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253 Commits

Author SHA1 Message Date
Michael Niedermayer
601e648458 Update for 0.9.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-05 03:02:35 +01:00
John Brooks
e540446f6c vc1dec: fix invalid memory access for small video dimensions
For small video dimensions, these calculations of the upper bound
for pixel access may have a negative result. Using an unsigned
comparison to bound a potentially negative value only works if
the greater operand is non-negative. Fixed by doing edge emulation
when the upper bound is probably negative, everywhere that this
pattern appears.

Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
(cherry picked from commit d209c27b09)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-05 02:23:08 +01:00
John Brooks
fc9c5ad9ea rv34: fix invalid memory access for small video dimensions
For small video dimensions calculations of the upper bound for pixel
access may result in negative value. Using an unsigned comparison
works only if the greater operand is non-negative. This is fixed by
doing edge emulation explicitly for such conditions.

Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
(cherry picked from commit aacf6b3a2f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-05 02:23:02 +01:00
Michael Niedermayer
68f1212696 Revert "rmdec: Avoid allocating huge packets"
This reverts commit 66f71f3b5e.

This causes regressions with RDT.
(cherry picked from commit c4eec85a1f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-05 01:06:13 +01:00
Michael Niedermayer
0a5bca957d lavf: use avg_frame_rate and packet number to exit find_stream_info
qatar does this too but clobbers the AVPacket.duration by approximate
values.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ad8e3304f7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-05 01:06:12 +01:00
Michael Niedermayer
0f61521f69 lavf: revert commit bb99ae3ae9
Author: Michael Niedermayer <michaelni@gmx.at>
	Date:   Thu Nov 3 22:38:10 2011 +0100

	    lavf: fix null pointer dereference in rdt

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

This is no longer needed and causes various problems with RTSP

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1171d938af)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-05 01:06:11 +01:00
Michael Niedermayer
58afa73338 vorbis: make sure ch is non zero before calling vorbis_residue_decode
This possibly makes part of the CVE-2011-3895 fix unneeded.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ff7f198d7f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:19:02 +01:00
Chris Evans
68ee43468e vorbis: An additional defense in the Vorbis codec.
BUG=101458
Review URL: http://codereview.chromium.org/8414025

Fixes second part of CVE-2011-3895

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f74ce3a60d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:19:00 +01:00
Chris Evans
80440c5b1d vorbis: Fix decoder bug.
BUG=101458
Review URL: http://codereview.chromium.org/8413019

This fixes part of 2011-3895

bigned-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 68226ed9ec)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:19:00 +01:00
Michael Niedermayer
89bd49b25c vorbisdec: Make sure blocksize is not set to an invalid value.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 405e99bdfd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:59 +01:00
Michael Niedermayer
d7e5301e43 vorbis: Fix last quarter of CVE-2011-3893
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6fcf2bb8af)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:58 +01:00
Chris Evans
6a3fed8749 ogg: Avoid the possibility to read out-of-bounds of a static global array in Vorbis
decoding.

BUG=100543
Review URL: http://codereview.chromium.org/8365014
This fixes 25% of CVE-2011-3893

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7149fce2ca)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:56 +01:00
Chris Evans
49b8709870 mkv: Fix a bug where a pointer was cached to an array that might later move due to
a realloc()

BUG=100492
Review URL: http://codereview.chromium.org/8366004
Fixes: 1 of 2 for CVE-2011-3893

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f35e037c93)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:55 +01:00
Philippe Saint-Pierre
1bd1103175 Fix possible infinite loop decoding als.
Reviewed-by: Thilo Borgmann
(cherry picked from commit f0f2babca2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:53 +01:00
Michael Niedermayer
66569b375c jpegdec: Fix vlc table check for progressive jpegs.
Fixes Ticket834

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 15b219fae9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:01 +01:00
Michael Niedermayer
9f9b6388a3 electronicarts: Pass error through ea_read_header().
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f37174bc19)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:01 +01:00
Oana Stratulat
32b62c7d6a electronicarts: Fix division by zero. Fixes Ticket #793
Signed-off-by: Oana Stratulat <oanaandreeastratulat@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0d3a51e5d2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-04 22:18:01 +01:00
Carl Eugen Hoyos
9f561ec398 Revert "wavpack: Clip samples after shifting"
This reverts commit 8d055e9079.

The original commit introduced ticket #871, do not import the problem
into the release branch.
2012-01-04 20:34:45 +01:00
Stefano Sabatini
103cb461b0 vf_drawtext: select YUV color for drawbox() in case YUV colorspace is used
Fix box alpha rendering when a YUV colorspace is selected, in particular
fix trac ticket #763.
(cherry picked from commit 1858a5c25e)
2012-01-04 00:25:55 +01:00
Hendrik Leppkes
7e97d98033 wavpack: determine sample_fmt before requesting a buffer
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit 14165fe125)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:53:17 +01:00
Michael Bradshaw
632fd58a8f Added yuva420p decoding support for libopenjpeg
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2ba3416362)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:52:44 +01:00
Michael Bradshaw
68874c42e7 Added RGBA and YUV440 decoding support for libopenjpeg decoder
Reviewed-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3f07ef1dff)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:52:39 +01:00
Michael Bradshaw
a598f0a5d7 Fixed openjpeg decoding bug with width/linesize issue
The original code wasn't taking into account the fact that linesize may not equal the frame's width.  This is to correct that.

Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d1669e5fe3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:52:21 +01:00
Michael Niedermayer
80695c9d1f jpegdec: non interleaved rgb ljpeg support.
Fixes Ticket856

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 45552371e3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:51:48 +01:00
Michael Niedermayer
17c3ec77c2 jpegdec: 9-16 bit yuv/gray ljpeg support.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 465eb0eb48)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:51:42 +01:00
Michael Niedermayer
e8c8b27f66 jpegdec: Only enable rgb mode when there are 3 components.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 24964f21e4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:51:28 +01:00
Justin Ruggles
4eff392a4b bmv audio: implement new audio decoding API
(cherry picked from commit 8893fbdf7a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:50:30 +01:00
Paul B Mahol
e5ae872309 y41p encoder and decoder
y41p is a packed 12-bit 4:1:1 YUV format used by Brooktree.

Fixes issue 1123 / ticket #102.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dfa77dead2)

Conflicts:

	Changelog
	libavcodec/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:50:03 +01:00
Jean First
3fe4055c5f j2kdec: av_log formatting use %tx instead of %x
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b103156235)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:48:40 +01:00
Carl Eugen Hoyos
86fbd610ef Update copyright year.
(cherry picked from commit 8dc973e6d1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:47:12 +01:00
Michael Niedermayer
f6e360770d mpegpsenc: Fix integer overflow of the muxrate calculation.
this should fix the failure of h264-bsf-mp4toannexb on freebsd/openbsd

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 490c97bdf5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:47:01 +01:00
Reimar Döffinger
b8cc9e206b Avoid uninitialized data in lcldec when ofs is 0 in MSZH.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit 42a1f1d7a8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:44:47 +01:00
Michael Niedermayer
4ae111cf71 lavf: ffio_limit(), treat avio_size()==0 as error.
This works around issues arising from inputs that claim to have a
filesize of 0.

Reported-by: buzz_
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e6362f3a3c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:44:34 +01:00
Michael Niedermayer
0bf48d24d7 flvdec: Check avpriv_mpeg4audio_get_config() return
value before using its output.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5500e65342)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:43:43 +01:00
Jean First
2dcacbb68e cache: use av_freep instead of av_free in cache_open
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 576ada791a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:43:26 +01:00
Michael Niedermayer
0064fcb486 mlp_parser: Fix infinite loop with 0 bytes_left.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e146ad95d7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:43:02 +01:00
Reimar Döffinger
a4269652ab Avoid av_memcpy_backptr hang without extra branch.
This only happens for a "back" value of 0 which is invalid anyway,
but lcldec does not properly validate input.
Also extend the documentation to specify valid values.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit 874da652b3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:42:12 +01:00
Michael Niedermayer
3ccbd6b06d ljpegdec: fix point transform injection.
Fix Ticket842

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b0143da806)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:41:59 +01:00
Michael Niedermayer
fa97a8030f flvdec: Check index for being valid
Fixes seeking in Enigma_Principles_of_Lust.flv

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit df0bff6643)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:41:40 +01:00
Michael Niedermayer
53c430415a nsvdec: Check av_malloc(string_size)
This can easily be NULL as string_size can be 2g in a damaged file.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b9e0e9537a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:41:16 +01:00
Michael Niedermayer
f246d46ee6 indeo5: Fix null pointer dereferences of ref_mb
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f41a6c8f3a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:40:58 +01:00
Michael Niedermayer
94773637ba h264_mp4toannexb_filter: pass error code through.
Bug-Found-by and Suggested bugfix: Tanami, Ohad
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7ae251b4d8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:40:49 +01:00
Michael Niedermayer
e1152971a6 issdemux: Check packet_size for validity
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bf5cc80519)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:40:24 +01:00
Michael Niedermayer
7cd9732b33 qpeg: Check for overread in qpeg_decode_intra.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e7c1e38ba6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:40:17 +01:00
Michael Niedermayer
acdc505b2a indeo5: fix division by 0 in ff_ivi_init_tiles()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 92e2b59dec)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:39:40 +01:00
Michael Niedermayer
f09bbd38b0 indeo5: Fix crash due to partially initialized gop vars.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d46bc4133c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:38:58 +01:00
Michael Niedermayer
690fda3ae4 indeo5: fix null pointer crash with ref_mb
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4b35ee0b7c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:38:35 +01:00
Michael Niedermayer
0280cf9aa7 vcr1dec: Check that there is sufficient input data.
Fixes crash.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8e09482e4d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:38:22 +01:00
Michael Niedermayer
a47c277205 v410dec: Check for sufficient input data.
Fixes crash

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 11ca3416f9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:38:16 +01:00
Kostya Shishkov
b0355d3253 Indeo 4 decoder
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
(cherry picked from commit adfe0c942e)

Conflicts:

	Changelog
	libavcodec/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:38:07 +01:00
Carl Eugen Hoyos
e9ce8a4480 Fix a crash when reading gray pam files.
Fixes ticket #837.
(cherry picked from commit 190a0998c3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:36:22 +01:00
Stefano Sabatini
e9378b42b1 lavfi: add missing check in avfilter_filter_samples()
Avoid out-of-buffer data access when nb_channels is 8.
(cherry picked from commit ae21776207)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:35:42 +01:00
Michael Niedermayer
6caca26533 shorten: validate values in fmt chunk search
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5e9a56a035)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:33:21 +01:00
Michael Niedermayer
446d11f5ed shorten: Fix invalid free()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 18bcfc912e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:33:15 +01:00
Michael Niedermayer
a80205ff82 segafilm: fail earlier in case theres not enough bytestream left for a
audio packet.
This prevents a potentially large memory allocation.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1795fed7bc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:32:39 +01:00
Michael Niedermayer
0231a68ecd avio: Fix ffio_limit() when due to seeking past the end less than 0 bytes remain.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 27d323577c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:32:28 +01:00
Michael Niedermayer
7c67d9c6fb golomb: Fix infinite loop in svq3_get_ue_golomb()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 964506bb97)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:32:19 +01:00
Michael Niedermayer
49db360005 ws_snd1: Fix wrong samples count and crash.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5257743aee)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:32:14 +01:00
Michael Niedermayer
42476635ba asfdec: fix endless loop on EOF
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0d4404ed65)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:32:04 +01:00
Michael Niedermayer
462ecdb9bb sierravmd: limit packetsize to the amount that could be read.
Fixes huge allocations.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 47c4713a23)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:31:57 +01:00
Michael Niedermayer
6210d62c5f vmdav: check that theres enough space for a chunk remaining.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2278ecc434)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:31:46 +01:00
Michael Niedermayer
a1666ae8f0 rmdec: Avoid allocating huge packets
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 66f71f3b5e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:31:35 +01:00
Michael Niedermayer
f69045793d mpeg4videodec: Fix division by zero in mpeg4_decode_sprite_trajectory()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fc5c49ab32)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:31:24 +01:00
Michael Niedermayer
adf0c1c70c rpl: Fix near infinite loop in index reading due to missing eof check.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 43abef9fde)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:31:16 +01:00
Michael Niedermayer
3073564a97 j2kdec: Fix crash in get_qcx
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 282bb02839)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:31:06 +01:00
Michael Niedermayer
5e3389b742 j2kdec: Check curtileno for validity
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3eedf9f716)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:30:59 +01:00
Michael Niedermayer
f41e935225 j2kdec: Check for interger overflow in tile array allocation
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3132999fdb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:30:47 +01:00
Michael Niedermayer
897088d604 j2kdec: fix division by zero, check tile dimensions for validity
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 628c9dcca3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:30:32 +01:00
Michael Niedermayer
7dbdf02abc lzo: fix memcpy_backptr() with 0 offset
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 58c41799ab)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:30:25 +01:00
Jean First
f21771260b libopenjpeg: support decoding with bits per pixel greater than 8
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Approved-by: Alex Zhukov
(cherry picked from commit 0d4a77472a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:30:08 +01:00
Michael Niedermayer
a9de82ac17 avio: fix handling of , in urls
Fixes Ticket805

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ad9e0ed170)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:29:21 +01:00
Carl Eugen Hoyos
340fee05f0 Release buffers after encoding svq1.
Fixes ticket #820.
(cherry picked from commit 8e88145d0b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:27:59 +01:00
Carl Eugen Hoyos
a5546736bd Fix a possible endless loop when decoding aac.
Fixes ticket #789.
(cherry picked from commit e5de928923)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:27:49 +01:00
Alexander Strange
3aa89662ce h264: Fix a possible overread in decode_nal_units()
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit bc6a3bd4a5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:27:36 +01:00
Michael Niedermayer
ed0dc01a47 h264: Decode short VUI without error
Fixes 47j9R7PXBep.mov

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ae7a6441df)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:27:02 +01:00
Sergey Radionov
4a4e4b8139 w32thread: call ResetEvent() in pthread_cond_broadcast().
Also add "volatile" to broadcast flag (since it is used from
multiple threads).

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 01eb9805f3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:26:47 +01:00
Michael Niedermayer
33636442e2 udp: support non blocking reads with fifo
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9f50dafe90)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:26:26 +01:00
Michael Niedermayer
3710f0b9ed udp: Replace double select() by select+mutex+cond.
When no data was available both the buffer thread as well as
the main thread would block in select(), when data becomes
available both should move forward and as data is read in the
buffer thread the main thread would block in select() later
the read data was put in the fifo but the main thread still
would be blocked in select() until either the timeout or
another packet would come in.

This is solved in this commit by using a mutex and a condition
variable

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bc900501e0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:26:20 +01:00
Michael Niedermayer
12d0e44bdb udp: return circular buffer error if such error happened.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3dcbafc777)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:26:06 +01:00
Michael Niedermayer
857d64a58c fifo: Make writes atomic.
Prior to this a X bytes write could be seen as less than X bytes being
available if the check was done at an unfortunate moment.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9eb0d8bab1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:25:53 +01:00
Gaurav Narula
a644934b80 kvmc: fix invalid reads
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit ad3161ec1d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:25:47 +01:00
Paul B Mahol
89cf156ace adpcm: fix IMA SMJPEG decoding
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit 01a01bf8bd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:25:03 +01:00
Janne Grunau
4f94de84e8 pthreads: reset active_thread_type when slice thread_init returrns early
(cherry picked from commit f77f640b30)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 22:24:57 +01:00
Justin Ruggles
9665ccda10 ra144enc: zero the reflection coeffs if the filter is unstable
fixes use of uninitialized values if the filter is still unstable after using
the previous frame lpc coefficients.
(cherry picked from commit 9a3f10695a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:53 +01:00
Paul B Mahol
8dce2dd9cc lavf: Add SMJPEG demuxer.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a4209ad631)

Conflicts:

	Changelog
	doc/general.texi
	libavformat/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:53 +01:00
Michael Niedermayer
532731369a ipmovie: fix FPE
Fixed Ticket807
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8d960fbc70)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:53 +01:00
Michael Niedermayer
1c930fe915 proresdec: Check yuv slice data sizes.
Fixes overread
Fixes Ticket812
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2b73cddd40)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:53 +01:00
Justin Ruggles
a25c7081ab truespeech: fix invalid reads in truespeech_apply_twopoint_filter()
fixes Bug 171
(cherry picked from commit f264d336fe)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:53 +01:00
Martin Storsjö
dfdb3ca341 libavcodec: Define a side data type for parameter changes
Also define a codec capability for codecs that can handle
parameters changed externally between decoded packets.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit fe75dc8583)

Conflicts:

	libavcodec/avcodec.h
	libavcodec/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:53 +01:00
Martin Storsjö
c21440637e libavcodec: Define a side data type for new extradata
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 195c123cc8)

Conflicts:

	libavcodec/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Nicolas George
0620b6636e doc: add a list of errno codes on some systems.
(cherry picked from commit 1f797a5826)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Michael Niedermayer
f45a5c60b9 mpeg12dec: reset first_field with picture_structure changes.
Fixes Ticket809
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a40f6a5c69)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Martin Storsjö
6ba87ba6f9 configure: Show whether the safe bitstream reader is enabled
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 9c984b1bd7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Michael Niedermayer
22721b2829 mpegaudiodec: disable unneeded and not working checked reader.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Michael Niedermayer
9bd47bbcc6 mpeg12: disable checked reader.
overreads in mpeg12 are not possible

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 549749c68e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Michael Niedermayer
42b69286de mpeg4: disable checked bitstream reader, overreads are not possible.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 06192e42fd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Michael Niedermayer
e8928f7e97 h263: disable slow checked reader, overreads are not possible in
ffmpegs h263 decoder

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8e7776036b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:52 +01:00
Michael Niedermayer
b1933a1e12 h264: disable checking reader, overreads are not possible
in ffmpegs h264 decoder.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 38331d2036)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:51:44 +01:00
Michael Niedermayer
36e30d7ab9 proresdec2: fix for removed a32 reader
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 21:01:08 +01:00
Ronald S. Bultje
ea199c707c get_bits: introduce safe bitreading to prevent overreads.
When turned on, H264/CAVLC gets ~15% (CVPCMNL1_SVA_C.264) slower for
ultra-high-bitrate files, or ~2.5% (CVFI1_SVA_C.264) for lower-bitrate
files. Other codecs are affected to a lesser extent because they are
less optimized; e.g., VC-1 slows down by less than 1% (all on x86).
The patch generated 3 extra instructions (cmp, cmovae and mov) per
call to get_bits().

The performance penalty on ARM is within the error margin for most
files, up to 4% in extreme cases such as CVPCMNL1_SVA_C.264.

Based on work (for GCI) by Aneesh Dogra <lionaneesh@gmail.com>, and
inspired by patch in Chromium by Chris Evans <cevans@chromium.org>.
(cherry picked from commit 8cfbbd928c)

Conflicts:

	configure

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:59:17 +01:00
Mans Rullgard
09ce851c7e get_bits: remove A32 variant
The A32 bitstream reader variant is only used on ARMv5 and for
Prores due to the larger bit cache this decoder requires.

In benchmarks on ARMv5 (Marvell Sheeva) with gcc 4.6, the only
statistically significant difference between ALT and A32 is
a 4% advantage for ALT in FLAC decoding.  There is thus no (longer)
any reason to keep the A32 reader from this point of view.

This patch adds an option to the ALT reader increasing the bit
cache to 32 bits as required by the Prores decoder.  Benchmarking
shows no significant change in speed on Intel i7.  Again, the
A32 reader fails to justify its existence.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit a1e98f198e)

Conflicts:

	libavcodec/get_bits.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:58:22 +01:00
Tomas Härdin
5c2097cff7 mxfdec: Sanity check PreviousPartition
Without this certain files could get the demuxer stuck in a loop
(cherry picked from commit 46d65fb8a5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:52:21 +01:00
Tomas Härdin
6a8826abf4 mxfdec: Move the current_partition check inside mxf_read_header()
This fixes SIGSEGV on files where this is actually the case, such as zzuf4.mxf
(cherry picked from commit 184f479096)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:51:52 +01:00
Tomas Härdin
46e4dd66da mxfdec: Check url_feof() in mxf_read_local_tags()
This fixes the infinite loop with zzuf2.mxf
(cherry picked from commit 2116e4ba91)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:50:30 +01:00
Tomas Härdin
7e35c3e177 mxfdec: Check for NULL component
This fixes SIGSEGV with zzuf1.mxf
(cherry picked from commit 5e5e69d078)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:50:20 +01:00
Sergey Radionov
0f38b11939 w32threads: wait for the waked thread in pthread_cond_signal.
This fixes a deadlock VLC triggered with multithreaded decoding. The
wait forces one of the current waiters to wake and not the thread
which calls pthread_cond_signal() itself.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 2003a29284)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:49:52 +01:00
Michael Niedermayer
fd552756d9 avio: remove ENOTSUP mention in doxy as its not available on all supported platforms
and thus cannot be used

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 077b086dea)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:49:24 +01:00
Luca Barbato
8056d020ad fate: account for NUT 10bit support
Some libavifilter tests use NUT as output even if the produced
files were not decodable. The support for 10bit introduced in
432f0e5b7d and 91b1e6f0c changed the hashes.
(cherry picked from commit 9f3c77dd2a)

Conflicts:

	tests/ref/lavfi/pixdesc
	tests/ref/lavfi/pixfmts_copy
	tests/ref/lavfi/pixfmts_null
	tests/ref/lavfi/pixfmts_scale
	tests/ref/lavfi/pixfmts_vflip

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:48:48 +01:00
Michael Niedermayer
d20e948f07 indeo3: Fix null ptr dereference
Fixes Ticket804
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 134aaa79f7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:01:50 +01:00
Michael Niedermayer
acbdee0a2b golomb: fix end of bitstream check
Fixes Ticket806
Bug found by Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fc739b79ad)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:01:45 +01:00
Rafaël Carré
fa86884dbc lavc: always align height by 32 pixel
Interlaced content for most codec requires it.
This patch is a stop-gap pending a serious rework to support
codecs with non 16 pixel macroblocks.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 37c0dc626d)

Conflicts:

	libavcodec/utils.c

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:01:13 +01:00
Luca Barbato
3d207417fb raw: add 10bit YUV definitions
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 91b1e6f0c6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:00:13 +01:00
Luca Barbato
cb75ed3014 nut: support 10bit YUV
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 432f0e5b7d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 20:00:05 +01:00
Michael Niedermayer
41ff9350e9 indeo5: Fix null pointer dereference
Fixes Ticket803
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f27930cd9a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:59:33 +01:00
Michael Niedermayer
f8320fd344 ff*: Fix duplicatedly printed version info with -version
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 452406bd5b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:59:26 +01:00
Aneesh Dogra
ae26aab2c3 bytestream: add a new set of bytestream functions with overread checking
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit fd22616c59)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:58:41 +01:00
Alex Converse
7a8fc9906f mpegts: Suppress invalid timebase warnings on DMB streams.
timestamp_len and timestamp_res intialize to zero.
(cherry picked from commit 41bdd4adc4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:58:16 +01:00
Alex Converse
c2507fb986 mpegts: Fix typo in handling sections in the PMT.
This was an error rebasing 4682a1dc3a for
commit. The "pes" variable guaranteed to be NULL in that block.
(cherry picked from commit d9b89b23df)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:56:57 +01:00
Michael Niedermayer
1db39228c6 roq: fix excessive memory alloc
Fixes Ticket802
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c1acd58198)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:56:16 +01:00
Michael Niedermayer
d8d25f143f alac: Check for bitstream overread
Fixes Ticket801
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a8469223f6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:56:06 +01:00
Michael Niedermayer
c550d56414 tm2: Check remaining size before init_get_bits()
Fixes a null pointer dereference.
Fixes 2nd half of Ticket800
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 65f0f9183b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:56:01 +01:00
Michael Niedermayer
d561025e49 avidec: move eof check before continue.
Fixes a infinite loop
Fixes half of Ticket800
Bug found by Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3c7f75bd84)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:55:54 +01:00
Stefano Sabatini
61fb7e17a2 lavd/lavfi: add check in case pix_fmts cannot be allocated in lavfi_read_header()
(cherry picked from commit 4d475f7eb3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:55:10 +01:00
Michael Niedermayer
5750a95f8b swr: fix assert failure
Bug found by: Oana Stratulat
GCI-id: 7211207

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit aa1c590b29)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:54:33 +01:00
Michael Niedermayer
a1a079dce4 siff: simplify code by using ffio_limit()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b08a742526)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Aneesh Dogra
2cdd89a561 xl: Fix overreads
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 00aad121d8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
c0c5189e66 soxdec: check av_malloc return.
Bug found by: durandal_1707

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1dcce49e10)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
125894f15f eamad: pass & check errors
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1d0ae92a25)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
54cfb532a0 eamad: fix excessive memory allocation.
Fixes Ticket792
Bug found by Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 03a9c9932d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
e9a7ab20be swscale: Fix several (real world) integer overflows.
Fixes Ticket795
Bug Found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 90cad5edf6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
0ab9072ee0 dsicin demuxer: Fix excessive malloc()
use ffio_limit()
Fixes Ticket 790
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9c5104e84e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
3461d27506 lavf: split out ffio_limit()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e39eeb10d1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:56 +01:00
Michael Niedermayer
9c338b7c4c tqi: Fix passing errors from the MB decode. This silences some valgrind warnings.
Fixes second half of Ticket794
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f85334f58e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
25cadd366b adpcm: Fix crash
Fixes half of Ticket794
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 608708009f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
f57c629b18 proresdec: add UPDATE_CACHE, should fix decoding with -Os
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 29179239f5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
0815a2907a rl2demux: Fix FPE
Fixes Ticket788
Bug found by Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f371396dfb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Paul B. Mahol
d1e1674f66 cljr: fix buf_size sanity check
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
(cherry picked from commit 2e7905eee8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
21ca96049f siff: Fix excessive memory allocation.
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1c73391d7d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
9f1dde03d5 mov: detect EOF in mov_read_dref()
This fixes a (near) infinite loop and Ticket783
Bug found by: cosminamironesei

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit af3f2a87da)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
d0269db8e3 adx_parser: rewrite.
The previous code ended in multiple different infinite
loops. See stl_ten_1_big.sfd as example with and without zzuf

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7859740c6a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
cb42ed171d adxdec: Fix division by zero
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6b6b84ae16)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
2bbda9872f adxdec: Do not require extradata.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bdd62a615a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Michael Niedermayer
9dd420ede1 adx_parser: Fix infinite loop.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3bf1d787b5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:49:55 +01:00
Carl Eugen Hoyos
a1974d1ea0 Assume architecture arm for machine hardware name iPad*.
uname -m returns for example "iPad2,1".
(cherry picked from commit b6b0067511)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:15:27 +01:00
Carl Eugen Hoyos
bf58e5422d Fix possible endless loop when decoding amr.
Fixes bug 151.

Reviewed-by: Vitor Sessak
(cherry picked from commit 290e7eb77b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:10:33 +01:00
Michael Niedermayer
88eaab2827 motion_est: avoid generating motion vectors that point between widthxheight and mb_widthxmb_height.
No difference in PSNR or bitrate in the printed precission with the matrix lobby scene at 322x242

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 56bf2c2a19)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:09:45 +01:00
Michael Niedermayer
0c72d64160 lagarith: rgb24 support
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0db7b30700)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:09:20 +01:00
Michael Niedermayer
1bb565360c msvideo1enc: Check that dimensions are a multiple of 4.
Fixes Ticket784
Found by: ami_stuff

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e09ffa4524)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:08:44 +01:00
Shitiz Garg
1acf921d6f adpcm: Check for channels to be a non-zero integer
channels would be 0 sometimes and would cause floating point exception
Fixes bugzilla #124

Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
(cherry picked from commit e614fac2e6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:07:30 +01:00
Michael Niedermayer
7bf3f79502 lavc: Check codec_ids against insertion/removial typos
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 83b46713f0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:05:38 +01:00
Michael Niedermayer
e3316723b6 avutil: check sampleformats enum, pict type enum and media type enum against insert/remove mistakes.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 75f2cc5134)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:05:32 +01:00
Michael Niedermayer
17d82125c5 pixfmts: Selfcheck against insertions / deletions in the enum.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 68230eb375)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:05:23 +01:00
Michael Niedermayer
513b0f4872 aacdec: Fix null pointer crash
Fixes bug170
Bug found by: Gautam Gupta

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f37b2d5a68)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:05:12 +01:00
Michael Niedermayer
cb2d66a470 avienc: Fix rawvideo 32bit
Bug found by: Rik Maes <rik.maes54@gmail.com>
Change based on suggestion by Rik Maes

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 53b6503184)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:04:56 +01:00
Michael Niedermayer
8c2fe23b1c 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>
2012-01-03 19:04:45 +01:00
Derek Buitenhuis
c2aa7b70d4 wavpack: Fix 32-bit clipping
In the case that (frame_flags & 0x03) == 3, hybrid_maxclip
may have had a signed integer overflow.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit bb9747c8ee)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:04:14 +01:00
Derek Buitenhuis
8d055e9079 wavpack: Clip samples after shifting
It doesn't make much sense to clip pre-shift,
nor is it correct for proper decoding.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 365e1ec26d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:04:11 +01:00
Michael Niedermayer
4f23f24e30 CODEC_ID_SOL_DPCM: Fix used write buffer.
Bug found by: Oana Stratulat

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6d8e6fe9db)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:03:39 +01:00
Shitiz Garg
34cee5bd37 cljr: Check if width or height are positive integers
width and height might get passed as 0 and would cause floating point
exceptions in decode_frame.
Fixes bugzilla #149

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4af0262f7d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:03:05 +01:00
Michael Niedermayer
95234da75e ffmpeg: Fix killing [Y/n] prompt with ctrl-c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7b92863f30)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:02:52 +01:00
Michael Niedermayer
cb7579ae6f ffmpeg: exit() on repeated ctrl-c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cf7076ee96)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:02:43 +01:00
Michael Niedermayer
5341734b03 h264: return the consumed amountg in case of Q264
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3d07e0aba0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:02:32 +01:00
Michael Niedermayer
3c7ba39d09 h264: retuen the amount read in case of NAL_END_SEQUENCE
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 227960427b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:02:26 +01:00
Michael Niedermayer
7c98d284ff h264: reset nal_unit_type so that decoding frames without any nal units
dont leave its value at something random.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 902c090413)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:02:17 +01:00
Michael Niedermayer
c55b158b87 mtvdemuxer: fix segfault caused by truncated packets.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f890cb948c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:02:00 +01:00
Michael Niedermayer
5ca4ec76b7 lavf: print an error if a packet has been truncated due to filesize
in av_get_packet()

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ee181f84a3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:01:49 +01:00
Michael Niedermayer
813c05fcd2 txd/westwood: remove demuxer specific overallocate solutions as the new generic code
handles it fine.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 207d9eab5a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:01:40 +01:00
Michael Niedermayer
dd5d2de5ec lavf: Update AVIOContext.maxsize when hitting the end.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 559ae20dda)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 19:01:30 +01:00
Michael Niedermayer
c40296d1ad avidec: Check that the header chunks fit in the available filesize.
Fixes Ticket771
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 62adc60b97)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:29:02 +01:00
Michael Niedermayer
cd5d5bf45c txddemux: Limit allocated packets to filesize.
Fixes Ticket772
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f72601d063)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:28:54 +01:00
Michael Niedermayer
aed8e8f9f3 thp: simplify overallocate checks.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a000975444)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:28:45 +01:00
Michael Niedermayer
349cca9f12 lavf: add internal AVIOContext.maxsize
This allows simple and generic limiting of allocations used for packets.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 47572323f2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:28:38 +01:00
Michael Niedermayer
58b65c1241 thpdemux: Check that packed sizes are within the files data size and filesize.
Fixes Ticket774
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 482ee63641)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:28:26 +01:00
Michael Niedermayer
44e7ef15ae avidec: Fix infinite loop caused by rounding of timestamps in non interleaved avis.
Fixes Ticket775
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e098fba5d9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:28:15 +01:00
Michael Niedermayer
e7cb6e1c04 j2kdec: Fix integer overflow leading to a segfault
Fixes Ticket776
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1f99939a63)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:27:59 +01:00
Michael Niedermayer
4af03698f3 smackerdemuxer: check some values before instead of just after malloc()
Fixes Ticket777
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c402c1c976)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:27:48 +01:00
Michael Niedermayer
8af76473b3 westwooddemux: dont require avio_size() functionality.
Found by reimar

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e2a16e82b1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:27:36 +01:00
Carl Eugen Hoyos
7d96f764d6 Fix typo in v210 decoder options.
(cherry picked from commit 56669837ce)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:26:57 +01:00
Michael Niedermayer
b32554175c ffmpeg: check return code from av_vsrc_buffer_add_frame()
Fixed Ticket770
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1f273c2bf2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:26:39 +01:00
Michael Niedermayer
134b7f57db smacker: Check for overread in smka_decode_frame()
Fixes a segfault with wetlog_fail.smk
Bug found by: Shitiz Garg

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d0f7927177)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:26:09 +01:00
Michael Niedermayer
0177ac9637 lavf: Fix fps misdetection when codec_info_duration is 0 and only a few frames are used to estimate fps.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d19ca77f42)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:25:38 +01:00
Michael Niedermayer
17044e1a2b mov: Fix near infinite loop in stsd parsing.
Fixes Ticket768
Bug found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5f95c130a0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:25:27 +01:00
Michael Niedermayer
19babec06f westwooddemuxer: use av_get_packet()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4255d6d96c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:25:19 +01:00
Michael Niedermayer
02798736bf westwooddemuxer: Fix 1gb alloc
Fixes Ticket765
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f68b19fc28)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:25:11 +01:00
Carl Eugen Hoyos
56e2190e10 Fix linesize for bmp with bitdepth < 8.
Fixes ticket #751.
(cherry picked from commit 6f2054c589)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:25:03 +01:00
Michael Niedermayer
54aa78adaa h264: fix midstream extardata handling with dump.f4v
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a4755df523)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:24:36 +01:00
Michael Niedermayer
c575d63ac2 ffmpeg: reset dts/pts after decoding the first subpacket
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6c39b3c278)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:24:04 +01:00
Martin Storsjö
3f1e77a44c flvenc: Support muxing 16 kHz nellymoser
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit bd2ff1a8bc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:23:17 +01:00
Mans Rullgard
923deca05f v410enc: fix output buffer size check
The encoder clearly needs width * height * 4 bytes.

Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit dabba0c676)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:22:26 +01:00
Michael Niedermayer
d1f74d838e jpegdec: check return value of mjpeg_decode_dc()
Fixes Ticket754

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 484b1cdd53)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:20:43 +01:00
Michael Niedermayer
07892e33ef h264: Fix build with --disable-everything --enable-decoder='h264' --disable-optimizations
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d998a6ddee)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:20:34 +01:00
Shitiz Garg
0c62a00e68 westwood: Make sure audio header info is present when parsing audio packets
Audio header information might get scrambled and would not parse,
yet wsqva_read_packet would try to parse audio packets causing
segfaults such as floating point exception.

Fixes bugzilla #141.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 566ee0eaf1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:19:40 +01:00
Martin Storsjö
1804becb30 libgsm: Reset the MS mode of GSM in the flush function
The mode is set in libgsm_decode_init, but the decoder
object is simply destroyed and recreated in the flush
function - therefore the mode has to be set again.

This fixes playback using the libgsm_ms decoder in avplay.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 1b35af3225)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:19:22 +01:00
Martin Storsjö
a11c9d7778 libgsm: Set options on the right object
This fixes regressions in decoding using the libgsm_ms decoder,
broken since 0eea212943.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 8637af8d24)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:18:39 +01:00
Michael Niedermayer
1d23f5bbef h263dec: Fix regression / crash with lowres.
Fixes Ticket757
Bug Found by: ami_stuff

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 21270cffae)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:18:13 +01:00
Michael Niedermayer
c0ff6b6378 flicvideo: input buffer pointer checks.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 30b996d443)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:18:01 +01:00
Michael Niedermayer
6a805e5334 flicvideo: fix overread.
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit afb2bac48d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:17:50 +01:00
Michael Niedermayer
6b7a14e599 aacsbr: Fix memory corruption.
Fixes Ticket760 and Ticket761
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 944f5b2779)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:17:42 +01:00
Michael Niedermayer
b6c19c9c91 h264: check chroma_format_idc range.
Fixes Ticket758
Bug found by: Diana Elena Muscalu

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

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:17:26 +01:00
Michael Niedermayer
e8bd1af095 indeo3: check return values of allocate_frame_buffers()
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cd645c15d8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:17:07 +01:00
Michael Niedermayer
09453e6bd3 mtv: Fix FPE with 0 dimensions.
Fixes Ticket755
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8b9b6332df)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:17:02 +01:00
Martin Storsjö
bf9af661b8 applehttp: Properly clean up if unable to probe a segment
This avoids a segfault if the probe function wasn't able to
determine the format.

The bug was found by Panagiotis H.M. Issaris.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 115a57302a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:12:33 +01:00
Martin Storsjö
63d5fa55fe applehttp: Avoid reading uninitialized memory
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit c41b9842ce)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:09:56 +01:00
Carl Eugen Hoyos
d344e45f19 Detect 'yuv2' as rawvideo also in avi.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 18:09:28 +01:00
Michael Niedermayer
485d088f94 fate: fix v410 on big endian
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a11e7ffba)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:57:50 +01:00
Derek Buitenhuis
84f3bdb8f4 fate: Add FATE tests for v410 encoder and decoder
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 64a0b826e1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:57:43 +01:00
Derek Buitenhuis
93442a27c5 v410 encoder and decoder
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 309ffa4fe3)

Conflicts:

	libavcodec/version.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:57:26 +01:00
Michael Niedermayer
bfe7d96ed2 MAINTAINERS: add 0.9
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 635a301683)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:56:28 +01:00
Reimar Döffinger
97e6a0fa2a Support "VLB " audio in NSV files.
See samples/nsv/vlb_audio.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit 36be045ed7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:55:05 +01:00
Michael Niedermayer
9e4d30397b decoding example: reset pts/dts after subpacket.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0e187fd13c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:52:37 +01:00
Gaurav Narula
399e5b675b ulti: Fix invalid reads
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit a99273ebf3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:52:21 +01:00
Michael Niedermayer
0a2fbb0a84 msrledec: Check for overreads
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 53be37e368)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:51:52 +01:00
Michael Niedermayer
b4ad641334 aascdec: Check input buffer size on raw data.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit be5db7004f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:51:42 +01:00
Anton Khirnov
60cb07196f lavf: add avformat_close_input().
It sets the supplied AVFormatContext pointer to NULL after freeing it,
which is safer and its name is consistent with other lavf functions.

Also deprecate av_close_input_file().
(cherry picked from commit 526604545f)

Conflicts:

	doc/APIchanges
	libavformat/utils.c
	libavformat/version.h
2012-01-03 17:47:08 +01:00
Michael Niedermayer
c30e59383b cljrdec: improve scaling somewhat
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7e5cbb3c2d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:20:40 +01:00
Michael Niedermayer
b9e09c9eab cljrenc: fix scaling
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a1ca45317d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 17:20:31 +01:00
Clément Bœsch
dfdf616773 mp3enc: fix playback with mp3 encoded files on some devices.
This partially reverts 1fea432b; header variable needs to keep its value
in case it triggers "if (needed <= c.frame_size)".
(cherry picked from commit e1a8164d62)

Conflicts:

	libavformat/mp3enc.c
2012-01-03 15:07:45 +01:00
Michael Niedermayer
46d0aee3c4 lavc: Do not fail audio decoding if user has set a custom get_buffer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c3846e3eba)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-03 15:07:45 +01:00
Carl Eugen Hoyos
55e2dc7f77 Fix channel order for some less common TrueHD layouts.
Reported and tested by TDiTP_

Fixes ticket #779.
(cherry picked from commit 1af0ace3a4)
2012-01-03 14:05:30 +01:00
Hendrik Leppkes
a46fa584c5 mlpdec: fix channel order for wide 7.1 truehd layouts
(cherry picked from commit b9d8af0327)
2012-01-03 14:05:21 +01:00
Carl Eugen Hoyos
980e4263bc Add missing PIX_FMT_NONE to libopenjpegenc
fixes ticket #813

Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Reviewed-by: Jean First <jeanfirst@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ef611095f0)
2012-01-03 13:54:11 +01:00
Michael Niedermayer
2a442eb70f h264: Fix frame sync / random access handling.
It seems I have misunderstood the spec when I implemented this
originally.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9ba43e50ef)
2012-01-03 13:34:37 +01:00
Michael Niedermayer
23ed56d307 h264: Fix seeking in 00017.MTS
Fixes Ticket852

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9d9316ae34)
2012-01-03 13:34:30 +01:00
Michael Niedermayer
b84683a906 h264: Fix recovery_frame initialization when recovery_frame is -1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fba6777363)
2012-01-03 13:34:27 +01:00
Michael Niedermayer
01ff5378f0 h264: reset picture->sync on allocation.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 36fa781bd6)
2012-01-03 13:34:23 +01:00
Michael Niedermayer
96545777d7 h264: add forgotten \n to "Invalid mix of idr and non-idr slices"
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ecb14b8af7)
2012-01-03 13:34:15 +01:00
Michael Niedermayer
1506becf36 h264: Print pass in NAL debug output
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 393253b4b3)
2012-01-03 13:26:56 +01:00
Tom Evans
f7c196a1f9 libavdevice: Fix v4l2 compilation under FreeBSD
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ffa1f541b3)
2011-12-28 02:10:31 +01:00
Michael Niedermayer
7614facf8a h264: fix init of topleft ref/mv.
Fixes Ticket778

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4d70023a2a)
2011-12-18 02:10:59 +01:00
Tomas Härdin
35a42ca0bf mxfdec: Only use the new demuxing logic for OPAtom
This is a stopgap for ticket #746.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:35 +01:00
Michael Niedermayer
de69052b1a ljpeg: Check that lowres is 0 as lowres is not possible with ljpeg.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Michael Niedermayer
bf928ebed2 4xm: check if there are bits left in decode_i_block()
Fixed Ticket753
Bug Found by: Diana Elena Muscalu

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e1ba29c764)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Michael Niedermayer
f52324629a riff: Fix freeing of random value.
Fixes Ticket752
Bug-found-by: Diana Elena Muscalu
(cherry picked from commit 76b9a0961c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Michael Niedermayer
eee78ef30d ffplay: clear pkt_temp when pkt is freed.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit efe8a1ba08)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Michael Niedermayer
f600d709f0 ffplay: Fix got_frame type.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit a5a1e3cb8a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Panagiotis H.M. Issaris
2a66a65341 ffplay: add 10 minute seek support to ffplay
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 91a3ea671a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Marton Balint
964d75735f ffplay: force setting video mode on fullscreen toggle
Fixes ticket #736. Based on the patch by glangshaw.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 9fb2b41258)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Matthieu Bouron
6db6fe82ca mpeg12enc: fix drop_frame_timecode value
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 366605ff5d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Michael Niedermayer
388e3e7853 ac3dec: update checked AV_EF flags.
A user who wishes to use default error concealment cannot set the
AV_EF_CRCCHECK flag because not every CRC in every format is a
reliable indicator of bitstream damage. In some formats crcrs
can be nonsensical in absence of any damage. We thus add the
AV_EF_CAREFUL flag in addition to the AV_EF_CRCCHECK flag to
allow a user to enable this reliable CRC check without having to
enable all CRC checks in all formats.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit eda3758c87)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-16 15:21:19 +01:00
Carl Eugen Hoyos
866d5c958f Export ff_vdpau_vc1_decode_picture().
XBMC's configure script checks for this function in installed
libavcodec.so to determine VDPAU support.

Fixes ticket #762 reported by Christian Marillat
(cherry picked from commit 9994643fdd)
2011-12-16 02:36:51 +01:00
Carl Eugen Hoyos
238c7ffd7d Do not reject LIST tags in wav with size 4.
Fixes ticket #745.
(cherry picked from commit a8e93ce4cb)
2011-12-14 15:18:11 +01:00
Clément Bœsch
9aaf250fb3 lavfi: skip the frame in case of error.
This fixes playback with some audio files through filters.
(cherry picked from commit 220481e1d6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 23:38:02 +01:00
Michael Niedermayer
834aef341d cleanup: for(int
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 16b771b1f4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:47:01 +01:00
Michael Niedermayer
657737e58d clenaup: #ifdec CONFIG*
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b39e01a6e8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:46:59 +01:00
Michael Niedermayer
b92ef485c7 Fix all GNU %Ld
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9aeaf62903)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:46:45 +01:00
Michael Niedermayer
6289b6c811 twinnvq: remove ;;
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 35f707abe1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:46:43 +01:00
Michael Niedermayer
849ee7ee82 build: Disable avconv more completely while leaving the user the option to enable it
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c83ef636a6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:46:26 +01:00
Tomas Härdin
5693443446 mxfdec: Check slice_count before freeing slice_offset_entries
This fixes SIGSEGV on sliceless (single stream) VBR files

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 89cc8c17df)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:33:24 +01:00
Tomas Härdin
57111c558e mxfdec: Bring back accumulated_offset that was removed in 5e67e3e
I thought it had to do with file offsets, but's actually the offset inside
the essence container.
In other words, unbreak multiple EditUnitByteCounts.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c0ee522df6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 21:33:15 +01:00
Michael Niedermayer
e6a5de4421 ffmpeg: fix codecids with ffserver + ffmpeg + ffm
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ee731c1ab0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-12 18:08:37 +01:00
Michael Niedermayer
b8c41e5c08 fate: replace forgotten avconv by ffmpeg in vsynth / asynth stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 02a01c2fa1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-12 02:05:30 +01:00
Michael Niedermayer
7881dad2b2 seek-test: use standard format specifiers in snprintf()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cd2f7ed000)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-12 02:05:12 +01:00
Michael Niedermayer
9f8d8c57fb update for 0.9
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-11 21:24:41 +01:00
5123 changed files with 294032 additions and 677119 deletions

1
.gitattributes vendored
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@@ -1 +0,0 @@
*.pnm -diff -text

129
.gitignore vendored
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@@ -1,88 +1,53 @@
*.a
.config
.version
*.o
*.d
*.def
*.dll
*.dylib
*.exe
*.exp
*.gcda
*.gcno
*.h.c
*.ilk
*.lib
*.pc
*.pdb
*.so
*.so.*
*.swp
*.ver
*.ho
*-example
*-test
*_g
/.config
/.version
/ffmpeg
/ffplay
/ffprobe
/ffserver
/config.*
/coverage.info
/doc/*.1
/doc/*.3
/doc/*.html
/doc/*.pod
/doc/config.texi
/doc/avoptions_codec.texi
/doc/avoptions_format.texi
/doc/doxy/html/
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding
/doc/examples/extract_mvs
/doc/examples/filter_audio
/doc/examples/filtering_audio
/doc/examples/filtering_video
/doc/examples/metadata
/doc/examples/muxing
/doc/examples/pc-uninstalled
/doc/examples/remuxing
/doc/examples/resampling_audio
/doc/examples/scaling_video
/doc/examples/transcode_aac
/doc/examples/transcoding
/doc/fate.txt
/doc/print_options
/lcov/
/libavcodec/*_tablegen
/libavcodec/*_tables.c
/libavcodec/*_tables.h
/libavutil/avconfig.h
/libavutil/ffversion.h
/tests/audiogen
/tests/base64
/tests/data/
/tests/pixfmts.mak
/tests/rotozoom
/tests/tiny_psnr
/tests/tiny_ssim
/tests/videogen
/tests/vsynth1/
/tools/aviocat
/tools/ffbisect
/tools/bisect.need
/tools/crypto_bench
/tools/cws2fws
/tools/fourcc2pixfmt
/tools/ffescape
/tools/ffeval
/tools/ffhash
/tools/graph2dot
/tools/ismindex
/tools/pktdumper
/tools/probetest
/tools/qt-faststart
/tools/trasher
/tools/seek_print
/tools/uncoded_frame
/tools/zmqsend
*.def
*.dll
*.lib
*.exp
config.*
doc/*.1
doc/*.html
doc/*.pod
doxy
ffmpeg
ffplay
ffprobe
ffserver
avconv
libavcodec/*_tablegen
libavcodec/*_tables.c
libavcodec/*_tables.h
libavcodec/codec_names.h
libavcodec/libavcodec*
libavcore/libavcore*
libavdevice/libavdevice*
libavfilter/libavfilter*
libavformat/libavformat*
libavutil/avconfig.h
libavutil/libavutil*
libpostproc/libpostproc*
libswresample/libswresample*
libswscale/libswscale*
tests/audiogen
tests/base64
tests/data
tests/rotozoom
tests/tiny_psnr
tests/videogen
tests/vsynth1
tests/vsynth2
tools/cws2fws
tools/graph2dot
tools/lavfi-showfiltfmts
tools/pktdumper
tools/probetest
tools/qt-faststart
tools/trasher
version.h

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

59
CREDITS
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@@ -1,6 +1,55 @@
See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
get the names of people who have contributed to FFmpeg.
This file contains the names of some of the people who have contributed to
FFmpeg. The names are sorted alphabetically by last name. As this file is
currently quite outdated and git serves as a much better tool for determining
authorship, it remains here for historical reasons only.
To check the log, you can type the command "git log" in the FFmpeg
source directory, or browse the online repository at
http://source.ffmpeg.org.
Dénes Balatoni
Michel Bardiaux
Fabrice Bellard
Patrice Bensoussan
Alex Beregszaszi
BERO
Thilo Borgmann
Mario Brito
Ronald Bultje
Alex Converse
Maarten Daniels
Reimar Doeffinger
Tim Ferguson
Brian Foley
Arpad Gereoffy
Philip Gladstone
Vladimir Gneushev
Roine Gustafsson
David Hammerton
Wolfgang Hesseler
Marc Hoffman
Falk Hueffner
Aurélien Jacobs
Steven Johnson
Zdenek Kabelac
Robin Kay
Todd Kirby
Nick Kurshev
Benjamin Larsson
Loïc Le Loarer
Daniel Maas
Mike Melanson
Loren Merritt
Jeff Muizelaar
Michael Niedermayer
François Revol
Peter Ross
Måns Rullgård
Stefano Sabatini
Roman Shaposhnik
Oded Shimon
Dieter Shirley
Konstantin Shishkov
Juan J. Sierralta
Ewald Snel
Sascha Sommer
Leon van Stuivenberg
Roberto Togni
Lionel Ulmer
Reynaldo Verdejo

