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Author SHA1 Message Date
Diego Biurrun
4466d0e84b truemotion1: remove disabled code 2012-05-20 15:04:43 +02:00
Kostya Shishkov
f05b69a79f fix typo in comment 2012-05-20 14:15:14 +02:00
Mans Rullgard
4852cb37ce fate: fix dependencies for non-SAMPLES avconv tests 
The encode/decode tests should all depend on avconv.  Since
avconv requires libavfilter, there is no need to enable those
tests selectively.

Signed-off-by: Mans Rullgard <mans@mansr.com>
 2012-05-20 12:54:31 +01:00
Nicolas George
ea07063fd8 cmdutils: allow to specify a stream by its ID. 
The stream can be specified as "#129" or "#0x81".
It is especially useful for VOBs dumped from a DVD,
where the language-id mapping is available externally
and the probing can find the streams in a random order.
 2012-05-20 12:47:27 +02:00
Michael Niedermayer
bf3dce6b59 ffmpeg: change AVFMT_RAWPICTURE pts to reduce diff to qatar 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 12:29:33 +02:00
Michael Niedermayer
13c254a216 ffmpeg: random cosmectics to reduce diff to qatar 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 12:24:49 +02:00
Michael Niedermayer
9ce3d03fc2 ffmpeg: cosmetics to reduce difference to qatar by 101 lines 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 12:12:13 +02:00
Michael Niedermayer
3ba90d9e97 ffmpeg: cosmetics to reduce difference to qatar by about 90 lines 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 12:12:13 +02:00
Michael Niedermayer
c53326afad ffmpeg: move ost->frame_rate setting code before the filtergraph stuff 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 11:55:25 +02:00
Michael Niedermayer
953df5d962 ffmpeg: drop pkt_pts 
Reduces diff to qatar by 9 lines
Should not change anything

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 11:40:05 +02:00
Michael Niedermayer
ef7ac8e8ec ffmpeg: set the video filters timebase to the demuxers. 
Inspired-by a change from Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 11:40:05 +02:00
Michael Niedermayer
3cf5591344 ffmpeg: reintroducing check_recording_time() 
This reduces the difference to qatar by 27 lines
No functional change, the code should be unused

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 11:35:08 +02:00
Michael Niedermayer
8364cb9719 ffmpeg: fix uninitialized variable warning 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 11:35:08 +02:00
Kostya Shishkov
cf61aaaca1 indeo: check for invalid motion vectors 2012-05-20 11:19:48 +02:00
Kostya Shishkov
9603738221 indeo: check that band output buffer exists 2012-05-20 11:19:48 +02:00
Kostya Shishkov
23ba1503f2 indeo: clear allocated band buffers 2012-05-20 11:19:48 +02:00
Kostya Shishkov
a6e4ac40a6 indeo: track tile macroblock size 2012-05-20 11:19:48 +02:00
Kostya Shishkov
fe7a37c36f indeo: check custom Huffman tables for errors 2012-05-20 11:19:47 +02:00
Kostya Shishkov
aa372cf470 factor out common decoding code for Indeo 4 and Indeo 5 2012-05-20 11:19:30 +02:00
Kostya Shishkov
b37d945dd4 mp3: fix start band index for block type 2 in 8kHz audio 
In hybrid frames long window part ends at 36 samples for most of the cases
but at 72 for 8kHz case. For some reason decoder assumed it's 48 or even 36
samples, which caused wrong bitstream decoding for such blocks.

l3_25207.mpg from conformance suite demonstrates it the best.
 2012-05-20 10:53:21 +02:00
Michael Niedermayer
31f3c94760 Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  fate: fix dependencies on SAMPLES being set
  fate: remove unnecessary libavfilter dependency

Conflicts:
	ffmpeg.c
	tests/Makefile
	tests/fate/aac.mak
	tests/fate/audio.mak
	tests/fate/demux.mak
	tests/fate/ea.mak
	tests/fate/image.mak
	tests/fate/lossless-audio.mak
	tests/fate/lossless-video.mak
	tests/fate/microsoft.mak
	tests/fate/mpc.mak
	tests/fate/probe.mak
	tests/fate/qt.mak
	tests/fate/real.mak
	tests/fate/screen.mak
	tests/fate/video.mak
	tests/fate/voice.mak
	tests/fate/vqf.mak
	tests/ref/fate/idroq-video-encode
	tests/ref/fate/vc1-ism

