.debian.org/561956 as side effect


Originally committed as revision 21613 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-02 20:48:21 +00:00
Andres Mejia
b18806f811 Add gnu/kfreebsd to list of recognized operating systems.
patch by Andres Mejia, mcitadel gmail com


Originally committed as revision 21612 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-02 20:44:47 +00:00
Reinhard Tartler
478394bab7 backport configure bits for cpu runtime detection for libpostproc and libswscale
Originally committed as revision 21611 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-02 20:30:20 +00:00
Reinhard Tartler
de3196da60 fix version script for libswscale
Originally committed as revision 21610 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-02 20:16:46 +00:00
Reinhard Tartler
49549033fd unbreak compilation with vhook enabled
the backported introduced a regression that prevents successfully
linking vhook shared objects on systems with gnu linkers. As the version
scripts only apply to shared objects against that applications are being
linked, this commit unties the VHOOKSHFLAGS and SHFLAGS variable.


Originally committed as revision 21609 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-02 19:58:05 +00:00
Reinhard Tartler
15dfbc503d missing hunk to unbreak linking with symbol versioning
this hunk is does not apply in trunk/ because there the dependencies for
this rule was reworked


Originally committed as revision 21608 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-02 19:57:40 +00:00
Reinhard Tartler
f5c694972e mention symbol versioning
Originally committed as revision 21596 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-01 16:02:46 +00:00
Reinhard Tartler
a9785f58c6 backport symbol versioning patch
Originally committed as revision 21595 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2010-02-01 16:00:09 +00:00
Diego Biurrun
7a5e131735 The license upgrade code was ported from trunk.
Originally committed as revision 20876 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-12-15 22:34:22 +00:00
Diego Biurrun
93229681b5 Merge remaining changes to make libswscale usable in LGPL mode from trunk.
Originally committed as revision 19352 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-07-05 20:35:02 +00:00
Stefano Sabatini
8819b9c600 Revert r19321.
The changes were not approved.


Originally committed as revision 19329 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-07-03 11:14:37 +00:00
Stefano Sabatini
5d62141092 Update ffmpeg documentation regarding metadata setting. -title,
-author, -copyright, -track, -album, and -year options have been
dropped in favor of -metadata.

Backfix of r19285, r19287, and r19320.


Originally committed as revision 19321 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-07-01 20:14:19 +00:00
Diego Biurrun
44b20d1d74 Fix OpenCORE build: Do not use new AVPacket infrastructure from trunk.
Originally committed as revision 19134 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-06-07 22:41:11 +00:00
Diego Biurrun
9ad437eafb Merge OpenCORE AMR support from trunk.
Originally committed as revision 19133 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-06-07 20:14:56 +00:00
Diego Biurrun
4fcef88c4d Merge recent libamr changes from trunk, as preparation for OpenCORE support.
Originally committed as revision 19131 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-06-07 16:14:50 +00:00
Diego Biurrun
dd2089dfd8 Merge (L)GPL upgrade code and related changes from trunk.
Originally committed as revision 19129 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-06-07 12:52:31 +00:00
Diego Biurrun
df0ff1a029 Ignore generated files.
Originally committed as revision 18939 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-25 09:17:17 +00:00
Diego Biurrun
41a4fd7a61 Merge more verbose licensing information output.
Originally committed as revision 18931 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-24 22:14:10 +00:00
Diego Biurrun
a4d8ebfaa1 Merge GPL --> LGPL conversion of AC-3 decoder from trunk.
Originally committed as revision 18915 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-23 12:58:44 +00:00
Diego Biurrun
0ae7dcae2c Mention post 0.5 commits in the changelog.
Originally committed as revision 18914 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-23 12:18:53 +00:00
Diego Biurrun
d6c23ec06a Merge explanation of changelog sort order from trunk.
Originally committed as revision 18913 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-23 12:04:11 +00:00
Diego Biurrun
3499f0f3e3 Merge fix for license check function from trunk.
Originally committed as revision 18782 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-10 11:02:03 +00:00
Ramiro Polla
7056dd763f Revert unapproved changes.
Originally committed as revision 18770 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-07 22:46:42 +00:00
Ramiro Polla
eade41f3ec Backport r17995 from trunk.
Originally committed as revision 18760 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-06 15:31:02 +00:00
Ramiro Polla
2f14399e40 Backport r18214 from trunk.
Originally committed as revision 18759 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-06 15:29:59 +00:00
Diego Biurrun
4f3ce00704 Merge marking of libfaac as non-free from trunk.
Originally committed as revision 18757 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-06 12:12:05 +00:00
Diego Biurrun
d8ef221893 Merge factorization of license check code from trunk.
Originally committed as revision 18756 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-05-06 12:10:27 +00:00
Diego Biurrun
8d003e22ca Merge automatic addition of -fno-common to CFLAGS for Windows from trunk.
Originally committed as revision 18619 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-04-19 13:22:08 +00:00
Diego Biurrun
b0b57fa13b Merge replacement of MPEG group reference DCT code.
Originally committed as revision 18492 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-04-13 10:23:10 +00:00
Diego Biurrun
266f6af570 Merge LGPL relicensing of AltiVec optimizations.
Originally committed as revision 18491 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-04-13 10:15:48 +00:00
Diego Biurrun
241c55aabe Merge fix for GPL code that erroneously made it into the LGPL build.
Originally committed as revision 18490 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-04-13 10:15:04 +00:00
Diego Biurrun
030896c76a Merge split of README <-> LICENSE files along with the clarifications.
Originally committed as revision 18489 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-04-13 10:11:31 +00:00
Diego Biurrun
beb93f987c Add a copy of libswscale into the branch instead of using svn:external.
This will allow merging some changes from trunk.


Originally committed as revision 18488 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-04-13 10:00:56 +00:00
Diego Biurrun
8e8813a0a1 Merge improved version number generation from trunk.
Originally committed as revision 18288 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-31 21:11:04 +00:00
Diego Biurrun
c3c2325adc Revert hackish release version number hardcoding in version.sh.
Originally committed as revision 18287 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-31 21:06:20 +00:00
Diego Biurrun
f8429ed58c Peg libswscale to the revision corresponding to the moment the branch was cut.
Originally committed as revision 17887 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-08 22:13:48 +00:00
Robert Swain
df4763a782 Correct grammar in one sentence and add a note about doc/APIchanges
Originally committed as revision 17805 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-04 16:52:37 +00:00
Robert Swain
58af0caf04 Add some release notes for this 0.5 release branch
Originally committed as revision 17787 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-03 23:04:47 +00:00
Diego Biurrun
3aafe82485 Output 0.5 as version string.
Originally committed as revision 17754 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-03 12:47:47 +00:00
Baptiste Coudurier
07679e680c revert r16717, r16718, r16719, EAGAIN handling, this causes FFserver to hang
Originally committed as revision 17737 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-02 20:32:24 +00:00
Diego Biurrun
da835cc8a3 Create 0.5 release branch.
Originally committed as revision 17727 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-02 08:32:29 +00:00
3006 changed files with 174095 additions and 530929 deletions

40
.gitignore vendored
View File

@@ -1,64 +1,46 @@
.config
.version
*.a
*.o
*.so
*.d
*.def
*.dll
*.exe
*.ho
*.lib
*.pc
*.so
*.so.*
*.ver
*-example
*-test
*_g
*.def
*.dll
*.lib
*.exp
config.*
doc/*.1
doc/*.html
doc/*.pod
doc/fate.txt
doxy
ffmpeg
ffplay
ffprobe
ffserver
avconv
doc/avoptions_codec.texi
doc/avoptions_format.texi
doc/print_options
doc/examples/decoding_encoding
doc/examples/filtering_audio
doc/examples/filtering_video
doc/examples/metadata
doc/examples/muxing
libavcodec/*_tablegen
libavcodec/*_tables.c
libavcodec/*_tables.h
libavcodec/codec_names.h
libavcodec/libavcodec*
libavcore/libavcore*
libavdevice/libavdevice*
libavfilter/libavfilter*
libavformat/libavformat*
libavutil/avconfig.h
libavutil/libavutil*
libpostproc/libpostproc*
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/ffeval
tools/graph2dot
tools/ismindex
tools/lavfi-showfiltfmts
tools/pktdumper
tools/probetest
tools/qt-faststart
tools/trasher
tools/trasher*.d
version.h

View File

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

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
@@ -9,10 +7,8 @@ Fabrice Bellard
Patrice Bensoussan
Alex Beregszaszi
BERO
Thilo Borgmann
Mario Brito
Ronald Bultje
Alex Converse
Maarten Daniels
Reimar Doeffinger
Tim Ferguson
@@ -41,7 +37,6 @@ Michael Niedermayer
François Revol
Peter Ross
Måns Rullgård
Stefano Sabatini
Roman Shaposhnik
Oded Shimon
Dieter Shirley