833
Changelog
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@@ -1,822 +1,11 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 2.4.10:
- diracdec: check if reference could not be allocated
- diracdec: avoid overflow of bytes*8 in decode_lowdelay
- diracdec: prevent overflow in data_unit_size check
- avformat/matroskadec: Use tracks[k]->stream instead of s->streams[k]
- matroskadec: check s->streams[k] before using it
- avcodec/ffv1dec: Check chroma shift parameters
- matroskadec: use uint64_t instead of int for index_scale
- avcodec/wavpack: Check L/R values before use to avoid harmless integer overflow and undefined behavior in fate
- nutdec: fix illegal count check in decode_main_header
- nutdec: check for negative frame rate in decode_info_header
- ffmpeg: remove incorrect network deinit
- OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c
- apedec: set s->samples only when init_frame_decoder succeeded
- OS/2:Makedef.cmd cleanup
- avcodec/golomb: get_ur_golomb_jpegls: Fix reading huge k values
- avformat/mov: Fix parsing short loci
- avformat/mov: Print reason of loci parsing failure
- avcodec/hevc: Fix typo in num_entry_point_offsets check
- avcodec/libtheoraenc: Check for av_malloc failure
- avcodec/dcadec: Check active_bands
- avcodec/dcadec: Check scale table index
- avcodec/sonic: More completely check sample_rate_index and channels
- avcodec/sonic: check memory allocations
- avcodec/smvjpegdec: check avcodec_decode_video2() return code
- avcodec/shorten: More complete pred_order check
- avcodec/shorten: Check skip_bytes()
- png: Set the color range as full range
- mpegts: Update the PSI/SI table only if the version change
- rtsp: Make sure we don't write too many transport entries into a fixed-size array
- rtpenc_jpeg: Handle case of picture dimensions not dividing by 8
- libvpx: Fix mixed use of av_malloc() and av_reallocp()
- aacpsy: correct calculation of minath in psy_3gpp_init
- avcodec/shorten: Fix code depending on signed overflow behavior
- avcodec/proresdec2: Reset slice_count on deallocation
- ffmpeg_opt: Fix -timestamp parsing
- avcodec/dcadec: Check subsubframes
- avcodec/dcadec: Check nchans
- hevc: make avcodec_decode_video2() fail if get_format() fails
- avcodec/cavsdec: Check esc_code
- avcodec/on2avc: Check run more carefully
- avcodec/mpeg4audio: add some padding/alignment to MAX_PCE_SIZE
- swr: fix alignment issue caused by 8ch sse functions
- avcodec/mjpegdec: fix len computation in ff_mjpeg_decode_dqt()
- avcodec/jpeg2000dec: fix boolean operator
- avcodec/hevc_ps: Explicitly check num_tile_* for negative values
- avcodec/hevc_ps: Check vps_num_hrd_parameters
- avcodec/hevc_ps: More completely check vps_num_layer_sets
- avcodec/hevc: Check num_entry_point_offsets
- avcodec/hevc: Check offset_len
- libswscale/x86/hscale_fast_bilinear_simd.c: Include BX in the clobber list on x86_64, because it isn't implicitly included when PIC is on.
- avcodec/hevc_sei: Check num_sps_ids_minus1 value
- avcodec/vqavideo: Check chunk size
- aacdec: don't return frames without data
- avformat/hevc: Check num_negative_pics and num_positive_pics
- avformat/hevc: Check cpb_cnt_minus1
- avformat/matroskadec: Cleanup error handling for bz2 & zlib
- avformat/nutdec: Fix use of uinitialized value
- avformat/rtpenc_jpeg: Check remaining buffer size for SOS
- avformat/rtpdec_xiph: Check upper bound on len in xiph_handle_packet()
- tools/graph2dot: use larger data types than int for array/string sizes
- avformat/matroskaenc: Check ff_vorbiscomment_length in put_flac_codecpriv()
- avformat/oggenc: Check ff_vorbiscomment_length in ogg_write_vorbiscomment()
- avformat/flacenc: Check length in flac_write_block_comment()
- avformat/subtitles: Use size_t for len
- avformat/url: Use size_t for len from strlen()
- avutil/avstring: Use size_t in av_strlcatf()
- avformat/vorbiscomment: Check entry length in ff_vorbiscomment_write()
- avutil/dict: Use size_t for appending strings
- libavutil/mem: use size_t for the length in av_strdup()
- aacsbr: break infinite loop in sbr_hf_calc_npatches
- diracdec: check that block length is valid
- ffmpeg_opt: Set the video VBV parameters only for the video stream from -target
- avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits()
- avcodec/put_bits: Assert that there is enough space left in skip_put_bytes()
- avcodec/mpegvideo_enc: Update the buffer size as more slices are merged
- avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size()
- avformat/wavdec: Increase dts packet threshold to fix more misdetections
- avformat/wavdec: Increase probe_packets limit
- nutdec: abort if EOF is reached in decode_info_header/read_sm_data
- nutdec: stop skipping bytes at EOF
- nutdec: fix infinite resync loops
- avformat/nutdec: Check X in 2nd branch of index reading
- avformat/nutdec: Fix recovery when immedeately after seeking a failure happens
- avformat/nutdec: Return error on EOF from get_str()
- avcodec/x86/cavsdsp: remove unneeded tmp
- nutdec: fix memleaks on error in nut_read_header
- avformat/avidec: print a warning for negative sample_size
- avidec: avoid infinite loop due to negative ast->sample_size
- nutdec: check chapter creation in decode_info_header
- apedec: prevent out of array writes in decode_array_0000
- avformat/mpegts: Detect changes in packet through CRC instead of just the 5bit version
- avformat/mpegts: reset last_ver on corrupted packets
- avformat/mpegts: Factorize version checking code out
- avformat/mpegts: Also parse the FMC descriptor if the codec has not been identified yet
- avformat/mpegts: reset last_version on seeking
- avformat/mp3dec: Check for avcodec_alloc_context3() failure
- avformat/mp3dec: properly allocate dummy AVCodecContext
- lavfi/fade: Do not overread input buffer.
- tests: drop bc dependency
- alsdec: limit avctx->bits_per_raw_sample to 32
version 2.4.9:
- alac: reject rice_limit 0 if compression is used
- lavf: Reset global flag on deinit
- bink: check vst->index_entries before using it
- mpeg4videodec: only allow a positive length
- alsdec: check sample pointer range in revert_channel_correlation
- avcodec/h264_refs: Do not set reference to things which do not exist
- avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode
- h264: avoid unnecessary calls to get_format
- avutil/pca: Check for av_malloc* failures
- alsdec: validate time diff index
- avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data
- alsdec: ensure channel reordering is reversible
- avcodec/atrac3plusdsp: fix on stack alignment
- ac3: validate end in ff_ac3_bit_alloc_calc_mask
- aacpsy: avoid psy_band->threshold becoming NaN
- aasc: return correct buffer size from aasc_decode_frame
- aacdec: consistently use avctx for logging in decode_eld_specific_config
- msrledec: use signed pixel_ptr in msrle_decode_pal4
- swresample/swresample-test: Randomly wipe out channel counts
- swresample: Check channel layouts and channels against each other and print human readable error messages
- swresample: Allow reinitialization without ever setting channel layouts
- swresample: Allow reinitialization without ever setting channel counts
- avcodec/h264: Do not fail with randomly truncated VUIs
- avcodec/h264_ps: Move truncation check from VUI to SPS
- avcodec/h264: Be more tolerant to changing pps id between slices
- avcodec/aacdec: Fix storing state before PCE decode
- avcodec/h264: reset the counts in the correct context
- avcodec/h264_slice: Do not reset mb_aff_frame per slice
- tests: Fix test name for pixfmts tests
- avcodec/h264: finish previous slices before switching to single thread mode
- avcodec/h264: Fix race between slices where one overwrites data from the next
- avformat/utils: avoid discarded streams in av_find_default_stream_index()
- avformat/utils: Ensure that AVFMT_FLAG_CUSTOM_IO is set before use
- avformat/img2dec: do not rewind custom io buffers
- fate: Include branch information in the payload header
version 2.4.8:
- avutil/cpu: add missing check for mmxext to av_force_cpu_flags
- avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
- avcodec/hevc_ps: Check cropping parameters more correctly
- avcodec/dnxhddec: Check that the frame is interlaced before using cur_field
- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
- avformat/mov: Check for string truncation in mov_open_dref()
- ac3_fixed: fix out-of-bound read
- avcodec/012v: redesign main loop
- avcodec/012v: Check dimensions more completely
- asfenc: fix leaking asf->index_ptr on error
- avcodec/options_table: remove extradata_size from the AVOptions table
- ffmdec: limit the backward seek to the last resync position
- ffmdec: make sure the time base is valid
- ffmdec: fix infinite loop at EOF
- avcodec/tiff: move bpp check to after "end:"
- avcodec/opusdec: Fix delayed sample value
- avcodec/utils: Align YUV411 by as much as the other YUV variants
- vp9: fix segmentation map retention with threading enabled.
- doc/protocols/tcp: fix units of listen_timeout option value, from microseconds to milliseconds
- fix VP9 packet decoder returning 0 instead of the used data size
- avformat/bit: only accept the g729 codec and 1 channel
- avformat/adxdec: check avctx->channels for invalid values
- Fix buffer_size argument to init_put_bits() in multiple encoders.
- mips/acelp_filters: fix incorrect register constraint
- avcodec/hevc_ps: Sanity checks for some log2_* values
- avcodec/zmbv: Check len before reading in decode_frame()
- avcodec/snowdec: Fix ref value check
- swscale/utils: More carefully merge and clear coefficients outside the input
- avcodec/a64multienc: fix use of uninitialized values in to_meta_with_crop
- avcodec/a64multienc: don't set incorrect packet size
- webp: ensure that each transform is only used once
- avcodec/hevc_ps: More complete window reset
- vp9: make above buffer pointer 32-byte aligned.
- avformat/rm: limit packet size
- avcodec/webp: validate the distance prefix code
- avcodec/gif: fix off by one in column offsetting finding
version 2.4.7:
- avcodec/flac_parser: fix handling EOF if no headers are found
- avfilter/vf_framepack: Check and update frame_rate
- avcodec/hevc: Fix handling of skipped_bytes() reallocation failures
- qpeg: avoid pointless invalid memcpy()
- avcodec/arm/videodsp_armv5te: Fix linking failure with "g++ -shared -D__STDC_CONSTANT_MACROS -o test.so ... libavcodec.a"
- avcodec/mjpegdec: Skip blocks which are outside the visible area
- lavc/aarch64: Do not use the neon horizontal chroma loop filter for H.264 4:2:2. (cherry picked from commit 4faea46bd906b3897018736208123aa36c3f45d5)
- avcodec/h264_slice: assert that reinit does not occur after the first slice
- avcodec/h264_slice: ignore SAR changes in slices after the first
- avcodec/h264_slice: Check picture structure before setting the related fields
- avcodec/h264_slice: Do not change frame_num after the first slice
- avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT
- avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT
- avcodec/h264: Be more strict on rejecting pps/sps changes
- avcodec/h264: Be more strict on rejecting pps_id changes
- avcodec/h264_ps: More completely check the bit depths
- avformat/thp: Check av_get_packet() for failure not only for partial output
- swscale/utils: Limit filter shifting so as not to read from prior the array
- avcodec/mpegvideo_motion: Fix gmc chroma dimensions
- avcodec/mjpegdec: Check number of components for JPEG-LS
- avcodec/mjpegdec: Check escape sequence validity
- avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior
- avformat/mpc8: fix broken pointer math
- avformat/mpc8: fix hang with fuzzed file
- avformat/tta: fix crash with corrupted files
version 2.4.6:
- doc/examples: fix lib math dep for decoding_encoding
- avformat/movenc: workaround bug in "PathScale EKOPath(tm) Compiler Suite Version 4.0.12.1"
- vp9: fix parser return values in error case
- ffmpeg: Clear error message array at init.
- avcodec/dvdsubdec: fix accessing dangling pointers
- avcodec/dvdsubdec: error on bitmaps with size 0
- avformat/mov: Fix mixed declaration and statement warning
- cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file()
- avformat/utils: Clear pointer in ff_alloc_extradata() to avoid leaving a stale pointer in memory
- avformat/matroskadec: Use av_freep() to avoid leaving stale pointers in memory
- lavfi: check av_strdup() return value
- mov: Fix negative size calculation in mov_read_default().
- avformat/mov: fix integer overflow in mov_read_udta_string()
- mov: Avoid overflow with mov_metadata_raw()
- avcodec/dvdsubdec: fix out of bounds accesses
- avfilter/vf_sab: fix filtering tiny images
- avformat/flvdec: Increase string array size
- avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream
- avformat/cdxl: Fix integer overflow of image_size
- avformat/segment: Use av_freep() avoid leaving stale pointers in memory
- avformat/mov: Fix memleaks for duplicate STCO/CO64/STSC atoms
- mov: avoid a memleak when multiple stss boxes are presen
version 2.4.5:
- lavu/frame: fix malloc error path in av_frame_copy_props()
- avformat/utils: Do not update programs streams from program-less streams in update_wrap_reference()
- avformat/aviobuf: Check that avio_seek() target is non negative
- swresample/soxr_resample: fix error handling
- avformat/flvdec: fix potential use of uninitialized variables
- avformat/matroskadec: fix handling of recursive SeekHead elements
- doc/examples/transcoding: check encoder before using it
- swscale/x86/rgb2rgb_template: fix crash with tiny size and nv12 output
- avformat/rmdec: Check codec_data_size
- avformat/aviobuf: Fix infinite loop in ff_get_line()
- vc1: Do not assume seek happens after decoding
- mmvideo: check frame dimensions
- jvdec: check frame dimensions
- avcodec/indeo3: ensure offsets are non negative
- avcodec/h264: Check *log2_weight_denom
- avcodec/hevc_ps: Check diff_cu_qp_delta_depth
- avcodec/h264: Clear delayed_pic on deallocation
- avcodec/hevc: clear filter_slice_edges() on allocation
- avcodec/dcadec: Check that the added xch channel isnt already there
- avcodec/indeo3: use signed variables to avoid underflow
- swscale: increase yuv2rgb table headroom
- avformat/mov: fix integer overflow of size
- avformat/mov: check atom nesting depth
- avcodec/utvideodec: Fix handling of slice_height=0
- avcodec/vmdvideo: Check len before using it in method 3
- avformat/flvdec: Use av_freep() avoid leaving stale pointers in memory
- avformat/hdsenc: Use av_freep() avoid leaving stale pointers in memory
- configure: create the tests directory like the doc directory
- v4l2: Make use of the VIDIOC_ENUM_FRAMESIZES ioctl on OpenBSD
- avcodec/motion_est: use 2x8x8 for interlaced qpel
- Treat all '*.pnm' files as non-text file
version 2.4.4:
- avformat: replace some odd 30-60 rates by higher less odd ones in get_std_framerate()
- swscale: fix yuv2yuvX_8 assembly on x86
- avcodec/hevc_ps: Check num_long_term_ref_pics_sps
- avcodec/mjpegdec: Fix integer overflow in shift
- avcodec/hevc_ps: Check return code from pps_range_extensions()
- avcodec/rawdec: Check the return code of avpicture_get_size()
- avcodec/pngdec: Check IHDR/IDAT order
- avcodec/flacdec: Call ff_flacdsp_init() unconditionally
- avcodec/utils: Check that the data is complete in avpriv_bprint_to_extradata()
- avcodec/mjpegdec: Fix context fields becoming inconsistent
- avcodec/mjpegdec: Check for pixfmtid 0x42111100 || 0x24111100 with more than 8 bits
- swscale/x86/rgb2rgb_template: handle the first 2 lines with C in rgb24toyv12_*()
- doc/APIchanges: Fix some wrong versions
- avformat/hlsenc: Free context after hls_append_segment
- avcodec/mpeg4video_parser: fix spurious extradata parse warnings
- lavu/opt: fix av_opt_get function
- avcodec/wmaprodec: Fix integer overflow in sfb_offsets initialization
- avcodec/utvideodec: fix assumtation that slice_height >= 1
- avcodec/options_table fix min of audio channels and sample rate
- libavutil/thread.h: Support OS/2 threads
- fix Makefile objects for pulseaudio support
- opusdec: make sure all substreams have the same number of coded samples
- lavu: add wrappers for the pthreads mutex API
- avformat/avidec: fix handling dv in avi
- avfilter/vf_lut: gammaval709()
- cinedec: report white balance gain coefficients using metadata
- swscale/utils: support bayer input + scaling, and bayer input + any supported output
- swscale: support internal scaler cascades
- avformat/dtsdec: dts_probe: check reserved bit, check lfe, check sr_code similarity
- avformat/segment: export inner muxer timebase
- Remove fminf() emulation, fix build issues
- avcodec/mpegaudio_parser: fix off by 1 error in bitrate calculation
- Use -fno-optimize-sibling-calls on parisc also for gcc 4.9.
- ffmpeg_opt: store canvas size in decoder context
- avcodec/mpeg12dec: do not trust AVCodecContext input dimensions
version 2.4.3:
- avcodec/svq1dec: zero terminate embedded message before printing
- avcodec/cook: check that the subpacket sizes fit in block_align
- avcodec/g2meet: check tile dimensions to avoid integer overflow
- avcodec/utils: Align dimensions by at least their chroma sub-sampling factors.
- avcodec/dnxhddec: treat pix_fmt like width/height
- avcodec/dxa: check dimensions
- avcodec/dirac_arith: fix integer overflow
- avcodec/diracdec: Tighter checks on CODEBLOCKS_X/Y
- avcodec/diracdec: Use 64bit in calculation of codeblock coordinates
- avcodec/sgidec: fix count check
- avcodec/sgidec: fix linesize for 16bit
- avcodec/hevc_ps: Check default display window bitstream and skip if invalid
- avcodec/tiffenc: properly compute packet size
- lavd: export all symbols with av_ prefix
- avformat/mxfdec: Fix termination of mxf_data_essence_container_uls
- postproc: fix qp count
- postproc/postprocess: fix quant store for fq mode
- vf_drawtext: add missing clear of pointers after av_expr_free()
- utvideoenc: properly set slice height/last line
- swresample: fix sample drop loop end condition
- resample: Avoid off-by-1 errors in PTS calcs.
- imc: fix order of operations in coefficients read
- hevc_mvs: make sure to always initialize the temporal MV fully
- hevc_mvs: initialize the temporal MV in case of missing reference
version 2.4.2:
- avcodec/on2avc: Check number of channels
- avcodec/hevc: fix chroma transform_add size
- avcodec/h264: Check mode before considering mixed mode intra prediction
- avformat/mpegts: use a padded buffer in read_sl_header()
- avformat/mpegts: Check desc_len / get8() return code
- avcodec/vorbisdec: Fix off by 1 error in ptns_to_read
- sdp: add support for H.261
- avcodec/svq3: Do not memcpy AVFrame
- avcodec/smc: fix off by 1 error
- avcodec/qpeg: fix off by 1 error in MV bounds check
- avcodec/gifdec: factorize interleave end handling out
- avcodec/cinepak: fix integer underflow
- avcodec/pngdec: Check bits per pixel before setting monoblack pixel format
- avcodec/pngdec: Calculate MPNG bytewidth more defensively
- avcodec/tiff: more completely check bpp/bppcount
- avcodec/mmvideo: Bounds check 2nd line of HHV Intra blocks
- avcodec/h263dec: Fix decoding messenger.h263
- avcodec/utils: Add case for jv to avcodec_align_dimensions2()
- avcodec/mjpegdec: check bits per pixel for changes similar to dimensions
- avcodec/jpeglsdec: Check run value more completely in ls_decode_line()
- avformat/hlsenc: export inner muxer timebase
- configure: add noexecstack to linker options if supported.
- avcodec/ac3enc_template: fix out of array read
- avutil/x86/cpu: fix cpuid sub-leaf selection
- avformat/img2dec: enable generic seeking for image pipes
- avformat/img2dec: initialize pkt->pos for image pipes
- avformat/img2dec: pass error code and signal EOF
- avformat/img2dec: fix error code at EOF for pipes
- libavutil/opt: fix av_opt_set_channel_layout() to access correct memory address
- tests/fate-run.sh: Cat .err file in case of error with V>0
- avformat/riffenc: Filter out "BottomUp" in ff_put_bmp_header()
- avcodec/webp: fix default palette color 0xff000000 -> 0x00000000
- avcodec/asvenc: fix AAN scaling
- Fix compile error on arm4/arm5 platform
version 2.4.1:
- swscale: Allow chroma samples to be above and to the left of luma samples
- avcodec/libilbc: support for latest git of libilbc
- avcodec/webp: treat out-of-bound palette index as translucent black
- vf_deshake: rename Transform.vector to Transform.vec to avoid compiler confusion
- apetag: Fix APE tag size check
- tools/crypto_bench: fix build when AV_READ_TIME is unavailable
version 2.4:
- Icecast protocol
- ported lenscorrection filter from frei0r filter
- large optimizations in dctdnoiz to make it usable
- ICY metadata are now requested by default with the HTTP protocol
- support for using metadata in stream specifiers in fftools
- LZMA compression support in TIFF decoder
- support for H.261 RTP payload format (RFC 4587)
- HEVC/H.265 RTP payload format (draft v6) depacketizer
- added codecview filter to visualize information exported by some codecs
- Matroska 3D support thorugh side data
- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
- silenceremove filter
version 2.3:
- AC3 fixed-point decoding
- shuffleplanes filter
- subfile protocol
- Phantom Cine demuxer
- replaygain data export
- VP7 video decoder
- Alias PIX image encoder and decoder
- Improvements to the BRender PIX image decoder
- Improvements to the XBM decoder
- QTKit input device
- improvements to OpenEXR image decoder
- support decoding 16-bit RLE SGI images
- GDI screen grabbing for Windows
- alternative rendition support for HTTP Live Streaming
- AVFoundation input device
- Direct Stream Digital (DSD) decoder
- Magic Lantern Video (MLV) demuxer
- On2 AVC (Audio for Video) decoder
- support for decoding through DXVA2 in ffmpeg
- libbs2b-based stereo-to-binaural audio filter
- libx264 reference frames count limiting depending on level
- native Opus decoder
- display matrix export and rotation API
- WebVTT encoder
- showcqt multimedia filter
- zoompan filter
- signalstats filter
- hqx filter (hq2x, hq3x, hq4x)
- flanger filter
- Image format auto-detection
- LRC demuxer and muxer
- Samba protocol (via libsmbclient)
- WebM DASH Manifest muxer
- libfribidi support in drawtext
version 2.2:
- HNM version 4 demuxer and video decoder
- Live HDS muxer
- setsar/setdar filters now support variables in ratio expressions
- elbg filter
- string validation in ffprobe
- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
- complete Voxware MetaSound decoder
- remove mp3_header_compress bitstream filter
- Windows resource files for shared libraries
- aeval filter
- stereoscopic 3d metadata handling
- WebP encoding via libwebp
- ATRAC3+ decoder
- VP8 in Ogg demuxing
- side & metadata support in NUT
- framepack filter
- XYZ12 rawvideo support in NUT
- Exif metadata support in WebP decoder
- OpenGL device
- Use metadata_header_padding to control padding in ID3 tags (currently used in
MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
- Mirillis FIC video decoder
- Support DNx444
- libx265 encoder
- dejudder filter
- Autodetect VDA like all other hardware accelerations
- aliases and defaults for Ogg subtypes (opus, spx)
version 2.1:
- aecho filter
- perspective filter ported from libmpcodecs
- ffprobe -show_programs option
- compand filter
- RTMP seek support
- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
even when used as an input option. Previous behavior can be restored with
the -noaccurate_seek option.
- ffmpeg -t option can now be used for inputs, to limit the duration of
data read from an input file
- incomplete Voxware MetaSound decoder
- read EXIF metadata from JPEG
- DVB teletext decoder
- phase filter ported from libmpcodecs
- w3fdif filter
- Opus support in Matroska
- FFV1 version 1.3 is stable and no longer experimental
- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
more consistent with other muxers.
- adelay filter
- pullup filter ported from libmpcodecs
- ffprobe -read_intervals option
- Lossless and alpha support for WebP decoder
- Error Resilient AAC syntax (ER AAC LC) decoding
- Low Delay AAC (ER AAC LD) decoding
- mux chapters in ASF files
- SFTP protocol (via libssh)
- libx264: add ability to encode in YUVJ422P and YUVJ444P
- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
- make decoding alpha optional for prores, ffv1 and vp6 by setting
the skip_alpha flag.
- ladspa wrapper filter
- native VP9 decoder
- dpx parser
- max_error_rate parameter in ffmpeg
- PulseAudio output device
- ReplayGain scanner
- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
- Linux framebuffer output device
- HEVC decoder
- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
- mergeplanes filter
version 2.0:
- curves filter
- reference-counting for AVFrame and AVPacket data
- ffmpeg now fails when input options are used for output file
or vice versa
- support for Monkey's Audio versions from 3.93
- perms and aperms filters
- audio filtering support in ffplay
- 10% faster aac encoding on x86 and MIPS
- sine audio filter source
- WebP demuxing and decoding support
- ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
- OpenCL support
- audio phaser filter
- separatefields filter
- libquvi demuxer
- uniform options syntax across all filters
- telecine filter
- interlace filter
- smptehdbars source
- inverse telecine filters (fieldmatch and decimate)
- colorbalance filter
- colorchannelmixer filter
- The matroska demuxer can now output proper verbatim ASS packets. It will
become the default at the next libavformat major bump.
- decent native animated GIF encoding
- asetrate filter
- interleave filter
- timeline editing with filters
- vidstabdetect and vidstabtransform filters for video stabilization using
the vid.stab library
- astats filter
- trim and atrim filters
- ffmpeg -t and -ss (output-only) options are now sample-accurate when
transcoding audio
- Matroska muxer can now put the index at the beginning of the file.
- extractplanes filter
- avectorscope filter
- ADPCM DTK decoder
- ADP demuxer
- RSD demuxer
- RedSpark demuxer
- ADPCM IMA Radical decoder
- zmq filters
- DCT denoiser filter (dctdnoiz)
- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
- Apple Intermediate Codec decoder
- Escape 130 video decoder
- FTP protocol support
- V4L2 output device
- 3D LUT filter (lut3d)
- SMPTE 302M audio encoder
- support for slice multithreading in libavfilter
- Hald CLUT support (generation and filtering)
- VC-1 interlaced B-frame support
- support for WavPack muxing (raw and in Matroska)
- XVideo output device
- vignette filter
- True Audio (TTA) encoder
- Go2Webinar decoder
- mcdeint filter ported from libmpcodecs
- sab filter ported from libmpcodecs
- ffprobe -show_chapters option
- WavPack encoding through libwavpack
- rotate filter
- spp filter ported from libmpcodecs
- libgme support
- psnr filter
version 1.2:
- VDPAU hardware acceleration through normal hwaccel
- SRTP support
- Error diffusion dither in Swscale
- Chained Ogg support
- Theora Midstream reconfiguration support
- EVRC decoder
- audio fade filter
- filtering audio with unknown channel layout
- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
and treble audio filter
- improved showspectrum filter, with multichannel support and sox-like colors
- histogram filter
- tee muxer
- il filter ported from libmpcodecs
- support ID3v2 tags in ASF files
- encrypted TTA stream decoding support
- RF64 support in WAV muxer
- noise filter ported from libmpcodecs
- Subtitles character encoding conversion
- blend filter
- stereo3d filter ported from libmpcodecs
version 1.1:
- stream disposition information printing in ffprobe
- filter for loudness analysis following EBU R128
- Opus encoder using libopus
- ffprobe -select_streams option
- Pinnacle TARGA CineWave YUV16 decoder
- TAK demuxer, decoder and parser
- DTS-HD demuxer
- remove -same_quant, it hasn't worked for years
- FFM2 support
- X-Face image encoder and decoder
- 24-bit FLAC encoding
- multi-channel ALAC encoding up to 7.1
- metadata (INFO tag) support in WAV muxer
- subtitles raw text decoder
- support for building DLLs using MSVC
- LVF demuxer
- ffescape tool
- metadata (info chunk) support in CAF muxer
- field filter ported from libmpcodecs
- AVR demuxer
- geq filter ported from libmpcodecs
- remove ffserver daemon mode
- AST muxer/demuxer
- new expansion syntax for drawtext
- BRender PIX image decoder
- ffprobe -show_entries option
- ffprobe -sections option
- ADPCM IMA Dialogic decoder
- BRSTM demuxer
- animated GIF decoder and demuxer
- PVF demuxer
- subtitles filter
- IRCAM muxer/demuxer
- Paris Audio File demuxer
- Virtual concatenation demuxer
- VobSub demuxer
- JSON captions for TED talks decoding support
- SOX Resampler support in libswresample
- aselect filter
- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
- Silicon Graphics Motion Video Compressor 1 & 2 decoder
- Silicon Graphics Movie demuxer
- apad filter
- Resolution & pixel format change support with multithreading for H.264
- documentation split into per-component manuals
- pp (postproc) filter ported from MPlayer
- NIST Sphere demuxer
- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
- Sony Wave64 muxer
- adobe and limelight publisher authentication in RTMP
- data: URI scheme
- support building on the Plan 9 operating system
- kerndeint filter ported from MPlayer
- histeq filter ported from VirtualDub
- Megalux Frame demuxer
- 012v decoder
- Improved AVC Intra decoding support
version 1.0:
- INI and flat output in ffprobe
- Scene detection in libavfilter
- Indeo Audio decoder
- channelsplit audio filter
- setnsamples audio filter
- atempo filter
- ffprobe -show_data option
- RTMPT protocol support
- iLBC encoding/decoding via libilbc
- Microsoft Screen 1 decoder
- join audio filter
- audio channel mapping filter
- Microsoft ATC Screen decoder
- RTSP listen mode
- TechSmith Screen Codec 2 decoder
- AAC encoding via libfdk-aac
- Microsoft Expression Encoder Screen decoder
- RTMPS protocol support
- RTMPTS protocol support
- RTMPE protocol support
- RTMPTE protocol support
- showwaves and showspectrum filter
- LucasArts SMUSH SANM playback support
- LucasArts SMUSH VIMA audio decoder (ADPCM)
- LucasArts SMUSH demuxer
- SAMI, RealText and SubViewer demuxers and decoders
- Heart Of Darkness PAF playback support
- iec61883 device
- asettb filter
- new option: -progress
- 3GPP Timed Text encoder/decoder
- GeoTIFF decoder support
- ffmpeg -(no)stdin option
- Opus decoder using libopus
- caca output device using libcaca
- alphaextract and alphamerge filters
- concat filter
- flite filter
- Canopus Lossless Codec decoder
- bitmap subtitles in filters (experimental and temporary)
- MP2 encoding via TwoLAME
- bmp parser
- smptebars source
- asetpts filter
- hue filter
- ICO muxer
- SubRip encoder and decoder without embedded timing
- edge detection filter
- framestep filter
- ffmpeg -shortest option is now per-output file
-pass and -passlogfile are now per-output stream
- volume measurement filter
- Ut Video encoder
- Microsoft Screen 2 decoder
- smartblur filter ported from MPlayer
- CPiA decoder
- decimate filter ported from MPlayer
- RTP depacketization of JPEG
- Smooth Streaming live segmenter muxer
- F4V muxer
- sendcmd and asendcmd filters
- WebVTT demuxer and decoder (simple tags supported)
- RTP packetization of JPEG
- faststart option in the MOV/MP4 muxer
- support for building with MSVC
version 0.11:
- Fixes: CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
- setfield filter
- CDXL demuxer and decoder
- Apple ProRes encoder
- ffprobe -count_packets and -count_frames options
- Sun Rasterfile Encoder
- ID3v2 attached pictures reading and writing
- WMA Lossless decoder
- bluray protocol
- blackdetect filter
- libutvideo encoder wrapper (--enable-libutvideo)
- swapuv filter
- bbox filter
- XBM encoder and decoder
- RealAudio Lossless decoder
- ZeroCodec decoder
- tile video filter
- Metal Gear Solid: The Twin Snakes demuxer
- OpenEXR image decoder
- removelogo filter
- drop support for ffmpeg without libavfilter
- drawtext video filter: fontconfig support
- ffmpeg -benchmark_all option
- super2xsai filter ported from libmpcodecs
- add libavresample audio conversion library for compatibility
- MicroDVD decoder
- Avid Meridien (AVUI) encoder and decoder
- accept + prefix to -pix_fmt option to disable automatic conversions.
- complete audio filtering in libavfilter and ffmpeg
- add fps filter
- vorbis parser
- png parser
- audio mix filter
- ffv1: support (draft) version 1.3
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
- support encoding and decoding 4-channel SGI images
version 0.9:
- Indeo 4 decoder
- SMJPEG demuxer
- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
- openal input device added
- boxblur filter added
- BWF muxer
@@ -939,7 +128,7 @@ easier to use. The changes are:
- pan audio filter
- IFF Amiga Continuous Bitmap (ACBM) decoder
- ass filter
- CRI ADX audio format muxer and demuxer
- CRI ADX audio format demuxer
- Playstation Portable PMP format demuxer
- Microsoft Windows ICO demuxer
- life source
@@ -948,13 +137,11 @@ easier to use. The changes are:
- new option: -report
- Dxtory capture format decoder
- cellauto source
- Simple segmenting muxer
- Indeo 4 decoder
- SMJPEG demuxer
- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
version 0.8:
- many many things we forgot because we rather write code than changelogs
- WebM support in Matroska de/muxer
- low overhead Ogg muxing
@@ -1113,7 +300,7 @@ version 0.6:
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- ATRAC1 decoder
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -1213,7 +400,7 @@ version 0.5:
- MXF demuxer
- VC-1/WMV3/WMV9 video decoder
- MacIntel support
- AviSynth support
- AVISynth support
- VMware video decoder
- VP5 video decoder
- VP6 video decoder
@@ -1241,7 +428,7 @@ version 0.5:
- Interplay C93 demuxer and video decoder
- Bethsoft VID demuxer and video decoder
- CRYO APC demuxer
- ATRAC3 decoder
- Atrac3 decoder
- V.Flash PTX decoder
- RoQ muxer, RoQ audio encoder
- Renderware TXD demuxer and decoder
@@ -1518,7 +705,7 @@ version 0.4.5:
- MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
- ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
- Windows porting of file converter
- added MJPEG raw format (input/output)
- added MJPEG raw format (input/ouput)
- added JPEG image format support (input/output)

View File

@@ -31,20 +31,14 @@ PROJECT_NAME = FFmpeg
# This could be handy for archiving the generated documentation or
# if some version control system is used.
PROJECT_NUMBER = 2.4.10
# With the PROJECT_LOGO tag one can specify a logo or icon that is included
# in the documentation. The maximum height of the logo should not exceed 55
# pixels and the maximum width should not exceed 200 pixels. Doxygen will
# copy the logo to the output directory.
PROJECT_LOGO =
PROJECT_NUMBER = 0.9.1
# The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute)
# base path where the generated documentation will be put.
# If a relative path is entered, it will be relative to the location
# where doxygen was started. If left blank the current directory will be used.
OUTPUT_DIRECTORY = doc/doxy
OUTPUT_DIRECTORY = doxy
# If the CREATE_SUBDIRS tag is set to YES, then doxygen will create
# 4096 sub-directories (in 2 levels) under the output directory of each output
@@ -277,7 +271,7 @@ SUBGROUPING = YES
# be useful for C code in case the coding convention dictates that all compound
# types are typedef'ed and only the typedef is referenced, never the tag name.
TYPEDEF_HIDES_STRUCT = YES
TYPEDEF_HIDES_STRUCT = NO
# The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
# determine which symbols to keep in memory and which to flush to disk.
@@ -288,7 +282,7 @@ TYPEDEF_HIDES_STRUCT = YES
# causing a significant performance penality.
# If the system has enough physical memory increasing the cache will improve the
# performance by keeping more symbols in memory. Note that the value works on
# a logarithmic scale so increasing the size by one will roughly double the
# a logarithmic scale so increasing the size by one will rougly double the
# memory usage. The cache size is given by this formula:
# 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
# corresponding to a cache size of 2^16 = 65536 symbols
@@ -409,7 +403,7 @@ INLINE_INFO = YES
# alphabetically by member name. If set to NO the members will appear in
# declaration order.
SORT_MEMBER_DOCS = NO
SORT_MEMBER_DOCS = YES
# If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
# brief documentation of file, namespace and class members alphabetically
@@ -489,6 +483,12 @@ MAX_INITIALIZER_LINES = 30
SHOW_USED_FILES = YES
# If the sources in your project are distributed over multiple directories
# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy
# in the documentation. The default is NO.
SHOW_DIRECTORIES = NO
# Set the SHOW_FILES tag to NO to disable the generation of the Files page.
# This will remove the Files entry from the Quick Index and from the
# Folder Tree View (if specified). The default is YES.
@@ -625,7 +625,8 @@ EXCLUDE_SYMLINKS = NO
# for example use the pattern */test/*
EXCLUDE_PATTERNS = *.git \
*.d
*.d \
avconv.c
# The EXCLUDE_SYMBOLS tag can be used to specify one or more symbol names
# (namespaces, classes, functions, etc.) that should be excluded from the
@@ -639,14 +640,15 @@ EXCLUDE_SYMBOLS =
# directories that contain example code fragments that are included (see
# the \include command).
EXAMPLE_PATH = doc/examples/
EXAMPLE_PATH = libavcodec/ \
libavformat/
# If the value of the EXAMPLE_PATH tag contains directories, you can use the
# EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
# and *.h) to filter out the source-files in the directories. If left
# blank all files are included.
EXAMPLE_PATTERNS = *.c
EXAMPLE_PATTERNS = *-example.c
# If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
# searched for input files to be used with the \include or \dontinclude
@@ -709,7 +711,7 @@ INLINE_SOURCES = NO
# doxygen to hide any special comment blocks from generated source code
# fragments. Normal C and C++ comments will always remain visible.
STRIP_CODE_COMMENTS = NO
STRIP_CODE_COMMENTS = YES
# If the REFERENCED_BY_RELATION tag is set to YES
# then for each documented function all documented
@@ -818,7 +820,7 @@ HTML_STYLESHEET =
# 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
# The allowed range is 0 to 359.
#HTML_COLORSTYLE_HUE = 120
HTML_COLORSTYLE_HUE = 120
# The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of
# the colors in the HTML output. For a value of 0 the output will use
@@ -841,6 +843,12 @@ HTML_COLORSTYLE_GAMMA = 80
HTML_TIMESTAMP = YES
# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
# files or namespaces will be aligned in HTML using tables. If set to
# NO a bullet list will be used.
HTML_ALIGN_MEMBERS = YES
# If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML
# documentation will contain sections that can be hidden and shown after the
# page has loaded. For this to work a browser that supports
@@ -1021,6 +1029,11 @@ ENUM_VALUES_PER_LINE = 4
GENERATE_TREEVIEW = NO
# By enabling USE_INLINE_TREES, doxygen will generate the Groups, Directories,
# and Class Hierarchy pages using a tree view instead of an ordered list.
USE_INLINE_TREES = NO
# If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
# used to set the initial width (in pixels) of the frame in which the tree
# is shown.
@@ -1056,7 +1069,7 @@ FORMULA_TRANSPARENT = YES
# typically be disabled. For large projects the javascript based search engine
# can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.
SEARCHENGINE = YES
SEARCHENGINE = NO
# When the SERVER_BASED_SEARCH tag is enabled the search engine will be
# implemented using a PHP enabled web server instead of at the web client
@@ -1356,11 +1369,14 @@ INCLUDE_FILE_PATTERNS =
# instead of the = operator.
PREDEFINED = "__attribute__(x)=" \
"RENAME(x)=x ## _TMPL" \
"DEF(x)=x ## _TMPL" \
HAVE_AV_CONFIG_H \
HAVE_MMX \
HAVE_MMX2 \
HAVE_AMD3DNOW \
"DECLARE_ALIGNED(a,t,n)=t n" \
"offsetof(x,y)=0x42" \
av_alloc_size \
AV_GCC_VERSION_AT_LEAST(x,y)=1 \
__GNUC__=1 \
"offsetof(x,y)=0x42"
# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
# this tag can be used to specify a list of macro names that should be expanded.

15
INSTALL Normal file
View File

@@ -0,0 +1,15 @@
1) Type './configure' to create the configuration. A list of configure
options is printed by running 'configure --help'.
'configure' can be launched from a directory different from the FFmpeg
sources to build the objects out of tree. To do this, use an absolute
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.

View File

@@ -1,17 +0,0 @@
#Installing FFmpeg:
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.
`configure` can be launched from a directory different from the FFmpeg
sources to build the objects out of tree. To do this, use an absolute
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.

46
LICENSE Normal file
View File

@@ -0,0 +1,46 @@
FFmpeg:
-------
Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
FFmpeg.
Some optional parts of FFmpeg are licensed under the GNU General Public License
version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
these parts are used by default, you have to explicitly pass --enable-gpl to
configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are
- libpostproc
- optional x86 optimizations in the files
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.c, libavcodec/jrevdct.c
are taken from libjpeg, see the top of the files for licensing details.
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter --enable-version3 will activate this licensing option
for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
external libraries:
-------------------
Some external libraries, e.g. libx264, are under GPL and can be used in
conjunction with FFmpeg. They require --enable-gpl to be passed to configure
as well.
The OpenCORE external libraries are under the Apache License 2.0. That license
is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
version needs to be upgraded by passing --enable-version3 to configure.
The nonfree external libraries libfaac and libaacplus can be hooked up in FFmpeg.
You need to pass --enable-nonfree to configure to enable it. Employ this option
with care as FFmpeg then becomes nonfree and unredistributable.

View File

@@ -1,105 +0,0 @@
#FFmpeg:
Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
FFmpeg.
Some optional parts of FFmpeg are licensed under the GNU General Public License
version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
these parts are used by default, you have to explicitly pass --enable-gpl to
configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are:
- libpostproc
- libmpcodecs
- optional x86 optimizations in the files
libavcodec/x86/flac_dsp_gpl.asm
libavcodec/x86/idct_mmx.c
- libutvideo encoding/decoding wrappers in
libavcodec/libutvideo*.cpp
- the X11 grabber in libavdevice/x11grab.c
- the swresample test app in
libswresample/swresample-test.c
- the texi2pod.pl tool
- the following filters in libavfilter:
- f_ebur128.c
- vf_blackframe.c
- vf_boxblur.c
- vf_colormatrix.c
- vf_cropdetect.c
- vf_decimate.c
- vf_delogo.c
- vf_geq.c
- vf_histeq.c
- vf_hqdn3d.c
- vf_interlace.c
- vf_kerndeint.c
- vf_mcdeint.c
- vf_mp.c
- vf_owdenoise.c
- vf_perspective.c
- vf_phase.c
- vf_pp.c
- vf_pullup.c
- vf_sab.c
- vf_smartblur.c
- vf_spp.c
- vf_stereo3d.c
- vf_super2xsai.c
- vf_tinterlace.c
- vsrc_mptestsrc.c
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter --enable-version3 will activate this licensing option
for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
There are a handful of files under other licensing terms, namely:
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
licensing details. Specifically note that you must credit the IJG in the
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
external libraries
==================
FFmpeg can be combined with a number of external libraries, which sometimes
affect the licensing of binaries resulting from the combination.
compatible libraries
--------------------
The following libraries are under GPL:
- frei0r
- libcdio
- libutvideo
- libvidstab
- libx264
- libx265
- libxavs
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing --enable-gpl to configure.
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing --enable-version3 to configure.
incompatible libraries
----------------------
The Fraunhofer AAC library, FAAC and aacplus are under licenses which
are incompatible with the GPLv2 and v3. We do not know for certain if their
licenses are compatible with the LGPL.
If you wish to enable these libraries, pass --enable-nonfree to configure.
But note that if you enable any of these libraries the resulting binary will
be under a complex license mix that is more restrictive than the LGPL and that
may result in additional obligations. It is possible that these
restrictions cause the resulting binary to be unredistributeable.