540
Changelog
View File

@@ -1,438 +1,127 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version next:
version 0.11.2:
- Several bugs and crashes have been fixed as well as build problems
with recent mingw64
version 0.5.9:
- dpcm: ignore extra unpaired bytes in stereo streams (CVE-2011-3951)
- h264: Add check for invalid chroma_format_idc (CVE-2012-0851)
- adpcm: ADPCM Electronic Arts has always two channels (CVE-2012-0852)
- kmvc: Check palsize (CVE-2011-3952)
- qdm2: clip array indices returned by qdm2_get_vlc()
- configure: properly check for mingw-w64 through installed headers
- Replace every usage of -lvfw32 with what is particularly necessary for that case
- mingw32: properly check if vfw capture is supported by the system headers
- mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
- vfwcap: Include windows.h before vfw.h since the latter requires defines from the former
- ea: check chunk_size for validity
- eatqi: move "block" variable into context to ensure sufficient alignment for idct_put
- tqi: Pass errors from the MB decoder
- png: check bit depth for PAL8/Y400A pixel formats.
version 0.11:
Fixes:CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
- setfield filter
- CDXL demuxer and decoder
- Apple ProRes encoder
- ffprobe -count_packets and -count_frames options
- Sun Rasterfile Encoder
- ID3v2 attached pictures reading and writing
- WMA Lossless decoder
- bluray protocol
- blackdetect filter
- libutvideo encoder wrapper (--enable-libutvideo)
- swapuv filter
- bbox filter
- XBM encoder and decoder
- RealAudio Lossless decoder
- ZeroCodec decoder
- tile video filter
- Metal Gear Solid: The Twin Snakes demuxer
- OpenEXR image decoder
- removelogo filter
- drop support for ffmpeg without libavfilter
- drawtext video filter: fontconfig support
- ffmpeg -benchmark_all option
- super2xsai filter ported from libmpcodecs
- add libavresample audio conversion library for compatibility
- MicroDVD decoder
- Avid Meridien (AVUI) encoder and decoder
- accept + prefix to -pix_fmt option to disable automatic conversions.
- complete audio filtering in libavfilter and ffmpeg
- add fps filter
- audio split filter
- vorbis parser
- png parser
- audio mix filter
version 0.5.8:
- id3v2: fix skipping extended header in id3v2.4
- nsvdec: Several bugfixes related to CVE-2011-3940
- dv: check stype
- dv: Fix null pointer dereference due to ach=0
- dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
- atrac3: Fix crash in tonal component decoding, fixes CVE-2012-0853
- mjpegbdec: Fix overflow in SOS, fixes CVE-2011-3947
- motionpixels: Clip YUV values after applying a gradient.
- vqavideo: return error if image size is not a multiple of block size,
fixes CVE-2012-0947.
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 0.5.7:
- 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 0.9:
- 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 0.5.6:
- svq1dec: call avcodec_set_dimensions() after dimensions changed. (NGS00148, CVE-2011-4579)
- vmd: fix segfaults on corruped streams (CVE-2011-4364)
- commits related to CVE-2011-4353:
- vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
- Plug some memory leaks in the VP6 decoder
- vp6: Reset the internal state when aborting key frames header parsing
- vp6: Fix illegal read.
- vp6: Fix illegal read.
- Fix out of bound reads in the QDM2 decoder.
- commits related to CVE-2011-4351:
- Check for out of bound writes in the QDM2 decoder.
- qdm2: check output buffer size before decoding
- Fix qdm2 decoder packet handling to match the api
version 0.8:
version 0.5.5:
- 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)
- 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
- Detect and handle overreads in the MJPEG decoder.
- multiple other security fixes.
version 0.7:
- all the changes for 0.8, but keeping API/ABI compatibility with the 0.6 release
version 0.5.4:
- Fix memory corruption in WMV parsing (addresses CVE-2010-3908)
- Fix heap corruption crashes (addresses CVE-2011-0722)
- Fix crashes in Vorbis decoding found by zzuf (addresses CVE-2010-4704)
- Fix another crash in Vorbis decoding (addresses CVE-2011-0480, Chrome issue 68115)
- 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 0.6:
- PB-frame decoding for H.263
- deprecated vhook subsystem removed
- deprecated old scaler removed
- VQF demuxer
- Alpha channel scaler
- PCX encoder
- RTP packetization of H.263
- RTP packetization of AMR
- RTP depacketization of Vorbis
- CorePNG decoding support
- Cook multichannel decoding support
- introduced avlanguage helpers in libavformat
- 8088flex TMV demuxer and decoder
- per-stream language-tags extraction in asfdec
- V210 decoder and encoder
version 0.5.3:
- build system improvements
- performance fix for seekable HTTP
- fix several potentially exploitable issues in the FLIC decoder
(addresses CVE-2010-3429)
version 0.5.2:
- Hurd support
- PowerPC without AltiVec compilation issues
- validate channels and samplerate in the Vorbis decoder
version 0.5.1:
- build system updates
- documentation updates
- libswscale now is LGPL except for x86 optimizations
- fix for GPL code in libswscale that was erroneously activated
- AltiVec code in libswscale is now LGPL
- remaining GPL parts in AC-3 decoder converted to LGPL
- QCP demuxer
- SoX native format muxer and demuxer
- (L)GPL license upgrade support
- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
- DPX image decoder
- Electronic Arts Madcow decoder
- DivX (XSUB) subtitle encoder
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- experimental AAC encoder
- RTP depacketization of ASF and RTSP from WMS servers
- RTMP support in libavformat
- noX handling for OPT_BOOL X options
- 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
- -formats option split into -formats, -codecs, -bsfs, and -protocols
- 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 depacketization of H.263
- Bink demuxer and audio/video decoders
- enable symbol versioning by default for linkers that support it
- IFF PBM/ILBM bitmap decoder
- concat protocol
- Indeo 5 decoder
- RTP depacketization of AMR
- WMA Voice decoder
- ffprobe tool
- AMR-NB decoder
- RTSP muxer
- HE-AAC v1 decoder
- Kega Game Video (KGV1) decoder
- VorbisComment writing for FLAC, Ogg FLAC and Ogg Speex files
- RTP depacketization of Theora
- HTTP Digest authentication
- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
- Psygnosis YOP demuxer and video decoder
- spectral extension support in the E-AC-3 decoder
- unsharp video filter
- RTP hinting in the mov/3gp/mp4 muxer
- Dirac in Ogg demuxing
- seek to keyframes in Ogg
- 4:2:2 and 4:4:4 Theora decoding
- 35% faster VP3/Theora decoding
- faster AAC decoding
- faster H.264 decoding
- RealAudio 1.0 (14.4K) encoder
- backport av_lockmgr_register(), see doc/APIchanges for details
- security fixes for:
- ASF, Ogg and MOV demuxers
- FFv1, H.264, HuffYUV, MLP, MPEG audio and Snow decoders
version 0.5:
- The "device" muxers and demuxers are now in a new libavdevice library
- DV50 AKA DVCPRO50 encoder, decoder, muxer and demuxer
- DV100 AKA DVCPRO HD decoder and demuxer
- TechSmith Camtasia (TSCC) video decoder
- IBM Ultimotion (ULTI) video decoder
- Sierra Online audio file demuxer and decoder
@@ -452,7 +141,7 @@ version 0.5:
- Fraps FPS1 video decoder
- Snow video encoder/decoder
- Sonic audio encoder/decoder
- Vorbis audio decoder
- Vorbis audio encoder/decoder
- Macromedia ADPCM decoder
- Duck TrueMotion 2 video decoder
- support for decoding FLX and DTA extensions in FLIC files
@@ -463,19 +152,18 @@ version 0.5:
- TrueSpeech audio decoder
- WMA2 audio decoder fixed, now all files should play correctly
- RealAudio 14.4 and 28.8 decoders fixed
- JPEG-LS decoder
- JPEG-LS encoder and decoder
- CamStudio video decoder
- build system improvements
- tabs and trailing whitespace removed from the codebase
- CamStudio video decoder
- AIFF/AIFF-C audio format, encoding and decoding
- ADTS AAC file reading and writing
- Creative VOC file reading and writing
- American Laser Games multimedia (*.mm) playback system
- Zip Motion Blocks Video decoder
- Zip Blocks Motion Video decoder and encoder
- improved Theora/VP3 decoder
- True Audio (TTA) decoder
- AVS demuxer and video decoder
- JPEG-LS encoder
- Smacker demuxer and decoder
- NuppelVideo/MythTV demuxer and RTjpeg decoder
- KMVC decoder
@@ -494,14 +182,12 @@ version 0.5:
- VP6 video decoder
- WavPack lossless audio decoder
- Targa (.TGA) picture decoder
- Vorbis audio encoder
- Delphine Software .cin demuxer/audio and video decoder
- Tiertex .seq demuxer/video decoder
- MTV demuxer
- TIFF picture encoder and decoder
- GIF picture decoder
- Intel Music Coder decoder
- Zip Motion Blocks Video encoder
- Musepack decoder
- Flash Screen Video encoder
- Theora encoding via libtheora
@@ -528,10 +214,10 @@ version 0.5:
- OS/2 support removed and restored again
- AC-3 decoder
- NUT muxer
- additional SPARC (VIS) optimizations
- Matroska muxer
- slice-based parallel H.264 decoding
- Monkey's Audio demuxer and decoder
- additional SPARC (VIS) optimizations
- AMV audio and video decoder
- DNxHD encoder
- H.264 PAFF decoding
@@ -540,7 +226,6 @@ version 0.5:
- libvorbis Vorbis decoding removed in favor of native decoder
- IntraX8 (J-Frame) subdecoder for WMV2 and VC-1
- Ogg (Theora, Vorbis and FLAC) muxer
- The "device" muxers and demuxers are now in a new libavdevice library
- PC Paintbrush PCX decoder
- Sun Rasterfile decoder
- TechnoTrend PVA demuxer
@@ -570,7 +255,6 @@ version 0.5:
- AAC decoder
- floating point PCM encoder/decoder
- MXF muxer
- DV100 AKA DVCPRO HD decoder and demuxer
- E-AC-3 support added to AC-3 decoder
- Nellymoser ASAO encoder
- ASS and SSA demuxer and muxer
@@ -578,9 +262,8 @@ version 0.5:
- SVQ3 watermark decoding support
- Speex decoding via libspeex
- Electronic Arts TGQ decoder
- RV40 decoder
- RV30 and RV40 decoder
- QCELP / PureVoice decoder
- RV30 decoder
- hybrid WavPack support
- R3D REDCODE demuxer
- ALSA support for playback and record
@@ -590,7 +273,6 @@ version 0.5:
- Gopher client support
- MXF D-10 muxer
- generic metadata API
- flash ScreenVideo2 encoder
version 0.4.9-pre1:
@@ -676,7 +358,6 @@ version 0.4.9-pre1:
- many optimizations and bugfixes
- FunCom ISS audio file demuxer and according ADPCM decoding
version 0.4.8:
- MPEG-2 video encoding (Michael)
@@ -687,7 +368,6 @@ version 0.4.8:
- Interplay MVE playback subsystem (Mike Melanson)
- 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
@@ -728,7 +408,6 @@ 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
@@ -765,7 +444,6 @@ version 0.4.6:
- simple stats output (Juanjo)
- 16-bit and 15-bit RGB/BGR/GBR support (Bisqwit)
version 0.4.5:
- some header fixes (Zdenek Kabelac <kabi at informatics.muni.cz>)
@@ -793,10 +471,9 @@ version 0.4.5:
- MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
- ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
- Windows porting of file converter
- added MJPEG raw format (input/output)
- added MJPEG raw format (input/ouput)
- added JPEG image format support (input/output)
version 0.4.4:
- fixed some std header definitions (Bjorn Lindgren
@@ -821,7 +498,6 @@ version 0.4.4:
- codecs are no longer searched by read_header() (should fix ffserver
segfault)
version 0.4.3:
- BGR24 patch (initial patch by Jeroen Vreeken <pe1rxq at amsat.org>)
@@ -842,7 +518,6 @@ version 0.4.3:
<lists at reserv.at>, but changed completely since we use a format
instead of a protocol)
version 0.4.2:
- added H.263/MPEG-4/MSMPEG4 decoding support. MPEG-4 decoding support
@@ -859,7 +534,6 @@ version 0.4.2:
- added file overwrite confirmation (can be disabled with -y)
- added custom size picture to H.263 using H.263+ (Juanjo)
version 0.4.1:
- added MSMPEG4 (aka DivX) compatible encoder. Changed default codec
@@ -872,7 +546,6 @@ version 0.4.1:
- improved dct_quantize speed
- factorized some motion estimation code
version 0.4.0:
- removing grab code from ffserver and moved it to ffmpeg. Added
@@ -900,12 +573,10 @@ version 0.4.0:
- changed directories: libav for format handling, libavcodec for
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
@@ -913,18 +584,15 @@ version 0.3.3:
- fixed rounding problems which caused quality problems at high
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
- added MPEG-4 codec compatible with OpenDivX. It is based on the H.263 codec
- added sound for flash format (not tested)
version 0.3: initial public release

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@@ -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.

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LICENSE
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@@ -14,18 +14,17 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are
- libpostproc
- some x86 optimizations in libswscale
- 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
There are a handful of files under other licensing terms, namely:
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
licensing details. Specifically note that you must credit the IJG in the
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
are taken from libjpeg, see the top of the files for licensing details.
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter --enable-version3 will activate this licensing option
@@ -45,6 +44,7 @@ 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 libraries libamrnb, libamrwb and libfaac can be hooked up
in FFmpeg. You need to pass --enable-nonfree to configure to enable them. Employ
this option with care as FFmpeg then becomes nonfree and unredistributable.
Note that libfaac claims to be LGPL, but is not.

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 up to 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
==============
@@ -24,11 +18,17 @@ Applications
ffmpeg:
ffmpeg.c Michael Niedermayer
ffplay:
ffplay.c Marton Balint
Video Hooks:
vhook
vhook/watermark.c Marcus Engene
vhook/ppm.c
vhook/drawtext.c
vhook/fish.c
vhook/null.c
vhook/imlib2.c
ffprobe:
ffprobe.c Stefano Sabatini
ffplay:
ffplay.c Michael Niedermayer
ffserver:
ffserver.c, ffserver.h Baptiste Coudurier
@@ -37,20 +37,19 @@ Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
QuickTime faststart:
tools/qt-faststart.c Baptiste Coudurier
qt-faststart.c Mike Melanson
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 +86,6 @@ 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:
@@ -109,31 +106,25 @@ Generic Parts:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
libpostproc/* Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
vdpau:
vdpau* Carl Eugen Hoyos
Codecs:
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aasc.c Kostya Shishkov
aac.[ch], aactab.[ch], aacdectab.h Robert Swain
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
bink.c Kostya Shishkov
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
celp_filters.* Vitor Sessak
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 +133,10 @@ Codecs:
dv.c Roman Shaposhnik
eacmv*, eaidct*, eat* Peter Ross
ffv1.c Michael Niedermayer
ffwavesynth.c Nicolas George
flac* Justin Ruggles
flacdec.c Alex Beregszaszi, Justin Ruggles
flacenc.c 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
@@ -156,32 +146,18 @@ Codecs:
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
indeo2* Kostya Shishkov
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
ivi* Kostya Shishkov
jacosub* Clément Bœsch
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
lcl*.c Roberto Togni
libgsm.c Michel Bardiaux
libdirac* David Conrad
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libutvideo* Derek Buitenhuis
libvorbis.c David Conrad
libxavs.c Stefan Gehrer
libx264.c Mans Rullgard, Jason Garrett-Glaser
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
mimic.c Ramiro Polla
mjpeg.c Michael Niedermayer
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpc* Kostya Shishkov
mpeg12.c, mpeg12data.h Michael Niedermayer
@@ -192,7 +168,6 @@ 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
@@ -208,11 +183,9 @@ Codecs:
rv3* Kostya Shishkov
rv4* Kostya Shishkov
s3tc* Ivo van Poorten
smacker.c Kostya Shishkov
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
@@ -221,41 +194,27 @@ Codecs:
truemotion2* Kostya Shishkov
truespeech.c Kostya Shishkov
tscc.c Kostya Shishkov
tta.c Alex Beregszaszi, Jaikrishnan Menon
tta.c Alex Beregszaszi
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
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
xl.c Kostya Shishkov
xvmc.c Ivan Kalvachev
zerocodec.c Derek Buitenhuis
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
libavdevice
===========
@@ -268,18 +227,6 @@ 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
===========
@@ -295,15 +242,11 @@ 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
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
ffm* Baptiste Coudurier
flac* Justin Ruggles
flic.c Mike Melanson
@@ -311,58 +254,44 @@ 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
jacosub* Clément Bœsch
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
mpc.c Kostya Shishkov
mpeg.c Michael Niedermayer
mpegenc.c Michael Niedermayer
mpegts* Baptiste Coudurier
mpegts* Mans Rullgard
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
oggdec.c, oggdec.h Mans Rullgard
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
oggparsevorbis.c Mans Rullgard
oggparseogm.c Mans Rullgard
psxstr.c Mike Melanson
pva.c Ivo van Poorten
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
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
rm.c Roberto Togni
rtp.c, rtpenc.c Luca Abeni
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
@@ -370,13 +299,9 @@ Muxers/Demuxers:
wav.c Michael Niedermayer
wc3movie.c Mike Melanson
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
Protocols:
bluray.c Petri Hintukainen
http.c Ronald S. Bultje
mms*.c Ronald S. Bultje
udp.c Luca Abeni
@@ -385,8 +310,7 @@ Operating systems / CPU architectures
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
@@ -397,41 +321,15 @@ Sparc Roman Shaposhnik
x86 Michael Niedermayer
Releases
========
GnuPG Fingerprints of maintainers and others who have svn write access
======================================================================
1.2 Michael Niedermayer
1.1 Michael Niedermayer
1.0 Michael Niedermayer
If you want to maintain an older release, please contact us
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
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

409
Makefile
View File

@@ -1,76 +1,54 @@
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 := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher
TOOLS-$(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)))
BASENAMES = ffmpeg ffplay ffprobe ffserver
ALLPROGS = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLPROGS_G = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
ALLMANPAGES = $(BASENAMES:%=%.1)
BASENAMES = ffmpeg ffplay ffserver
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-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
FFLIBS := avdevice avformat avcodec avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile
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_VHOOK) += videohook
ALL_TARGETS-$(BUILD_DOC) += documentation
$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@$(PROGSSUF)
$(STRIP) $@$(PROGSSUF)
INSTALL_TARGETS-$(CONFIG_VHOOK) += install-vhook
ifneq ($(PROGS),)
INSTALL_TARGETS-yes += install-progs install-data
INSTALL_TARGETS-$(BUILD_DOC) += install-man
endif
INSTALL_PROGS_TARGETS-$(BUILD_SHARED) = install-libs
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
tools/cws2fws$(EXESUF): ELIBS = -lz
$(PROGS): %$(EXESUF): %_g$(EXESUF)
cp -p $< $@
$(STRIP) $@
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
SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \
ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ALTIVEC-OBJS ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS MMI-OBJS \
MMX-OBJS NEON-OBJS VIS-OBJS YASM-OBJS \
OBJS TESTOBJS
SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTS
define RESET
$(1) :=
@@ -80,104 +58,303 @@ 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)
$(CC) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
OBJDIRS += tools
output_example$(EXESUF): output_example.o $(FF_DEP_LIBS)
$(CC) $(CFLAGS) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
-include $(wildcard tools/*.d)
tools/%$(EXESUF): tools/%.c
$(CC) $(CFLAGS) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
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: M=@
cmdutils.o cmdutils.d: version.h
version.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
$(Q)touch .version
alltools: $(addsuffix $(EXESUF),$(addprefix tools/, cws2fws pktdumper qt-faststart trasher))
# force version.sh to run whenever version might have changed
-include .version
VHOOKCFLAGS += $(filter-out -mdynamic-no-pic,$(CFLAGS))
ifdef PROGS
install: install-progs install-data
endif
BASEHOOKS = fish null watermark
ALLHOOKS = $(BASEHOOKS) drawtext imlib2 ppm
ALLHOOKS_SRCS = $(addprefix vhook/, $(addsuffix .c, $(ALLHOOKS)))
install: install-libs install-headers
HOOKS-$(HAVE_FORK) += ppm
HOOKS-$(HAVE_IMLIB2) += imlib2
HOOKS-$(HAVE_FREETYPE2) += drawtext
install-libs: install-libs-yes
HOOKS = $(addprefix vhook/, $(addsuffix $(SLIBSUF), $(BASEHOOKS) $(HOOKS-yes)))
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
VHOOKCFLAGS-$(HAVE_IMLIB2) += `imlib2-config --cflags`
LIBS_imlib2$(SLIBSUF) = `imlib2-config --libs`
install-progs: install-progs-yes $(PROGS)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
VHOOKCFLAGS-$(HAVE_FREETYPE2) += `freetype-config --cflags`
LIBS_drawtext$(SLIBSUF) = `freetype-config --libs`
install-data: $(DATA_FILES) $(EXAMPLES_FILES)
$(Q)mkdir -p "$(DATADIR)/examples"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
VHOOKCFLAGS += $(VHOOKCFLAGS-yes)
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
vhook/%.o vhook/%.d: CFLAGS:=$(VHOOKCFLAGS)
# vhooks compile fine without libav*, but need them nonetheless.
videohook: $(FF_DEP_LIBS) $(HOOKS)
$(eval VHOOKSHFLAGS=$(VHOOKSHFLAGS))
vhook/%$(SLIBSUF): vhook/%.o
$(CC) $(LDFLAGS) -o $@ $(VHOOKSHFLAGS) $< $(VHOOKLIBS) $(LIBS_$(@F))
VHOOK_DEPS = $(HOOKS:$(SLIBSUF)=.d)
depend dep: $(VHOOK_DEPS)
documentation: $(addprefix doc/, ffmpeg-doc.html faq.html ffserver-doc.html \
ffplay-doc.html general.html hooks.html \
$(ALLMANPAGES))
doc/%.html: doc/%.texi
texi2html -monolithic -number $<
mv $(@F) $@
doc/%.pod: doc/%-doc.texi
doc/texi2pod.pl $< $@
doc/%.1: doc/%.pod
pod2man --section=1 --center=" " --release=" " $< > $@
install: $(INSTALL_TARGETS-yes)
install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
install -d "$(BINDIR)"
install -c -m 755 $(PROGS) "$(BINDIR)"
install-data: $(DATA_FILES)
install -d "$(DATADIR)"
install -m 644 $(DATA_FILES) "$(DATADIR)"
install-man: $(MANPAGES)
install -d "$(MANDIR)/man1"
install -m 644 $(MANPAGES) "$(MANDIR)/man1"
install-vhook: videohook
install -d "$(SHLIBDIR)/vhook"
install -m 755 $(HOOKS) "$(SHLIBDIR)/vhook"
uninstall: uninstall-progs uninstall-data uninstall-man uninstall-vhook
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
rm -f $(addprefix "$(BINDIR)/", $(ALLPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
rm -rf "$(DATADIR)"
clean::
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(TOOLS)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) coverage.info
$(RM) -r coverage-html
uninstall-man:
rm -f $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
uninstall-vhook:
rm -f $(addprefix "$(SHLIBDIR)/",$(ALLHOOKS_SRCS:.c=$(SLIBSUF)))
-rmdir "$(SHLIBDIR)/vhook/"
testclean:
rm -rf tests/vsynth1 tests/vsynth2 tests/data tests/asynth1.sw tests/*~
clean:: testclean
rm -f $(ALLPROGS) $(ALLPROGS_G) output_example$(EXESUF)
rm -f doc/*.html doc/*.pod doc/*.1
rm -f $(addprefix tests/,$(addsuffix $(EXESUF),audiogen videogen rotozoom seek_test tiny_psnr))
rm -f $(addprefix tools/,$(addsuffix $(EXESUF),cws2fws pktdumper qt-faststart trasher))
rm -f vhook/*.o vhook/*~ vhook/*.so vhook/*.dylib vhook/*.dll
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .version version.h libavutil/avconfig.h
rm -f version.h config.* vhook/*.d
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
# regression tests
# 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' > $@
check: test checkheaders
coverage-html: coverage.info
$(Q)mkdir -p $@
$(Q)genhtml -o $@ $<
$(Q)touch $@
fulltest test: codectest libavtest seektest
check: all alltools checkheaders examples testprogs fate
FFMPEG_REFFILE = $(SRC_PATH)/tests/ffmpeg.regression.ref
FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
LIBAV_REFFILE = $(SRC_PATH)/tests/libav.regression.ref
ROTOZOOM_REFFILE = $(SRC_PATH)/tests/rotozoom.regression.ref
SEEK_REFFILE = $(SRC_PATH)/tests/seek.regression.ref
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/tests/Makefile
CODEC_TESTS = $(addprefix regtest-, \
mpeg \
mpeg2 \
mpeg2thread \
msmpeg4v2 \
msmpeg4 \
wmv1 \
wmv2 \
h261 \
h263 \
h263p \
mpeg4 \
huffyuv \
rc \
mpeg4adv \
mpeg4thread \
error \
mpeg4nr \
mpeg1b \
mjpeg \
ljpeg \
jpegls \
rv10 \
rv20 \
asv1 \
asv2 \
flv \
ffv1 \
snow \
snowll \
dv \
dv50 \
svq1 \
flashsv \
mp2 \
ac3 \
g726 \
adpcm_ima_wav \
adpcm_ima_qt \
adpcm_ms \
adpcm_yam \
adpcm_swf \
flac \
wma \
pcm \
)
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
LAVF_TESTS = $(addprefix regtest-, \
avi \
asf \
rm \
mpg \
ts \
swf \
ffm \
flv_fmt \
mov \
dv_fmt \
gxf \
nut \
mkv \
pbmpipe \
pgmpipe \
ppmpipe \
gif \
yuv4mpeg \
pgm \
ppm \
bmp \
tga \
tiff \
sgi \
jpg \
wav \
alaw \
mulaw \
au \
mmf \
aiff \
voc \
ogg \
pixfmt \
)
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
REGFILES = $(addprefix tests/data/,$(addsuffix .$(1),$(2:regtest-%=%)))
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
CODEC_ROTOZOOM = $(call REGFILES,rotozoom.regression,$(CODEC_TESTS))
CODEC_VSYNTH = $(call REGFILES,vsynth.regression,$(CODEC_TESTS))
.PHONY: all all-yes alltools check *clean config install*
.PHONY: testprogs uninstall*
LAVF_REGFILES = $(call REGFILES,lavf.regression,$(LAVF_TESTS))
LAVF_REG = tests/data/lavf.regression
ROTOZOOM_REG = tests/data/rotozoom.regression
VSYNTH_REG = tests/data/vsynth.regression
ifneq ($(CONFIG_SWSCALE),yes)
servertest codectest $(CODEC_TESTS) libavtest: swscale-error
swscale-error:
@echo
@echo "This regression test requires --enable-swscale."
@echo
@exit 1
endif
ifneq ($(CONFIG_ZLIB),yes)
regtest-flashsv codectest: zlib-error
endif
zlib-error:
@echo
@echo "This regression test requires zlib."
@echo
@exit 1
codectest: $(VSYNTH_REG) $(ROTOZOOM_REG)
diff -u -w $(FFMPEG_REFFILE) $(VSYNTH_REG)
diff -u -w $(ROTOZOOM_REFFILE) $(ROTOZOOM_REG)
libavtest: $(LAVF_REG)
diff -u -w $(LIBAV_REFFILE) $(LAVF_REG)
$(VSYNTH_REG) $(ROTOZOOM_REG) $(LAVF_REG):
cat $^ > $@
$(LAVF_REG): $(LAVF_REGFILES)
$(ROTOZOOM_REG): $(CODEC_ROTOZOOM)
$(VSYNTH_REG): $(CODEC_VSYNTH)
$(CODEC_VSYNTH) $(CODEC_ROTOZOOM): $(CODEC_TESTS)
$(LAVF_REGFILES): $(LAVF_TESTS)
$(CODEC_TESTS) $(LAVF_TESTS): regtest-ref
regtest-ref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm tests/asynth1.sw
$(CODEC_TESTS) regtest-ref: tests/tiny_psnr$(EXESUF)
$(SRC_PATH)/tests/regression.sh $@ vsynth tests/vsynth1 a "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(SRC_PATH)/tests/regression.sh $@ rotozoom tests/vsynth2 a "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(LAVF_TESTS):
$(SRC_PATH)/tests/regression.sh $@ lavf tests/vsynth1 b "$(TARGET_EXEC)" "$(TARGET_PATH)"
seektest: codectest libavtest tests/seek_test$(EXESUF)
$(SRC_PATH)/tests/seek_test.sh $(SEEK_REFFILE) "$(TARGET_EXEC)" "$(TARGET_PATH)"
servertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/asynth1.sw
@echo
@echo "Unfortunately ffserver is broken and therefore its regression"
@echo "test fails randomly. Treat the results accordingly."
@echo
$(SRC_PATH)/tests/server-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/test.conf
tests/vsynth1/00.pgm: tests/videogen$(EXESUF)
mkdir -p tests/vsynth1
$(BUILD_ROOT)/$< 'tests/vsynth1/'
tests/vsynth2/00.pgm: tests/rotozoom$(EXESUF)
mkdir -p tests/vsynth2
$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
tests/asynth1.sw: tests/audiogen$(EXESUF)
$(BUILD_ROOT)/$< $@
tests/%$(EXESUF): tests/%.c
$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
tests/seek_test$(EXESUF): tests/seek_test.c $(FF_DEP_LIBS)
$(CC) $(FF_LDFLAGS) $(CFLAGS) -o $@ $< $(FF_EXTRALIBS)
.PHONY: lib videohook documentation *test regtest-* swscale-error zlib-error alltools check
-include $(VHOOK_DEPS)

8
README
View File

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

235
RELEASE
View File

@@ -1 +1,234 @@
0.11.5
Release Notes
=============
* 0.5 "Bike Shed World Domination" March 3, 2009
General notes
-------------
It has been so long since the last release that this should be considered the
first FFmpeg release of recent times. Because of the way things have unfolded to
date, the notes for this version cannot be entirely conventional.
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 notes
---------
In the next release, it is intended to remove a number of deprecated APIs. We
decided to put out a release that includes said APIs for the benefit of third
party software.
As such, this release:
- provides a sync point for said APIs
- increases awareness of API changes
- allows the next release to detail how to transition from the old to the new
The deprecated APIs to be removed are:
- imgconvert (to be replaced by libswscale)
- vhook (to be replaced by libavfilter)
If at all possible, do not use the deprecated APIs. All notes on API changes
should appear in doc/APIchanges.
* 0.5.1 March 2, 2010
General notes
-------------
This point release includes some minor updates to make the 0.5 release series
usable for users that need to retain the existing behavior as closely as
possible. The changes follow below:
Security fixes
--------------
Various 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:
- the Vorbis audio codec
- the FF Video 1 codec
- the MPEG audio codec
- the H264 video codec
- the MLP codec
- the HuffYUV codec
- the ASF demuxer
- the Ogg container implementation
- the MOV container implementation
Symbol Versioning enabled
-------------------------
The backported symbol versioning change is enabled on platforms that support
it. This allows users to upgrade from 0.5.1 to the upcoming 0.6 release
without having to recompile their applications. Please note that distributors
have to recompile applications against 0.5.1 before upgrading to 0.6.
libx264.c backport
------------------
This release includes a backport to the libx264 wrapper that allows FFmpeg to
be compiled against newer versions of libx264 up to API version 85.
licensing changes
-----------------
Previously both libswscale and our AC-3 decoder had GPLed parts. These have
been replaced by fresh LGPL code. x86 optimizations for libswscale remain GPL,
but the C code is fully functional. Optimizations for other architectures have
been relicensed to LGPL.
AMR-NB decoding/encoding and AMR-WB decoding is now possible through the free
software OpenCORE libraries as an alternative to the non-free libamr libraries.
We found out that libfaac contains non-free parts and is not LGPL as previously
claimed. We have changed configure to reflect this. You now have to pass the
--enable-nonfree option if you wish to compile with libfaac support enabled.
Furthermore the non-free bits in libavcodec/fdctref.c have been rewritten. Note
well that they were only used in a test program and never compiled into any
FFmpeg library.
* 0.5.2 May 25, 2010
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.
* 0.5.3 Oct 18, 2010
General notes
-------------
This is (again) another maintenance-only release that addresses a fix
for seekable HTTP and an exploitable bug in the FLIC decoder
(cf. CVE-2010-3429 for details). Distributors and system integrators are
encouraged to update and share their patches against this branch.
* 0.5.4 Mar 17, 2011
General notes
-------------
This is the first release that we cut after git migration. It is another
maintenance-only release that addresses several security issues that were
brought to our attention. In detail, fixes for RV30/40, WMV, Vorbis and
VC-1 have been backported from trunk. Distributors and system integrators
are encouraged to update and share their patches against this branch.
* 0.5.5 Nov 6, 2011
General notes
-------------
This maintenance-only release addresses several security issues that
were brought to our attention. In detail, fixes for the MJPEG decoder,
the CAVS decoder (CVE-2011-3362, CVE-2011-3973, CVE-2011-3974), and the
Matroska decoder (MSVR11-011/CVE-2011-3504) and many others have been
corrected. Additional, this release contains fixes for compilation with
gcc-4.6. Distributors and system integrators are encouraged to update
and share their patches against this branch.
* 0.5.6 Nov 21, 2011
General notes
-------------
This maintenance-only release addresses several security issues that
were brought to our attention.
* 0.5.7 Dec 25, 2011
General notes
-------------
This maintenance-only release addresses several security issues that
were brought to our attention. In details, it features fixes for the
QDM2 decoder (CVE-2011-4351), DoS in the VP5/VP6 decoders
(CVE-2011-4353), and a buffer overflow in the Sierra VMD decoder
CVE-2011-4364, and a safety fix in the SVQ1 decoder (CVE-2011-4579).
CVE-2011-4352, a bug in the VP3 decoder, is not known to affect this
release.
Distributors and system integrators are encouraged to update and share
their patches against this branch.
* 0.5.8 Jan 12, 2012
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.
* 0.5.9 May 11, 2012
General notes
-------------
This maintenance-only release that addresses a number a number of
security issues that have been brought to our attention. Among other
(rather minor) fixes, this release features fixes for the DV decoder
(CVE-2011-3929 and CVE-2011-3936), nsvdec (CVE-2011-3940), Atrac3
(CVE-2012-0853), mjpegdec (CVE-2011-3947) and the VQA video decoder
(CVE-2012-0947).
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.
* 0.5.10 Jun 09, 2012
General notes
-------------
This mostly maintenance-only release addresses a number a number of bugs
such as security and compilation issues that have been brought to our
attention. Among other fixes, this release features includes security
updates for the DPCM codecs (CVE-2011-3951), H.264 (CVE-2012-0851),
ADPCM (CVE-2012-0852), and the KMVC decoder (CVE-2011-3952).
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 or the git commit history.