View File

@@ -4,11 +4,11 @@ FFmpeg maintainers
Below is a list of the people maintaining different parts of the
FFmpeg code.
Please try to keep entries where you are the maintainer up to date!
Please try to keep entries where you are the maintainer upto date!
Names in () mean that the maintainer currently has no time to maintain the code.
A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
patches and related discussions.
A CC after the name means that the maintainer prefers to be CC-ed on patches
and related discussions.
Project Leader
@@ -31,7 +31,7 @@ ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c Reynaldo H. Verdejo Pinochet
ffserver.c, ffserver.h Baptiste Coudurier
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@@ -43,26 +43,16 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu
build system (configure, makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
documentation Mike Melanson
website Robert Swain, Lou Logan
build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
Communication
=============
website Deby Barbara Lepage
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Michael Niedermayer, Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan
Launchpad Timothy Gu
libavutil
=========
@@ -72,24 +62,11 @@ Internal Interfaces:
libavutil/common.h Michael Niedermayer
Other:
bprint Nicolas George
bswap.h
des Reimar Doeffinger
dynarray.h Nicolas George
eval.c, eval.h Michael Niedermayer
float_dsp Loren Merritt
hash Reimar Doeffinger
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
mathematics.c, mathematics.h Michael Niedermayer
mem.c, mem.h Michael Niedermayer
opencl.c, opencl.h Wei Gao
opt.c, opt.h Michael Niedermayer
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
timecode Clément Bœsch
mathematics.c, mathematics.h Michael Niedermayer
integer.c, integer.h Michael Niedermayer
bswap.h
libavcodec
@@ -100,14 +77,16 @@ Generic Parts:
avcodec.h Michael Niedermayer
utility code:
utils.c Michael Niedermayer
mem.c Michael Niedermayer
opt.c, opt.h Michael Niedermayer
arithmetic expression evaluator:
eval.c Michael Niedermayer
audio and video frame extraction:
parser.c Michael Niedermayer
bitstream reading:
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:
@@ -130,9 +109,6 @@ Generic Parts:
libpostproc/* Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
fixed point FFT:
fft* Zeljko Lukac
Text Subtitles Clément Bœsch
Codecs:
4xm.c Michael Niedermayer
@@ -146,87 +122,69 @@ Codecs:
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3* Benjamin Larsson
atrac3plus* Maxim Poliakovski
bgmc.c, bgmc.h Thilo Borgmann
bink.c Kostya Shishkov
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
cljr Alex Beregszaszi
cllc.c Derek Buitenhuis
cook.c, cookdata.h Benjamin Larsson
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
dca.c Kostya Shishkov, Benjamin Larsson
dnxhd* Baptiste Coudurier
dpcm.c Mike Melanson
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
dxa.c Kostya Shishkov
dv.c Roman Shaposhnik
eacmv*, eaidct*, eat* Peter Ross
exif.c, exif.h Thilo Borgmann
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
fic.c Derek Buitenhuis
ffv1.c Michael Niedermayer
flac* Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
g722.c Martin Storsjo
g726.c Roman Shaposhnik
gifdec.c Baptiste Coudurier
h264* Loren Merritt, Michael Niedermayer
h261* Michael Niedermayer
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
huffyuv* Michael Niedermayer, Christophe Gisquet
huffyuv.c Michael Niedermayer
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
indeo2* Kostya Shishkov
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
ivi* Kostya Shishkov
jacosub* Clément Bœsch
jpeg2000* Nicolas Bertrand
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libdirac* David Conrad
libgsm.c Michel Bardiaux
libdirac* David Conrad
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libutvideo* Derek Buitenhuis
libvorbis.c David Conrad
libvpx* James Zern
libx264.c Mans Rullgard, Jason Garrett-Glaser
libx265.c Derek Buitenhuis
libxavs.c Stefan Gehrer
libzvbi-teletextdec.c Marton Balint
libx264.c Mans Rullgard, Jason Garrett-Glaser
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
mimic.c Ramiro Polla
mjpeg*.c Michael Niedermayer
mjpeg.c Michael Niedermayer
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpc* Kostya Shishkov
mpeg12.c, mpeg12data.h Michael Niedermayer
mpegvideo.c, mpegvideo.h Michael Niedermayer
mqc* Nicolas Bertrand
msmpeg4.c, msmpeg4data.h Michael Niedermayer
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nellymoserdec.c Benjamin Larsson
nuv.c Reimar Doeffinger
paf.* Paul B Mahol
pcx.c Ivo van Poorten
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
@@ -242,17 +200,15 @@ Codecs:
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
rv3* Kostya Shishkov
rv4* Kostya Shishkov, Christophe Gisquet
rv4* Kostya Shishkov
s3tc* Ivo van Poorten
smacker.c Kostya Shishkov
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow* Michael Niedermayer, Loren Merritt
snow.c Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
targa.c Kostya Shishkov
tiff.c Kostya Shishkov
truemotion1* Mike Melanson
@@ -260,24 +216,20 @@ Codecs:
truespeech.c Kostya Shishkov
tscc.c Kostya Shishkov
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov, Christophe Gisquet
vc1* Kostya Shishkov
vcr1.c Michael Niedermayer
vda_h264_dec.c Xidorn Quan
vima.c Paul B Mahol
vmnc.c Kostya Shishkov
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
vorbis_enc.c Oded Shimon
vorbis_dec.c Denes Balatoni, David Conrad
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje
vp9 Ronald Bultje, Clément Bœsch
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
@@ -285,12 +237,8 @@ Codecs:
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
xbm* Paul B Mahol
xface Stefano Sabatini
xl.c Kostya Shishkov
xvmc.c Ivan Kalvachev
xwd* Paul B Mahol
zerocodec.c Derek Buitenhuis
zmbv* Kostya Shishkov
Hardware acceleration:
@@ -308,65 +256,18 @@ libavdevice
libavdevice/avdevice.h
avfoundation.m Thilo Borgmann
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
iec61883.c Georg Lippitsch
lavfi Stefano Sabatini
libdc1394.c Roman Shaposhnik
opengl_enc.c Lukasz Marek
pulse_audio_enc.c Lukasz Marek
qtkit.m Thilo Borgmann
sdl Stefano Sabatini
v4l2.c Giorgio Vazzana
v4l2.c Luca Abeni
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
libavfilter
===========
Generic parts:
graphdump.c Nicolas George
Filters:
af_adelay.c Paul B Mahol
af_aecho.c Paul B Mahol
af_afade.c Paul B Mahol
af_amerge.c Nicolas George
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_astreamsync.c Nicolas George
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_compand.c Paul B Mahol
af_ladspa.c Paul B Mahol
af_pan.c Nicolas George
af_silenceremove.c Paul B Mahol
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_colorbalance.c Paul B Mahol
vf_dejudder.c Nicholas Robbins
vf_delogo.c Jean Delvare (CC <khali@linux-fr.org>)
vf_drawbox.c/drawgrid Andrey Utkin
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_lenscorrection.c Daniel Oberhoff
vf_mergeplanes.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_scale.c Michael Niedermayer
vf_separatefields.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
Video filters:
vsrc_mandelbrot.c Michael Niedermayer
vf_yadif.c Michael Niedermayer
Sources:
vsrc_mandelbrot.c Michael Niedermayer
libavformat
===========
@@ -381,79 +282,58 @@ Generic parts:
Muxers/Demuxers:
4xm.c Mike Melanson
adtsenc.c Robert Swain
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
aiff.c Baptiste Coudurier
ape.c Kostya Shishkov
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
avi* Michael Niedermayer
avisynth.c AvxSynth Team (avxsynth.testing at gmail dot com)
avr.c Paul B Mahol
bink.c Peter Ross
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flac* Justin Ruggles
flic.c Mike Melanson
flvdec.c, flvenc.c Michael Niedermayer
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Anssi Hannula
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
img2*.c Michael Niedermayer
ipmovie.c Mike Melanson
ircam* Paul B Mahol
img2.c Michael Niedermayer
iss.c Stefan Gehrer
jacosub* Clément Bœsch
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
matroska subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Michael Niedermayer, Baptiste Coudurier
movenc.c Baptiste Coudurier, Matthieu Bouron
movenc.c Michael Niedermayer, Baptiste Coudurier
mpc.c Kostya Shishkov
mpeg.c Michael Niedermayer
mpegenc.c Michael Niedermayer
mpegts.c Marton Balint
mpegtsenc.c Baptiste Coudurier
mpegts* Baptiste Coudurier
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
mxfdec.c Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
nut.c Michael Niedermayer
nuv.c Reimar Doeffinger
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
paf.c Paul B Mahol
psxstr.c Mike Melanson
pva.c Ivo van Poorten
pvfdec.c Paul B Mahol
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
@@ -461,71 +341,42 @@ Muxers/Demuxers:
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rtp.c, rtpenc.c Martin Storsjo
rtpdec_h261.*, rtpenc_h261.* Thomas Volkert
rtpdec_hevc.* Thomas Volkert
rtpdec_asf.* Ronald S. Bultje
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
rtsp.c Luca Barbato
sbgdec.c Nicolas George
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
smjpeg* Paul B Mahol
spdif* Anssi Hannula
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
takdec.c Paul B Mahol
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
voc.c Aurelien Jacobs
wav.c Michael Niedermayer
wc3movie.c Mike Melanson
webm dash (matroskaenc.c) Vignesh Venkatasubramanian
webvtt* Matthew J Heaney
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
wvenc.c Paul B Mahol
Protocols:
bluray.c Petri Hintukainen
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
mms*.c Ronald S. Bultje
udp.c Luca Abeni
libswresample
=============
Generic parts:
audioconvert.c Michael Niedermayer
dither.c Michael Niedermayer
rematrix*.c Michael Niedermayer
swresample*.c Michael Niedermayer
Resamplers:
resample*.c Michael Niedermayer
soxr_resample.c Rob Sykes
Operating systems / CPU architectures
=====================================
Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard, Nedeljko Babic
MIPS Mans Rullgard
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Luca Barbato
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver
Windows ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
x86 Michael Niedermayer
@@ -534,47 +385,38 @@ x86 Michael Niedermayer
Releases
========
2.4 Michael Niedermayer
2.2 Michael Niedermayer
1.2 Michael Niedermayer
0.5 *Deprecated/Unmaintained*
0.6 *Deprecated/Unmaintained*
0.7 Michael Niedermayer
0.8 Michael Niedermayer
0.9 Michael Niedermayer
If you want to maintain an older release, please contact us
GnuPG Fingerprints of maintainers and contributors
==================================================
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
Ash Hughes 694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
Ben Littler 3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Clément Bœsch 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Luca Barbato 6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
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 Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
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 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
Tiancheng "Timothy" Gu 9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
Tim Nicholson 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
Stefano Sabatini 9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
Tomas Härdin D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD

148
Makefile
View File

@@ -4,70 +4,59 @@ include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(SRC_PATH)
vpath %.m $(SRC_PATH)
vpath %.S $(SRC_PATH)
vpath %.asm $(SRC_PATH)
vpath %.rc $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %/fate_config.sh.template $(SRC_PATH)
AVPROGS-$(CONFIG_FFMPEG) += ffmpeg
AVPROGS-$(CONFIG_FFPLAY) += ffplay
AVPROGS-$(CONFIG_FFPROBE) += ffprobe
AVPROGS-$(CONFIG_FFSERVER) += ffserver
PROGS-$(CONFIG_FFMPEG) += ffmpeg
PROGS-$(CONFIG_AVCONV) += avconv
PROGS-$(CONFIG_FFPLAY) += ffplay
PROGS-$(CONFIG_FFPROBE) += ffprobe
PROGS-$(CONFIG_FFSERVER) += ffserver
AVPROGS := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
INSTPROGS = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
PROGS += $(AVPROGS)
AVBASENAMES = ffmpeg ffplay ffprobe ffserver
ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o
OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_vda.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher uncoded_frame
PROGS := $(PROGS-yes:%=%$(EXESUF))
INSTPROGS = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
OBJS = $(PROGS-yes:%=%.o) cmdutils.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr base64
HOSTPROGS := $(TESTTOOLS:%=tests/%)
TOOLS = qt-faststart trasher
TOOLS-$(CONFIG_ZLIB) += cws2fws
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
BASENAMES = ffmpeg avconv ffplay ffprobe ffserver
ALLPROGS = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLPROGS_G = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
ALLMANPAGES = $(BASENAMES:%=%.1)
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset)
SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
SKIPHEADERS = cmdutils_common_opts.h
include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
all: $(AVPROGS)
all: $(PROGS)
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@$(PROGSSUF)
$(STRIP) $@$(PROGSSUF)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
tools/cws2fws$(EXESUF): ELIBS = -lz
config.h: .config
.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -75,12 +64,9 @@ config.h: .config
@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \
HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS \
ALTIVEC-OBJS MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS \
OBJS SLIBOBJS HOSTOBJS TESTOBJS
SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS ARMV6-OBJS TOOLS
define RESET
$(1) :=
@@ -91,34 +77,16 @@ define DOSUBDIR
$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/library.mak
endef
$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
include $(SRC_PATH)/doc/Makefile
ffplay.o: CFLAGS += $(SDL_CFLAGS)
ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)
define DOPROG
OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
$$(OBJS-$(1)): CFLAGS += $(CFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
-include $$(OBJS-$(1):.o=.d)
endef
$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@
$(STRIP) $@
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
OBJDIRS += tools
@@ -130,14 +98,14 @@ GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
.version: M=@
libavutil/ffversion.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
version.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
$(Q)touch .version
# force version.sh to run whenever version might have changed
-include .version
ifdef AVPROGS
ifdef PROGS
install: install-progs install-data
endif
@@ -148,39 +116,47 @@ install-libs: install-libs-yes
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
install-progs: install-progs-yes $(AVPROGS)
install-progs: install-progs-yes $(PROGS)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
install-data: $(DATA_FILES) $(EXAMPLES_FILES)
$(Q)mkdir -p "$(DATADIR)/examples"
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(TOOLS)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) coverage.info
$(RM) -r coverage-html
$(RM) -rf coverage.info lcov
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
$(RM) config.* .version version.h libavutil/avconfig.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
check: all alltools examples testprogs fate
# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
coverage-html: coverage.info
$(Q)mkdir -p $@
$(Q)genhtml -o $@ $<
$(Q)touch $@
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/tests/Makefile
$(sort $(OBJDIRS)):
@@ -194,5 +170,5 @@ $(sort $(OBJDIRS)):
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools check *clean config install*
.PHONY: all all-yes alltools *clean config examples install*
.PHONY: testprogs uninstall*

12
README Normal file
View File

@@ -0,0 +1,12 @@
FFmpeg README
-------------
1) Documentation
----------------
* Read the documentation in the doc/ directory.
2) Licensing
------------
* See the LICENSE file.

View File

@@ -1,40 +0,0 @@
FFmpeg README
=============
FFmpeg is a collection of libraries and tools to process multimedia content
such as audio, video, subtitles and related metadata.
## Libraries
* `libavcodec` provides implementation of a wider range of codecs.
* `libavformat` implements streaming protocols, container formats and basic I/O access.
* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
* `libavdevice` provides an abstraction to access capture and playback devices.
* `libswresample` implements audio mixing and resampling routines.
* `libswscale` implements color conversion and scaling routines.
## Tools
* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
manipulate, convert and stream multimedia content.
* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analisys tool to inspect
multimedia content.
* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
## Documentation
The offline documentation is available in the **doc/** directory.
The online documentation is available in the main [website](http://ffmpeg.org)
and in the [wiki](http://trac.ffmpeg.org).
### Examples
Conding examples are available in the **doc/example** directory.
## License
FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
GPL. Please refer to the LICENSE file for detailed information.

View File

@@ -1 +1 @@
2.4.10
0.9.1

View File

@@ -1,83 +0,0 @@
┌────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 2.4 "Fresnel" │
└────────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 2.4 "Fresnel", just 2 months
after the release of 2.3. Since this wasn't a long time ago, the Changelog
is a bit short this time.
The most important thing in this release is the major version bump of the
libraries. This means that this release is neither ABI-compatible nor
fully API-compatible. But on the other hand it is aligned with the Libav
11 release series, and will as a result probably end up being maintained for
a long time.
As usual, if you have any question on this release or any FFmpeg related
topic, feel free to join us on the #ffmpeg IRC channel (on
irc.freenode.net).
┌────────────────────────────┐
│ 🔨 API Information │
└────────────────────────────┘
FFmpeg 2.4 includes the following library versions:
• libavutil 54.7.100
• libavcodec 56.1.100
• libavformat 56.4.101
• libavdevice 56.0.100
• libavfilter 5.1.100
• libswscale 3.0.100
• libswresample 1.1.100
• libpostproc 53.0.100
Important API changes since 2.3:
• The new field mime_type was added to AVProbeData, which can
cause crashes, if it is not initialized.
• Some deprecated functions were removed.
• The avfilter_graph_parse function was made compatible with Libav.
• The Matroska demuxer now outputs verbatim ASS packets.
Please refer to the doc/APIchanges file for more information.
┌────────────────────────────┐
│ ★ List of New Features │
└────────────────────────────┘
┌────────────────────────────┐
│ libavformat │
└────────────────────────────┘
• Icecast protocol.
• API for live metadata updates through event flags.
• UTF-16 support in text subtitles formats.
• The ASS muxer now reorders the Dialogue events properly.
• support for H.261 RTP payload format (RFC 4587)
• HEVC/H.265 RTP payload format (draft v6) depacketizer
┌────────────────────────────┐
│ libavfilter │
└────────────────────────────┘
• Ported lenscorrection filter from frei0r filter.
• Large optimizations in dctdnoiz to make it usable.
• Added codecview filter to visualize information exported by some codecs.
• Added silenceremove filter.
┌────────────────────────────┐
│ libavutil │
└────────────────────────────┘
• Added clip() function in eval.
┌────────────────────────────┐
│ ⚠ Behaviour changes │
└────────────────────────────┘
• dctdnoiz filter now uses a block size of 8x8 instead of 16x16 by default
• -vismv option is deprecated in favor of the codecview filter
• libmodplug is now detected through pkg-config
• HTML documentation generation through texi2html is deprecated in
favor of makeinfo/texi2any
• ICY metadata are now requested by default with the HTTP protocol

1
VERSION Normal file
View File

@@ -0,0 +1 @@
0.9.1

View File

@@ -1,15 +0,0 @@
OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
OBJS-$(HAVE_ARMV6) += $(ARMV6-OBJS) $(ARMV6-OBJS-yes)
OBJS-$(HAVE_ARMV8) += $(ARMV8-OBJS) $(ARMV8-OBJS-yes)
OBJS-$(HAVE_VFP) += $(VFP-OBJS) $(VFP-OBJS-yes)
OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_MIPS32R2) += $(MIPS32R2-OBJS) $(MIPS32R2-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS) $(MIPSDSPR1-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

4397
avconv.c Normal file

File diff suppressed because it is too large Load Diff

1683
cmdutils.c

File diff suppressed because it is too large Load Diff

View File

@@ -24,13 +24,12 @@
#include <stdint.h>
#include "config.h"
#include "libavcodec/avcodec.h"
#include "libavfilter/avfilter.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#ifdef _WIN32
#ifdef __MINGW32__
#undef main /* We don't want SDL to override our main() */
#endif
@@ -47,19 +46,7 @@ extern const int program_birth_year;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern struct SwsContext *sws_opts;
extern AVDictionary *swr_opts;
extern AVDictionary *format_opts, *codec_opts, *resample_opts;
extern int hide_banner;
/**
* Register a program-specific cleanup routine.
*/
void register_exit(void (*cb)(int ret));
/**
* Wraps exit with a program-specific cleanup routine.
*/
void exit_program(int ret) av_noreturn;
extern AVDictionary *format_opts, *codec_opts;
/**
* Initialize the cmdutils option system, in particular
@@ -78,38 +65,25 @@ void uninit_opts(void);
*/
void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
/**
* Override the cpuflags.
*/
int opt_cpuflags(void *optctx, const char *opt, const char *arg);
/**
* Fallback for options that are not explicitly handled, these will be
* parsed through AVOptions.
*/
int opt_default(void *optctx, const char *opt, const char *arg);
int opt_default(const char *opt, const char *arg);
/**
* Set the libav* libraries log level.
*/
int opt_loglevel(void *optctx, const char *opt, const char *arg);
int opt_loglevel(const char *opt, const char *arg);
int opt_report(const char *opt);
int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_codec_debug(void *optctx, const char *opt, const char *arg);
#if CONFIG_OPENCL
int opt_opencl(void *optctx, const char *opt, const char *arg);
int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
#endif
int opt_codec_debug(const char *opt, const char *arg);
/**
* Limit the execution time.
*/
int opt_timelimit(void *optctx, const char *opt, const char *arg);
int opt_timelimit(const char *opt, const char *arg);
/**
* Parse a string and return its corresponding value as a double.
@@ -117,15 +91,14 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg);
* 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
@@ -133,16 +106,15 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
* 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
* date
*
* @see av_parse_time()
* @see parse_date()
*/
int64_t parse_time_or_die(const char *context, const char *timestr,
int is_duration);
int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);
typedef struct SpecifierOpt {
char *specifier; /**< stream/chapter/program/... specifier */
@@ -155,7 +127,7 @@ typedef struct SpecifierOpt {
} u;
} SpecifierOpt;
typedef struct OptionDef {
typedef struct {
const char *name;
int flags;
#define HAS_ARG 0x0001
@@ -164,42 +136,31 @@ typedef struct OptionDef {
#define OPT_STRING 0x0008
#define OPT_VIDEO 0x0010
#define OPT_AUDIO 0x0020
#define OPT_GRAB 0x0040
#define OPT_INT 0x0080
#define OPT_FLOAT 0x0100
#define OPT_SUBTITLE 0x0200
#define OPT_INT64 0x0400
#define OPT_EXIT 0x0800
#define OPT_DATA 0x1000
#define OPT_PERFILE 0x2000 /* the option is per-file (currently ffmpeg-only).
implied by OPT_OFFSET or OPT_SPEC */
#define OPT_FUNC2 0x2000
#define OPT_OFFSET 0x4000 /* option is specified as an offset in a passed optctx */
#define OPT_SPEC 0x8000 /* option is to be stored in an array of SpecifierOpt.
Implies OPT_OFFSET. Next element after the offset is
an int containing element count in the array. */
#define OPT_TIME 0x10000
#define OPT_DOUBLE 0x20000
#define OPT_INPUT 0x40000
#define OPT_OUTPUT 0x80000
union {
void *dst_ptr;
int (*func_arg)(void *, const char *, const char *);
int (*func_arg)(const char *, const char *);
int (*func2_arg)(void *, const char *, const char *);
size_t off;
} u;
const char *help;
const char *argname;
} OptionDef;
/**
* Print help for all options matching specified flags.
*
* @param options a list of options
* @param msg title of this group. Only printed if at least one option matches.
* @param req_flags print only options which have all those flags set.
* @param rej_flags don't print options which have any of those flags set.
* @param alt_flags print only options that have at least one of those flags set
*/
void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int rej_flags, int alt_flags);
void show_help_options(const OptionDef *options, const char *msg, int mask, int value);
/**
* Show help for all options with given flags in class and all its
@@ -207,23 +168,10 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
*/
void show_help_children(const AVClass *class, int flags);
/**
* Per-fftool specific help handler. Implemented in each
* fftool, called by show_help().
*/
void show_help_default(const char *opt, const char *arg);
/**
* Generic -h handler common to all fftools.
*/
int show_help(void *optctx, const char *opt, const char *arg);
/**
* Parse the command line arguments.
*
* @param optctx an opaque options context
* @param argc number of command line arguments
* @param argv values of command line arguments
* @param options Array with the definitions required to interpret every
* option of the form: -option_name [argument]
* @param parse_arg_function Name of the function called to process every
@@ -238,115 +186,13 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
*
* @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);
int parse_option(void *optctx, const char *opt, const char *arg, const OptionDef *options);
/**
* An option extracted from the commandline.
* Cannot use AVDictionary because of options like -map which can be
* used multiple times.
*/
typedef struct Option {
const OptionDef *opt;
const char *key;
const char *val;
} Option;
typedef struct OptionGroupDef {
/**< group name */
const char *name;
/**
* Option to be used as group separator. Can be NULL for groups which
* are terminated by a non-option argument (e.g. ffmpeg output files)
*/
const char *sep;
/**
* Option flags that must be set on each option that is
* applied to this group
*/
int flags;
} OptionGroupDef;
typedef struct OptionGroup {
const OptionGroupDef *group_def;
const char *arg;
Option *opts;
int nb_opts;
AVDictionary *codec_opts;
AVDictionary *format_opts;
AVDictionary *resample_opts;
struct SwsContext *sws_opts;
AVDictionary *swr_opts;
} OptionGroup;
/**
* A list of option groups that all have the same group type
* (e.g. input files or output files)
*/
typedef struct OptionGroupList {
const OptionGroupDef *group_def;
OptionGroup *groups;
int nb_groups;
} OptionGroupList;
typedef struct OptionParseContext {
OptionGroup global_opts;
OptionGroupList *groups;
int nb_groups;
/* parsing state */
OptionGroup cur_group;
} OptionParseContext;
/**
* Parse an options group and write results into optctx.
*
* @param optctx an app-specific options context. NULL for global options group
*/
int parse_optgroup(void *optctx, OptionGroup *g);
/**
* Split the commandline into an intermediate form convenient for further
* processing.
*
* The commandline is assumed to be composed of options which either belong to a
* group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
* (everything else).
*
* A group (defined by an OptionGroupDef struct) is a sequence of options
* terminated by either a group separator option (e.g. -i) or a parameter that
* is not an option (doesn't start with -). A group without a separator option
* must always be first in the supplied groups list.
*
* All options within the same group are stored in one OptionGroup struct in an
* OptionGroupList, all groups with the same group definition are stored in one
* OptionGroupList in OptionParseContext.groups. The order of group lists is the
* same as the order of group definitions.
*/
int split_commandline(OptionParseContext *octx, int argc, char *argv[],
const OptionDef *options,
const OptionGroupDef *groups, int nb_groups);
/**
* Free all allocated memory in an OptionParseContext.
*/
void uninit_parse_context(OptionParseContext *octx);
/**
* Find the '-loglevel' option in the command line args and apply it.
* Find the '-loglevel' option in the commandline 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.
*
@@ -364,16 +210,11 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
* Create a new options dictionary containing only the options from
* opts which apply to the codec with ID codec_id.
*
* @param opts dictionary to place options in
* @param codec_id ID of the codec that should be filtered for
* @param s Corresponding format context.
* @param st A stream from s for which the options should be filtered.
* @param codec The particular codec for which the options should be filtered.
* If null, the default one is looked up according to the codec id.
* @return a pointer to the created dictionary
*/
AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
AVFormatContext *s, AVStream *st, AVCodec *codec);
AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec, AVFormatContext *s, AVStream *st);
/**
* Setup AVCodecContext options for avformat_find_stream_info().
@@ -386,8 +227,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
* @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);
AVDictionary **setup_find_stream_info_opts(AVFormatContext *s, AVDictionary *codec_opts);
/**
* Print an error message to stderr, indicating filename and a human
@@ -413,101 +253,62 @@ void show_banner(int argc, char **argv, const OptionDef *options);
* libraries.
* This option processing function does not utilize the arguments.
*/
int show_version(void *optctx, const char *opt, const char *arg);
/**
* Print the build configuration of the program to stdout. The contents
* depend on the definition of FFMPEG_CONFIGURATION.
* This option processing function does not utilize the arguments.
*/
int show_buildconf(void *optctx, const char *opt, const char *arg);
int opt_version(const char *opt, const char *arg);
/**
* Print the license of the program to stdout. The license depends on
* the license of the libraries compiled into the program.
* This option processing function does not utilize the arguments.
*/
int show_license(void *optctx, const char *opt, const char *arg);
int opt_license(const char *opt, const char *arg);
/**
* Print a listing containing all the formats supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the devices supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_devices(void *optctx, const char *opt, const char *arg);
int opt_formats(const char *opt, const char *arg);
/**
* Print a listing containing all the codecs supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_codecs(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the decoders supported by the
* program.
*/
int show_decoders(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the encoders supported by the
* program.
*/
int show_encoders(void *optctx, const char *opt, const char *arg);
int opt_codecs(const char *opt, const char *arg);
/**
* Print a listing containing all the filters supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_filters(void *optctx, const char *opt, const char *arg);
int opt_filters(const char *opt, const char *arg);
/**
* Print a listing containing all the bit stream filters supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_bsfs(void *optctx, const char *opt, const char *arg);
int opt_bsfs(const char *opt, const char *arg);
/**
* Print a listing containing all the protocols supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_protocols(void *optctx, const char *opt, const char *arg);
int opt_protocols(const char *opt, const char *arg);
/**
* Print a listing containing all the pixel formats supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_pix_fmts(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the standard channel layouts supported by
* the program.
* This option processing function does not utilize the arguments.
*/
int show_layouts(void *optctx, const char *opt, const char *arg);
int opt_pix_fmts(const char *opt, const char *arg);
/**
* Print a listing containing all the sample formats supported by the
* program.
*/
int show_sample_fmts(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the color names and values recognized
* by the program.
*/
int show_colors(void *optctx, const char *opt, const char *arg);
int show_sample_fmts(const char *opt, const char *arg);
/**
* Return a positive value if a line read from standard input
@@ -519,10 +320,9 @@ int read_yesno(void);
* Read the file with name filename, and put its content in a newly
* allocated 0-terminated buffer.
*
* @param filename file to read from
* @param bufptr location where pointer to buffer is returned
* @param size location where size of buffer is returned
* @return >= 0 in case of success, a negative value corresponding to an
* @return 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
@@ -536,7 +336,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
* at configuration time or in a "ffpresets" folder along the executable
* on win32, in that order. If no such file is found and
* codec_name is defined, then search for a file named
* codec_name-preset_name.avpreset in the above-mentioned directories.
* codec_name-preset_name.ffpreset in the above-mentioned directories.
*
* @param filename buffer where the name of the found filename is written
* @param filename_size size in bytes of the filename buffer
@@ -548,39 +348,20 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
FILE *get_preset_file(char *filename, size_t filename_size,
const char *preset_name, int is_path, const char *codec_name);
/**
* Do all the necessary cleanup and abort.
* This function is implemented in the avtools, not cmdutils.
*/
void exit_program(int ret);
/**
* Realloc array to hold new_size elements of elem_size.
* Calls exit() on failure.
* Calls exit_program() on failure.
*
* @param array array to reallocate
* @param elem_size size in bytes of each element
* @param size new element count will be written here
* @param new_size number of elements to place in reallocated array
* @return reallocated array
*/
void *grow_array(void *array, int elem_size, int *size, int new_size);
#define media_type_string av_get_media_type_string
#define GROW_ARRAY(array, nb_elems)\
array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
#define GET_PIX_FMT_NAME(pix_fmt)\
const char *name = av_get_pix_fmt_name(pix_fmt);
#define GET_SAMPLE_FMT_NAME(sample_fmt)\
const char *name = av_get_sample_fmt_name(sample_fmt)
#define GET_SAMPLE_RATE_NAME(rate)\
char name[16];\
snprintf(name, sizeof(name), "%d", rate);
#define GET_CH_LAYOUT_NAME(ch_layout)\
char name[16];\
snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
#define GET_CH_LAYOUT_DESC(ch_layout)\
char name[128];\
av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
#endif /* CMDUTILS_H */