View File

@@ -1 +1 @@
0.11.5
0.5.10

View File

@@ -1,13 +0,0 @@
OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
OBJS-$(HAVE_ARMV6) += $(ARMV6-OBJS) $(ARMV6-OBJS-yes)
OBJS-$(HAVE_ARMVFP) += $(ARMVFP-OBJS) $(ARMVFP-OBJS-yes)
OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_MMI) += $(MMI-OBJS) $(MMI-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_VIS) += $(VIS-OBJS) $(VIS-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

1271
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().
*/
@@ -43,33 +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 AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern const char **opt_names;
extern AVCodecContext *avctx_opts[CODEC_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern struct SwsContext *sws_opts;
extern struct SwrContext *swr_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
@@ -78,66 +51,35 @@ 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.
*/
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_cpuflags(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
* @param is_duration a flag which tells how to interpret \p timestr, if
* not zero \p timestr is interpreted as a duration, otherwise as a
* date
*
* @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;
@@ -152,238 +94,62 @@ 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*));
/**
* 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);
void set_context_opts(void *ctx, void *opts_ctx, int flags);
/**
* Find the '-loglevel' option in the command line args and apply it.
*/
void parse_loglevel(int argc, char **argv, const OptionDef *options);
/**
* Return index of option opt in argv or 0 if not found.
*/
int locate_option(int argc, char **argv, const OptionDef *options,
const char *optname);
/**
* Check if the given stream matches a stream specifier.
*
* @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
* readable description of the error code err.
*
* If strerror_r() is not available the use of this function in a
* multithreaded application may be unsafe.
*
* @see av_strerror()
*/
void print_error(const char *filename, int err);
/**
* 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.
*/
int read_yesno(void);
/**
* Read the file with name filename, and put 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
* @return 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
/**
* Get a file corresponding to a preset file.
*
* 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.avpreset 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

@@ -1,20 +0,0 @@
{ "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" },
{ "fdebug", 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" },
{ "cpuflags", HAS_ARG | OPT_EXPERT, {(void*)opt_cpuflags}, "force specific cpu flags", "flags" },

View File

@@ -2,119 +2,116 @@
# common bits used by all libraries
#
# first so "all" becomes default target
all: all-yes
all: # make "all" default target
ifndef SUBDIR
vpath %.c $(SRC_DIR)
vpath %.h $(SRC_DIR)
vpath %.S $(SRC_DIR)
vpath %.asm $(SRC_DIR)
vpath %.v $(SRC_DIR)
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX 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))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
BUILD_ROOT_REL = .
else
BUILD_ROOT_REL = ..
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CFLAGS)
CXXFLAGS := $(CFLAGS) $(CXXFLAGS)
YASMFLAGS += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
HOSTCFLAGS += $(IFLAGS)
LDFLAGS := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
define COMPILE
$($(1)DEP)
$($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
CFLAGS := -DHAVE_AV_CONFIG_H -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE \
-I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(OPTFLAGS)
%.o: %.c
$(COMPILE_C)
%.o: %.cpp
$(COMPILE_CXX)
%.s: %.c
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
$(CC) $(CFLAGS) $(LIBOBJFLAGS) -c -o $@ $<
%.o: %.S
$(COMPILE_S)
$(CC) $(CFLAGS) $(LIBOBJFLAGS) -c -o $@ $<
%.ho: %.h
$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
$(CC) $(CFLAGS) $(LIBOBJFLAGS) -Wno-unused -c -o $@ -x c $<
%.d: %.c
$(DEPEND_CMD) > $@
%.d: %.S
$(DEPEND_CMD) > $@
%.d: %.cpp
$(DEPEND_CMD) > $@
%.o: %.d
%$(EXESUF): %.c
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
# Do not delete intermediate files from chains of implicit rules
$(OBJS):
SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
ifeq ($(wildcard $(SVN_ENTRIES)),$(SVN_ENTRIES))
$(BUILD_ROOT_REL)/version.h: $(SVN_ENTRIES)
endif
include $(SRC_PATH)/arch.mak
$(BUILD_ROOT_REL)/version.h: $(SRC_PATH_BARE)/version.sh
$< $(SRC_PATH) $@ $(EXTRA_VERSION)
OBJS += $(OBJS-yes)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
install: install-libs install-headers
FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(EXTRALIBS)
uninstall: uninstall-libs uninstall-headers
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))
.PHONY: all depend dep clean distclean install* uninstall* tests
endif
DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
CFLAGS += $(CFLAGS-yes)
OBJS += $(OBJS-yes)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTS += $(TESTS-yes)
FFEXTRALIBS := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
FFLDFLAGS := $(addprefix -L$(BUILD_ROOT)/lib,$(FFLIBS)) $(LDFLAGS)
OBJS := $(addprefix $(SUBDIR),$(OBJS))
TESTS := $(addprefix $(SUBDIR),$(TESTS))
DEP_LIBS:=$(foreach NAME,$(FFLIBS),lib$(NAME)/$($(BUILD_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)%)
checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
checkheaders: $(filter-out %_template.ho,$(ALLHEADERS:.h=.ho))
alltools: $(TOOLS)
DEPS := $(OBJS:.o=.d)
depend dep: $(DEPS)
$(HOSTOBJS): %.o: %.c
$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
CLEANSUFFIXES = *.o *~ *.ho *.ver
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp *.map
DISTCLEANSUFFIXES = *.d *.pc
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
define RULES
$(SUBDIR)%$(EXESUF): $(SUBDIR)%.o
$(CC) $(FFLDFLAGS) -o $$@ $$^ $(SUBDIR)$(LIBNAME) $(FFEXTRALIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
$(SUBDIR)%-test.o: $(SUBDIR)%.c
$(CC) $(CFLAGS) -DTEST -c -o $$@ $$^
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
$(SUBDIR)%-test.o: $(SUBDIR)%-test.c
$(CC) $(CFLAGS) -DTEST -c -o $$@ $$^
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
$(SUBDIR)x86/%.o: $(SUBDIR)x86/%.asm
$(YASM) $(YASMFLAGS) -I $$(<D)/ -o $$@ $$<
-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))
$(SUBDIR)x86/%.d: $(SUBDIR)x86/%.asm
$(YASM) $(YASMFLAGS) -I $$(<D)/ -M -o $$(@:%.d=%.o) $$< > $$@
clean::
rm -f $(TESTS) $(addprefix $(SUBDIR),$(CLEANFILES) $(CLEANSUFFIXES) $(LIBSUFFIXES)) \
$(addprefix $(SUBDIR), $(foreach suffix,$(CLEANSUFFIXES),$(addsuffix /$(suffix),$(DIRS))))
distclean:: clean
rm -f $(addprefix $(SUBDIR),$(DISTCLEANSUFFIXES)) \
$(addprefix $(SUBDIR), $(foreach suffix,$(DISTCLEANSUFFIXES),$(addsuffix /$(suffix),$(DIRS))))
endef
$(eval $(RULES))
tests: $(TESTS)
-include $(DEPS)

3662
configure vendored

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@@ -1,73 +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 \
doc/syntax.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/print_options.o: libavformat/options_table.h libavcodec/options_table.h
GENTEXI = format codec
GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
$(GENTEXI): TAG = GENTEXI
$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)$(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
$(DOCS): | doc/
install-progs-$(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/%) doc/avoptions_*.texi
-include $(wildcard $(DOCS:%=%.d))
.PHONY: documentation

View File

@@ -1,16 +0,0 @@
Release Notes
=============
* 0.11 "Happiness" May, 2012
General notes
-------------
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.

90
doc/TODO Normal file
View File

@@ -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

View File

@@ -1,182 +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.
@item #@var{stream_id}
Matches the stream by format-specific ID.
@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.
@item -cpuflags flags (@emph{global})
Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you're doing.
@example
ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
@end example
@end table
@section AVOptions
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.
@include avoptions_codec.texi
@include avoptions_format.texi

View File

@@ -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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@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 -bsf:v 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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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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@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.
@section libcelt
libcelt decoder wrapper
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libcelt}.
@section libgsm
libgsm decoder wrapper
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
the presence of the libgsm headers and library during configuration. You need
to explicitly configure the build with @code{--enable-libgsm}.
This decoder supports both the ordinary GSM and the Microsoft variant.
@section libilbc
libilbc decoder wrapper
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
audio codec. Requires the presence of the libilbc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libilbc}.
@subsection Options
The following option is supported by the libilbc wrapper.
@table @option
@item enhance
Enable the enhancement of the decoded audio when set to 1. The default
value is 0 (disabled).
@end table
@section libopencore-amrnb
libopencore-amrnb decoder wrapper
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrnb}.
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
without this library.
@section libopencore-amrwb
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrwb}.
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
without this library.
@c man end AUDIO DECODERS

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@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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\input texinfo @c -*- texinfo -*-
@settitle Developer Documentation
@titlepage
@center @titlefont{Developer Documentation}
@end titlepage
@top
@contents
@chapter Developers Guide
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
it.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{doc/examples/muxing.c} to use it to generate audio or video
streams.
@end itemize
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@section Contributing
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:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
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
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
* MPEG codec.
* @@author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
int var3;
@} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @@param my_parameter description of my_parameter
* @@return return value description
*/
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
set cindent
set cinoptions=(0
" allow tabs in Makefiles
autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode nil)
(show-trailing-whitespace t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end example
@section Development Policy
@enumerate
@item
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder).
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@anchor{Submitting patches}
@section Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
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.
Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch will not be trashed during
transmission) to the ffmpeg-devel mailing list, see
@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@section New codecs or formats checklist
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@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},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
@item
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)?
@end enumerate
@section patch submission checklist
@enumerate
@item
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
@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.
@item
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
@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
@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.
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
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
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
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.
[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
accordingly].
@bye

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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
@section libmp3lame
LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper
Requires the presence of the libmp3lame headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libmp3lame}.
@subsection Option Mapping
The following options are supported by the libmp3lame wrapper,
the LAME-equivalent options follow the FFmpeg ones.
@multitable @columnfractions .2 .2
@item FFmpeg @tab LAME
@item b @tab b
FFmpeg @code{b} option is expressed in bits/s, lame @code{bitrate}
in kilobits/s.
@item q @tab V
Quality setting for VBR.
@item compression_level @tab q
Algorithm quality. Valid options are integers from 0-9.
@item reservoir @tab N.A.
Enable use of bit reservoir. LAME has this enabled by default.
@item joint_stereo @tab -m j
Enables the the encoder to use (on a frame by frame basis) either L/R
stereo or mid/side stereo.
@end multitable
@section libopencore-amrnb
OpenCORE Adaptive Multi-Rate Narrowband encoder.
Requires the presence of the libopencore-amrnb headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopencore-amrnb --enable-version3}.
This is a mono-only encoder. Officially it only supports 8000Hz sample rate,
but you can override it by setting @option{strict} to @samp{unofficial} or
lower.
@subsection Options
@table @option
@item b
Set bitrate in bits per second. Only the following bitrates are supported,
otherwise libavcodec will round to the nearest valid bitrate.
@table @option
@item 4750
@item 5150
@item 5900
@item 6700
@item 7400
@item 7950
@item 10200
@item 12200
@end table
@item dtx
Allow discontinuous transmission (generate comfort noise) when set to 1. The
default value is 0 (disabled).
@end table
@section libvo-aacenc
VisualOn AAC encoder
Requires the presence of the libvo-aacenc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libvo-aacenc --enable-version3}.
@subsection Options
The VisualOn AAC encoder only support encoding AAC-LC and up to 2
channels. It is also CBR-only. It is considered to be worse than the
native experimental FFmpeg AAC encoder.
@table @option
@item b
Bitrate.
@end table
@section libvo-amrwbenc
VisualOn Adaptive Multi-Rate Wideband encoder
Requires the presence of the libvo-amrwbenc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libvo-amrwbenc --enable-version3}.
This is a mono-only encoder. Officially it only supports 16000Hz sample
rate, but you can override it by setting @option{strict} to
@samp{unofficial} or lower.
@subsection Options
@table @option
@item b
Set bitrate in bits/s. Only the following bitrates are supported, otherwise
libavcodec will round to the nearest valid bitrate.
@table @samp
@item 6600
@item 8850
@item 12650
@item 14250
@item 15850
@item 18250
@item 19850
@item 23050
@item 23850
@end table
@item dtx
Allow discontinuous transmission (generate comfort noise) when set to 1. The
default value is 0 (disabled).
@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}
@section libxvid
Xvid MPEG-4 Part 2 encoder wrapper.
This encoder requires the presence of the libxvidcore headers and library
during configuration. You need to explicitly configure the build with
@code{--enable-libxvid --enable-gpl}.
The native @code{mpeg4} encoder supports the MPEG-4 Part 2 format, so
users can encode to this format without this library.
@subsection Options
The following options are supported by the libxvid wrapper. Some of
the following options are listed but are not documented, and
correspond to shared codec options. The other shared options
which are not listed have no effect for the libxvid encoder.
@table @option
@item b
@item g
@item qmin
@item qmax
@item mpeg_quant
@item threads
@item bf
@item b_qfactor
@item b_qoffset
@item flags
Set specific encoding flags. Possible values:
@table @samp
@item mv4
Use four motion vector by macroblock.
@item aic
Enable high quality AC prediction.
@item gray
Only encode grayscale.
@item gmc
Enable the use of global motion compensation (GMC).
@item qpel
Enable quarter-pixel motion compensation.
@item cgop
Enable closed GOP.
@item global_header
Place global headers in extradata instead of every keyframe.
@end table
@item trellis
@item me_method
Set motion estimation method. Possible values in decreasing order of
speed and increasing order of quality:
@table @samp
@item zero
Use no motion estimation (default).
@item phods
@item x1
@item log
Enable advanced diamond zonal search for 16x16 blocks and half-pixel
refinement for 16x16 blocks. @samp{x1} and @samp{log} are aliases for
@samp{phods}.
@item epzs
Enable all of the things described above, plus advanced diamond zonal
search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion
estimation on chroma planes.
@item full
Enable all of the things described above, plus extended 16x16 and 8x8
blocks search.
@end table
@item mbd
Set macroblock decision algorithm. Possible values in the increasing
order of quality:
@table @samp
@item simple
Use macroblock comparing function algorithm (default).
@item bits
Enable rate distortion-based half pixel and quarter pixel refinement for
16x16 blocks.
@item rd
Enable all of the things described above, plus rate distortion-based
half pixel and quarter pixel refinement for 8x8 blocks, and rate
distortion-based search using square pattern.
@end table
@end table
@c man end VIDEO ENCODERS

View File

@@ -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

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@@ -1,204 +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.
@item taylor(expr, x) taylor(expr, x, id)
Evaluate a taylor series at x.
expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
then 0 is assumed.
note, when you have the derivatives at y instead of 0
taylor(expr, x-y) can be used
When the series does not converge the results are undefined.
@item root(expr, max)
Finds x where f(x)=0 in the interval 0..max.
f() must be continuous or the result is undefined.
@end table
The following constants are available:
@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

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@@ -1,32 +0,0 @@
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libavresample \
libswresample \
libswscale \
libavutil \
CFLAGS += -Wall -O2 -g
CFLAGS += $(shell pkg-config --cflags $(FFMPEG_LIBS))
LDLIBS += $(shell pkg-config --libs $(FFMPEG_LIBS))
EXAMPLES= decoding_encoding \
filtering_video \
filtering_audio \
metadata \
muxing \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
.phony: all clean
all: $(OBJS) $(EXAMPLES)
clean:
rm -rf $(EXAMPLES) $(OBJS)

View File

@@ -1,235 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
const char *filter_descr = "aresample=8000,aconvert=s16:mono";
const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
const int64_t *chlayouts = avfilter_all_channel_layouts;
AVABufferSinkParams *abuffersink_params;
const AVFilterLink *outlink;
filter_graph = avfilter_graph_alloc();
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64,
dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
return ret;
}
/* buffer audio sink: to terminate the filter chain. */
abuffersink_params = av_abuffersink_params_alloc();
abuffersink_params->sample_fmts = sample_fmts;
abuffersink_params->channel_layouts = chlayouts;
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, abuffersink_params, filter_graph);
av_free(abuffersink_params);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
return 0;
}
static void print_samplesref(AVFilterBufferRef *samplesref)
{
const AVFilterBufferRefAudioProps *props = samplesref->audio;
const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
const uint16_t *p = (uint16_t*)samplesref->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame frame;
int got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
AVFilterBufferRef *samplesref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
av_free_packet(&packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
if (samplesref) {
print_samplesref(samplesref);
avfilter_unref_buffer(samplesref);
}
}
}
}
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

View File

@@ -1,231 +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 <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.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;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the 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_open2(dec_ctx, dec, NULL)) < 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, filters_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
return 0;
}
static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
{
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) {
frame.pts = av_frame_get_best_effort_timestamp(&frame);
/* push the decoded frame into the filtergraph */
av_vsrc_buffer_add_frame(buffersrc_ctx, &frame, 0);
/* pull filtered pictures from the filtergraph */
while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
av_buffersink_get_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);
avformat_close_input(&fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

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_close_input(&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?
@@ -101,25 +138,6 @@ Notice that @samp{%d} is replaced by the image number.
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
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"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
@section How do I encode movie to single pictures?
@@ -135,15 +153,15 @@ 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 +176,87 @@ quite logical that there is a small reduction of quality. This is not a bug.
Use @file{-} as file name.
@section Why does FFmpeg not decode audio in VOB files?
The audio is AC-3 (a.k.a. A/52). AC-3 decoding is an optional component in FFmpeg
as the component that handles AC-3 decoding is currently released under the GPL.
Enable AC-3 decoding with @code{./configure --enable-gpl}. Take care: By
enabling AC-3, you automatically change the license of libavcodec from
LGPL to GPL.
@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 -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 -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 -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 +275,19 @@ 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 +300,13 @@ 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.
@@ -238,18 +316,14 @@ 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
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
must use command grouping, @code{@{ ;@}}, to background properly.
For example, let's say we want to join two FLV files into an output.flv file:
@@ -263,52 +337,32 @@ mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
ffmpeg -i input2.flv -an -f yuv4mpegpipe - > temp2.v < /dev/null &
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 +410,34 @@ 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 +445,23 @@ 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 +469,7 @@ 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 +477,12 @@ 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

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@@ -1,176 +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.
@item CPUFLAGS
Specify CPU flags.
@end table
Example:
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
@end example

969
doc/ffmpeg-doc.texi Normal file
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@@ -0,0 +1,969 @@
\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
@section Main options
@table @option
@item -L
Show license.
@item -h
Show help.
@item -version
Show version.
@item -formats
Show available formats, codecs, bitstream filters, protocols, and frame size and frame rate abbreviations.
The fields preceding the format and 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 -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 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.
@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 (only available when compiled with swscale support).
@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 -vhook @var{module}
Insert video processing @var{module}. @var{module} contains the module
name and its parameters separated by spaces.
@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
Add timings for benchmarking.
@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 packet size in bits.
@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. Video will be stretched/squeezed to match the timestamps,
it is done by duplicating and dropping frames. 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} 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 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{$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}.
Finally, if the above rules failed and the argument specifies an
absolute pathname, ffmpeg will search for that filename. This way you
can specify the absolute and complete filename of the preset file, for
example @file{./ffpresets/libx264-max.ffpreset}.
@node 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 -formats' 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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@@ -1,47 +0,0 @@
:
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-:--- | : :
OutputFile 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

@@ -1,102 +1,76 @@
\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
@section Main options
@table @option
@item -h
Show help.
@item -version
Show version.
@item -L
Show license.
@item -formats
Show available formats, codecs, protocols, ...
@item -x @var{width}
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
Disable video.
@item -ss @var{pos}
Seek to a given position in seconds.
@item -t @var{duration}
play <duration> seconds of audio/video
@item -bytes
Seek by bytes.
@item -nodisp
Disable graphical display.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
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 +102,6 @@ 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 +134,6 @@ 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.
@@ -178,27 +141,17 @@ Seek to percentage in file corresponding to fraction of width.
@c man end
@include syntax.texi
@include eval.texi
@include decoders.texi
@include demuxers.texi
@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

View File

@@ -1,372 +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_format_entry @var{name}
Like @option{-show_format}, but only prints the specified entry of the
container format information, rather than all. This option may be given more
than once, then all specified entries will be shown.
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
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 -count_frames
Count the number of frames per stream and report it in the
corresponding stream section.
@item -count_packets
Count the number of packets per stream and report it in the
corresponding stream section.
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
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:".
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
A description of the accepted options follows.
@table @option
@item nokey, nk
If set to 1 specify not to print the key of each field. Default value
is 0.
@item noprint_wrappers, nw
If set to 1 specify not to print the section header and footer.
Default value is 0.
@end table
@section compact
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.
An updated version of the schema can be retrieved at the url
@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
latest schema committed into the FFmpeg development source code tree.
Note that the output issued will be compliant to the
@file{ffprobe.xsd} schema only when no special global output options
(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@option{sexagesimal} etc.) are specified.
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@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/}.
@c man end WRITERS
@chapter Timecode
@c man begin TIMECODE
@command{ffprobe} supports Timecode extraction:
@itemize
@item
MPEG1/2 timecode is extracted from the GOP, and is available in the video
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 TIMECODE
@include syntax.texi
@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

@@ -1,167 +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="bit_rate" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<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.
@@ -238,20 +226,31 @@ 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
@section Main options
@table @option
@item -version
Show version.
@item -L
Show license.
@item -formats
Show available formats, codecs, protocols, ...
@item -h
Show help.
@item -f @var{configfile}
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 +261,15 @@ 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
#
@@ -373,3 +352,5 @@ ACL allow 192.168.0.0 192.168.255.255
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>

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@@ -1,116 +0,0 @@
Filter design
=============
This document explains guidelines that should be observed (or ignored with
good reason) when writing filters for libavfilter.
In this document, the word “frame” indicates either a video frame or a group
of audio samples, as stored in an AVFilterBuffer structure.
Format negotiation
==================
The query_formats method should set, for each input and each output links,
the list supported formats.
For video links, that means pixel format. For audio links, that means
channel layout, and sample format (the sample packing is implied by the
sample format).
The lists are not just lists, they are references to shared objects. When
the negotiation mechanism computes the intersection of the formats
supported at each ends of a link, all references to both lists are
replaced with a reference to the intersection. And when a single format is
eventually chosen for a link amongst the remaining list, again, all
references to the list are updated.
That means that if a filter requires that its input and output have the
same format amongst a supported list, all it have to do is use a reference
to the same list of formats.
Buffer references ownership and permissions
===========================================
TODO
Frame scheduling
================
The purpose of these rules is to ensure that frames flow in the filter
graph without getting stuck and accumulating somewhere.
Simple filters that output one frame for each input frame should not have
to worry about it.
start_frame / filter_samples
----------------------------
These methods are called when a frame is pushed to the filter's input.
They can be called at any time except in a reentrant way.
If the input frame is enough to produce output, then the filter should
push the output frames on the output link immediately.
As an exception to the previous rule, if the input frame is enough to
produce several output frames, then the filter needs output only at
least one per link. The additional frames can be left buffered in the
filter; these buffered frames must be flushed immediately if a new input
produces new output.
(Example: framerate-doubling filter: start_frame must (1) flush the
second copy of the previous frame, if it is still there, (2) push the
first copy of the incoming frame, (3) keep the second copy for later.)
If the input frame is not enough to produce output, the filter must not
call request_frame to get more. It must just process the frame or queue
it. The task of requesting more frames is left to the filter's
request_frame method or the application.
If a filter has several inputs, the filter must be ready for frames
arriving randomly on any input. Therefore, any filter with several input
will most likely require some kind of queuing mechanism. It is perfectly
acceptable to have a limited queue and to drop frames when the inputs
are too unbalanced.
request_frame
-------------
This method is called when a frame is wanted on an output.
For an input, it should directly call start_frame or filter_samples on
the corresponding output.
For a filter, if there are queued frames already ready, one of these
frames should be pushed. If not, the filter should request a frame on
one of its input, repeatedly until at least one frame has been pushed.
Return values:
if request_frame could produce a frame, it should return 0;
if it could not for temporary reasons, it should return AVERROR(EAGAIN);
if it could not because there are no more frames, it should return
AVERROR_EOF.
The typical implementation of request_frame for a filter with several
inputs will look like that:
if (frames_queued) {
push_one_frame();
return 0;
}
while (!frame_pushed) {
input = input_where_a_frame_is_most_needed();
ret = avfilter_request_frame(input);
if (ret == AVERROR_EOF) {
process_eof_on_input();
} else if (ret < 0) {
return ret;
}
}
return 0;
Note that, except for filters that can have queued frames, request_frame
does not push frames: it requests them to its input, and as a reaction,
the start_frame / filter_samples method will be called and do the work.

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@@ -1,352 +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.
Make sure that you do not have Windows line endings in your checkouts,
otherwise you may experience spurious compilation failures. One way to
achieve this is to run
@example
git config --global core.autocrlf false
@end example
@section Updating the source tree to the latest revision
@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.

299
doc/hooks.texi Normal file
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\input texinfo @c -*- texinfo -*-
@settitle Video Hook Documentation
@titlepage
@sp 7
@center @titlefont{Video Hook Documentation}
@sp 3
@end titlepage
@chapter Introduction
@var{Please be aware that vhook is deprecated, and hence its development is
frozen (bug fixes are still accepted).
The substitute will be 'libavfilter', the result of our 'Video Filter API'
Google Summer of Code project. You may monitor its progress by subscribing to
the ffmpeg-soc mailing list at
@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc}.}
The video hook functionality is designed (mostly) for live video. It allows
the video to be modified or examined between the decoder and the encoder.
Any number of hook modules can be placed inline, and they are run in the
order that they were specified on the ffmpeg command line.
The video hook modules are provided for use as a base for your own modules,
and are described below.
Modules are loaded using the -vhook option to ffmpeg. The value of this parameter
is a space separated list of arguments. The first is the module name, and the rest
are passed as arguments to the Configure function of the module.
The modules are dynamic libraries: They have different suffixes (.so, .dll, .dylib)
depending on your platform. And your platform dictates if they need to be
somewhere in your PATH, or in your LD_LIBRARY_PATH. Otherwise you will need to
specify the full path of the vhook file that you are using.
@section null.c
This does nothing. Actually it converts the input image to RGB24 and then converts
it back again. This is meant as a sample that you can use to test your setup.
@section fish.c
This implements a 'fish detector'. Essentially it converts the image into HSV
space and tests whether more than a certain percentage of the pixels fall into
a specific HSV cuboid. If so, then the image is saved into a file for processing
by other bits of code.
Why use HSV? It turns out that HSV cuboids represent a more compact range of
colors than would an RGB cuboid.
@section imlib2.c
This module implements a text overlay for a video image. Currently it
supports a fixed overlay or reading the text from a file. The string
is passed through strftime() so that it is easy to imprint the date and
time onto the image.
This module depends on the external library imlib2, available on
Sourceforge, among other places, if it is not already installed on
your system.
You may also overlay an image (even semi-transparent) like TV stations do.
You may move either the text or the image around your video to create
scrolling credits, for example.
The font file used is looked for in a FONTPATH environment variable, and
prepended to the point size as a command line option and can be specified
with the full path to the font file, as in:
@example
-F /usr/X11R6/lib/X11/fonts/TTF/VeraBd.ttf/20
@end example
where 20 is the point size.
You can specify the filename to read RGB color names from. If it is not
specified, these defaults are used: @file{/usr/share/X11/rgb.txt} and
@file{/usr/lib/X11/rgb.txt}
Options:
@multitable @columnfractions .2 .8
@item @option{-C <rgb.txt>} @tab The filename to read RGB color names from
@item @option{-c <color>} @tab The color of the text
@item @option{-F <fontname>} @tab The font face and size
@item @option{-t <text>} @tab The text
@item @option{-f <filename>} @tab The filename to read text from
@item @option{-x <expression>}@tab x coordinate of text or image
@item @option{-y <expression>}@tab y coordinate of text or image
@item @option{-i <filename>} @tab The filename to read a image from
@item @option{-R <expression>}@tab Value for R color
@item @option{-G <expression>}@tab Value for G color
@item @option{-B <expression>}@tab Value for B color
@item @option{-A <expression>}@tab Value for Alpha channel
@end multitable
Expressions are functions of these variables:
@multitable @columnfractions .2 .8
@item @var{N} @tab frame number (starting at zero)
@item @var{H} @tab frame height
@item @var{W} @tab frame width
@item @var{h} @tab image height
@item @var{w} @tab image width
@item @var{X} @tab previous x coordinate of text or image
@item @var{Y} @tab previous y coordinate of text or image
@end multitable
You may also use the constants @var{PI}, @var{E}, and the math functions available at the
FFmpeg formula evaluator at (@url{ffmpeg-doc.html#SEC13}), except @var{bits2qp(bits)}
and @var{qp2bits(qp)}.
Usage examples:
@example
# Remember to set the path to your fonts
FONTPATH="/cygdrive/c/WINDOWS/Fonts/"
FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/"
FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/"
export FONTPATH
# Bulb dancing in a Lissajous pattern
ffmpeg -i input.avi -vhook \
'vhook/imlib2.dll -x W*(0.5+0.25*sin(N/47*PI))-w/2 -y H*(0.5+0.50*cos(N/97*PI))-h/2 -i /usr/share/imlib2/data/images/bulb.png' \
-acodec copy -sameq output.avi
# Text scrolling
ffmpeg -i input.avi -vhook \
'vhook/imlib2.dll -c red -F Vera.ttf/20 -x 150+0.5*N -y 70+0.25*N -t Hello' \
-acodec copy -sameq output.avi
# Date and time stamp, security-camera style:
ffmpeg -r 29.97 -s 320x256 -f video4linux -i /dev/video0 \
-vhook 'vhook/imlib2.so -x 0 -y 0 -i black-260x20.png' \
-vhook 'vhook/imlib2.so -c white -F VeraBd.ttf/12 -x 0 -y 0 -t %A-%D-%T' \
output.avi
In this example the video is captured from the first video capture card as a
320x256 AVI, and a black 260 by 20 pixel PNG image is placed in the upper
left corner, with the day, date and time overlaid on it in Vera Bold 12
point font. A simple black PNG file 260 pixels wide and 20 pixels tall
was created in the GIMP for this purpose.
# Scrolling credits from a text file
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -c white -F VeraBd.ttf/16 -x 100 -y -1.0*N -f credits.txt' \
-sameq output.avi
In this example, the text is stored in a file, and is positioned 100
pixels from the left hand edge of the video. The text is scrolled from the
bottom up. Making the y factor positive will scroll from the top down.
Increasing the magnitude of the y factor makes the text scroll faster,
decreasing it makes it scroll slower. Hint: Blank lines containing only
a newline are treated as end-of-file. To create blank lines, use lines
that consist of space characters only.
# Scrolling credits with custom color from a text file
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -C rgb.txt -c CustomColor1 -F VeraBd.ttf/16 -x 100 -y -1.0*N -f credits.txt' \
-sameq output.avi
This example does the same as the one above, but specifies an rgb.txt file
to be used, which has a custom-made color in it.
# Variable colors
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -t Hello -R abs(255*sin(N/47*PI)) -G abs(255*sin(N/47*PI)) -B abs(255*sin(N/47*PI))' \
-sameq output.avi
In this example, the color for the text goes up and down from black to
white.
# Text fade-out
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -t Hello -A max(0,255-exp(N/47))' \
-sameq output.avi
In this example, the text fades out in about 10 seconds for a 25 fps input
video file.
# scrolling credits from a graphics file
ffmpeg -sameq -i input.avi \
-vhook 'vhook/imlib2.so -x 0 -y -1.0*N -i credits.png' output.avi
In this example, a transparent PNG file the same width as the video
(e.g. 320 pixels), but very long, (e.g. 3000 pixels), was created, and
text, graphics, brushstrokes, etc, were added to the image. The image
is then scrolled up, from the bottom of the frame.
@end example
@section ppm.c
It's basically a launch point for a PPM pipe, so you can use any
executable (or script) which consumes a PPM on stdin and produces a PPM
on stdout (and flushes each frame). The Netpbm utilities are a series of
such programs.
A list of them is here:
@url{http://netpbm.sourceforge.net/doc/directory.html}
Usage example:
@example
ffmpeg -i input -vhook "/path/to/ppm.so some-ppm-filter args" output
@end example
@section drawtext.c
This module implements a text overlay for a video image. Currently it
supports a fixed overlay or reading the text from a file. The string
is passed through strftime() so that it is easy to imprint the date and
time onto the image.
Features:
@itemize @minus
@item TrueType, Type1 and others via the FreeType2 library
@item Font kerning (better output)
@item Line Wrap (put the text that doesn't fit one line on the next line)
@item Background box (currently in development)
@item Outline
@end itemize
Options:
@multitable @columnfractions .2 .8
@item @option{-c <color>} @tab Foreground color of the text ('internet' way) <#RRGGBB> [default #FFFFFF]
@item @option{-C <color>} @tab Background color of the text ('internet' way) <#RRGGBB> [default #000000]
@item @option{-f <font-filename>} @tab font file to use
@item @option{-t <text>} @tab text to display
@item @option{-T <filename>} @tab file to read text from
@item @option{-x <pos>} @tab x coordinate of the start of the text
@item @option{-y <pos>} @tab y coordinate of the start of the text
@end multitable
Text fonts are being looked for in a FONTPATH environment variable.
If the FONTPATH environment variable is not available, or is not checked by
your target (i.e. Cygwin), then specify the full path to the font file as in:
@example
-f /usr/X11R6/lib/X11/fonts/TTF/VeraBd.ttf
@end example
Usage Example:
@example
# Remember to set the path to your fonts
FONTPATH="/cygdrive/c/WINDOWS/Fonts/"
FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/"
FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/"
export FONTPATH
# Time and date display
ffmpeg -f video4linux2 -i /dev/video0 \
-vhook 'vhook/drawtext.so -f VeraBd.ttf -t %A-%D-%T' movie.mpg
This example grabs video from the first capture card and outputs it to an
MPEG video, and places "Weekday-dd/mm/yy-hh:mm:ss" at the top left of the
frame, updated every second, using the Vera Bold TrueType Font, which
should exist in: /usr/X11R6/lib/X11/fonts/TTF/
@end example
Check the man page for strftime() for all the various ways you can format
the date and time.
@section watermark.c
Command Line options:
@multitable @columnfractions .2 .8
@item @option{-m [0|1]} @tab Mode (default: 0, see below)
@item @option{-t 000000 - FFFFFF} @tab Threshold, six digit hex number
@item @option{-f <filename>} @tab Watermark image filename, must be specified!
@end multitable
MODE 0:
The watermark picture works like this (assuming color intensities 0..0xFF):
Per color do this:
If mask color is 0x80, no change to the original frame.
If mask color is < 0x80 the absolute difference is subtracted from the
frame. If result < 0, result = 0.
If mask color is > 0x80 the absolute difference is added to the
frame. If result > 0xFF, result = 0xFF.
You can override the 0x80 level with the -t flag. E.g. if threshold is
000000 the color value of watermark is added to the destination.
This way a mask that is visible both in light and dark pictures can be made
(e.g. by using a picture generated by the Gimp and the bump map tool).
An example watermark file is at:
@url{http://engene.se/ffmpeg_watermark.gif}
MODE 1:
Per color do this:
If mask color > threshold color then the watermark pixel is used.
Example usage:
@example
ffmpeg -i infile -vhook '/path/watermark.so -f wm.gif' -an out.mov
ffmpeg -i infile -vhook '/path/watermark.so -f wm.gif -m 1 -t 222222' -an out.mov
@end example
@bye