View File

@@ -1,29 +1,17 @@
{ "L" , OPT_EXIT, {.func_arg = show_license}, "show license" },
{ "h" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "?" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "help" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "-help" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "version" , OPT_EXIT, {.func_arg = show_version}, "show version" },
{ "buildconf" , OPT_EXIT, {.func_arg = show_buildconf}, "show build configuration" },
{ "formats" , OPT_EXIT, {.func_arg = show_formats }, "show available formats" },
{ "devices" , OPT_EXIT, {.func_arg = show_devices }, "show available devices" },
{ "codecs" , OPT_EXIT, {.func_arg = show_codecs }, "show available codecs" },
{ "decoders" , OPT_EXIT, {.func_arg = show_decoders }, "show available decoders" },
{ "encoders" , OPT_EXIT, {.func_arg = show_encoders }, "show available encoders" },
{ "bsfs" , OPT_EXIT, {.func_arg = show_bsfs }, "show available bit stream filters" },
{ "protocols" , OPT_EXIT, {.func_arg = show_protocols}, "show available protocols" },
{ "filters" , OPT_EXIT, {.func_arg = show_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {.func_arg = show_pix_fmts }, "show available pixel formats" },
{ "layouts" , OPT_EXIT, {.func_arg = show_layouts }, "show standard channel layouts" },
{ "L", OPT_EXIT, {(void*)opt_license}, "show license" },
{ "h", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "?", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "help", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "-help", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "version", OPT_EXIT, {(void*)opt_version}, "show version" },
{ "formats" , OPT_EXIT, {(void*)opt_formats }, "show available formats" },
{ "codecs" , OPT_EXIT, {(void*)opt_codecs }, "show available codecs" },
{ "bsfs" , OPT_EXIT, {(void*)opt_bsfs }, "show available bit stream filters" },
{ "protocols", OPT_EXIT, {(void*)opt_protocols}, "show available protocols" },
{ "filters", OPT_EXIT, {(void*)opt_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {(void*)opt_pix_fmts }, "show available pixel formats" },
{ "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
{ "colors" , OPT_EXIT, {.func_arg = show_colors }, "show available color names" },
{ "loglevel" , HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" },
{ "v", HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" },
{ "report" , 0, {(void*)opt_report}, "generate a report" },
{ "max_alloc" , HAS_ARG, {.func_arg = opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
{ "cpuflags" , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" },
#if CONFIG_OPENCL
{ "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" },
#endif
{ "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
{ "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
{ "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
{ "report", 0, {(void*)opt_report}, "generate a report" },

View File

@@ -1,276 +0,0 @@
/*
* Copyright (C) 2013 Lenny Wang
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavutil/log.h"
#include "libavutil/opencl.h"
#include "libavutil/avstring.h"
#include "cmdutils.h"
typedef struct {
int platform_idx;
int device_idx;
char device_name[64];
int64_t runtime;
} OpenCLDeviceBenchmark;
const char *ocl_bench_source = AV_OPENCL_KERNEL(
inline unsigned char clip_uint8(int a)
{
if (a & (~0xFF))
return (-a)>>31;
else
return a;
}
kernel void unsharp_bench(
global unsigned char *src,
global unsigned char *dst,
global int *mask,
int width,
int height)
{
int i, j, local_idx, lc_idx, sum = 0;
int2 thread_idx, block_idx, global_idx, lm_idx;
thread_idx.x = get_local_id(0);
thread_idx.y = get_local_id(1);
block_idx.x = get_group_id(0);
block_idx.y = get_group_id(1);
global_idx.x = get_global_id(0);
global_idx.y = get_global_id(1);
local uchar data[32][32];
local int lc[128];
for (i = 0; i <= 1; i++) {
lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
for (j = 0; j <= 1; j++) {
lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
}
}
local_idx = thread_idx.y*16 + thread_idx.x;
if (local_idx < 128)
lc[local_idx] = mask[local_idx];
barrier(CLK_LOCAL_MEM_FENCE);
\n#pragma unroll\n
for (i = -4; i <= 4; i++) {
lm_idx.y = 8 + i + thread_idx.y;
\n#pragma unroll\n
for (j = -4; j <= 4; j++) {
lm_idx.x = 8 + j + thread_idx.x;
lc_idx = (i + 4)*8 + j + 4;
sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
}
}
int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
if (global_idx.x < width && global_idx.y < height)
dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
}
);
#define OCLCHECK(method, ... ) \
do { \
status = method(__VA_ARGS__); \
if (status != CL_SUCCESS) { \
av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n", \
av_opencl_errstr(status)); \
ret = AVERROR_EXTERNAL; \
goto end; \
} \
} while (0)
#define CREATEBUF(out, flags, size) \
do { \
out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
if (status != CL_SUCCESS) { \
av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n"); \
ret = AVERROR_EXTERNAL; \
goto end; \
} \
} while (0)
static void fill_rand_int(int *data, int n)
{
int i;
srand(av_gettime());
for (i = 0; i < n; i++)
data[i] = rand();
}
#define OPENCL_NB_ITER 5
static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
{
int i, arg = 0, width = 1920, height = 1088;
int64_t start, ret = 0;
cl_int status;
size_t kernel_len;
char *inbuf;
int *mask;
int buf_size = width * height * sizeof(char);
int mask_size = sizeof(uint32_t) * 128;
cl_mem cl_mask, cl_inbuf, cl_outbuf;
cl_kernel kernel = NULL;
cl_program program = NULL;
size_t local_work_size_2d[2] = {16, 16};
size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
ret = AVERROR(ENOMEM);
goto end;
}
fill_rand_int((int*)inbuf, buf_size/4);
fill_rand_int(mask, mask_size/4);
CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
kernel_len = strlen(ocl_bench_source);
program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
&kernel_len, &status);
if (status != CL_SUCCESS || !program) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
ret = AVERROR_EXTERNAL;
goto end;
}
status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
if (status != CL_SUCCESS) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
ret = AVERROR_EXTERNAL;
goto end;
}
kernel = clCreateKernel(program, "unsharp_bench", &status);
if (status != CL_SUCCESS) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
ret = AVERROR_EXTERNAL;
goto end;
}
OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
buf_size, inbuf, 0, NULL, NULL);
OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
mask_size, mask, 0, NULL, NULL);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
start = av_gettime_relative();
for (i = 0; i < OPENCL_NB_ITER; i++)
OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
clFinish(ext_opencl_env->command_queue);
ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
end:
if (kernel)
clReleaseKernel(kernel);
if (program)
clReleaseProgram(program);
if (cl_inbuf)
clReleaseMemObject(cl_inbuf);
if (cl_outbuf)
clReleaseMemObject(cl_outbuf);
if (cl_mask)
clReleaseMemObject(cl_mask);
av_free(inbuf);
av_free(mask);
return ret;
}
static int compare_ocl_device_desc(const void *a, const void *b)
{
return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
}
int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
{
int i, j, nb_devices = 0, count = 0;
int64_t score = 0;
AVOpenCLDeviceList *device_list;
AVOpenCLDeviceNode *device_node = NULL;
OpenCLDeviceBenchmark *devices = NULL;
cl_platform_id platform;
av_opencl_get_device_list(&device_list);
for (i = 0; i < device_list->platform_num; i++)
nb_devices += device_list->platform_node[i]->device_num;
if (!nb_devices) {
av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
return AVERROR(EINVAL);
}
if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
return AVERROR(ENOMEM);
}
for (i = 0; i < device_list->platform_num; i++) {
for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
device_node = device_list->platform_node[i]->device_node[j];
platform = device_list->platform_node[i]->platform_id;
score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
if (score > 0) {
devices[count].platform_idx = i;
devices[count].device_idx = j;
devices[count].runtime = score;
av_strlcpy(devices[count].device_name, device_node->device_name,
sizeof(devices[count].device_name));
count++;
}
}
}
qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
for (i = 0; i < count; i++)
fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
devices[i].platform_idx, devices[i].device_idx,
devices[i].device_name, devices[i].runtime);
av_opencl_free_device_list(&device_list);
av_free(devices);
return 0;
}
int opt_opencl(void *optctx, const char *opt, const char *arg)
{
char *key, *value;
const char *opts = arg;
int ret = 0;
while (*opts) {
ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
if (ret < 0)
return ret;
ret = av_opencl_set_option(key, value);
if (ret < 0)
return ret;
if (*opts)
opts++;
}
return ret;
}

View File

@@ -10,9 +10,8 @@ ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
BRIEF = CC CXX AS YASM AR LD HOSTCC STRIP CP
SILENT = DEPCC YASMDEP RM RANLIB
MSG = $@
M = @$(call ECHO,$(TAG),$@);
$(foreach VAR,$(BRIEF), \
@@ -21,29 +20,26 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CPPFLAGS) $(CFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS += $(CPPFLAGS) $(CFLAGS)
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
CCFLAGS = $(CFLAGS)
CXXFLAGS := $(CFLAGS) $(CXXFLAGS)
YASMFLAGS += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
HOSTCFLAGS += $(IFLAGS)
LDFLAGS := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
define COMPILE
$(call $(1)DEP,$(1))
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
$($(1)DEP)
$($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.c
$(COMPILE_C)
@@ -51,26 +47,14 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.cpp
$(COMPILE_CXX)
%.o: %.m
$(COMPILE_C)
%.s: %.c
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
%.o: %.S
$(COMPILE_S)
%_host.o: %.c
$(COMPILE_HOSTC)
%.o: %.rc
$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
%.i: %.c
$(CC) $(CCFLAGS) $(CC_E) $<
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.ho: %.h
$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
@@ -89,18 +73,16 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
$(OBJS):
endif
include $(SRC_PATH)/arch.mak
OBJS-$(HAVE_MMX) += $(MMX-OBJS-yes)
OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(EXTRALIBS)
EXAMPLES := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -108,47 +90,31 @@ HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
HEADERS += $(HEADERS-yes)
PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
SRC_DIR := $(SRC_PATH)/lib$(NAME)
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c)
checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
alltools: $(TOOLS)
$(HOSTOBJS): %.o: %.c
$(COMPILE_HOSTC)
$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
define RULES
clean::
$(RM) $(OBJS) $(OBJS:.o=.d)
$(RM) $(HOSTPROGS)
$(RM) $(TOOLS)
endef
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))

View File

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

View File

@@ -1,880 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
// MA 02110-1301 USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
// NOTE: this is a partial update of the Avisynth C interface to recognize
// new color spaces added in Avisynth 2.60. By no means is this document
// completely Avisynth 2.60 compliant.
#ifndef __AVISYNTH_C__
#define __AVISYNTH_C__
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
typedef unsigned char BYTE;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVISYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 4 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
AVS_PLANAR_A=1<<4,
AVS_PLANAR_R=1<<5,
AVS_PLANAR_G=1<<6,
AVS_PLANAR_B=1<<7,
AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 1 << 3,
AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
AVS_CS_UPLANEFIRST = 1 << 4, // I420
AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
AVS_CS_PLANAR_FILTER = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
// AVS_CS_YV12 = 1<<3 Reserved
// AVS_CS_I420 = 1<<4 Reserved
AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
AVS_CS_YV24 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1, // YVU 4:4:4 planar
AVS_CS_YV16 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:2 planar
AVS_CS_YV12 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:0 planar
AVS_CS_I420 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YUV 4:2:0 planar
AVS_CS_IYUV = AVS_CS_I420,
AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:1 planar
AVS_CS_YUV9 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:0 planar
AVS_CS_Y8 = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 // Y 4:0:0 planar
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoInfo
//
// AVS_VideoInfo is layed out identicly to VideoInfo
typedef struct AVS_VideoInfo {
int width, height; // width=0 means no video
unsigned fps_numerator, fps_denominator;
int num_frames;
int pixel_type;
int audio_samples_per_second; // 0 means no audio
int sample_type;
INT64 num_audio_samples;
int nchannels;
// Imagetype properties
int image_type;
} AVS_VideoInfo;
// useful functions of the above
AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
{ return (p->width!=0); }
AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
{ return (p->audio_samples_per_second!=0); }
AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
{
switch (p->sample_type) {
case AVS_SAMPLE_INT8: return sizeof(signed char);
case AVS_SAMPLE_INT16: return sizeof(signed short);
case AVS_SAMPLE_INT24: return 3;
case AVS_SAMPLE_INT32: return sizeof(signed int);
case AVS_SAMPLE_FLOAT: return sizeof(float);
default: return 0;
}
}
AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
{ return p->nchannels*avs_bytes_per_channel_sample(p);}
AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
{ return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
{ return bytes / avs_bytes_per_audio_sample(p); }
AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return samples * avs_bytes_per_audio_sample(p); }
AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
{ return p->nchannels; }
AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
{ return p->sample_type;}
// useful mutator
AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
{ p->image_type|=property; }
AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
{ p->image_type&=~property; }
AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
{ if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
{
unsigned x=numerator, y=denominator;
while (y) { // find gcd
unsigned t = x%y; x = y; y = t;
}
p->fps_numerator = numerator/x;
p->fps_denominator = denominator/x;
}
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
BYTE * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
volatile long sequence_number;
volatile long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
volatile long refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
int row_sizeUV, heightUV;
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return p->pitch;}
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_sizeUV;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_sizeUV;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->heightUV;
return 0;
}
return p->height;}
AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
return p->vfb->data + p->offset;}
AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on an AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, int frame_range);
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
AVS_CPUF_SSE3 = 0x100, // PIV+, K8 Venice
AVS_CPUF_SSSE3 = 0x200, // Core 2
AVS_CPUF_SSE4 = 0x400, // Penryn, Wolfdale, Yorkfield
AVS_CPUF_SSE4_1 = 0x400,
AVS_CPUF_SSE4_2 = 0x800, // Nehalem
};
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#ifdef AVSC_NO_DECLSPEC
// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
/*
The following functions needs to have been declared, probably from windows.h
void* malloc(size_t)
void free(void*);
HMODULE LoadLibrary(const char*);
void* GetProcAddress(HMODULE, const char*);
FreeLibrary(HMODULE);
*/
typedef struct AVS_Library AVS_Library;
#define AVSC_DECLARE_FUNC(name) name##_func name
struct AVS_Library {
HMODULE handle;
AVSC_DECLARE_FUNC(avs_add_function);
AVSC_DECLARE_FUNC(avs_at_exit);
AVSC_DECLARE_FUNC(avs_bit_blt);
AVSC_DECLARE_FUNC(avs_check_version);
AVSC_DECLARE_FUNC(avs_clip_get_error);
AVSC_DECLARE_FUNC(avs_copy_clip);
AVSC_DECLARE_FUNC(avs_copy_value);
AVSC_DECLARE_FUNC(avs_copy_video_frame);
AVSC_DECLARE_FUNC(avs_create_script_environment);
AVSC_DECLARE_FUNC(avs_delete_script_environment);
AVSC_DECLARE_FUNC(avs_function_exists);
AVSC_DECLARE_FUNC(avs_get_audio);
AVSC_DECLARE_FUNC(avs_get_cpu_flags);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_get_frame);
AVSC_DECLARE_FUNC(avs_get_parity);
AVSC_DECLARE_FUNC(avs_get_var);
AVSC_DECLARE_FUNC(avs_get_version);
AVSC_DECLARE_FUNC(avs_get_video_info);
AVSC_DECLARE_FUNC(avs_invoke);
AVSC_DECLARE_FUNC(avs_make_writable);
AVSC_DECLARE_FUNC(avs_new_c_filter);
AVSC_DECLARE_FUNC(avs_new_video_frame_a);
AVSC_DECLARE_FUNC(avs_release_clip);
AVSC_DECLARE_FUNC(avs_release_value);
AVSC_DECLARE_FUNC(avs_release_video_frame);
AVSC_DECLARE_FUNC(avs_save_string);
AVSC_DECLARE_FUNC(avs_set_cache_hints);
AVSC_DECLARE_FUNC(avs_set_global_var);
AVSC_DECLARE_FUNC(avs_set_memory_max);
AVSC_DECLARE_FUNC(avs_set_to_clip);
AVSC_DECLARE_FUNC(avs_set_var);
AVSC_DECLARE_FUNC(avs_set_working_dir);
AVSC_DECLARE_FUNC(avs_sprintf);
AVSC_DECLARE_FUNC(avs_subframe);
AVSC_DECLARE_FUNC(avs_subframe_planar);
AVSC_DECLARE_FUNC(avs_take_clip);
AVSC_DECLARE_FUNC(avs_vsprintf);
};
#undef AVSC_DECLARE_FUNC
AVSC_INLINE AVS_Library * avs_load_library() {
AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
if (!library)
return NULL;
library->handle = LoadLibrary("avisynth");
if (library->handle == NULL)
goto fail;
#define __AVSC_STRINGIFY(x) #x
#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
#define AVSC_LOAD_FUNC(name) {\
library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
if (library->name == NULL)\
goto fail;\
}
AVSC_LOAD_FUNC(avs_add_function);
AVSC_LOAD_FUNC(avs_at_exit);
AVSC_LOAD_FUNC(avs_bit_blt);
AVSC_LOAD_FUNC(avs_check_version);
AVSC_LOAD_FUNC(avs_clip_get_error);
AVSC_LOAD_FUNC(avs_copy_clip);
AVSC_LOAD_FUNC(avs_copy_value);
AVSC_LOAD_FUNC(avs_copy_video_frame);
AVSC_LOAD_FUNC(avs_create_script_environment);
AVSC_LOAD_FUNC(avs_delete_script_environment);
AVSC_LOAD_FUNC(avs_function_exists);
AVSC_LOAD_FUNC(avs_get_audio);
AVSC_LOAD_FUNC(avs_get_cpu_flags);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_get_frame);
AVSC_LOAD_FUNC(avs_get_parity);
AVSC_LOAD_FUNC(avs_get_var);
AVSC_LOAD_FUNC(avs_get_version);
AVSC_LOAD_FUNC(avs_get_video_info);
AVSC_LOAD_FUNC(avs_invoke);
AVSC_LOAD_FUNC(avs_make_writable);
AVSC_LOAD_FUNC(avs_new_c_filter);
AVSC_LOAD_FUNC(avs_new_video_frame_a);
AVSC_LOAD_FUNC(avs_release_clip);
AVSC_LOAD_FUNC(avs_release_value);
AVSC_LOAD_FUNC(avs_release_video_frame);
AVSC_LOAD_FUNC(avs_save_string);
AVSC_LOAD_FUNC(avs_set_cache_hints);
AVSC_LOAD_FUNC(avs_set_global_var);
AVSC_LOAD_FUNC(avs_set_memory_max);
AVSC_LOAD_FUNC(avs_set_to_clip);
AVSC_LOAD_FUNC(avs_set_var);
AVSC_LOAD_FUNC(avs_set_working_dir);
AVSC_LOAD_FUNC(avs_sprintf);
AVSC_LOAD_FUNC(avs_subframe);
AVSC_LOAD_FUNC(avs_subframe_planar);
AVSC_LOAD_FUNC(avs_take_clip);
AVSC_LOAD_FUNC(avs_vsprintf);
#undef __AVSC_STRINGIFY
#undef AVSC_STRINGIFY
#undef AVSC_LOAD_FUNC
return library;
fail:
free(library);
return NULL;
}
AVSC_INLINE void avs_free_library(AVS_Library *library) {
if (!library)
return;
FreeLibrary(library->handle);
free(library);
}
#endif
#endif

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

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@@ -1,728 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
// MA 02110-1301 USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef __AVXSYNTH_C__
#define __AVXSYNTH_C__
#include "windowsPorts/windows2linux.h"
#include <stdarg.h>
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVXSYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 3 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar
AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar
AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR // same as above
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoInfo
//
// AVS_VideoInfo is layed out identicly to VideoInfo
typedef struct AVS_VideoInfo {
int width, height; // width=0 means no video
unsigned fps_numerator, fps_denominator;
int num_frames;
int pixel_type;
int audio_samples_per_second; // 0 means no audio
int sample_type;
INT64 num_audio_samples;
int nchannels;
// Imagetype properties
int image_type;
} AVS_VideoInfo;
// useful functions of the above
AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
{ return (p->width!=0); }
AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
{ return (p->audio_samples_per_second!=0); }
AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
{
switch (p->sample_type) {
case AVS_SAMPLE_INT8: return sizeof(signed char);
case AVS_SAMPLE_INT16: return sizeof(signed short);
case AVS_SAMPLE_INT24: return 3;
case AVS_SAMPLE_INT32: return sizeof(signed int);
case AVS_SAMPLE_FLOAT: return sizeof(float);
default: return 0;
}
}
AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
{ return p->nchannels*avs_bytes_per_channel_sample(p);}
AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
{ return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
{ return bytes / avs_bytes_per_audio_sample(p); }
AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return samples * avs_bytes_per_audio_sample(p); }
AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
{ return p->nchannels; }
AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
{ return p->sample_type;}
// useful mutator
AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
{ p->image_type|=property; }
AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
{ p->image_type&=~property; }
AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
{ if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
{
unsigned x=numerator, y=denominator;
while (y) { // find gcd
unsigned t = x%y; x = y; y = t;
}
p->fps_numerator = numerator/x;
p->fps_denominator = denominator/x;
}
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
unsigned char * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
long sequence_number;
long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
int refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return p->pitch;}
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_size>>1;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_size>>1;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->height>>1;
return 0;
}
return p->height;}
AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
return p->vfb->data + p->offset;}
AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
INT64 integer64; // match addition of __int64 to avxplugin.h
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
#if defined __cplusplus
}
#endif // __cplusplus
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on am AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, size_t frame_range);
#if defined __cplusplus
}
#endif // __cplusplus
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
#if defined __cplusplus
}
#endif // __cplusplus
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
};
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
#if defined __cplusplus
}
#endif // __cplusplus
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#if defined __cplusplus
}
#endif // __cplusplus
#endif //__AVXSYNTH_C__

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

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

View File

@@ -1,35 +0,0 @@
/*
* Work around broken floating point limits on some systems.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include_next <float.h>
#ifdef FLT_MAX
#undef FLT_MAX
#define FLT_MAX 3.40282346638528859812e+38F
#undef FLT_MIN
#define FLT_MIN 1.17549435082228750797e-38F
#undef DBL_MAX
#define DBL_MAX ((double)1.79769313486231570815e+308L)
#undef DBL_MIN
#define DBL_MIN ((double)2.22507385850720138309e-308L)
#endif

View File

@@ -1,22 +0,0 @@
/*
* Work around broken floating point limits on some systems.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include_next <limits.h>
#include <float.h>

View File

@@ -1,84 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* This file was copied from the following newsgroup posting:
*
* Newsgroups: mod.std.unix
* Subject: public domain AT&T getopt source
* Date: 3 Nov 85 19:34:15 GMT
*
* Here's something you've all been waiting for: the AT&T public domain
* source for getopt(3). It is the code which was given out at the 1985
* UNIFORUM conference in Dallas. I obtained it by electronic mail
* directly from AT&T. The people there assure me that it is indeed
* in the public domain.
*/
#include <stdio.h>
#include <string.h>
static int opterr = 1;
static int optind = 1;
static int optopt;
static char *optarg;
static int getopt(int argc, char *argv[], char *opts)
{
static int sp = 1;
int c;
char *cp;
if (sp == 1) {
if (optind >= argc ||
argv[optind][0] != '-' || argv[optind][1] == '\0')
return EOF;
else if (!strcmp(argv[optind], "--")) {
optind++;
return EOF;
}
}
optopt = c = argv[optind][sp];
if (c == ':' || !(cp = strchr(opts, c))) {
fprintf(stderr, ": illegal option -- %c\n", c);
if (argv[optind][++sp] == '\0') {
optind++;
sp = 1;
}
return '?';
}
if (*++cp == ':') {
if (argv[optind][sp+1] != '\0')
optarg = &argv[optind++][sp+1];
else if(++optind >= argc) {
fprintf(stderr, ": option requires an argument -- %c\n", c);
sp = 1;
return '?';
} else
optarg = argv[optind++];
sp = 1;
} else {
if (argv[optind][++sp] == '\0') {
sp = 1;
optind++;
}
optarg = NULL;
}
return c;
}

View File

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

View File

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

View File

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

View File

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

View File

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

View File

@@ -1,93 +0,0 @@
/*
* C99-compatible strtod() implementation
* Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <limits.h>
#include <stdlib.h>
#include "libavutil/avstring.h"
#include "libavutil/mathematics.h"
static char *check_nan_suffix(char *s)
{
char *start = s;
if (*s++ != '(')
return start;
while ((*s >= 'a' && *s <= 'z') || (*s >= 'A' && *s <= 'Z') ||
(*s >= '0' && *s <= '9') || *s == '_')
s++;
return *s == ')' ? s + 1 : start;
}
#undef strtod
double strtod(const char *, char **);
double avpriv_strtod(const char *nptr, char **endptr)
{
char *end;
double res;
/* Skip leading spaces */
while (av_isspace(*nptr))
nptr++;
if (!av_strncasecmp(nptr, "infinity", 8)) {
end = nptr + 8;
res = INFINITY;
} else if (!av_strncasecmp(nptr, "inf", 3)) {
end = nptr + 3;
res = INFINITY;
} else if (!av_strncasecmp(nptr, "+infinity", 9)) {
end = nptr + 9;
res = INFINITY;
} else if (!av_strncasecmp(nptr, "+inf", 4)) {
end = nptr + 4;
res = INFINITY;
} else if (!av_strncasecmp(nptr, "-infinity", 9)) {
end = nptr + 9;
res = -INFINITY;
} else if (!av_strncasecmp(nptr, "-inf", 4)) {
end = nptr + 4;
res = -INFINITY;
} else if (!av_strncasecmp(nptr, "nan", 3)) {
end = check_nan_suffix(nptr + 3);
res = NAN;
} else if (!av_strncasecmp(nptr, "+nan", 4) ||
!av_strncasecmp(nptr, "-nan", 4)) {
end = check_nan_suffix(nptr + 4);
res = NAN;
} else if (!av_strncasecmp(nptr, "0x", 2) ||
!av_strncasecmp(nptr, "-0x", 3) ||
!av_strncasecmp(nptr, "+0x", 3)) {
/* FIXME this doesn't handle exponents, non-integers (float/double)
* and numbers too large for long long */
res = strtoll(nptr, &end, 16);
} else {
res = strtod(nptr, &end);
}
if (endptr)
*endptr = end;
return res;
}

View File

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

View File

@@ -1,29 +0,0 @@
/*
* MSVC Compatible va_copy macro
* Copyright (c) 2012 Derek Buitenhuis
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdarg.h>
#if !defined(va_copy) && defined(_MSC_VER)
#define va_copy(dst, src) ((dst) = (src))
#endif
#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
#define va_copy(dst, src) __va_copy(dst, src)
#endif

View File

@@ -1,132 +0,0 @@
#!/bin/sh
# Copyright (c) 2013, Derek Buitenhuis
#
# Permission to use, copy, modify, and/or distribute this software for any
# purpose with or without fee is hereby granted, provided that the above
# copyright notice and this permission notice appear in all copies.
#
# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
# mktemp isn't POSIX, so supply an implementation
mktemp() {
echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
}
if [ $# -lt 2 ]; then
echo "Usage: makedef <version_script> <objects>" >&2
exit 0
fi
vscript=$1
shift
if [ ! -f "$vscript" ]; then
echo "Version script does not exist" >&2
exit 1
fi
for object in "$@"; do
if [ ! -f "$object" ]; then
echo "Object does not exist: ${object}" >&2
exit 1
fi
done
# Create a lib temporarily to dump symbols from.
# It's just much easier to do it this way
libname=$(mktemp -u "library").lib
trap 'rm -f -- $libname' EXIT
lib -out:${libname} $@ >/dev/null
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
exit 1
fi
IFS='
'
# Determine if we're building for x86 or x86_64 and
# set the symbol prefix accordingly.
prefix=""
arch=$(dumpbin -headers ${libname} |
tr '\t' ' ' |
grep '^ \+.\+machine \+(.\+)' |
head -1 |
sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
if [ "${arch}" = "x86" ]; then
prefix="_"
else
if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
echo "Unknown machine type." >&2
exit 1
fi
fi
started=0
regex="none"
for line in $(cat ${vscript} | tr '\t' ' '); do
# We only care about global symbols
echo "${line}" | grep -q '^ \+global:'
if [ $? = 0 ]; then
started=1
line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
else
echo "${line}" | grep -q '^ \+local:'
if [ $? = 0 ]; then
started=0
fi
fi
if [ ${started} = 0 ]; then
continue
fi
# Handle multiple symbols on one line
IFS=';'
# Work around stupid expansion to filenames
line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
for exp in ${line}; do
# Remove leading and trailing whitespace
exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
if [ "${regex}" = "none" ]; then
regex="${exp}"
else
regex="${regex};${exp}"
fi
done
IFS='
'
done
dump=$(dumpbin -linkermember:1 ${libname})
rm ${libname}
IFS=';'
list=""
for exp in ${regex}; do
list="${list}"'
'$(echo "${dump}" |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3 |
grep "^${exp}" |
sed -e 's/^/ /')
done
echo "EXPORTS"
echo "${list}" | sort | uniq | tail -n +2

4775
configure vendored

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@@ -1,182 +1,53 @@
LIBRARIES-$(CONFIG_AVUTIL) += libavutil
LIBRARIES-$(CONFIG_SWSCALE) += libswscale
LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
LIBRARIES-$(CONFIG_AVCODEC) += libavcodec
LIBRARIES-$(CONFIG_AVFORMAT) += libavformat
LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice
LIBRARIES-$(CONFIG_AVFILTER) += libavfilter
COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils
COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler
COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters
COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols
COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices
COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters
MANPAGES1 = $(AVPROGS-yes:%=doc/%.1) $(AVPROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1)
MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3)
MANPAGES = $(MANPAGES1) $(MANPAGES3)
PODPAGES = $(AVPROGS-yes:%=doc/%.pod) $(AVPROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
MANPAGES = $(PROGS-yes:%=doc/%.1)
PODPAGES = $(PROGS-yes:%=doc/%.pod)
HTMLPAGES = $(PROGS-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/nut.html \
doc/libavfilter.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
all-$(CONFIG_DOC): documentation
DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
DOCS-$(CONFIG_PODPAGES) += $(PODPAGES)
DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec
DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
PROGS += $(DOC_EXAMPLES)
all-$(CONFIG_DOC): doc
doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
examples: $(DOC_EXAMPLES)
TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
GENTEXI = format codec
GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
$(GENTEXI): TAG = GENTEXI
$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
TEXIDEP = awk '/^@include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
endif
$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
doc/%.pod: doc/%.texi
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
doc/%-all.pod: TAG = POD
doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
$(DOCS): | doc
OBJDIRS += doc
DOXY_INPUT = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
install-doc: install-html install-man
install-html:
install-man:
ifdef CONFIG_HTMLPAGES
install-progs-$(CONFIG_DOC): install-html
install-html: $(HTMLPAGES)
$(Q)mkdir -p "$(DOCDIR)"
$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
endif
ifdef CONFIG_MANPAGES
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
$(Q)mkdir -p "$(MANDIR)/man3"
$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
endif
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
uninstall: uninstall-doc
uninstall-doc: uninstall-html uninstall-man
uninstall-html:
$(RM) -r "$(DOCDIR)"
uninstall: uninstall-man
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
clean:: docclean
distclean:: docclean
$(RM) doc/config.texi
examplesclean:
$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
docclean: examplesclean
$(RM) $(CLEANSUFFIXES:%=doc/%)
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
$(RM) -r doc/doxy/html
clean::
$(RM) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
-include $(wildcard $(DOCS:%=%.d))
.PHONY: apidoc doc documentation
.PHONY: documentation

17
doc/RELEASE_NOTES Normal file
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@@ -0,0 +1,17 @@
Release Notes
=============
* 0.9 "Harmony" December, 2011
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.

View File

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

1041
doc/avconv.texi Normal file

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@@ -0,0 +1,167 @@
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".
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
@code{a:1} stream specifer, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several stream, the option is then applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams, for example @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data and 't' for attachments. If @var{stream_index} is given, then
matches stream number @var{stream_index} of this type. Otherwise matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then matches stream number @var{stream_index} in
program with id @var{program_id}. Otherwise matches all streams in this program.
@end table
@section Generic options
These options are shared amongst the av* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help
Show help.
@item -version
Show version.
@item -formats
Show available formats.
The fields preceding the format names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@end table
@item -codecs
Show available codecs.
The fields preceding the codec names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@item V/A/S
Video/audio/subtitle codec
@item S
Codec supports slices
@item D
Codec supports direct rendering
@item T
Codec can handle input truncated at random locations instead of only at frame boundaries
@end table
@item -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -loglevel @var{loglevel} | -v @var{loglevel}
Set the logging level used by the library.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
@item panic
@item fatal
@item error
@item warning
@item info
@item verbose
@item debug
@end table
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{FFMPEG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{FFMPEG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Note: setting the environment variable @code{FFREPORT} to any value has the
same effect.
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are obviously per-stream, so the chapter on stream
specifiers applies to them
Note @option{-nooption} syntax cannot be used for boolean AVOptions,
use @option{-option 0}/@option{-option 1}.
Note2 old undocumented way of specifying per-stream AVOptions by prepending
v/a/s to the options name is now obsolete and will be removed soon.