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@@ -1,624 +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 the mingw-w64 project.
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 video4linux2
Video4Linux2 input video device.
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.
Video4Linux2 devices usually support a limited set of
@var{width}x@var{height} sizes and framerates. You can check which are
supported using @command{-list_formats all} for Video4Linux2 devices.
Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
The time base for the timestamps is 1 microsecond. Depending on the kernel
version and configuration, the timestamps may be derived from the real time
clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
boot time, unaffected by NTP or manual changes to the clock). The
@option{-timestamps abs} or @option{-ts abs} option can be used to force
conversion into the real time clock.
Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
option), it will always be used.
@example
# Grab and show the input of a video4linux2 device.
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
# Grab and record the input of a video4linux2 device, leave the
framerate and size as previously set.
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@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

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

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

@@ -1,498 +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{aiff}
@section aiff
Audio Interchange File Format muxer.
It accepts the following options:
@table @option
@item write_id3v2
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
@item id3v2_version
Select ID3v2 version to write. Currently only version 3 and 4 (aka.
ID3v2.3 and ID3v2.4) are supported. The default is version 4.
@end table
@anchor{crc}
@section crc
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-packet CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each audio
and video packet. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a line for each audio and video
packet of the form:
@example
@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC}
@end example
@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
CRC of the packet.
For example to compute the CRC of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
in the file @file{out.crc}:
@example
ffmpeg -i INPUT -f framecrc out.crc
@end example
To print the information to stdout, use the command:
@example
ffmpeg -i INPUT -f framecrc -
@end example
With @command{ffmpeg}, you can select the output format to which the
audio and video frames are encoded before computing the CRC for each
packet by specifying the audio and video codec. For example, to
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{framemd5}
@section framemd5
Per-packet MD5 testing format.
This muxer computes and prints the MD5 hash for each audio
and video packet. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
hash.
The output of the muxer consists of a line for each audio and video
packet of the form:
@example
@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
@end example
@var{MD5} is a hexadecimal number representing the computed MD5 hash
for the packet.
For example to compute the MD5 of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
in the file @file{out.md5}:
@example
ffmpeg -i INPUT -f framemd5 out.md5
@end example
To print the information to stdout, use the command:
@example
ffmpeg -i INPUT -f framemd5 -
@end example
See also the @ref{md5} muxer.
@anchor{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.
@anchor{md5}
@section md5
MD5 testing format.
This muxer computes and prints the MD5 hash of all the input audio
and video frames. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
hash.
The output of the muxer consists of a single line of the form:
MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing
the computed MD5 hash.
For example to compute the MD5 hash of the input converted to raw
audio and video, and store it in the file @file{out.md5}:
@example
ffmpeg -i INPUT -f md5 out.md5
@end example
You can print the MD5 to stdout with the command:
@example
ffmpeg -i INPUT -f md5 -
@end example
See also the @ref{framemd5} muxer.
@section MOV/MP4/ISMV
The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
file has all the metadata about all packets stored in one location
(written at the end of the file, it can be moved to the start for
better playback using the @command{qt-faststart} tool). A fragmented
file consists of a number of fragments, where packets and metadata
about these packets are stored together. Writing a fragmented
file has the advantage that the file is decodable even if the
writing is interrupted (while a normal MOV/MP4 is undecodable if
it is not properly finished), and it requires less memory when writing
very long files (since writing normal MOV/MP4 files stores info about
every single packet in memory until the file is closed). The downside
is that it is less compatible with other applications.
Fragmentation is enabled by setting one of the AVOptions that define
how to cut the file into fragments:
@table @option
@item -moov_size @var{bytes}
Reserves space for the moov atom at the beginning of the file instead of placing the
moov atom at the end. If the space reserved is insufficient, muxing will fail.
@item -movflags frag_keyframe
Start a new fragment at each video keyframe.
@item -frag_duration @var{duration}
Create fragments that are @var{duration} microseconds long.
@item -frag_size @var{size}
Create fragments that contain up to @var{size} bytes of payload data.
@item -movflags frag_custom
Allow the caller to manually choose when to cut fragments, by
calling @code{av_write_frame(ctx, NULL)} to write a fragment with
the packets written so far. (This is only useful with other
applications integrating libavformat, not from @command{ffmpeg}.)
@item -min_frag_duration @var{duration}
Don't create fragments that are shorter than @var{duration} microseconds long.
@end table
If more than one condition is specified, fragments are cut when
one of the specified conditions is fulfilled. The exception to this is
@code{-min_frag_duration}, which has to be fulfilled for any of the other
conditions to apply.
Additionally, the way the output file is written can be adjusted
through a few other options:
@table @option
@item -movflags empty_moov
Write an initial moov atom directly at the start of the file, without
describing any samples in it. Generally, an mdat/moov pair is written
at the start of the file, as a normal MOV/MP4 file, containing only
a short portion of the file. With this option set, there is no initial
mdat atom, and the moov atom only describes the tracks but has
a zero duration.
Files written with this option set do not work in QuickTime.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags separate_moof
Write a separate moof (movie fragment) atom for each track. Normally,
packets for all tracks are written in a moof atom (which is slightly
more efficient), but with this option set, the muxer writes one moof/mdat
pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@end table
Smooth Streaming content can be pushed in real time to a publishing
point on IIS with this muxer. Example:
@example
ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example
@section mpegts
MPEG transport stream muxer.
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.
@item segment_wrap @var{limit}
Wrap around segment index once it reaches @var{limit}.
@end table
@example
ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
@end example
@section mp3
The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
not written by default, but may be enabled with the @code{write_id3v1} option.
For seekable output the muxer also writes a Xing frame at the beginning, which
contains the number of frames in the file. It is useful for computing duration
of VBR files.
The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
are supplied to the muxer in form of a video stream with a single packet. There
can be any number of those streams, each will correspond to a single APIC frame.
The stream metadata tags @var{title} and @var{comment} map to APIC
@var{description} and @var{picture type} respectively. See
@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
Note that the APIC frames must be written at the beginning, so the muxer will
buffer the audio frames until it gets all the pictures. It is therefore advised
to provide the pictures as soon as possible to avoid excessive buffering.
Examples:
Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@example
ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
@end example
Attach a picture to an mp3:
@example
ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover"
-metadata:s:v comment="Cover (Front)" out.mp3
@end example
@c man end MUXERS

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@@ -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

@@ -1,380 +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
(@command{gmake}).
@section (Open)Solaris
GNU Make is required to build FFmpeg, so you have to invoke (@command{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 versions 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 during
@code{make install}).
@item In order to compile FFplay, you must have the MinGW development library
of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
@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 @emph{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 @command{lib.exe}:
@example
lib /machine:i386 /def:..\lib\foo-version.def /out:foo.lib
@end example
Replace @code{foo-version} and @code{foo} with the respective library names.
@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.
To build shared libraries add a special compiler flag to work around current
@code{gcc4-core} package bugs in addition to the normal configure flags:
@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, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
@end example
The recommendation for x264 is to build it from source, as it evolves too
quickly for Cygwin Ports to be up to date.
@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

View File

@@ -1,125 +0,0 @@
/*
* Copyright (c) 2012 Anton Khirnov
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* generate texinfo manpages for avoptions
*/
#include <stddef.h>
#include <string.h>
#include <float.h>
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavutil/opt.h"
static void print_usage(void)
{
fprintf(stderr, "Usage: enum_options type\n"
"type: format codec\n");
exit(1);
}
static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
{
printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
switch (o->type) {
case AV_OPT_TYPE_BINARY: printf("hexadecimal string"); break;
case AV_OPT_TYPE_STRING: printf("string"); break;
case AV_OPT_TYPE_INT:
case AV_OPT_TYPE_INT64: printf("integer"); break;
case AV_OPT_TYPE_FLOAT:
case AV_OPT_TYPE_DOUBLE: printf("float"); break;
case AV_OPT_TYPE_RATIONAL: printf("rational number"); break;
case AV_OPT_TYPE_FLAGS: printf("flags"); break;
default: printf("value"); break;
}
printf("} (@emph{");
if (o->flags & AV_OPT_FLAG_DECODING_PARAM) {
printf("input");
if (o->flags & AV_OPT_FLAG_ENCODING_PARAM)
printf("/");
}
if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output");
if (o->flags & AV_OPT_FLAG_AUDIO_PARAM) printf(",audio");
if (o->flags & AV_OPT_FLAG_VIDEO_PARAM) printf(",video");
if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles");
printf("})\n");
if (o->help)
printf("%s\n", o->help);
if (o->unit) {
const AVOption *u;
printf("\nPossible values:\n@table @samp\n");
for (u = opts; u->name; u++) {
if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit))
printf("@item %s\n%s\n", u->name, u->help ? u->help : "");
}
printf("@end table\n");
}
}
static void show_opts(const AVOption *opts, int per_stream)
{
const AVOption *o;
printf("@table @option\n");
for (o = opts; o->name; o++) {
if (o->type != AV_OPT_TYPE_CONST)
print_option(opts, o, per_stream);
}
printf("@end table\n");
}
static void show_format_opts(void)
{
#include "libavformat/options_table.h"
printf("@section Format AVOptions\n");
show_opts(options, 0);
}
static void show_codec_opts(void)
{
#include "libavcodec/options_table.h"
printf("@section Codec AVOptions\n");
show_opts(options, 1);
}
int main(int argc, char **argv)
{
if (argc < 2)
print_usage();
printf("@c DO NOT EDIT THIS FILE!\n"
"@c It was generated by print_options.\n\n");
if (!strcmp(argv[1], "format"))
show_format_opts();
else if (!strcmp(argv[1], "codec"))
show_codec_opts();
else
print_usage();
return 0;
}

View File

@@ -1,526 +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 bluray
Read BluRay playlist.
The accepted options are:
@table @option
@item angle
BluRay angle
@item chapter
Start chapter (1...N)
@item playlist
Playlist to read (BDMV/PLAYLIST/?????.mpls)
@end table
Examples:
Read longest playlist from BluRay mounted to /mnt/bluray:
@example
bluray:/mnt/bluray
@end example
Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
@example
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
@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 hls
Read Apple HTTP Live Streaming compliant segmented stream as
a uniform one. The M3U8 playlists describing the segments can be
remote HTTP resources or local files, accessed using the standard
file protocol.
The nested protocol is declared by specifying
"+@var{proto}" after the hls URI scheme name, where @var{proto}
is either "file" or "http".
@example
hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8
@end example
Using this protocol is discouraged - the hls demuxer should work
just as well (if not, please report the issues) and is more complete.
To use the hls demuxer instead, simply use the direct URLs to the
m3u8 files.
@section http
HTTP (Hyper Text Transfer Protocol).
@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). This
can be disabled by setting the maximum demuxing delay to zero (via
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

@@ -1,61 +0,0 @@
A Quick Description Of Rate Distortion Theory.
We want to encode a video, picture or piece of music optimally. What does
"optimally" really mean? It means that we want to get the best quality at a
given filesize OR we want to get the smallest filesize at a given quality
(in practice, these 2 goals are usually the same).
Solving this directly is not practical; trying all byte sequences 1
megabyte in length and selecting the "best looking" sequence will yield
256^1000000 cases to try.
But first, a word about quality, which is also called distortion.
Distortion can be quantified by almost any quality measurement one chooses.
Commonly, the sum of squared differences is used but more complex methods
that consider psychovisual effects can be used as well. It makes no
difference in this discussion.
First step: that rate distortion factor called lambda...
Let's consider the problem of minimizing:
distortion + lambda*rate
rate is the filesize
distortion is the quality
lambda is a fixed value choosen as a tradeoff between quality and filesize
Is this equivalent to finding the best quality for a given max
filesize? The answer is yes. For each filesize limit there is some lambda
factor for which minimizing above will get you the best quality (using your
chosen quality measurement) at the desired (or lower) filesize.
Second step: splitting the problem.
Directly splitting the problem of finding the best quality at a given
filesize is hard because we do not know how many bits from the total
filesize should be allocated to each of the subproblems. But the formula
from above:
distortion + lambda*rate
can be trivially split. Consider:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
This creates a problem made of 2 independent subproblems. The subproblems
might be 2 16x16 macroblocks in a frame of 32x16 size. To minimize:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
we just have to minimize:
distortion0 + lambda*rate0
and
distortion1 + lambda*rate1
I.e, the 2 problems can be solved independently.
Author: Michael Niedermayer
Copyright: LGPL

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 skipped 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

@@ -30,7 +30,7 @@ slices, that is, consecutive non-overlapping rectangles of dimension
special converter
These generally are unscaled converters of common
formats, like YUV 4:2:0/4:2:2 -> RGB12/15/16/24/32. Though it could also
formats, like YUV 4:2:0/4:2:2 -> RGB15/16/24/32. Though it could also
in principle contain scalers optimized for specific common cases.
Main path
@@ -96,3 +96,4 @@ would benefit from it.
Also, as already hinted at, initFilter() accepts an optional convolutional
filter as input that can be used for contrast, saturation, blur, sharpening
shift, chroma vs. luma shift, ...

View File

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

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

@@ -1,70 +0,0 @@
Writing a table generator
This documentation is preliminary.
Parts of the API are not good and should be changed.
Basic concepts
A table generator consists of two files, *_tablegen.c and *_tablegen.h.
The .h file will provide the variable declarations and initialization
code for the tables, the .c calls the initialization code and then prints
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.
The .c file
This file should include the *_tablegen.h and tableprint.h files and
anything else it needs as long as it does not depend on config.h or
avconfig.h.
In addition to that it must contain a main() function which initializes
all tables by calling the init functions from the .h file and then prints
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);
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
for standard types - they print only the "core" parts so they are easier
to reuse for multi-dimensional arrays so the outermost {} must be printed
separately.
If there's no standard function for printing the type you need, the
WRITE_1D_FUNC_ARGV macro is a very quick way to create one.
See libavcodec/dv_tablegen.c for an example.
The .h file
This file should contain:
- one or more initialization functions
- the table variable declarations
If CONFIG_HARDCODED_TABLES is set, the initialization functions should
not do anything, and instead of the variable declarations the
generated *_tables.h file should be included.
Since that will be generated in the build directory, the path must be
included, i.e.
#include "libavcodec/example_tables.h"
not
#include "example_tables.h"
Makefile changes
To make the automatic table creation work, you must manually declare the
new dependency.
For this add a line similar to this:
$(SUBDIR)example.o: $(SUBDIR)example_tables.h
under the "ifdef CONFIG_HARDCODED_TABLES" section in the Makefile.

View File

@@ -1,4 +1,4 @@
#! /usr/bin/perl
#! /usr/bin/perl -w
# Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
@@ -23,13 +23,10 @@
# markup to Perl POD format. It's intended to be used to extract
# something suitable for a manpage from a Texinfo document.
use warnings;
$output = 0;
$skipping = 0;
%chapters = ();
@chapters_sequence = ();
$chapter = "";
%sects = ();
$section = "";
@icstack = ();
@endwstack = ();
@skstack = ();
@@ -38,7 +35,7 @@ $shift = "";
%defs = ();
$fnno = 1;
$inf = "";
@ibase = ();
$ibase = "";
while ($_ = shift) {
if (/^-D(.*)$/) {
@@ -54,8 +51,6 @@ while ($_ = shift) {
die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
unless $flag =~ /^[a-zA-Z0-9_-]+$/;
$defs{$flag} = $value;
} elsif (/^-I(.*)$/) {
push @ibase, $1 ne "" ? $1 : shift;
} elsif (/^-/) {
usage();
} else {
@@ -65,12 +60,10 @@ while ($_ = shift) {
}
}
push @ibase, ".";
if (defined $in) {
$inf = gensym();
open($inf, "<$in") or die "opening \"$in\": $!\n";
push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|;
$ibase = $1 if $in =~ m|^(.+)/[^/]+$|;
} else {
$inf = \*STDIN;
}
@@ -80,7 +73,7 @@ if (defined $out) {
}
while(defined $inf) {
INF: while(<$inf>) {
while(<$inf>) {
# Certain commands are discarded without further processing.
/^\@(?:
[a-z]+index # @*index: useful only in complete manual
@@ -106,32 +99,21 @@ INF: while(<$inf>) {
next;
};
/^\@include\s+(.+)$/ and do {
push @instack, $inf;
$inf = gensym();
for (@ibase) {
open($inf, "<" . $_ . "/" . $1) and next INF;
}
die "cannot open $1: $!\n";
};
/^\@chapter\s+([A-Za-z ]+)/ and do {
# close old chapter
$chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);
# start new chapter
$chapter_name = $1, push (@chapters_sequence, $chapter_name) unless $skipping;
$chapters{$chapter_name} = "" unless exists $chapters{$chapter_name};
$chapter = "";
$output = 1;
# 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-Z]+)\s+([A-Za-z0-9-]+)/ and do {
$output = 1 if exists $defs{$2};
$sect = $1;
next;
};
/^\@bye/ and do {
# close old chapter
$chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);
last INF;
/^\@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);
$section = "";
$output = 0;
next;
};
# handle variables
@@ -156,20 +138,20 @@ INF: while(<$inf>) {
# Ignore @end foo, where foo is not an operation which may
# cause us to skip, if we are presently skipping.
my $ended = $1;
next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/;
next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex)$/;
die "\@end $ended without \@$ended at line $.\n" unless defined $endw;
die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw;
$endw = pop @endwstack;
if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) {
if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex)$/) {
$skipping = pop @skstack;
next;
} elsif ($ended =~ /^(?:example|smallexample|display)$/) {
$shift = "";
$_ = ""; # need a paragraph break
} elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
} elsif ($ended =~ /^(?:itemize|enumerate|[fv]?table)$/) {
$_ = "\n=back\n";
$ic = pop @icstack;
} else {
@@ -196,11 +178,11 @@ INF: while(<$inf>) {
next;
};
/^\@(ignore|menu|iftex|ifhtml|ifnothtml)\b/ and do {
/^\@(ignore|menu|iftex)\b/ and do {
push @endwstack, $endw;
push @skstack, $skipping;
$endw = $1;
$skipping = $endw !~ /ifnothtml/;
$skipping = 1;
next;
};
@@ -217,6 +199,7 @@ INF: while(<$inf>) {
s/\@TeX\{\}/TeX/g;
s/\@pounds\{\}/\#/g;
s/\@minus(?:\{\})?/-/g;
s/\\,/,/g;
# Now the ones that have to be replaced by special escapes
# (which will be turned back into text by unmunge())
@@ -241,18 +224,27 @@ INF: 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";
$endw = "itemize";
};
@@ -269,16 +261,15 @@ INF: while(<$inf>) {
$endw = "enumerate";
};
/^\@((?:multi|[fv])?table)\s+(\@[a-z]+)/ and do {
/^\@([fv]?table)\s+(\@[a-z]+)/ and do {
push @endwstack, $endw;
push @icstack, $ic;
$endw = $1;
$ic = $2;
$ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env|command)/B/;
$ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env)/B/;
$ic =~ s/\@(?:code|kbd)/C/;
$ic =~ s/\@(?:dfn|var|emph|cite|i)/I/;
$ic =~ s/\@(?:file)/F/;
$ic =~ s/\@(?:columnfractions)//;
$_ = "\n=over 4\n";
};
@@ -289,21 +280,6 @@ INF: while(<$inf>) {
$_ = ""; # need a paragraph break
};
/^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
my $columns = $1;
$columns =~ s/\@tab/ : /;
$_ = "\n=item B&LT;". $columns ."&GT;\n";
};
/^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
my $columns = $1;
$columns =~ s/\@tab/ : /;
$_ = " : ". $columns;
$chapter =~ s/\n+\s+$//;
};
/^\@itemx?\s*(.+)?$/ and do {
if (defined $1) {
# Entity escapes prevent munging by the <> processing below.
@@ -315,7 +291,7 @@ INF: while(<$inf>) {
}
};
$chapter .= $shift.$_."\n";
$section .= $shift.$_."\n";
}
# End of current file.
close($inf);
@@ -324,15 +300,16 @@ $inf = pop @instack;
die "No filename or title\n" unless defined $fn && defined $tl;
$chapters{NAME} = "$fn \- $tl\n";
$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
$sects{NAME} = "$fn \- $tl\n";
$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES};
unshift @chapters_sequence, "NAME";
for $chapter (@chapters_sequence) {
if (exists $chapters{$chapter}) {
$head = uc($chapter);
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/;
print "=head1 $head\n\n";
print scalar unmunge ($chapters{$chapter});
print scalar unmunge ($sects{$sect});
print "\n";
}
}
@@ -373,12 +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/\@ref\{([^\}]*)\}/$1/g;
s/\@noindent\s*//g;
s/\@refill//g;
s/\@gol//g;
@@ -387,7 +361,7 @@ sub postprocess
# @uref can take one, two, or three arguments, with different
# semantics each time. @url and @email are just like @uref with
# one argument, for our purposes.
s/\@(?:uref|url|email)\{([^\},]*),?[^\}]*\}/&lt;B<$1>&gt;/g;
s/\@(?:uref|url|email)\{([^\},]*)\}/&lt;B<$1>&gt;/g;
s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g;
s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g;
@@ -431,13 +405,13 @@ sub unmunge
sub add_footnote
{
unless (exists $chapters{FOOTNOTES}) {
$chapters{FOOTNOTES} = "\n=over 4\n\n";
unless (exists $sects{FOOTNOTES}) {
$sects{FOOTNOTES} = "\n=over 4\n\n";
}
$chapters{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++;
$chapters{FOOTNOTES} .= $_[0];
$chapters{FOOTNOTES} .= "\n\n";
$sects{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++;
$sects{FOOTNOTES} .= $_[0];
$sects{FOOTNOTES} .= "\n\n";
}
# stolen from Symbol.pm

View File

@@ -1,109 +0,0 @@
This is a quick description of the viterbi aka dynamic programing
algorthm.
Its reason for existence is that wikipedia has become very poor on
describing algorithms in a way that makes it useable for understanding
them or anything else actually. It tends now to describe the very same
algorithm under 50 different names and pages with few understandable
by even people who fully understand the algorithm and the theory behind.
Problem description: (that is what it can solve)
assume we have a 2d table, or you could call it a graph or matrix if you
prefer
O O O O O O O
O O O O O O O
O O O O O O O
O O O O O O O
That table has edges connecting points from each column to the next column
and each edge has a score like: (only some edge and scores shown to keep it
readable)
O--5--O-----O-----O-----O-----O
2 / 7 / \ / \ / \ /
\ / \ / \ / \ / \ /
O7-/--O--/--O--/--O--/--O--/--O
\/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
/\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
O3-/--O--/--O--/--O--/--O--/--O
/ \ / \ / \ / \ / \
1 \ 9 \ / \ / \ / \
O--2--O--1--O--5--O--3--O--8--O
Our goal is to find a path from left to right through it which
minimizes the sum of the score of all edges.
(and of course left/right is just a convention here it could be top down too)
Similarly the minimum could be the maximum by just fliping the sign,
Example of a path with scores:
O O O O O O O
>---O. O O .O-2-O O O
5. .7 .
O O-1-O O O 8 O O
.
O O O O O O-1-O---> (sum here is 24)
The viterbi algorthm now solves this simply column by column
For the previous column each point has a best path and a associated
score:
O-----5 O
\
\
O \ 1 O
\/
/\
O / 2 O
/
/
O-----2 O
To move one column forward we just need to find the best path and associated
scores for the next column
here are some edges we could choose from:
O-----5--3--O
\ \8
\ \
O \ 1--9--O
\/ \3
/\ \
O / 2--1--O
/ \2
/ \
O-----2--4--O
Finding the new best pathes and scores for each point of our new column is
trivial given we know the previous column best pathes and scores:
O-----0-----8
\
\
O \ 0----10
\/
/\
O / 0-----3
/ \
/ \
O 0 4
the viterbi algorthm continues exactly like this column for column until the
end and then just picks the path with the best score (above that would be the
one with score 3)
Author: Michael niedermayer
Copyright LGPL

9156
ffmpeg.c

File diff suppressed because it is too large Load Diff

2800
ffplay.c

File diff suppressed because it is too large Load Diff

View File

@@ -0,0 +1,4 @@
coder=0
bf=0
flags2=-wpred-dct8x8+mbtree
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+mbtree
wpredp=2

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+mbtree
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+mbtree
wpredp=2

View File

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

View File

@@ -0,0 +1,8 @@
coder=0
bf=0
refs=1
flags2=-wpred-dct8x8+mbtree
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+mbtree
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+mbtree
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+mbtree
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+mbtree
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+mbtree
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+mbtree
cqp=0

View File

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

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+mbtree
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+mbtree
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=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=3
trellis=0
flags2=+wpred+dct8x8+fastpskip+mbtree
wpredp=2

1823
ffprobe.c

File diff suppressed because it is too large Load Diff

1584
ffserver.c

File diff suppressed because it is too large Load Diff

View File

@@ -1,5 +1,6 @@
/*
* GSM common header
* Multiple format streaming server
* copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
@@ -17,15 +18,11 @@
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef FFMPEG_FFSERVER_H
#define FFMPEG_FFSERVER_H
#ifndef AVCODEC_GSM_H
#define AVCODEC_GSM_H
/* interface between ffserver and modules */
/* bytes per block */
#define GSM_BLOCK_SIZE 33
#define GSM_MS_BLOCK_SIZE 65
void ffserver_module_init(void);
/* samples per block */
#define GSM_FRAME_SIZE 160
#endif /* AVCODEC_GSM_H */
#endif /* FFMPEG_FFSERVER_H */