36
doc/avutil.txt Normal file
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@@ -0,0 +1,36 @@
AVUtil
======
libavutil is a small lightweight library of generally useful functions.
It is not a library for code needed by both libavcodec and libavformat.
Overview:
=========
adler32.c adler32 checksum
aes.c AES encryption and decryption
fifo.c resizeable first in first out buffer
intfloat_readwrite.c portable reading and writing of floating point values
log.c "printf" with context and level
md5.c MD5 Message-Digest Algorithm
rational.c code to perform exact calculations with rational numbers
tree.c generic AVL tree
crc.c generic CRC checksumming code
integer.c 128bit integer math
lls.c
mathematics.c greatest common divisor, integer sqrt, integer log2, ...
mem.c memory allocation routines with guaranteed alignment
Headers:
bswap.h big/little/native-endian conversion code
x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code
avutil.h
common.h
intreadwrite.h reading and writing of unaligned big/little/native-endian integers
Goals:
======
* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
* Small (source and object)
* Efficient (low CPU and memory usage)
* Useful (avoid useless features almost no one needs)

View File

@@ -17,75 +17,13 @@ Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
bitstream filter.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
ADTS header and removes the ADTS header.
This is required for example when copying an AAC stream from a raw
ADTS AAC container to a FLV or a MOV/MP4 file.
@section chomp
Remove zero padding at the end of a packet.
@section dump_extra
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item a
add extradata to all key packets, but only if @var{local_header} is
set in the @option{flags2} codec context field
@item k
add extradata to all key packets
@item e
add extradata to all packets
@end table
If not specified it is assumed @samp{k}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the @code{libx264} encoder, but corrects them by adding
the header stored in extradata to the key packets:
@example
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dump_extradata
@section h264_mp4toannexb
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 imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut
Pro decoder. This filter only applies to the mpeg2video codec, and is
likely not needed for Final Cut Pro 7 and newer with the appropriate
@option{-tag:v}.
For example, to remux 30 MB/sec NTSC IMX to MOV:
@example
ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
@end example
@section imx_dump_header
@section mjpeg2jpeg
@@ -119,7 +57,7 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
@example
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
ffmpeg -i mjpeg-movie.avi -c:v copy -vbsf mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@@ -128,17 +66,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@section movsub
@section mp3_header_compress
@section mp3_header_decompress
@section noise
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
@example
ffmpeg -i INPUT -c copy -bsf noise output.mkv
@end example
@section remove_extra
@section remove_extradata
@c man end BITSTREAM FILTERS

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@@ -25,9 +25,6 @@ fate-list
install
Install headers, libraries and programs.
examples
Build all examples located in doc/examples.
libavformat/output-example
Build the libavformat basic example.
@@ -37,9 +34,6 @@ libavcodec/api-example
libswscale/swscale-test
Build the swscale self-test (useful also as example).
config
Reconfigure the project with current configuration.
Useful standard make commands:
make -t <target>

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@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-decoders} of the ff* tools will display the list of
The option @code{-codecs} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@@ -48,189 +48,3 @@ top-field-first is assumed
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
A description of some of the currently available audio decoders
follows.
@section ac3
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
@subsection AC-3 Decoder Options
@table @option
@item -drc_scale @var{value}
Dynamic Range Scale Factor. The factor to apply to dynamic range values
from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
DRC disabled. Produces full range audio.
@item 0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value.
Audio reproduction is between full range and full compression.
@item drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically.
Loud sounds are fully compressed. Soft sounds are enhanced.
@end table
@end table
@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.
@section libopus
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
also be found in VobSub file pairs and in some Matroska files.
@subsection Options
@table @option
@item palette
Specify the global palette used by the bitmaps. When stored in VobSub, the
palette is normally specified in the index file; in Matroska, the palette is
stored in the codec extra-data in the same format as in VobSub. In DVDs, the
palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@end table
@section libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libzvbi}.
@subsection Options
@table @option
@item txt_page
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
Y offset of generated bitmaps, default is 0.
@item txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext charactes.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
miliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque (black) background.
@end table
@c man end SUBTILES DECODERS

View File

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

View File

@@ -1,23 +1,69 @@
@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg that can read the
Demuxers are configured elements in FFmpeg which allow to read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option @code{--list-demuxers}.
configure option "--list-demuxers".
You can disable all the demuxers using the configure option
@code{--disable-demuxers}, and selectively enable a single demuxer with
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
with the option @code{--disable-demuxer=@var{DEMUXER}}.
"--disable-demuxers", and selectively enable a single demuxer with
the option "--enable-demuxer=@var{DEMUXER}", or disable it
with the option "--disable-demuxer=@var{DEMUXER}".
The option @code{-formats} of the ff* tools will display the list of
The option "-formats" of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
@section image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The pattern may contain the string "%d" or "%0@var{N}d", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0@var{N}d" is used, the string representing the number in each
filename is 0-padded and @var{N} is the total number of 0-padded
digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between 0 and 4, all the following numbers must
be sequential. This limitation may be hopefully fixed.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
The following example shows how to use @command{ffmpeg} for creating a
video from the images in the file sequence @file{img-001.jpeg},
@file{img-002.jpeg}, ..., assuming an input frame rate of 10 frames per
second:
@example
ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
@end example
Note that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
ffmpeg -i img.jpeg img.png
@end example
@section applehttp
Apple HTTP Live Streaming demuxer.
@@ -29,407 +75,4 @@ the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
@table @option
@item -no_resync_search @var{bool}
Do not try to resynchronize by looking for a certain optional start code.
@end table
@anchor{concat}
@section concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packet had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
and each next file starts where the previous one finishes. Note that it is
done globally and may cause gaps if all streams do not have exactly the same
length.
All files must have the same streams (same codecs, same time base, etc.).
The duration of each file is used to adjust the timestamps of the next file:
if the duration is incorrect (because it was computed using the bit-rate or
because the file is truncated, for example), it can cause artifacts. The
@code{duration} directive can be used to override the duration stored in
each file.
@subsection Syntax
The script is a text file in extended-ASCII, with one directive per line.
Empty lines, leading spaces and lines starting with '#' are ignored. The
following directive is recognized:
@table @option
@item @code{file @var{path}}
Path to a file to read; special characters and spaces must be escaped with
backslash or single quotes.
All subsequent file-related directives apply to that file.
@item @code{ffconcat version 1.0}
Identify the script type and version. It also sets the @option{safe} option
to 1 if it was to its default -1.
To make FFmpeg recognize the format automatically, this directive must
appears exactly as is (no extra space or byte-order-mark) on the very first
line of the script.
@item @code{duration @var{dur}}
Duration of the file. This information can be specified from the file;
specifying it here may be more efficient or help if the information from the
file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the
whole concatenated video.
@item @code{stream}
Introduce a stream in the virtual file.
All subsequent stream-related directives apply to the last introduced
stream.
Some streams properties must be set in order to allow identifying the
matching streams in the subfiles.
If no streams are defined in the script, the streams from the first file are
copied.
@item @code{exact_stream_id @var{id}}
Set the id of the stream.
If this directive is given, the string with the corresponding id in the
subfiles will be used.
This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
@end table
@subsection Options
This demuxer accepts the following option:
@table @option
@item safe
If set to 1, reject unsafe file paths. A file path is considered safe if it
does not contain a protocol specification and is relative and all components
only contain characters from the portable character set (letters, digits,
period, underscore and hyphen) and have no period at the beginning of a
component.
If set to 0, any file name is accepted.
The default is -1, it is equivalent to 1 if the format was automatically
probed and 0 otherwise.
@item auto_convert
If set to 1, try to perform automatic conversions on packet data to make the
streams concatenable.
Currently, the only conversion is adding the h264_mp4toannexb bitstream
filter to H.264 streams in MP4 format. This is necessary in particular if
there are resolution changes.
@end table
@section flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams.
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@end table
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{http://code.google.com/p/game-music-emu/} for more information.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
For very large files, the @option{max_size} option may have to be adjusted.
@section libquvi
Play media from Internet services using the quvi project.
The demuxer accepts a @option{format} option to request a specific quality. It
is by default set to @var{best}.
See @url{http://quvi.sourceforge.net/} for more information.
FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
enabled.
@section gif
Animated GIF demuxer.
It accepts the following options:
@table @option
@item min_delay
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@item ignore_loop
GIF files can contain information to loop a certain number of times (or
infinitely). If @option{ignore_loop} is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set to 0,
then looping will occur and will cycle the number of times according to
the GIF. Default value is 1.
@end table
For example, with the overlay filter, place an infinitely looping GIF
over another video:
@example
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
Note that in the above example the shortest option for overlay filter is
used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
@section image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The syntax and meaning of the pattern is specified by the
option @var{pattern_type}.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
This demuxer accepts the following options:
@table @option
@item framerate
Set the frame rate for the video stream. It defaults to 25.
@item loop
If set to 1, loop over the input. Default value is 0.
@item pattern_type
Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.
A sequence pattern may contain the string "%d" or "%0@var{N}d", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0@var{N}d" is used, the string representing the number in each
filename is 0-padded and @var{N} is the total number of 0-padded
digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between @var{start_number} and
@var{start_number}+@var{start_number_range}-1, and all the following
numbers must be sequential.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
Note that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
ffmpeg -i img.jpeg img.png
@end example
@item glob
Select a glob wildcard pattern type.
The pattern is interpreted like a @code{glob()} pattern. This is only
selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
interpreted like a @code{glob()} pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters @code{%*?[]@{@}} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern @code{foo-%*.jpeg} will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
Default value is @var{glob_sequence}.
@item pixel_format
Set the pixel format of the images to read. If not specified the pixel
format is guessed from the first image file in the sequence.
@item start_number
Set the index of the file matched by the image file pattern to start
to read from. Default value is 0.
@item start_number_range
Set the index interval range to check when looking for the first image
file in the sequence, starting from @var{start_number}. Default value
is 5.
@item ts_from_file
If set to 1, will set frame timestamp to modification time of image file. Note
that monotonity of timestamps is not provided: images go in the same order as
without this option. Default value is 0.
If set to 2, will set frame timestamp to the modification time of the image file in
nanosecond precision.
@item video_size
Set the video size of the images to read. If not specified the video
size is guessed from the first image file in the sequence.
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} for creating a video from the images in the file
sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
input frame rate of 10 frames per second:
@example
ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
@end example
@item
As above, but start by reading from a file with index 100 in the sequence:
@example
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
@end example
@item
Read images matching the "*.png" glob pattern , that is all the files
terminating with the ".png" suffix:
@example
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section mpegts
MPEG-2 transport stream demuxer.
@table @option
@item fix_teletext_pts
Overrides teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
not discarded. Default value is 1, set this option to 0 if you want your
teletext packet PTS and DTS values untouched.
@end table
@section rawvideo
Raw video demuxer.
This demuxer allows one to read raw video data. Since there is no header
specifying the assumed video parameters, the user must specify them
in order to be able to decode the data correctly.
This demuxer accepts the following options:
@table @option
@item framerate
Set input video frame rate. Default value is 25.
@item pixel_format
Set the input video pixel format. Default value is @code{yuv420p}.
@item video_size
Set the input video size. This value must be specified explicitly.
@end table
For example to read a rawvideo file @file{input.raw} with
@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
size of @code{320x240}, and a frame rate of 10 images per second, use
the command:
@example
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
@section sbg
SBaGen script demuxer.
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.
@section tedcaptions
JSON captions used for @url{http://www.ted.com/, TED Talks}.
TED does not provide links to the captions, but they can be guessed from the
page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
contains a bookmarklet to expose them.
This demuxer accepts the following option:
@table @option
@item start_time
Set the start time of the TED talk, in milliseconds. The default is 15000
(15s). It is used to sync the captions with the downloadable videos, because
they include a 15s intro.
@end table
Example: convert the captions to a format most players understand:
@example
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@c man end DEMUXERS
@c man end INPUT DEVICES

View File

@@ -11,23 +11,28 @@
@chapter Developers Guide
@section Notes for external developers
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in @file{doc/examples} and in the source code to
see how the public API is employed.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{libavformat/output-example.c} to use it to generate
audio or video streams.
You can use the FFmpeg libraries in your commercial program, but you
are encouraged to @emph{publish any patch you make}. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
@end itemize
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{http://ffmpeg.org/legal.html}.
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@section Contributing
@@ -51,16 +56,13 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@@ -92,7 +94,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
* more text ...
* ...
*/
typedef struct Foobar @{
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
@@ -114,17 +116,13 @@ int myfunc(int my_parameter)
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
@@ -136,72 +134,46 @@ clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@subsection Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in the CamelCase
There are the following conventions for naming variables and functions:
There are following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For file-scope variables and functions declared as @code{static}, no prefix
is required.
For variables and functions declared as @code{static} no prefixes are required.
@item
For variables and functions visible outside of file scope, but only used
internally by a library, an @code{ff_} prefix should be used,
e.g. @samp{ff_w64_demuxer}.
For variables and functions used internally by the library, @code{ff_} prefix
should be used.
For example, @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_aac_parse_header}.
For variables and functions used internally across multiple libraries, use
@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
@item
Each library has its own prefix for public symbols, in addition to the
commonly used @code{av_} (@code{avformat_} for libavformat,
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
Check the existing code and choose names accordingly.
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
@code{lib<name>/lib<name>.v} files.
For exported names, each library has its own prefixes. Just check the existing
code and name accordingly.
@end itemize
Furthermore, name space reserved for the system should not be invaded.
Identifiers ending in @code{_t} are reserved by
@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
Also avoid names starting with @code{__} or @code{_} followed by an uppercase
letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@subsection Miscellaneous conventions
@subsection Miscellanous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@@ -215,10 +187,8 @@ the following snippet into your @file{.vimrc}:
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" allow tabs in Makefiles
autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
@@ -228,177 +198,134 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
(setq c-default-style "k&r")
(setq-default c-basic-offset 4)
(setq-default indent-tabs-mode nil)
(setq-default show-trailing-whitespace t)
@end example
@section Development Policy
@enumerate
@item
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
a gift-style license, the
@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
@uref{http://mit-license.org/, MIT} license.
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
@example
area changed: Short 1 line description
details describing what and why and giving references.
@end example
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder).
@item
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help
finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@anchor{Submitting patches}
@section Submitting patches
@@ -418,9 +345,14 @@ 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
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()')
@@ -428,13 +360,6 @@ and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to the
@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
@@ -449,51 +374,38 @@ send a reminder by email. Your patch should eventually be dealt with.
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
Did you add the CodecID to @file{avcodec.h}?
@item
If it has a FourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
If it has a fourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you @code{git add} the appropriate files before committing?
Did you @code{git add} the appropriate files before committing?
@item
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate
@@ -501,109 +413,73 @@ Did you make sure it compiles standalone, i.e. with
@enumerate
@item
Does @code{make fate} pass with the patch applied?
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
Was the patch generated with git format-patch or send-email?
@item
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
Did you sign off your patch? (git commit -s)
See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Does the patch not mix functional and cosmetic changes?
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider to add a regression test for your code.
Consider to add a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
If you added YASM code please check that things still work with --disable-yasm
@end enumerate
@section Patch review process
@@ -632,169 +508,15 @@ 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.
Running 'make fate' accomplishes this, please see @file{doc/fate.txt} 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].
@subsection Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be inlcuded in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductionary message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
the following steps:
@enumerate
@item
Configure to compile with instrumentation enabled:
@code{configure --toolchain=gcov}.
@item
Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.
@item
Run @code{make lcov} to generate coverage data in HTML format.
@item
View @code{lcov/index.html} in your preferred HTML viewer.
@end enumerate
You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@subsection Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
to your configure line, and reasonable defaults will be set for running
FATE under the supervision of either the @strong{memcheck} or the
@strong{massif} tool of the valgrind suite.
In case you need finer control over how valgrind is invoked, use the
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
your configure line instead.
@anchor{Release process}
@section Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a @strong{release
manager} prepares, tests and publishes tarballs on the
@url{http://ffmpeg.org} website.
There are two kinds of releases:
@enumerate
@item
@strong{Major releases} always include the latest and greatest
features and functionality.
@item
@strong{Point releases} are cut from @strong{release} branches,
which are named @code{release/X}, with @code{X} being the release
version number.
@end enumerate
Note that we promise to our users that shared libraries from any FFmpeg
release never break programs that have been @strong{compiled} against
previous versions of @strong{the same release series} in any case!
However, from time to time, we do make API changes that require adaptations
in applications. Such changes are only allowed in (new) major releases and
require further steps such as bumping library version numbers and/or
adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@subsection Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@enumerate
@item
Fixes a security issue, preferably identified by a @strong{CVE
number} issued by @url{http://cve.mitre.org/}.
@item
Fixes a documented bug in @url{https://trac.ffmpeg.org}.
@item
Improves the included documentation.
@item
Retains both source code and binary compatibility with previous
point releases of the same release branch.
@end enumerate
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@subsection Release Checklist
The release process involves the following steps:
@enumerate
@item
Ensure that the @file{RELEASE} file contains the version number for
the upcoming release.
@item
Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
@item
Announce the intent to do a release to the mailing list.
@item
Make sure all relevant security fixes have been backported. See
@url{https://ffmpeg.org/security.html}.
@item
Ensure that the FATE regression suite still passes in the release
branch on at least @strong{i386} and @strong{amd64}
(cf. @ref{Regression tests}).
@item
Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
supplementing files that contain @code{gpg} signatures
@item
Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
push an annotated tag in the form @code{nX}, with @code{X}
containing the version number.
@item
Propose and send a patch to the @strong{ffmpeg-devel} mailing list
with a news entry for the website.
@item
Publish the news entry.
@item
Send announcement to the mailing list.
@end enumerate
@bye

View File

@@ -1,25 +0,0 @@
@chapter Device Options
@c man begin DEVICE OPTIONS
The libavdevice library provides the same interface as
libavformat. Namely, an input device is considered like a demuxer, and
an output device like a muxer, and the interface and generic device
options are the same provided by libavformat (see the ffmpeg-formats
manual).
In addition each input or output device may support so-called private
options, which are specific for that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the device
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
@c man end DEVICE OPTIONS
@ifclear config-writeonly
@include indevs.texi
@end ifclear
@ifclear config-readonly
@include outdevs.texi
@end ifclear

View File

@@ -1,13 +0,0 @@
#!/bin/sh
SRC_PATH="${1}"
DOXYFILE="${2}"
DOXYGEN="${3}"
shift 3
$DOXYGEN - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
EXAMPLE_PATH = ${SRC_PATH}/doc/examples
EOF

File diff suppressed because it is too large Load Diff

183
doc/eval.texi Normal file
View File

@@ -0,0 +1,183 @@
@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.
@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
Note that:
@code{*} works like AND
@code{+} works like OR
thus
@example
if A then B else C
@end example
is equivalent to
@example
A*B + not(A)*C
@end example
In your C code, you can extend the list of unary and binary functions,
and define recognized constants, so that they are available for your
expressions.
The evaluator also recognizes the International System number
postfixes. If 'i' is appended after the postfix, powers of 2 are used
instead of powers of 10. The 'B' postfix multiplies the value for 8,
and can be appended after another postfix or used alone. This allows
using for example 'KB', 'MiB', 'G' and 'B' as postfix.
Follows the list of available International System postfixes, with
indication of the corresponding powers of 10 and of 2.
@table @option
@item y
-24 / -80
@item z
-21 / -70
@item a
-18 / -60
@item f
-15 / -50
@item p
-12 / -40
@item n
-9 / -30
@item u
-6 / -20
@item m
-3 / -10
@item c
-2
@item d
-1
@item h
2
@item k
3 / 10
@item K
3 / 10
@item M
6 / 20
@item G
9 / 30
@item T
12 / 40
@item P
15 / 40
@item E
18 / 50
@item Z
21 / 60
@item Y
24 / 70
@end table
@c man end

View File

@@ -1,44 +1,21 @@
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
# use pkg-config for getting CFLAGS abd LDFLAGS
FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
CFLAGS+=$(shell pkg-config --cflags $(FFMPEG_LIBS))
LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
filtering_video \
filtering_audio \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
EXAMPLES=decoding_encoding filtering metadata muxing
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
%: %.o
$(CC) $< $(LDFLAGS) -o $@
.phony: all clean-test clean
%.o: %.c
$(CC) $< $(CFLAGS) -c -o $@
.phony: all clean
all: $(OBJS) $(EXAMPLES)
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)
clean:
rm -rf $(EXAMPLES) $(OBJS)

View File

@@ -1,23 +0,0 @@
FFmpeg examples README
----------------------
Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
Method 2: build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then just run "make examples".
This will build the examples using the FFmpeg build system. You can clean those
examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make.

View File

@@ -1,134 +0,0 @@
/*
* Copyright (c) 2014 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat AVIOContext API example.
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
* @example avio_reading.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
struct buffer_data {
uint8_t *ptr;
size_t size; ///< size left in the buffer
};
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
{
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int main(int argc, char *argv[])
{
AVFormatContext *fmt_ctx = NULL;
AVIOContext *avio_ctx = NULL;
uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
size_t buffer_size, avio_ctx_buffer_size = 4096;
char *input_filename = NULL;
int ret = 0;
struct buffer_data bd = { 0 };
if (argc != 2) {
fprintf(stderr, "usage: %s input_file\n"
"API example program to show how to read from a custom buffer "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
input_filename = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
goto end;
/* fill opaque structure used by the AVIOContext read callback */
bd.ptr = buffer;
bd.size = buffer_size;
if (!(fmt_ctx = avformat_alloc_context())) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
if (!avio_ctx_buffer) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
ret = AVERROR(ENOMEM);
goto end;
}
fmt_ctx->pb = avio_ctx;
ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open input\n");
goto end;
}
ret = avformat_find_stream_info(fmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Could not find stream information\n");
goto end;
}
av_dump_format(fmt_ctx, 0, input_filename, 0);
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

View File

@@ -24,79 +24,21 @@
* @file
* libavcodec API use example.
*
* @example decoding_encoding.c
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
*/
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#include "libavutil/imgutils.h"
#include "libavutil/opt.h"
#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
@@ -104,134 +46,64 @@ static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
int frame_size, i, j, out_size, outbuf_size;
FILE *f;
uint16_t *samples;
short *samples;
float t, tincr;
uint8_t *outbuf;
printf("Encode audio file %s\n", filename);
printf("Audio encoding\n");
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
codec = avcodec_find_encoder(CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_rate = 44100;
c->channels = 2;
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* the codec gives us the frame size, in samples */
frame_size = c->frame_size;
samples = malloc(frame_size * 2 * c->channels);
outbuf_size = 10000;
outbuf = malloc(outbuf_size);
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
if (buffer_size < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
exit(1);
}
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
for(i=0;i<200;i++) {
for(j=0;j<frame_size;j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
samples[2*j+1] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
fwrite(outbuf, 1, out_size, f);
}
fclose(f);
free(outbuf);
free(samples);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
@@ -251,30 +123,26 @@ static void audio_decode_example(const char *outfilename, const char *filename)
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
printf("Audio decoding\n");
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
codec = avcodec_find_decoder(CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
@@ -291,11 +159,12 @@ static void audio_decode_example(const char *outfilename, const char *filename)
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
if (!(decoded_frame = avcodec_alloc_frame())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
}
} else
avcodec_get_frame_defaults(decoded_frame);
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
@@ -307,11 +176,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
@@ -337,7 +201,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
avcodec_close(c);
av_free(c);
av_frame_free(&decoded_frame);
av_free(decoded_frame);
}
/*
@@ -347,26 +211,22 @@ static void video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y, got_output;
int i, out_size, size, x, y, outbuf_size;
FILE *f;
AVFrame *frame;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
AVFrame *picture;
uint8_t *outbuf;
printf("Encode video file %s\n", filename);
printf("Video encoding\n");
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
picture= avcodec_alloc_frame();
/* put sample parameters */
c->bit_rate = 400000;
@@ -374,114 +234,82 @@ static void video_encode_example(const char *filename, int codec_id)
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1,25};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
c->time_base= (AVRational){1,25};
c->gop_size = 10; /* emit one intra frame every ten frames */
c->max_b_frames=1;
c->pix_fmt = PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
if(codec_id == CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
/* alloc image and output buffer */
outbuf_size = 100000;
outbuf = malloc(outbuf_size);
/* the image can be allocated by any means and av_image_alloc() is
* just the most convenient way if av_malloc() is to be used */
ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
c->pix_fmt, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw picture buffer\n");
exit(1);
}
av_image_alloc(picture->data, picture->linesize,
c->width, c->height, c->pix_fmt, 1);
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for(i=0;i<25;i++) {
fflush(stdout);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
for(y=0;y<c->height;y++) {
for(x=0;x<c->width;x++) {
picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
for(y=0;y<c->height/2;y++) {
for(x=0;x<c->width/2;x++) {
picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
printf("encoding frame %3d (size=%5d)\n", i, out_size);
fwrite(outbuf, 1, out_size, f);
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
for(; out_size; i++) {
fflush(stdout);
ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
printf("write frame %3d (size=%5d)\n", i, out_size);
fwrite(outbuf, 1, out_size, f);
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
outbuf[0] = 0x00;
outbuf[1] = 0x00;
outbuf[2] = 0x01;
outbuf[3] = 0xb7;
fwrite(outbuf, 1, 4, f);
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
av_frame_free(&frame);
av_free(picture->data[0]);
av_free(picture);
printf("\n");
}
@@ -495,49 +323,22 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
f=fopen(filename,"w");
fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
for(i=0;i<ysize;i++)
fwrite(buf + i * wrap,1,xsize,f);
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
pkt->size -= len;
pkt->data += len;
}
return 0;
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_count;
int frame, got_picture, len;
FILE *f;
AVFrame *frame;
AVFrame *picture;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
char buf[1024];
AVPacket avpkt;
av_init_packet(&avpkt);
@@ -545,20 +346,17 @@ static void video_decode_example(const char *outfilename, const char *filename)
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Decode video file %s to %s\n", filename, outfilename);
printf("Video decoding\n");
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
picture= avcodec_alloc_frame();
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
@@ -568,25 +366,21 @@ static void video_decode_example(const char *outfilename, const char *filename)
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* the codec gives us the frame size, in samples */
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame_count = 0;
for (;;) {
frame = 0;
for(;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
@@ -607,9 +401,26 @@ static void video_decode_example(const char *outfilename, const char *filename)
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0)
if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
while (avpkt.size > 0) {
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", frame);
exit(1);
}
if (got_picture) {
printf("saving frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
avpkt.size -= len;
avpkt.data += len;
}
}
/* some codecs, such as MPEG, transmit the I and P frame with a
@@ -617,48 +428,50 @@ static void video_decode_example(const char *outfilename, const char *filename)
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (got_picture) {
printf("saving last frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&frame);
av_free(picture);
printf("\n");
}
int main(int argc, char **argv)
{
const char *output_type;
const char *filename;
/* must be called before using avcodec lib */
avcodec_init();
/* register all the codecs */
avcodec_register_all();
if (argc < 2) {
printf("usage: %s output_type\n"
"API example program to decode/encode a media stream with libavcodec.\n"
"This program generates a synthetic stream and encodes it to a file\n"
"named test.h264, test.mp2 or test.mpg depending on output_type.\n"
"The encoded stream is then decoded and written to a raw data output.\n"
"output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
argv[0]);
return 1;
}
output_type = argv[1];
if (argc <= 1) {
audio_encode_example("/tmp/test.mp2");
audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
if (!strcmp(output_type, "h264")) {
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.sw", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");
video_encode_example("/tmp/test.h264", CODEC_ID_H264);
video_encode_example("/tmp/test.mpg", CODEC_ID_MPEG1VIDEO);
filename = "/tmp/test.mpg";
} else {
fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
output_type);
return 1;
filename = argv[1];
}
// audio_decode_example("/tmp/test.sw", filename);
video_decode_example("/tmp/test%d.pgm", filename);
return 0;
}

View File

@@ -1,386 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* The different ways of decoding and managing data memory. You are not
* supposed to support all the modes in your application but pick the one most
* appropriate to your needs. Look for the use of api_mode in this example to
* see what are the differences of API usage between them */
enum {
API_MODE_OLD = 0, /* old method, deprecated */
API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */
API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
};
static int api_mode = API_MODE_OLD;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,
av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
/* If we use the new API with reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the decoders, with or without reference counting */
if (api_mode == API_MODE_NEW_API_REF_COUNT)
av_dict_set(&opts, "refcounted_frames", "1", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
"input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call. If unset, it's using\n"
"the classic old method.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5) {
const char *mode = argv[1] + strlen("-refcount=");
if (!strcmp(mode, "old")) api_mode = API_MODE_OLD;
else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT;
else {
fprintf(stderr, "unknow mode '%s'\n", mode);
exit(1);
}
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
ret = av_image_alloc(video_dst_data, video_dst_linesize,
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
/* When using the new API, you need to use the libavutil/frame.h API, while
* the classic frame management is available in libavcodec */
if (api_mode == API_MODE_OLD)
frame = avcodec_alloc_frame();
else
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_close(video_dec_ctx);
avcodec_close(audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
if (api_mode == API_MODE_OLD)
avcodec_free_frame(&frame);
else
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}

View File

@@ -1,185 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
* Copyright (c) 2014 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.
*/
#include <libavutil/motion_vector.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL;
static AVStream *video_stream = NULL;
static const char *src_filename = NULL;
static int video_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int decode_packet(int *got_frame, int cached)
{
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
int i;
AVFrameSideData *sd;
video_frame_count++;
sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
if (sd) {
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
}
}
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
exit(1);
}
src_filename = argv[1];
av_register_all();
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
}
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!video_stream) {
fprintf(stderr, "Could not find video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
end:
avcodec_close(video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;
}

View File

@@ -1,365 +0,0 @@
/*
* copyright (c) 2013 Andrew Kelley
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* libavfilter API usage example.
*
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
*
* abuffer: This provides the endpoint where you can feed the decoded samples.
* volume: In this example we hardcode it to 0.90.
* aformat: This converts the samples to the samplefreq, channel layout,
* and sample format required by the audio device.
* abuffersink: This provides the endpoint where you can read the samples after
* they have passed through the filter chain.
*/
#include <inttypes.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
AVFilterContext **sink)
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
AVFilter *abuffer;
AVFilterContext *volume_ctx;
AVFilter *volume;
AVFilterContext *aformat_ctx;
AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
uint8_t ch_layout[64];
int err;
/* Create a new filtergraph, which will contain all the filters. */
filter_graph = avfilter_graph_alloc();
if (!filter_graph) {
fprintf(stderr, "Unable to create filter graph.\n");
return AVERROR(ENOMEM);
}
/* Create the abuffer filter;
* it will be used for feeding the data into the graph. */
abuffer = avfilter_get_by_name("abuffer");
if (!abuffer) {
fprintf(stderr, "Could not find the abuffer filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
if (!abuffer_ctx) {
fprintf(stderr, "Could not allocate the abuffer instance.\n");
return AVERROR(ENOMEM);
}
/* Set the filter options through the AVOptions API. */
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
/* Now initialize the filter; we pass NULL options, since we have already
* set all the options above. */
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffer filter.\n");
return err;
}
/* Create volume filter. */
volume = avfilter_get_by_name("volume");
if (!volume) {
fprintf(stderr, "Could not find the volume filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
if (!volume_ctx) {
fprintf(stderr, "Could not allocate the volume instance.\n");
return AVERROR(ENOMEM);
}
/* A different way of passing the options is as key/value pairs in a
* dictionary. */
av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
err = avfilter_init_dict(volume_ctx, &options_dict);
av_dict_free(&options_dict);
if (err < 0) {
fprintf(stderr, "Could not initialize the volume filter.\n");
return err;
}
/* Create the aformat filter;
* it ensures that the output is of the format we want. */
aformat = avfilter_get_by_name("aformat");
if (!aformat) {
fprintf(stderr, "Could not find the aformat filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
if (!aformat_ctx) {
fprintf(stderr, "Could not allocate the aformat instance.\n");
return AVERROR(ENOMEM);
}
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t)AV_CH_LAYOUT_STEREO);
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
return err;
}
/* Finally create the abuffersink filter;
* it will be used to get the filtered data out of the graph. */
abuffersink = avfilter_get_by_name("abuffersink");
if (!abuffersink) {
fprintf(stderr, "Could not find the abuffersink filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
if (!abuffersink_ctx) {
fprintf(stderr, "Could not allocate the abuffersink instance.\n");
return AVERROR(ENOMEM);
}
/* This filter takes no options. */
err = avfilter_init_str(abuffersink_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffersink instance.\n");
return err;
}
/* Connect the filters;
* in this simple case the filters just form a linear chain. */
err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
if (err >= 0)
err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
if (err >= 0)
err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
if (err < 0) {
fprintf(stderr, "Error connecting filters\n");
return err;
}
/* Configure the graph. */
err = avfilter_graph_config(filter_graph, NULL);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
return err;
}
*graph = filter_graph;
*src = abuffer_ctx;
*sink = abuffersink_ctx;
return 0;
}
/* Do something useful with the filtered data: this simple
* example just prints the MD5 checksum of each plane to stdout. */
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
int i, j;
for (i = 0; i < planes; i++) {
uint8_t checksum[16];
av_md5_init(md5);
av_md5_sum(checksum, frame->extended_data[i], plane_size);
fprintf(stdout, "plane %d: 0x", i);
for (j = 0; j < sizeof(checksum); j++)
fprintf(stdout, "%02X", checksum[j]);
fprintf(stdout, "\n");
}
fprintf(stdout, "\n");
return 0;
}
/* Construct a frame of audio data to be filtered;
* this simple example just synthesizes a sine wave. */
static int get_input(AVFrame *frame, int frame_num)
{
int err, i, j;
#define FRAME_SIZE 1024
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
err = av_frame_get_buffer(frame, 0);
if (err < 0)
return err;
/* Fill the data for each channel. */
for (i = 0; i < 5; i++) {
float *data = (float*)frame->extended_data[i];
for (j = 0; j < frame->nb_samples; j++)
data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
}
return 0;
}
int main(int argc, char *argv[])
{
struct AVMD5 *md5;
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
if (argc < 2) {
fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
return 1;
}
duration = atof(argv[1]);
nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
if (nb_frames <= 0) {
fprintf(stderr, "Invalid duration: %s\n", argv[1]);
return 1;
}
avfilter_register_all();
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Error allocating the frame\n");
return 1;
}
md5 = av_md5_alloc();
if (!md5) {
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
fprintf(stderr, "Unable to init filter graph:");
goto fail;
}
/* the main filtering loop */
for (i = 0; i < nb_frames; i++) {
/* get an input frame to be filtered */
err = get_input(frame, i);
if (err < 0) {
fprintf(stderr, "Error generating input frame:");
goto fail;
}
/* Send the frame to the input of the filtergraph. */
err = av_buffersrc_add_frame(src, frame);
if (err < 0) {
av_frame_unref(frame);
fprintf(stderr, "Error submitting the frame to the filtergraph:");
goto fail;
}
/* Get all the filtered output that is available. */
while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
/* now do something with our filtered frame */
err = process_output(md5, frame);
if (err < 0) {
fprintf(stderr, "Error processing the filtered frame:");
goto fail;
}
av_frame_unref(frame);
}
if (err == AVERROR(EAGAIN)) {
/* Need to feed more frames in. */
continue;
} else if (err == AVERROR_EOF) {
/* Nothing more to do, finish. */
break;
} else if (err < 0) {
/* An error occurred. */
fprintf(stderr, "Error filtering the data:");
goto fail;
}
}
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
return 0;
fail:
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}

229
doc/examples/filtering.c Normal file
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@@ -0,0 +1,229 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/vsrc_buffer.h>
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret, i;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = av_find_stream_info(fmt_ctx)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
/* init the video decoder */
if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args), "%d:%d:%d:%d:%d:%d:%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
return ret;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, pix_fmts, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
}
static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (picref->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(picref->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = picref->pts;
}
/* Trivial ASCII grayscale display. */
p0 = picref->data[0];
puts("\033c");
for (y = 0; y < picref->video->h; y++) {
p = p0;
for (x = 0; x < picref->video->w; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += picref->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame frame;
int got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
AVFilterBufferRef *picref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
av_free_packet(&packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
if (got_frame) {
if (frame.pts == AV_NOPTS_VALUE)
frame.pts = frame.pkt_dts == AV_NOPTS_VALUE ?
frame.pkt_dts : frame.pkt_pts;
/* push the decoded frame into the filtergraph */
av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);
/* pull filtered pictures from the filtergraph */
while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
av_vsink_buffer_get_video_buffer_ref(buffersink_ctx, &picref, 0);
if (picref) {
display_picref(picref, buffersink_ctx->inputs[0]->time_base);
avfilter_unref_buffer(picref);
}
}
}
}
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
av_close_input_file(fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

View File

@@ -1,280 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* 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);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet0, packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
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 */
packet0.data = NULL;
packet.data = NULL;
while (1) {
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
packet0 = packet;
}
if (packet.stream_index == audio_stream_index) {
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
if (packet.size <= 0)
av_free_packet(&packet0);
} else {
/* discard non-wanted packets */
av_free_packet(&packet0);
}
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

View File

@@ -1,261 +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
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.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;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* 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 = 0;
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 AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
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");
goto end;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &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 */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

View File

@@ -23,7 +23,6 @@
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
@@ -51,6 +50,6 @@ int main (int argc, char **argv)
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
avformat_free_context(fmt_ctx);
return 0;
}

File diff suppressed because it is too large Load Diff

View File

@@ -1,165 +0,0 @@
/*
* Copyright (c) 2013 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
tag,
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
int main(int argc, char **argv)
{
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
if (argc < 3) {
printf("usage: %s input output\n"
"API example program to remux a media file with libavformat and libavcodec.\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return 1;
}
in_filename = argv[1];
out_filename = argv[2];
av_register_all();
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
fprintf(stderr, "Failed to retrieve input stream information");
goto end;
}
av_dump_format(ifmt_ctx, 0, in_filename, 0);
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
fprintf(stderr, "Could not create output context\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *in_stream = ifmt_ctx->streams[i];
AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
if (ret < 0) {
fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
goto end;
}
out_stream->codec->codec_tag = 0;
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open output file '%s'", out_filename);
goto end;
}
}
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file\n");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
if (ret < 0)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
pkt.pos = -1;
log_packet(ofmt_ctx, &pkt, "out");
ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
av_free_packet(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_close(ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

View File

@@ -1,214 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

View File

@@ -1,140 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}

View File

@@ -1,755 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 48000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
/**
* Convert an error code into a text message.
* @param error Error code to be converted
* @return Corresponding error text (not thread-safe)
*/
static const char *get_error_text(const int error)
{
static char error_buffer[255];
av_strerror(error, error_buffer, sizeof(error_buffer));
return error_buffer;
}
/** Open an input file and the required decoder. */
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodec *input_codec;
int error;
/** Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
filename, get_error_text(error));
*input_format_context = NULL;
return error;
}
/** Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
*input_codec_context = (*input_format_context)->streams[0]->codec;
return 0;
}
/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
/** Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
filename, get_error_text(error));
return error;
}
/** Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/** Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/** Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
/** Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
/** Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/** Save the encoder context for easiert access later. */
*output_codec_context = stream->codec;
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
(*output_codec_context)->channels = OUTPUT_CHANNELS;
(*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
(*output_codec_context)->sample_rate = input_codec_context->sample_rate;
(*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
(*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
/**
* Some container formats (like MP4) require global headers to be present
* Mark the encoder so that it behaves accordingly.
*/
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
(*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
return 0;
cleanup:
avio_close((*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/** Initialize one data packet for reading or writing. */
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/** Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/**
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/**
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/** Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/** Initialize a FIFO buffer for the audio samples to be encoded. */
static int init_fifo(AVAudioFifo **fifo)
{
/** Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/** Write the header of the output file container. */
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
fprintf(stderr, "Could not write output file header (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Decode one audio frame from the input file. */
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/** Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
/** Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are the the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
get_error_text(error));
return error;
}
}
/**
* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
if ((error = avcodec_decode_audio4(input_codec_context, frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
get_error_text(error));
av_free_packet(&input_packet);
return error;
}
/**
* If the decoder has not been flushed completely, we are not finished,
* so that this function has to be called again.
*/
if (*finished && *data_present)
*finished = 0;
av_free_packet(&input_packet);
return 0;
}
/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/**
* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
get_error_text(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is specified
* by frame_size.
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context)
{
int error;
/** Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Add converted input audio samples to the FIFO buffer for later processing. */
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/** Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decodes, converts and stores
* it in the FIFO buffer.
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished)
{
/** Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/** Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int ret = AVERROR_EXIT;
/** Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/** Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/**
* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error.
*/
if (*finished && !data_present) {
ret = 0;
goto cleanup;
}
/** If there is decoded data, convert and store it */
if (data_present) {
/** Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/** Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/** Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity.
*/
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
get_error_text(error));
av_frame_free(frame);
return error;
}
return 0;
}
/** Encode one frame worth of audio to the output file. */
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/** Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
get_error_text(error));
av_free_packet(&output_packet);
return error;
}
/** Write one audio frame from the temporary packet to the output file. */
if (*data_present) {
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
get_error_text(error));
av_free_packet(&output_packet);
return error;
}
av_free_packet(&output_packet);
}
return 0;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/** Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/**
* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size
*/
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/** Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/** Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/** Write the trailer of the output file container. */
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
fprintf(stderr, "Could not write output file trailer (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Convert an audio file to an AAC file in an MP4 container. */
int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/** Register all codecs and formats so that they can be used. */
av_register_all();
/** Open the input file for reading. */
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/** Open the output file for writing. */
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/** Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/** Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo))
goto cleanup;
/** Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/**
* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither.
*/
while (1) {
/** Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
while (av_audio_fifo_size(fifo) < output_frame_size) {
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/**
* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file.
*/
if (finished)
break;
}
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/**
* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish.
*/
if (finished) {
int data_written;
/** Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/** Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_close(output_codec_context);
if (output_format_context) {
avio_close(output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_close(input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

View File

@@ -1,601 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2014 Andrey Utkin
*
* 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 demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream;
AVCodecContext *codec_ctx;
stream = ifmt_ctx->streams[i];
codec_ctx = stream->codec;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* Open decoder */
ret = avcodec_open2(codec_ctx,
avcodec_find_decoder(codec_ctx->codec_id), NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = in_stream->codec;
enc_ctx = out_stream->codec;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
enc_ctx->pix_fmt = encoder->pix_fmts[0];
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = dec_ctx->time_base;
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
return ret;
}
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
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");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* 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 (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
ofmt_ctx->streams[i]->codec, filter_spec);
if (ret)
return ret;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codec->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
return ret;
if (!(*got_frame))
return 0;
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.duration = av_rescale_q(enc_pkt.duration,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
CODEC_CAP_DELAY))
return 0;
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
goto end;
if ((ret = init_filters()) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_free_packet(&packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_free_packet(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_close(ifmt_ctx->streams[i]->codec);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
avcodec_close(ofmt_ctx->streams[i]->codec);
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
}
av_free(filter_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_close(ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}

View File

@@ -79,17 +79,6 @@ not a bug they should fix:
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.
To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.
@chapter Usage
@section ffmpeg does not work; what is wrong?
@@ -105,21 +94,12 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
@example
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example
Notice that @samp{%d} is replaced by the image number.
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
Use the @option{-start_number} option to declare a starting number for
the sequence. This is useful if your sequence does not start with
@file{img001.jpg} but is still in a numerical order. The following
example will start with @file{img100.jpg}:
@example
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
@@ -128,7 +108,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@@ -137,23 +117,17 @@ If you want to sequence them by oldest modified first, substitute
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
You can also use @command{cat} to pipe images to ffmpeg:
@example
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example
@section How do I encode movie to single pictures?
Use:
@example
ffmpeg -i movie.mpg movie%d.jpg
ffmpeg -i movie.mpg movie%d.jpg
@end example
The @file{movie.mpg} used as input will be converted to
@@ -169,7 +143,7 @@ to force the encoding.
Applying that to the previous example:
@example
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,79 +201,20 @@ then you may use any file that DirectShow can read as input.
Just create an "input.avs" text file with this single line ...
@example
DirectShowSource("C:\path to your file\yourfile.asf")
DirectShowSource("C:\path to your file\yourfile.asf")
@end example
... and then feed that text file to ffmpeg:
@example
ffmpeg -i input.avs
ffmpeg -i input.avs
@end example
For ANY other help on AviSynth, please visit the
@uref{http://www.avisynth.org/, AviSynth homepage}.
For ANY other help on Avisynth, please visit the
@uref{http://www.avisynth.org/, Avisynth homepage}.
@section How can I join video files?
To "join" video files is quite ambiguous. The following list explains the
different kinds of "joining" and points out how those are addressed in
FFmpeg. To join video files may mean:
@itemize
@item
To put them one after the other: this is called to @emph{concatenate} them
(in short: concat) and is addressed
@ref{How can I concatenate video files, in this very faq}.
@item
To put them together in the same file, to let the user choose between the
different versions (example: different audio languages): this is called to
@emph{multiplex} them together (in short: mux), and is done by simply
invoking ffmpeg with several @option{-i} options.
@item
For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.
@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@anchor{How can I concatenate video files}
@section How can I concatenate video files?
There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
video by merely concatenating the files containing them.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by
merely concatenating them.
Hence you may concatenate your multimedia files by first transcoding them to
these privileged formats, then using the humble @code{cat} command (or the
@@ -307,38 +222,28 @@ equally humble @code{copy} under Windows), and finally transcoding back to your
format of choice.
@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i input1.avi -same_quant intermediate1.mpg
ffmpeg -i input2.avi -same_quant intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
ffmpeg -i intermediate_all.mpg -same_quant output.avi
@end example
Additionally, you can use the @code{concat} protocol instead of @code{cat} or
@code{copy} which will avoid creation of a potentially huge intermediate file.
Notice that you should either use @code{-same_quant} or set a reasonably high
bitrate for your intermediate and output files, if you want to preserve
video quality.
@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example
Note that you may need to escape the character "|" which is special for many
shells.
Another option is usage of named pipes, should your platform support it:
Also notice that you may avoid the huge intermediate files by taking advantage
of named pipes, should your platform support it:
@example
mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
ffmpeg -i input1.avi -same_quant -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
ffmpeg -f mpeg -i - -same_quant -c:v mpeg4 -acodec libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
@@ -346,8 +251,7 @@ from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
must use command grouping, @code{@{ ;@}}, to background properly.
For example, let's say we want to concatenate two FLV files into an
output.flv file:
For example, let's say we want to join two FLV files into an output.flv file:
@example
mkfifo temp1.a
@@ -364,70 +268,15 @@ cat temp1.a temp2.a > all.a &
cat temp1.v temp2.v > all.v &
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
-same_quant -y output.flv
rm temp[12].[av] all.[av]
@end example
@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 aresample}. Try to understand
why the converting filter was needed at that place.
Just before the output is a likely place, as @option{-f lavfi} currently
only support packed S16.
Then insert the correct @code{aformat} explicitly in the filtergraph,
specifying the exact format.
@example
aformat=sample_fmts=s16:channel_layouts=stereo
@end example
@section Why does FFmpeg not see the subtitles in my VOB file?
VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initially detected.
Some applications, including the @code{ffmpeg} command-line tool, can only
work with streams that were detected during the initial scan; streams that
are detected later are ignored.
The size of the initial scan is controlled by two options: @code{probesize}
(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
The @option{-sameq} option meant "same quantizer", and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
"same quality" and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.
Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
Yes. Check the @file{doc/examples} directory in the source
repository, also available online at:
@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
Examples are also installed by default, usually in
@code{$PREFIX/share/ffmpeg/examples}.
Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively,
examine the source code for one of the many open source projects that
already incorporate FFmpeg at (@url{projects.html}).
@@ -439,8 +288,31 @@ with @code{#ifdef}s related to the compiler.
@section Is Microsoft Visual C++ supported?
Yes. Please see the @uref{platform.html, Microsoft Visual C++}
section in the FFmpeg documentation.
No. Microsoft Visual C++ is not compliant to the C99 standard and does
not - among other things - support the inline assembly used in FFmpeg.
If you wish to use MSVC++ for your
project then you can link the MSVC++ code with libav* as long as
you compile the latter with a working C compiler. For more information, see
the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
documentation.
There have been efforts to make FFmpeg compatible with MSVC++ in the
past. However, they have all been rejected as too intrusive, especially
since MinGW does the job adequately. None of the core developers
work with MSVC++ and thus this item is low priority. Should you find
the silver bullet that solves this problem, feel free to shoot it at us.
We strongly recommend you to move over from MSVC++ to MinGW tools.
@section Can I use FFmpeg or libavcodec under Windows?
Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
Read the @emph{Windows} section in the FFmpeg documentation to find more
information.
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@section Can you add automake, libtool or autoconf support?
@@ -455,10 +327,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?
The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
contain full debug information. Those binaries are stripped to create
@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
the *_g versions.
The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
you need the debug information, use the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
@@ -466,24 +337,6 @@ Yes, as long as the code is optional and can easily and cleanly be placed
under #if CONFIG_GPL without breaking anything. So, for example, a new codec
or filter would be OK under GPL while a bug fix to LGPL code would not.
@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
FFmpeg builds static libraries by default. In static libraries, dependencies
are not handled. That has two consequences. First, you must specify the
libraries in dependency order: @code{-lavdevice} must come before
@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
Second, external libraries that are used in FFmpeg have to be specified too.
An easy way to get the full list of required libraries in dependency order
is to use @code{pkg-config}.
@example
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example
See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
more details.
@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
FFmpeg is a pure C project, so to use the libraries within your C++ application
@@ -499,8 +352,12 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
You have to create a custom AVIOContext using @code{avio_alloc_context},
see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
You have to implement a URLProtocol, see @file{libavformat/file.c} in
FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
@section Where can I find libav* headers for Pascal/Delphi?
see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
@@ -514,23 +371,10 @@ In this specific case please look at RFC 4629 to see how it should be done.
@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
r_frame_rate is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
@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.
For example, if you have mixed 25 and 30 fps content, then r_frame_rate
will be 150.
@bye

View File

@@ -1,205 +1,135 @@
\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Automated Testing Environment
@settitle FATE Automated Testing Environment
@titlepage
@center @titlefont{FFmpeg Automated Testing Environment}
@center @titlefont{FATE Automated Testing Environment}
@end titlepage
@node Top
@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.
FATE provides a regression testsuite embedded within the FFmpeg build system.
It can be run locally and optionally configured to send reports to a web
aggregator and viewer @url{http://fate.ffmpeg.org}.
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.
It is advised to run FATE before submitting patches to the current codebase
and provide new tests when submitting patches to add additional features.
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@chapter Running FATE
@url{http://fate.ffmpeg.org/}
@section Samples and References
In order to run, FATE needs a large amount of data (samples and references)
that is provided separately from the actual source distribution.
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
To inform the build system about the testsuite location, pass
@option{--samples=<path to the samples>} to @command{configure} or set the
@var{SAMPLES} Make variable or the @var{FATE_SAMPLES} environment variable
to a suitable value.
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:
The dataset is available through @command{rsync}, is possible to fetch
the current sample using the straight rsync command or through a specific
@ref{Makefile target}.
@example
make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
# rsync -aL rsync://fate.ffmpeg.org/fate-suite/ 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
# make fate-rsync SAMPLES=fate-suite
@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
To use a custom wrapper to run the test, pass @option{--target-exec} to
@command{configure} or set the @var{TARGET_EXEC} Make variable.
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
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{doc/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
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 pair
and send the public key to the FATE server administrator who can be contacted
at the email address @email{fate-admin@@ffmpeg.org}.
Configure your SSH client to use public key authentication with that key
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:
@chapter Manual Run
FATE regression test can be run through @command{make}.
Specific Makefile targets and Makefile variables are available:
@anchor{Makefile target}
@section FATE Makefile targets
@table @option
@item RSA
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
If you have problems connecting to the FATE server, it may help to try out
the @command{ssh} command with one or more @option{-v} options. You should
get detailed output concerning your SSH configuration and the authentication
process.
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.
List all fate/regression test targets.
@item fate-rsync
Shortcut to download the fate test samples to the specified testsuite location.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@section Fate 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
@table @option
@item 0
show just the test arguments
@item 1
show just the command used in the test
@item 2
show everything
@end table
@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 THREAD_TYPE
Specify which threading strategy test, either @var{slice} or @var{frame},
by default @var{slice+frame}
@item CPUFLAGS
Specify CPU flags.
@item TARGET_EXEC
Specify or override the wrapper used to run the tests.
The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
@item GEN
Set to @var{1} to generate the missing or mismatched references.
@end table
@section Examples
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
@end example
@chapter Automated Tests
In order to automatically testing specific configurations, e.g. multiple
compilers, @command{tests/fate.sh} is provided.
This shell script builds FFmpeg, runs the regression tests and prepares a
report that can be sent to @url{fate.ffmpeg.org} or directly examined locally.
@section Testing Profiles
The configuration file passed to @command{fate.sh} is shell scripts as well.
It must provide at least a @var{slot} identifier, the @var{repo} from
which fetch the sources, the @var{samples} directory, a @var{workdir} with
enough space to build and run all the tests.
Optional submit command @var{fate_recv} and a @var{comment} to describe
the testing profile are available.
Additional optional parameter to tune the FFmpeg building and reporting process
can be passed.
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
samples=/path/to/fate/samples
workdir= # directory in which to do all the work
fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
comment= # optional description
# the following are optional and map to configure options
arch=
cpu=
cross_prefix=
cc=
target_os=
sysroot=
target_exec=
target_path=
extra_cflags=
extra_ldflags=
extra_libs=
extra_conf= # extra configure options not covered above
#make= # name of GNU make if not 'make'
makeopts= # extra options passed to 'make'
#tar= # command to create a tar archive from its arguments on
# stdout, defaults to 'tar c'
@end example
@section Submitting Reports
In order to send reports you need to create an @command{ssh} key and send it
to the fate server administrator.
The current server fingerprint is @var{b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92}

139
doc/fate.txt Normal file
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@@ -0,0 +1,139 @@
FATE Automated Testing Environment
==================================
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:
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.
1. 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:
# make fate-rsync SAMPLES=fate-suite/
# make fate SAMPLES=fate-suite/
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:
# ./configure --samples=fate-suite/
# make fate-rsync
# make fate
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.
# FATE_SAMPLES=fate-suite/ make fate
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.
2. 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.
# tests/fate.sh /path/to/fate_config
A configuration file template with comments describing the individual
configuration variables can be found at tests/fate_config.sh.template .
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:
- configure.log
- compile.log
- test.log
- report
- version
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.
3. FATE makefile targets and variables
--------------------------------------
FATE Makefile targets:
fate-list
Will list all fate/regression test targets.
fate
Run the FATE test suite (requires the fate-suite dataset).
FATE Makefile variables:
V
Verbosity level, can be set to 0, 1 or 2.
* 0: show just the test arguments
* 1: show just the command used in the test
* 2: show everything
SAMPLES
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
Example:
make V=1 SAMPLES=/var/fate/samples THREADS=2 fate

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

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

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

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

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

4561
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@@ -1,42 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Protocols Documentation
@titlepage
@center @titlefont{FFmpeg Protocols Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the input and output protocols provided by the
libavformat library.
@c man end DESCRIPTION
@include protocols.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-protocols
@settitle FFmpeg protocols
@end ignore
@bye

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

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

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

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

View File

@@ -11,7 +11,11 @@
@chapter Synopsis
ffplay [@var{options}] [@file{input_file}]
@example
@c man begin SYNOPSIS
ffplay [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
@@ -24,7 +28,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@@ -73,25 +77,11 @@ Default value is "video", if video is not present or cannot be played
You can interactively cycle through the available show modes by
pressing the key @key{w}.
@item -vf @var{filtergraph}
Create the filtergraph specified by @var{filtergraph} and use it to
filter the video stream.
@var{filtergraph} is a description of the filtergraph to apply to
the stream, and must have a single video input and a single video
output. In the filtergraph, the input is associated to the label
@code{in}, and the output to the label @code{out}. See the
ffmpeg-filters manual for more information about the filtergraph
syntax.
You can specify this parameter multiple times and cycle through the specified
filtergraphs along with the show modes by pressing the key @key{w}.
@item -af @var{filtergraph}
@var{filtergraph} is a description of the filtergraph to apply to
the input audio.
@item -vf @var{filter_graph}
@var{filter_graph} is a description of the filter graph to apply to
the input video.
Use the option "-filters" to show all the available filters (including
sources and sinks).
also sources and sinks).
@item -i @var{input_file}
Read @var{input_file}.
@@ -102,13 +92,9 @@ Read @var{input_file}.
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
Show the stream duration, the codec parameters, the current position in
the stream and the audio/video synchronisation drift.
@item -bug
Work around bugs.
@item -fast
@@ -148,24 +134,8 @@ Exit when video is done playing.
Exit if any key is pressed.
@item -exitonmousedown
Exit if any mouse button is pressed.
@item -codec:@var{media_specifier} @var{codec_name}
Force a specific decoder implementation for the stream identified by
@var{media_specifier}, which can assume the values @code{a} (audio),
@code{v} (video), and @code{s} subtitle.
@item -acodec @var{codec_name}
Force a specific audio decoder.
@item -vcodec @var{codec_name}
Force a specific video decoder.
@item -scodec @var{codec_name}
Force a specific subtitle decoder.
@item -autorotate
Automatically rotate the video according to presentation metadata. Set by
default, use -noautorotate to disable.
@item -codec:@var{stream_type}
Force a specific decoder implementation
@end table
@section While playing
@@ -181,25 +151,16 @@ Toggle full screen.
Pause.
@item a
Cycle audio channel in the current program.
Cycle audio channel.
@item v
Cycle video channel.
@item t
Cycle subtitle channel in the current program.
@item c
Cycle program.
Cycle subtitle channel.
@item w
Cycle video filters or show modes.
@item s
Step to the next frame.
Pause if the stream is not already paused, step to the next video
frame, and pause.
Show audio waves.
@item left/right
Seek backward/forward 10 seconds.
@@ -208,8 +169,6 @@ Seek backward/forward 10 seconds.
Seek backward/forward 1 minute.
@item page down/page up
Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
@item mouse click
@@ -219,75 +178,28 @@ Seek to percentage in file corresponding to fraction of width.
@c man end
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include eval.texi
@include decoders.texi
@include demuxers.texi
@include muxers.texi
@include indevs.texi
@include outdevs.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffplay.html,ffplay},
@end ifset
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffplay(1),
@end ifset
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@c man begin SEEALSO
ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye

View File

@@ -11,7 +11,13 @@
@chapter Synopsis
ffprobe [@var{options}] [@file{input_file}]
The generic syntax is:
@example
@c man begin SYNOPSIS
ffprobe [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
@@ -39,20 +45,15 @@ ffprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of a form defined by the selected
writer, which is specified by the @option{print_format} option.
Sections may contain other nested sections, and are identified by a
name (which may be shared by other sections), and an unique
name. See the output of @option{sections}.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
section.
and printed in the corresponding "FORMAT" or "STREAM" section.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@@ -79,7 +80,7 @@ Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
Prettify the format of the displayed values, it corresponds to the
options "-unit -prefix -byte_binary_prefix -sexagesimal".
@item -of, -print_format @var{writer_name}[=@var{writer_options}]
@item -print_format @var{writer_name}[=@var{writer_options}]
Set the output printing format.
@var{writer_name} specifies the name of the writer, and
@@ -93,41 +94,6 @@ For example for printing the output in JSON format, specify:
For more details on the available output printing formats, see the
Writers section below.
@item -sections
Print sections structure and section information, and exit. The output
is not meant to be parsed by a machine.
@item -select_streams @var{stream_specifier}
Select only the streams specified by @var{stream_specifier}. This
option affects only the options related to streams
(e.g. @code{show_streams}, @code{show_packets}, etc.).
For example to show only audio streams, you can use the command:
@example
ffprobe -show_streams -select_streams a INPUT
@end example
To show only video packets belonging to the video stream with index 1:
@example
ffprobe -show_packets -select_streams v:1 INPUT
@end example
@item -show_data
Show payload data, as a hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.
The dump is printed as the "data" field. It may contain newlines.
@item -show_data_hash @var{algorithm}
Show a hash of payload data, for packets with @option{-show_packets} and for
codec extradata with @option{-show_streams}.
@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.
@@ -135,64 +101,6 @@ stream.
All the container format information is printed within a section with
name "FORMAT".
@item -show_format_entry @var{name}
Like @option{-show_format}, but only prints the specified entry of the
container format information, rather than all. This option may be given more
than once, then all specified entries will be shown.
This option is deprecated, use @code{show_entries} instead.
@item -show_entries @var{section_entries}
Set list of entries to show.
Entries are specified according to the following
syntax. @var{section_entries} contains a list of section entries
separated by @code{:}. Each section entry is composed by a section
name (or unique name), optionally followed by a list of entries local
to that section, separated by @code{,}.
If section name is specified but is followed by no @code{=}, all
entries are printed to output, together with all the contained
sections. Otherwise only the entries specified in the local section
entries list are printed. In particular, if @code{=} is specified but
the list of local entries is empty, then no entries will be shown for
that section.
Note that the order of specification of the local section entries is
not honored in the output, and the usual display order will be
retained.
The formal syntax is given by:
@example
@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
@end example
For example, to show only the index and type of each stream, and the PTS
time, duration time, and stream index of the packets, you can specify
the argument:
@example
packet=pts_time,duration_time,stream_index : stream=index,codec_type
@end example
To show all the entries in the section "format", but only the codec
type in the section "stream", specify the argument:
@example
format : stream=codec_type
@end example
To show all the tags in the stream and format sections:
@example
stream_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
sections:
@example
stream_tags=title
@end example
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
@@ -200,13 +108,6 @@ stream.
The information for each single packet is printed within a dedicated
section with name "PACKET".
@item -show_frames
Show information about each frame and subtitle contained in the input
multimedia stream.
The information for each single frame is printed within a dedicated
section with name "FRAME" or "SUBTITLE".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
@@ -214,117 +115,6 @@ multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -show_programs
Show information about programs and their streams contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "PROGRAM_STREAM".
@item -show_chapters
Show information about chapters stored in the format.
Each chapter is printed within a dedicated section with name "CHAPTER".
@item -count_frames
Count the number of frames per stream and report it in the
corresponding stream section.
@item -count_packets
Count the number of packets per stream and report it in the
corresponding stream section.
@item -read_intervals @var{read_intervals}
Read only the specified intervals. @var{read_intervals} must be a
sequence of interval specifications separated by ",".
@command{ffprobe} will seek to the interval starting point, and will
continue reading from that.
Each interval is specified by two optional parts, separated by "%".
The first part specifies the interval start position. It is
interpreted as an abolute position, or as a relative offset from the
current position if it is preceded by the "+" character. If this first
part is not specified, no seeking will be performed when reading this
interval.
The second part specifies the interval end position. It is interpreted
as an absolute position, or as a relative offset from the current
position if it is preceded by the "+" character. If the offset
specification starts with "#", it is interpreted as the number of
packets to read (not including the flushing packets) from the interval
start. If no second part is specified, the program will read until the
end of the input.
Note that seeking is not accurate, thus the actual interval start
point may be different from the specified position. Also, when an
interval duration is specified, the absolute end time will be computed
by adding the duration to the interval start point found by seeking
the file, rather than to the specified start value.
The formal syntax is given by:
@example
@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
@end example
A few examples follow.
@itemize
@item
Seek to time 10, read packets until 20 seconds after the found seek
point, then seek to position @code{01:30} (1 minute and thirty
seconds) and read packets until position @code{01:45}.
@example
10%+20,01:30%01:45
@end example
@item
Read only 42 packets after seeking to position @code{01:23}:
@example
01:23%+#42
@end example
@item
Read only the first 20 seconds from the start:
@example
%+20
@end example
@item
Read from the start until position @code{02:30}:
@example
%02:30
@end example
@end itemize
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
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 -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@item -i @var{input_file}
Read @var{input_file}.
@@ -334,45 +124,11 @@ Read @var{input_file}.
@chapter Writers
@c man begin WRITERS
A writer defines the output format adopted by @command{ffprobe}, and will be
A writer defines the output format adopted by @file{ffprobe}, and will be
used for printing all the parts of the output.
A writer may accept one or more arguments, which specify the options
to adopt. The options are specified as a list of @var{key}=@var{value}
pairs, separated by ":".
All writers support the following options:
@table @option
@item string_validation, sv
Set string validation mode.
The following values are accepted.
@table @samp
@item fail
The writer will fail immediately in case an invalid string (UTF-8)
sequence or code point is found in the input. This is especially
useful to validate input metadata.
@item ignore
Any validation error will be ignored. This will result in possibly
broken output, especially with the json or xml writer.
@item replace
The writer will substitute invalid UTF-8 sequences or code points with
the string specified with the @option{string_validation_replacement}.
@end table
Default value is @samp{replace}.
@item string_validation_replacement, svr
Set replacement string to use in case @option{string_validation} is
set to @samp{replace}.
In case the option is not specified, the writer will assume the empty
string, that is it will remove the invalid sequences from the input
strings.
@end table
A writer may accept one or more arguments, which specify the options to
adopt.
A description of the currently available writers follows.
@@ -388,27 +144,11 @@ keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
PROGRAM_STREAM section, and are prefixed by the string "TAG:".
Metadata tags are printed as a line in the corresponding FORMAT or
STREAM section, and are prefixed by the string "TAG:".
A description of the accepted options follows.
@table @option
@item nokey, nk
If set to 1 specify not to print the key of each field. Default value
is 0.
@item noprint_wrappers, nw
If set to 1 specify not to print the section header and footer.
Default value is 0.
@end table
@section compact, csv
Compact and CSV format.
The @code{csv} writer is equivalent to @code{compact}, but supports
different defaults.
@section compact
Compact format.
Each section is printed on a single line.
If no option is specifid, the output has the form:
@@ -420,29 +160,30 @@ Metadata tags are printed in the corresponding "format" or "stream"
section. A metadata tag key, if printed, is prefixed by the string
"tag:".
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@item item_sep, s
Specify the character to use for separating fields in the output line.
It must be a single printable character, it is "|" by default ("," for
the @code{csv} writer).
It must be a single printable character, it is "|" by default.
@item nokey, nk
If set to 1 specify not to print the key of each field. Its default
value is 0 (1 for the @code{csv} writer).
value is 0.
@item escape, e
Set the escape mode to use, default to "c" ("csv" for the @code{csv}
writer).
Set the escape mode to use, default to "c".
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline ('\n'), carriage
return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
Perform C-like escaping. Strings containing a newline ('\n') or
carriage return ('\r'), the escaping character ('\') or the item
separator character @var{SEP} are escaped using C-like fashioned
escaping, so that a newline is converted to the sequence "\n", a
carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
converted to "\@var{SEP}".
@@ -456,220 +197,41 @@ containing a newline ('\n'), a carriage return ('\r'), a double quote
Perform no escaping.
@end table
@item print_section, p
Print the section name at the begin of each line if the value is
@code{1}, disable it with value set to @code{0}. Default value is
@code{1}.
@end table
@section flat
Flat format.
@section csv
CSV format.
A free-form output where each line contains an explicit key=value, such as
"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be
directly embedded in sh scripts as long as the separator character is an
alphanumeric character or an underscore (see @var{sep_char} option).
The description of the accepted options follows.
@table @option
@item sep_char, s
Separator character used to separate the chapter, the section name, IDs and
potential tags in the printed field key.
Default value is '.'.
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
set to 1, and if there is more than one section in the current
chapter, the section name will be prefixed by the name of the
chapter. A value of 0 will disable this behavior.
Default value is 1.
@end table
@section ini
INI format output.
Print output in an INI based format.
The following conventions are adopted:
@itemize
@item
all key and values are UTF-8
@item
'.' is the subgroup separator
@item
newline, '\t', '\f', '\b' and the following characters are escaped
@item
'\' is the escape character
@item
'#' is the comment indicator
@item
'=' is the key/value separator
@item
':' is not used but usually parsed as key/value separator
@end itemize
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
set to 1, and if there is more than one section in the current
chapter, the section name will be prefixed by the name of the
chapter. A value of 0 will disable this behavior.
Default value is 1.
@end table
This writer is equivalent to
@code{compact=item_sep=,:nokey=1:escape=csv}.
@section json
JSON based format.
Each section is printed using JSON notation.
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.
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, GXF and AVI timecodes are available in format metadata
(@option{-show_format}, see @var{TAG:timecode}).
@end itemize
@c man end TIMECODE
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include decoders.texi
@include demuxers.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffprobe.html,ffprobe},
@end ifset
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffprobe(1),
@end ifset
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@include indevs.texi
@ignore
@setfilename ffprobe
@settitle ffprobe media prober
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye

View File

@@ -1,280 +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="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<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="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
</xsd: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:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</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="convergence_duration" type="xsd:long" />
<xsd:attribute name="convergence_duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<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="best_effort_timestamp" type="xsd:long" />
<xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
<xsd:attribute name="pkt_duration" type="xsd:long" />
<xsd:attribute name="pkt_duration_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
<xsd:attribute name="pkt_size" type="xsd:int" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<xsd:attribute name="channels" type="xsd:int" />
<xsd:attribute name="channel_layout" type="xsd:string"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<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:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="format" type="xsd:int" />
<xsd:attribute name="start_display_time" type="xsd:int" />
<xsd:attribute name="end_display_time" type="xsd:int" />
<xsd:attribute name="num_rects" 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="programsType">
<xsd:sequence>
<xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
<xsd:attribute name="original" type="xsd:int" use="required" />
<xsd:attribute name="comment" type="xsd:int" use="required" />
<xsd:attribute name="lyrics" type="xsd:int" use="required" />
<xsd:attribute name="karaoke" type="xsd:int" use="required" />
<xsd:attribute name="forced" type="xsd:int" use="required" />
<xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="codec_name" type="xsd:string" />
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="profile" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_time_base" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- 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="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<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="channel_layout" type="xsd:string"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration_ts" type="xsd:long"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="programType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="end_time" type="xsd:float"/>
<xsd:attribute name="end_pts" type="xsd:long"/>
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<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="nb_programs" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
<xsd:attribute name="probe_score" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="tagType">
<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_ident" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="chaptersType">
<xsd:sequence>
<xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="chapterType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="id" type="xsd:int" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="end" type="xsd:int" use="required"/>
<xsd:attribute name="end_time" type="xsd:float" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
<xsd:attribute name="ident" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

View File

@@ -1,11 +1,11 @@
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
HTTPPort 8090
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
HTTPBindAddress 0.0.0.0
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
@@ -25,6 +25,10 @@ MaxBandwidth 1000
# '-' is the standard output.
CustomLog -
# Suppress that if you want to launch ffserver as a daemon.
NoDaemon
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
@@ -235,7 +239,7 @@ StartSendOnKey
#<Stream test.ogg>
#Feed feed1.ffm
#Metadata title "Stream title"
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
@@ -280,10 +284,10 @@ StartSendOnKey
#<Stream file.asf>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Metadata author "Me"
#Metadata copyright "Super MegaCorp"
#Metadata title "Test stream from disk"
#Metadata comment "Test comment"
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</Stream>
@@ -369,3 +373,5 @@ ACL allow 192.168.0.0 192.168.255.255
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>

View File

@@ -9,159 +9,57 @@
@contents
@chapter Synopsis
@chapter Synopsys
ffserver [@var{options}]
The generic syntax is:
@example
@c man begin SYNOPSIS
ffserver [options]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
@command{ffserver} is a streaming server for both audio and video.
It supports several live feeds, streaming from files and time shifting
on live feeds. You can seek to positions in the past on each live
feed, provided you specify a big enough feed storage.
ffserver is a streaming server for both audio and video. It supports
@command{ffserver} is configured through a configuration file, which
is read at startup. If not explicitly specified, it will read from
@file{/etc/ffserver.conf}.
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).
@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
RTP/RTSP/HTTP.
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.
An @command{ffserver} instance will listen on some port as specified
in the configuration file. You can launch one or more instances of
@command{ffmpeg} and send one or more FFM streams to the port where
ffserver is expecting to receive them. Alternately, you can make
@command{ffserver} launch such @command{ffmpeg} instances at startup.
This documentation covers only the streaming aspects of ffserver /
ffmpeg. All questions about parameters for ffmpeg, codec questions,
etc. are not covered here. Read @file{ffmpeg.html} for more
information.
Input streams are called feeds, and each one is specified by a
@code{<Feed>} section in the configuration file.
@section How does it work?
ffserver receives prerecorded files or FFM streams from some ffmpeg
instance as input, then streams them over RTP/RTSP/HTTP.
An ffserver instance will listen on some port as specified in the
configuration file. You can launch one or more instances of ffmpeg and
send one or more FFM streams to the port where ffserver is expecting
to receive them. Alternately, you can make ffserver launch such ffmpeg
instances at startup.
Input streams are called feeds, and each one is specified by a <Feed>
section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a @code{<Stream>} section in the
configuration file.
@chapter Detailed description
@command{ffserver} works by forwarding streams encoded by
@command{ffmpeg}, or pre-recorded streams which are read from disk.
Precisely, @command{ffserver} acts as an HTTP server, accepting POST
requests from @command{ffmpeg} to acquire the stream to publish, and
serving RTSP clients or HTTP clients GET requests with the stream
media content.
A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
a port where @command{ffserver} is listening.
Each feed is identified by a unique name, corresponding to the name
of the resource published on @command{ffserver}, and is configured by
a dedicated @code{Feed} section in the configuration file.
The feed publish URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
@end example
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{HTTPPort} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@command{ffserver}, and made accessible through the HTTP protocol to
clients.
A stream can be connected to a feed, or to a file. In the first case,
the published stream is forwarded from the corresponding feed
generated by a running instance of @command{ffmpeg}, in the second
case the stream is read from a pre-recorded file.
Each stream is identified by a unique name, corresponding to the name
of the resource served by @command{ffserver}, and is configured by
a dedicated @code{Stream} section in the configuration file.
The stream access HTTP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
@end example
The stream access RTSP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
@end example
@var{stream_name} is the name of the corresponding stream defined in
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{HTTPPort} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
must be configured in the stream configuration. They are sent to
@command{ffmpeg} when setting up the encoding. This allows
@command{ffserver} to define the encoding parameters used by
the @command{ffmpeg} encoders.
The @command{ffmpeg} @option{override_ffserver} commandline option
allows one to override the encoding parameters set by the server.
Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@example
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
\ |_________|\ |__________|
\ \
\ \ __________
\ \ | |
\ \| stream 2 |
\ |__________|
\
\ _________ __________
\ | | | |
\| feed 2 |-----| stream 3 |
|_________| |__________|
_________ __________
| | | |
ffmpeg 2 -----| feed 3 |-----| stream 4 |
|_________| |__________|
_________ __________
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end example
@anchor{FFM}
@section FFM, FFM2 formats
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
formats, each one specified by a <Stream> section in the configuration
file.
@section Status stream
@command{ffserver} supports an HTTP interface which exposes the
current status of the server.
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
specified in the configuration file.
@@ -180,8 +78,35 @@ ACL allow 192.168.0.0 192.168.255.255
then the server will post a page with the status information when
the special stream @file{status.html} is requested.
@section What can this do?
When properly configured and running, you can capture video and audio in real
time from a suitable capture card, and stream it out over the Internet to
either Windows Media Player or RealAudio player (with some restrictions).
It can also stream from files, though that is currently broken. Very often, a
web server can be used to serve up the files just as well.
It can stream prerecorded video from .ffm files, though it is somewhat tricky
to make it work correctly.
@section What do I need?
I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm
using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
I needed some special drivers for my motherboard-based sound card.]
I understand that FreeBSD systems work just fine as well.
@section How do I make it work?
First, build the kit. It *really* helps to have installed LAME first. Then when
you run the ffserver ./configure, make sure that you have the
@code{--enable-libmp3lame} flag turned on.
LAME is important as it allows for streaming audio to Windows Media Player.
Don't ask why the other audio types do not work.
As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@@ -203,9 +128,40 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
You should edit the @file{ffserver.conf} file to suit your needs (in
terms of frame rates etc). Then install @command{ffserver} and
@command{ffmpeg}, write a script to start them up, and off you go.
@section What happens next?
You should edit the ffserver.conf file to suit your needs (in terms of
frame rates etc). Then install ffserver and ffmpeg, write a script to start
them up, and off you go.
@section Troubleshooting
@subsection I don't hear any audio, but video is fine.
Maybe you didn't install LAME, or got your ./configure statement wrong. Check
the ffmpeg output to see if a line referring to MP3 is present. If not, then
your configuration was incorrect. If it is, then maybe your wiring is not
set up correctly. Maybe the sound card is not getting data from the right
input source. Maybe you have a really awful audio interface (like I do)
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 loose sync after a while.
Yes, they do.
@subsection After a long while, the video update rate goes way down in WMP.
Yes, it does. Who knows why?
@subsection WMP 6.4 behaves differently to WMP 7.
Yes, it does. Any thoughts on this would be gratefully received. These
differences extend to embedding WMP into a web page. [There are two
object IDs that you can use: The old one, which does not play well, and
the new one, which does (both tested on the same system). However,
I suspect that the new one is not available unless you have installed WMP 7].
@section What else can it do?
@@ -246,6 +202,9 @@ specify a time. In addition, ffserver will skip frames until a key_frame
is found. This further reduces the startup delay by not transferring data
that will be discarded.
* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
the amount of bandwidth consumed by live streams.
@section Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully
@@ -282,626 +241,39 @@ For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@table @option
@item -f @var{configfile}
Read configuration file @file{configfile}. If not specified it will
read by default from @file{/etc/ffserver.conf}.
Use @file{configfile} instead of @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the @code{Launch}
directives within the various @code{<Feed>} sections. Since
@command{ffserver} will not launch any @command{ffmpeg} instances, you
will have to launch them manually.
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.
@item -d
Enable debug mode. This option increases log verbosity, and directs
log messages to stdout. When specified, the @option{CustomLog} option
is ignored.
Enable debug mode. This option increases log verbosity, directs log
messages to stdout and causes ffserver to run in the foreground
rather than as a daemon.
@end table
@chapter Configuration file syntax
@command{ffserver} reads a configuration file containing global
options and settings for each stream and feed.
The configuration file consists of global options and dedicated
sections, which must be introduced by "<@var{SECTION_NAME}
@var{ARGS}>" on a separate line and must be terminated by a line in
the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
Currently the following sections are recognized: @samp{Feed},
@samp{Stream}, @samp{Redirect}.
A line starting with @code{#} is ignored and treated as a comment.
Name of options and sections are case-insensitive.
@section ACL syntax
An ACL (Access Control List) specifies the address which are allowed
to access a given stream, or to write a given feed.
It accepts the folling forms
@itemize
@item
Allow/deny access to @var{address}.
@example
ACL ALLOW <address>
ACL DENY <address>
@end example
@item
Allow/deny access to ranges of addresses from @var{first_address} to
@var{last_address}.
@example
ACL ALLOW <first_address> <last_address>
ACL DENY <first_address> <last_address>
@end example
@end itemize
You can repeat the ACL allow/deny as often as you like. It is on a per
stream basis. The first match defines the action. If there are no matches,
then the default is the inverse of the last ACL statement.
Thus 'ACL allow localhost' only allows access from localhost.
'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
allow everybody else.
@section Global options
@table @option
@item HTTPPort @var{port_number}
@item Port @var{port_number}
@item RTSPPort @var{port_number}
@var{HTTPPort} sets the HTTP server listening TCP port number,
@var{RTSPPort} sets the RTSP server listening TCP port number.
@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
You must select a different port from your standard HTTP web server if
it is running on the same computer.
If not specified, no corresponding server will be created.
@item HTTPBindAddress @var{ip_address}
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
deprecated.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
since it defines the @option{MaxClients} maximum limit.
Default value is 2000.
@item MaxClients @var{n}
Set number of simultaneous requests that can be handled. Since
@command{ffserver} is very fast, it is more likely that you will want
to leave this high and use @option{MaxBandwidth}.
Default value is 5.
@item MaxBandwidth @var{kbps}
Set the maximum amount of kbit/sec that you are prepared to consume
when streaming to clients.
Default value is 1000.
@item CustomLog @var{filename}
Set access log file (uses standard Apache log file format). '-' is the
standard output.
If not specified @command{ffserver} will produce no log.
In case the commandline option @option{-d} is specified this option is
ignored, and the log is written to standard output.
@item NoDaemon
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@end table
@section Feed section
A Feed section defines a feed provided to @command{ffserver}.
Each live feed contains one video and/or audio sequence coming from an
@command{ffmpeg} encoder or another @command{ffserver}. This sequence
may be encoded simultaneously with several codecs at several
resolutions.
A feed instance specification is introduced by a line in the form:
@example
<Feed FEED_FILENAME>
@end example
where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
The following options are recognized within a Feed section.
@table @option
@item File @var{filename}
@item ReadOnlyFile @var{filename}
Set the path where the feed file is stored on disk.
If not specified, the @file{/tmp/FEED.ffm} is assumed, where
@var{FEED} is the feed name.
If @option{ReadOnlyFile} is used the file is marked as read-only and
it will not be deleted or updated.
@item Truncate
Truncate the feed file, rather than appending to it. By default
@command{ffserver} will append data to the file, until the maximum
file size value is reached (see @option{FileMaxSize} option).
@item FileMaxSize @var{size}
Set maximum size of the feed file in bytes. 0 means unlimited. The
postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
recognized.
Default value is 5M.
@item Launch @var{args}
Launch an @command{ffmpeg} command when creating @command{ffserver}.
@var{args} must be a sequence of arguments to be provided to an
@command{ffmpeg} instance. The first provided argument is ignored, and
it is replaced by a path with the same dirname of the @command{ffserver}
instance, followed by the remaining argument and terminated with a
path corresponding to the feed.
When the launched process exits, @command{ffserver} will launch
another program instance.
In case you need a more complex @command{ffmpeg} configuration,
e.g. if you need to generate multiple FFM feeds with a single
@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
This option is ignored in case the commandline option @option{-n} is
specified.
@item ACL @var{spec}
Specify the list of IP address which are allowed or denied to write
the feed. Multiple ACL options can be specified.
@end table
@section Stream section
A Stream section defines a stream provided by @command{ffserver}, and
identified by a single name.
The stream is sent when answering a request containing the stream
name.
A stream section must be introduced by the line:
@example
<Stream STREAM_NAME>
@end example
where @var{STREAM_NAME} specifies the unique name of the stream.
The following options are recognized within a Stream section.
Encoding options are marked with the @emph{encoding} tag, and they are
used to set the encoding parameters, and are mapped to libavcodec
encoding options. Not all encoding options are supported, in
particular it is not possible to set encoder private options. In order
to override the encoding options specified by @command{ffserver}, you
can use the @command{ffmpeg} @option{override_ffserver} commandline
option.
Only one of the @option{Feed} and @option{File} options should be set.
@table @option
@item Feed @var{feed_name}
Set the input feed. @var{feed_name} must correspond to an existing
feed defined in a @code{Feed} section.
When this option is set, encoding options are used to setup the
encoding operated by the remote @command{ffmpeg} process.
@item File @var{filename}
Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input
file content is re-streamed as is.
@item Format @var{format_name}
Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to
@samp{status}, it is treated as a status stream.
@item InputFormat @var{format_name}
Set input format. If not specified, it is automatically guessed.
@item Preroll @var{n}
Set this to the number of seconds backwards in time to start. Note that
most players will buffer 5-10 seconds of video, and also you need to allow
for a keyframe to appear in the data stream.
Default value is 0.
@item StartSendOnKey
Do not send stream until it gets the first key frame. By default
@command{ffserver} will send data immediately.
@item MaxTime @var{n}
Set the number of seconds to run. This value set the maximum duration
of the stream a client will be able to receive.
A value of 0 means that no limit is set on the stream duration.
@item ACL @var{spec}
Set ACL for the stream.
@item DynamicACL @var{spec}
@item RTSPOption @var{option}
@item MulticastAddress @var{address}
@item MulticastPort @var{port}
@item MulticastTTL @var{integer}
@item NoLoop
@item FaviconURL @var{url}
Set favicon (favourite icon) for the server status page. It is ignored
for regular streams.
@item Author @var{value}
@item Comment @var{value}
@item Copyright @var{value}
@item Title @var{value}
Set metadata corresponding to the option. All these options are
deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item NoAudio
@item NoVideo
Suppress audio/video.
@item AudioCodec @var{codec_name} (@emph{encoding,audio})
Set audio codec.
@item AudioBitRate @var{rate} (@emph{encoding,audio})
Set bitrate for the audio stream in kbits per second.
@item AudioChannels @var{n} (@emph{encoding,audio})
Set number of audio channels.
@item AudioSampleRate @var{n} (@emph{encoding,audio})
Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
Set generic option for audio stream.
@item AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@item VideoCodec @var{codec_name} (@emph{encoding,video})
Set video codec.
@item VideoBitRate @var{n} (@emph{encoding,video})
Set bitrate for the video stream in kbits per second.
@item VideoBitRateRange @var{range} (@emph{encoding,video})
Set video bitrate range.
A range must be specified in the form @var{minrate}-@var{maxrate}, and
specifies the @option{minrate} and @option{maxrate} encoding options
expressed in kbits per second.
@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
Set video bitrate tolerance in kbits per second.
@item PixelFormat @var{pixel_format} (@emph{encoding,video})
Set video pixel format.
@item Debug @var{integer} (@emph{encoding,video})
Set video @option{debug} encoding option.
@item Strict @var{integer} (@emph{encoding,video})
Set video @option{strict} encoding option.
@item VideoBufferSize @var{n} (@emph{encoding,video})
Set ratecontrol buffer size, expressed in KB.
@item VideoFrameRate @var{n} (@emph{encoding,video})
Set number of video frames per second.
@item VideoSize (@emph{encoding,video})
Set size of the video frame, must be an abbreviation or in the form
@var{W}x@var{H}. See @ref{video size syntax,,the Video size section
in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Default value is @code{160x128}.
@item VideoIntraOnly (@emph{encoding,video})
Transmit only intra frames (useful for low bitrates, but kills frame rate).
@item VideoGopSize @var{n} (@emph{encoding,video})
If non-intra only, an intra frame is transmitted every VideoGopSize
frames. Video synchronization can only begin at an intra frame.
@item VideoTag @var{tag} (@emph{encoding,video})
Set video tag.
@item VideoHighQuality (@emph{encoding,video})
@item Video4MotionVector (@emph{encoding,video})
@item BitExact (@emph{encoding,video})
Set bitexact encoding flag.
@item IdctSimple (@emph{encoding,video})
Set simple IDCT algorithm.
@item Qscale @var{n} (@emph{encoding,video})
Enable constant quality encoding, and set video qscale (quantization
scale) value, expressed in @var{n} QP units.
@item VideoQMin @var{n} (@emph{encoding,video})
@item VideoQMax @var{n} (@emph{encoding,video})
Set video qmin/qmax.
@item VideoQDiff @var{integer} (@emph{encoding,video})
Set video @option{qdiff} encoding option.
@item LumiMask @var{float} (@emph{encoding,video})
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
Set generic option for video stream.
@item AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.
@var{preset} must be the path of a preset file.
@end table
@subsection Server status stream
A server status stream is a special stream which is used to show
statistics about the @command{ffserver} operations.
It must be specified setting the option @option{Format} to
@samp{status}.
@section Redirect section
A redirect section specifies where to redirect the requested URL to
another page.
A redirect section must be introduced by the line:
@example
<Redirect NAME>
@end example
where @var{NAME} is the name of the page which should be redirected.
It only accepts the option @option{URL}, which specify the redirection
URL.
@chapter Stream examples
@itemize
@item
Multipart JPEG
@example
<Stream test.mjpg>
Feed feed1.ffm
Format mpjpeg
VideoFrameRate 2
VideoIntraOnly
NoAudio
Strict -1
</Stream>
@end example
@item
Single JPEG
@example
<Stream test.jpg>
Feed feed1.ffm
Format jpeg
VideoFrameRate 2
VideoIntraOnly
VideoSize 352x240
NoAudio
Strict -1
</Stream>
@end example
@item
Flash
@example
<Stream test.swf>
Feed feed1.ffm
Format swf
VideoFrameRate 2
VideoIntraOnly
NoAudio
</Stream>
@end example
@item
ASF compatible
@example
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
@end example
@item
MP3 audio
@example
<Stream test.mp3>
Feed feed1.ffm
Format mp2
AudioCodec mp3
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Ogg Vorbis audio
@example
<Stream test.ogg>
Feed feed1.ffm
Metadata title "Stream title"
AudioBitRate 64
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Real with audio only at 32 kbits
@example
<Stream test.ra>
Feed feed1.ffm
Format rm
AudioBitRate 32
NoVideo
</Stream>
@end example
@item
Real with audio and video at 64 kbits
@example
<Stream test.rm>
Feed feed1.ffm
Format rm
AudioBitRate 32
VideoBitRate 128
VideoFrameRate 25
VideoGopSize 25
</Stream>
@end example
@item
For stream coming from a file: you only need to set the input filename
and optionally a new format.
@example
<Stream file.rm>
File "/usr/local/httpd/htdocs/tlive.rm"
NoAudio
</Stream>
@end example
@example
<Stream file.asf>
File "/usr/local/httpd/htdocs/test.asf"
NoAudio
Metadata author "Me"
Metadata copyright "Super MegaCorp"
Metadata title "Test stream from disk"
Metadata comment "Test comment"
</Stream>
@end example
@end itemize
@c man end
@include config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffserver.html,ffserver},
@end ifset
@ifset config-not-all
@url{ffserver-all.html,ffserver-all},
@end ifset
the @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffserver(1),
@end ifset
@ifset config-not-all
ffserver-all(1),
@end ifset
the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffserver
@settitle ffserver video server
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
example and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye

View File

@@ -1,323 +0,0 @@
All the numerical options, if not specified otherwise, accept a string
representing a number as input, which may be followed by one of the SI
unit prefixes, for example: 'K', 'M', or 'G'.
If 'i' is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
prefix multiplies the value by 8. This allows using, for example:
'KB', 'MiB', 'G' and 'B' as number suffixes.
Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
the option name with "no". For example using "-nofoo"
will set the boolean option with name "foo" to false.
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) a given option belongs to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
would select the ac3 codec for the second audio stream.
A stream specifier can match several streams, so that the option is applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams. For example, @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
program.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
Note that in @command{ffmpeg}, matching by metadata will only work properly for
input files.
@end table
@section Generic options
These options are shared amongst the ff* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help [@var{arg}]
Show help. An optional parameter may be specified to print help about a specific
item. If no argument is specified, only basic (non advanced) tool
options are shown.
Possible values of @var{arg} are:
@table @option
@item long
Print advanced tool options in addition to the basic tool options.
@item full
Print complete list of options, including shared and private options
for encoders, decoders, demuxers, muxers, filters, etc.
@item decoder=@var{decoder_name}
Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@item encoder=@var{encoder_name}
Print detailed information about the encoder named @var{encoder_name}. Use the
@option{-encoders} option to get a list of all encoders.
@item demuxer=@var{demuxer_name}
Print detailed information about the demuxer named @var{demuxer_name}. Use the
@option{-formats} option to get a list of all demuxers and muxers.
@item muxer=@var{muxer_name}
Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.
@item filter=@var{filter_name}
Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
@end table
@item -version
Show version.
@item -formats
Show available formats.
@item -codecs
Show all codecs known to libavcodec.
Note that the term 'codec' is used throughout this documentation as a shortcut
for what is more correctly called a media bitstream format.
@item -decoders
Show available decoders.
@item -encoders
Show all available encoders.
@item -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -layouts
Show channel names and standard channel layouts.
@item -colors
Show recognized color names.
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
to the first line and the "Last message repeated n times" line will be
omitted. "repeat" can also be used alone.
If "repeat" is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
'repeat' will not change the loglevel.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
Show nothing at all; be silent.
@item panic
Only show fatal errors which could lead the process to crash, such as
and assert failure. This is not currently used for anything.
@item fatal
Only show fatal errors. These are errors after which the process absolutely
cannot continue after.
@item error
Show all errors, including ones which can be recovered from.
@item warning
Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
@item info
Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
@item verbose
Same as @code{info}, except more verbose.
@item debug
Show everything, including debugging information.
@end table
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Setting the environment variable @code{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
options will affect the report; options values must be escaped if they
contain special characters or the options delimiter ':' (see the
``Quoting and escaping'' section in the ffmpeg-utils manual). The
following option is recognized:
@table @option
@item file
set the file name to use for the report; @code{%p} is expanded to the name
of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
to a plain @code{%}
@item level
set the log level
@end table
Errors in parsing the environment variable are not fatal, and will not
appear in the report.
@item -hide_banner
Suppress printing banner.
All FFmpeg tools will normally show a copyright notice, build options
and library versions. This option can be used to suppress printing
this information.
@item -cpuflags flags (@emph{global})
Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you're doing.
@example
ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
@end example
Possible flags for this option are:
@table @samp
@item x86
@table @samp
@item mmx
@item mmxext
@item sse
@item sse2
@item sse2slow
@item sse3
@item sse3slow
@item ssse3
@item atom
@item sse4.1
@item sse4.2
@item avx
@item avx2
@item xop
@item fma3
@item fma4
@item 3dnow
@item 3dnowext
@item bmi1
@item bmi2
@item cmov
@end table
@item ARM
@table @samp
@item armv5te
@item armv6
@item armv6t2
@item vfp
@item vfpv3
@item neon
@item setend
@end table
@item AArch64
@table @samp
@item armv8
@item vfp
@item neon
@end table
@item PowerPC
@table @samp
@item altivec
@end table
@item Specific Processors
@table @samp
@item pentium2
@item pentium3
@item pentium4
@item k6
@item k62
@item athlon
@item athlonxp
@item k8
@end table
@end table
@item -opencl_bench
Benchmark all available OpenCL devices and show the results. This option
is only available when FFmpeg has been compiled with @code{--enable-opencl}.
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.
@var{options} must be a list of @var{key}=@var{value} option pairs
separated by ':'. See the ``OpenCL Options'' section in the
ffmpeg-utils manual for the list of supported options.
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them.
Note: the @option{-nooption} syntax cannot be used for boolean
AVOptions, use @option{-option 0}/@option{-option 1}.
Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.

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