File diff suppressed because it is too large Load Diff

View File

@@ -20,14 +20,14 @@
*/
/**
* @file
* @file libavcodec/8bps.c
* QT 8BPS Video Decoder by Roberto Togni
* For more information about the 8BPS format, visit:
* http://www.pcisys.net/~melanson/codecs/
*
* Supports: PAL8 (RGB 8bpp, paletted)
* : BGR24 (RGB 24bpp) (can also output it as RGB32)
* : RGB32 (RGB 32bpp, 4th plane is alpha)
* : RGB32 (RGB 32bpp, 4th plane is probably alpha and it's ignored)
*
*/
@@ -38,20 +38,18 @@
#include "avcodec.h"
static const enum PixelFormat pixfmt_rgb24[] = {
PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE };
static const enum PixelFormat pixfmt_rgb24[] = {PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE};
/*
* Decoder context
*/
typedef struct EightBpsContext {
AVCodecContext *avctx;
AVFrame pic;
unsigned char planes;
unsigned char planemap[4];
AVCodecContext *avctx;
AVFrame pic;
uint32_t pal[256];
unsigned char planes;
unsigned char planemap[4];
} EightBpsContext;
@@ -60,90 +58,89 @@ typedef struct EightBpsContext {
* Decode a frame
*
*/
static int decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, const uint8_t *buf, int buf_size)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
EightBpsContext * const c = avctx->priv_data;
const unsigned char *encoded = buf;
unsigned char *pixptr, *pixptr_end;
unsigned int height = avctx->height; // Real image height
unsigned int dlen, p, row;
const unsigned char *lp, *dp;
unsigned char count;
unsigned int planes = c->planes;
unsigned char *planemap = c->planemap;
EightBpsContext * const c = avctx->priv_data;
const unsigned char *encoded = buf;
unsigned char *pixptr, *pixptr_end;
unsigned int height = avctx->height; // Real image height
unsigned int dlen, p, row;
const unsigned char *lp, *dp;
unsigned char count;
unsigned int px_inc;
unsigned int planes = c->planes;
unsigned char *planemap = c->planemap;
if (c->pic.data[0])
avctx->release_buffer(avctx, &c->pic);
if(c->pic.data[0])
avctx->release_buffer(avctx, &c->pic);
c->pic.reference = 0;
c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
if (avctx->get_buffer(avctx, &c->pic) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
c->pic.reference = 0;
c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
if(avctx->get_buffer(avctx, &c->pic) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
/* Set data pointer after line lengths */
dp = encoded + planes * (height << 1);
/* Set data pointer after line lengths */
dp = encoded + planes * (height << 1);
for (p = 0; p < planes; p++) {
/* Lines length pointer for this plane */
lp = encoded + p * (height << 1);
/* Ignore alpha plane, don't know what to do with it */
if (planes == 4)
planes--;
/* Decode a plane */
for (row = 0; row < height; row++) {
pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
pixptr_end = pixptr + c->pic.linesize[0];
dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
/* Decode a row of this plane */
while (dlen > 0) {
if (dp + 1 >= buf + buf_size)
return -1;
if ((count = *dp++) <= 127) {
count++;
dlen -= count + 1;
if (pixptr + count * planes > pixptr_end)
break;
if (dp + count > buf + buf_size)
return -1;
while (count--) {
*pixptr = *dp++;
pixptr += planes;
}
} else {
count = 257 - count;
if (pixptr + count * planes > pixptr_end)
break;
while (count--) {
*pixptr = *dp;
pixptr += planes;
}
dp++;
dlen -= 2;
px_inc = planes + (avctx->pix_fmt == PIX_FMT_RGB32);
for (p = 0; p < planes; p++) {
/* Lines length pointer for this plane */
lp = encoded + p * (height << 1);
/* Decode a plane */
for(row = 0; row < height; row++) {
pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
pixptr_end = pixptr + c->pic.linesize[0];
dlen = be2me_16(*(const unsigned short *)(lp+row*2));
/* Decode a row of this plane */
while(dlen > 0) {
if(dp + 1 >= buf+buf_size) return -1;
if ((count = *dp++) <= 127) {
count++;
dlen -= count + 1;
if (pixptr + count * px_inc > pixptr_end)
break;
if(dp + count > buf+buf_size) return -1;
while(count--) {
*pixptr = *dp++;
pixptr += px_inc;
}
} else {
count = 257 - count;
if (pixptr + count * px_inc > pixptr_end)
break;
while(count--) {
*pixptr = *dp;
pixptr += px_inc;
}
dp++;
dlen -= 2;
}
}
}
}
}
}
if (avctx->bits_per_coded_sample <= 8) {
const uint8_t *pal = av_packet_get_side_data(avpkt,
AV_PKT_DATA_PALETTE,
NULL);
if (pal) {
c->pic.palette_has_changed = 1;
memcpy(c->pal, pal, AVPALETTE_SIZE);
}
memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
}
if (avctx->palctrl) {
memcpy (c->pic.data[1], avctx->palctrl->palette, AVPALETTE_SIZE);
if (avctx->palctrl->palette_changed) {
c->pic.palette_has_changed = 1;
avctx->palctrl->palette_changed = 0;
} else
c->pic.palette_has_changed = 0;
}
*data_size = sizeof(AVFrame);
*(AVFrame*)data = c->pic;
*data_size = sizeof(AVFrame);
*(AVFrame*)data = c->pic;
/* always report that the buffer was completely consumed */
return buf_size;
/* always report that the buffer was completely consumed */
return buf_size;
}
@@ -154,47 +151,54 @@ static int decode_frame(AVCodecContext *avctx, void *data,
*/
static av_cold int decode_init(AVCodecContext *avctx)
{
EightBpsContext * const c = avctx->priv_data;
EightBpsContext * const c = avctx->priv_data;
c->avctx = avctx;
c->pic.data[0] = NULL;
c->avctx = avctx;
avcodec_get_frame_defaults(&c->pic);
switch (avctx->bits_per_coded_sample) {
case 8:
avctx->pix_fmt = PIX_FMT_PAL8;
c->planes = 1;
c->planemap[0] = 0; // 1st plane is palette indexes
break;
case 24:
avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
c->planes = 3;
c->planemap[0] = 2; // 1st plane is red
c->planemap[1] = 1; // 2nd plane is green
c->planemap[2] = 0; // 3rd plane is blue
break;
case 32:
avctx->pix_fmt = PIX_FMT_RGB32;
c->planes = 4;
#if HAVE_BIGENDIAN
c->planemap[0] = 1; // 1st plane is red
c->planemap[1] = 2; // 2nd plane is green
c->planemap[2] = 3; // 3rd plane is blue
c->planemap[3] = 0; // 4th plane is alpha
#else
c->planemap[0] = 2; // 1st plane is red
c->planemap[1] = 1; // 2nd plane is green
c->planemap[2] = 0; // 3rd plane is blue
c->planemap[3] = 3; // 4th plane is alpha
#endif
break;
default:
av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n",
avctx->bits_per_coded_sample);
return -1;
c->pic.data[0] = NULL;
if (avcodec_check_dimensions(avctx, avctx->width, avctx->height) < 0) {
return 1;
}
return 0;
switch (avctx->bits_per_coded_sample) {
case 8:
avctx->pix_fmt = PIX_FMT_PAL8;
c->planes = 1;
c->planemap[0] = 0; // 1st plane is palette indexes
if (avctx->palctrl == NULL) {
av_log(avctx, AV_LOG_ERROR, "Error: PAL8 format but no palette from demuxer.\n");
return -1;
}
break;
case 24:
avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
c->planes = 3;
c->planemap[0] = 2; // 1st plane is red
c->planemap[1] = 1; // 2nd plane is green
c->planemap[2] = 0; // 3rd plane is blue
break;
case 32:
avctx->pix_fmt = PIX_FMT_RGB32;
c->planes = 4;
#ifdef WORDS_BIGENDIAN
c->planemap[0] = 1; // 1st plane is red
c->planemap[1] = 2; // 2nd plane is green
c->planemap[2] = 3; // 3rd plane is blue
c->planemap[3] = 0; // 4th plane is alpha???
#else
c->planemap[0] = 2; // 1st plane is red
c->planemap[1] = 1; // 2nd plane is green
c->planemap[2] = 0; // 3rd plane is blue
c->planemap[3] = 3; // 4th plane is alpha???
#endif
break;
default:
av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n", avctx->bits_per_coded_sample);
return -1;
}
return 0;
}
@@ -207,24 +211,25 @@ static av_cold int decode_init(AVCodecContext *avctx)
*/
static av_cold int decode_end(AVCodecContext *avctx)
{
EightBpsContext * const c = avctx->priv_data;
EightBpsContext * const c = avctx->priv_data;
if (c->pic.data[0])
avctx->release_buffer(avctx, &c->pic);
if (c->pic.data[0])
avctx->release_buffer(avctx, &c->pic);
return 0;
return 0;
}
AVCodec ff_eightbps_decoder = {
.name = "8bps",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_8BPS,
.priv_data_size = sizeof(EightBpsContext),
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
AVCodec eightbps_decoder = {
"8bps",
CODEC_TYPE_VIDEO,
CODEC_ID_8BPS,
sizeof(EightBpsContext),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
};

View File

@@ -1,6 +1,6 @@
/*
* 8SVX audio decoder
* Copyright (C) 2008 Jaikrishnan Menon
* Copyright (C) 2011 Stefano Sabatini
*
* This file is part of FFmpeg.
*
@@ -20,226 +20,92 @@
*/
/**
* @file
* @file libavcodec/8svx.c
* 8svx audio decoder
* @author Jaikrishnan Menon
*
* supports: fibonacci delta encoding
* : exponential encoding
*
* For more information about the 8SVX format:
* http://netghost.narod.ru/gff/vendspec/iff/iff.txt
* http://sox.sourceforge.net/AudioFormats-11.html
* http://aminet.net/package/mus/misc/wavepak
* http://amigan.1emu.net/reg/8SVX.txt
*
* Samples can be found here:
* http://aminet.net/mods/smpl/
*/
#include "avcodec.h"
/** decoder context */
typedef struct EightSvxContext {
AVFrame frame;
const int8_t *table;
/* buffer used to store the whole audio decoded/interleaved chunk,
* which is sent with the first packet */
uint8_t *samples;
int64_t samples_size;
int samples_idx;
int16_t fib_acc;
const int16_t *table;
} EightSvxContext;
static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
#define MAX_FRAME_SIZE 2048
/**
* Interleave samples in buffer containing all left channel samples
* at the beginning, and right channel samples at the end.
* Each sample is assumed to be in signed 8-bit format.
*
* @param size the size in bytes of the dst and src buffer
*/
static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
{
uint8_t *dst_end = dst + size;
size = size>>1;
while (dst < dst_end) {
*dst++ = *src;
*dst++ = *(src+size);
src++;
}
}
/**
* Delta decode the compressed values in src, and put the resulting
* decoded n samples in dst.
*
* @param val starting value assumed by the delta sequence
* @param table delta sequence table
* @return size in bytes of the decoded data, must be src_size*2
*/
static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
int8_t val, const int8_t *table)
{
int n = src_size;
int8_t *dst0 = dst;
while (n--) {
uint8_t d = *src++;
val = av_clip(val + table[d & 0x0f], -127, 128);
*dst++ = val;
val = av_clip(val + table[d >> 4] , -127, 128);
*dst++ = val;
}
return dst-dst0;
}
static const int16_t fibonacci[16] = { -34<<8, -21<<8, -13<<8, -8<<8, -5<<8, -3<<8, -2<<8, -1<<8,
0, 1<<8, 2<<8, 3<<8, 5<<8, 8<<8, 13<<8, 21<<8 };
static const int16_t exponential[16] = { -128<<8, -64<<8, -32<<8, -16<<8, -8<<8, -4<<8, -2<<8, -1<<8,
0, 1<<8, 2<<8, 4<<8, 8<<8, 16<<8, 32<<8, 64<<8 };
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
const uint8_t *buf, int buf_size)
{
EightSvxContext *esc = avctx->priv_data;
int n, out_data_size, ret;
uint8_t *src, *dst;
int16_t *out_data = data;
int consumed = buf_size;
const uint8_t *buf_end = buf + buf_size;
/* decode and interleave the first packet */
if (!esc->samples && avpkt) {
uint8_t *deinterleaved_samples, *p = NULL;
if((*data_size >> 2) < buf_size)
return -1;
esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
if (!(esc->samples = av_malloc(esc->samples_size)))
return AVERROR(ENOMEM);
/* decompress */
if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int n = esc->samples_size;
if (buf_size < 2) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
return AVERROR(EINVAL);
}
if (!(deinterleaved_samples = av_mallocz(n)))
return AVERROR(ENOMEM);
p = deinterleaved_samples;
/* the uncompressed starting value is contained in the first byte */
if (avctx->channels == 2) {
delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
buf += buf_size/2;
delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
} else
delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
} else {
deinterleaved_samples = avpkt->data;
}
if (avctx->channels == 2)
interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
else
memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
av_freep(&p);
if(avctx->frame_number == 0) {
esc->fib_acc = buf[1] << 8;
buf_size -= 2;
buf += 2;
}
/* get output buffer */
esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
*data_size = buf_size << 2;
while(buf < buf_end) {
uint8_t d = *buf++;
esc->fib_acc += esc->table[d & 0x0f];
*out_data++ = esc->fib_acc;
esc->fib_acc += esc->table[d >> 4];
*out_data++ = esc->fib_acc;
}
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
dst = esc->frame.data[0];
src = esc->samples + esc->samples_idx;
out_data_size = esc->frame.nb_samples * avctx->channels;
for (n = out_data_size; n > 0; n--)
*dst++ = *src++ + 128;
esc->samples_idx += out_data_size;
return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
(avctx->frame_number == 0)*2 + out_data_size / 2 :
out_data_size;
return consumed;
}
/** initialize 8svx decoder */
static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
return AVERROR_INVALIDDATA;
switch(avctx->codec->id) {
case CODEC_ID_8SVX_FIB:
esc->table = fibonacci;
break;
case CODEC_ID_8SVX_EXP:
esc->table = exponential;
break;
default:
return -1;
}
switch (avctx->codec->id) {
case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
case CODEC_ID_PCM_S8_PLANAR:
case CODEC_ID_8SVX_RAW: esc->table = NULL; break;
default:
av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
avcodec_get_frame_defaults(&esc->frame);
avctx->coded_frame = &esc->frame;
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
av_freep(&esc->samples);
esc->samples_size = 0;
esc->samples_idx = 0;
return 0;
}
AVCodec ff_eightsvx_fib_decoder = {
AVCodec eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_8SVX_FIB,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};
AVCodec ff_eightsvx_exp_decoder = {
AVCodec eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_8SVX_EXP,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};
AVCodec ff_pcm_s8_planar_decoder = {
.name = "pcm_s8_planar",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_PCM_S8_PLANAR,
.priv_data_size = sizeof(EightSvxContext),
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
};

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