Merged-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-20 10:21:52 +02:00
Michael Niedermayer
d1384c00d4 swr: skip soft compensation when its disabled. 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 22:42:32 +02:00
Michael Niedermayer
c5371f775c mpegaudidec: 10l forgot buf_size 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 22:22:45 +02:00
Michael Niedermayer
43bc5cf9f4 mpegaudiodec: skip initial zeros. 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 22:15:21 +02:00
Mans Rullgard
68aef0b481 lavf: change some (de)muxer names to lowercase 
This is consistent with other format names.

Signed-off-by: Mans Rullgard <mans@mansr.com>
 2012-05-19 19:44:16 +01:00
Mans Rullgard
81ad97eeda lavf: make output format matching case insensitive 
This is consistent with how input formats are matched.

Signed-off-by: Mans Rullgard <mans@mansr.com>
 2012-05-19 19:44:15 +01:00
Michael Niedermayer
040a796dab Merge commit '755cd4197d53946208e042f095b930dca18d9430' 
* commit '755cd4197d53946208e042f095b930dca18d9430':
  mov: enable parsing for VC-1.
  lavfi: Add fps filter.
  lavfi: initialize pts to AV_NOPTS_VALUE when creating new buffer refs.
  avconv: add support for audio in complex filtergraphs.

Conflicts:
	ffmpeg.c
	libavfilter/version.h
	libavformat/mov.c
	tests/ref/fate/vc1-ism

Merged-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 20:31:24 +02:00
Carl Eugen Hoyos
612abe2773 Use a variable instead of a repeated calculation when decoding AVUI. 2012-05-19 20:07:28 +02:00
Carl Eugen Hoyos
f1892348c2 avuidec.c: Fix indentation. 2012-05-19 20:07:28 +02:00
Carl Eugen Hoyos
d9b2410f75 Simplify AVUI decoding, one skip value is enough. 2012-05-19 20:07:28 +02:00
Carl Eugen Hoyos
2596d7261d Fix interlaced AVUI decoding. 2012-05-19 20:07:28 +02:00
Michael Niedermayer
9d30a96aa1 ffmpeg: inject a aresample filter for -async 
Based on similar code from anton

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:38 +02:00
Michael Niedermayer
21d8a80e30 af_aresample: use new swr API to pass and compensate PTS 
This code is not only much more powerfull its also simpler

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:38 +02:00
Michael Niedermayer
087d09b6d5 af_aresample: allocate a larger buffer so accumulated data can be returned quicker 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:38 +02:00
Michael Niedermayer
e17a039647 af_aresample: use extended_data 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:38 +02:00
Michael Niedermayer
72a242c998 swr: add swr_next_pts() 
parameter descriptions partly reuse text from af_asyncts

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
f88f705abc swr: add swr_drop_output() 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
7a59964ba9 swr: add swr_inject_silence() 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
72872d0201 swr: Fix issue with injecting more data than can be used up in the next call. 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
c5c355546d swr: add some asserts to resample() 
These check that the internal and input buffers match in format

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
75e25f8540 swr: add reversefill_audiodata() 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
5027e3daf7 swr: fix buf_set() so it works with input == output 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
bdbfd4569d swr: make literal 32 a named constant in realloc_audio() 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Michael Niedermayer
186be49fd5 swr: fill_audio_data() support NULL 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-05-19 19:23:37 +02:00
Nicolas George
3d2515a8f3 f_split: fix output type. 
The previous version checked the filter context name,
instead of checking the filter name.
The new version just uses the same type as the input.

Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
 2012-05-19 19:23:37 +02:00
Nicolas George
5ac9ef6493 avfilter: make AVFilterFormats compatible with libav. 
The list goes back to being simple integers,
and avfilter_make_format64_list is fixed to work
with the correct structure directly.
 2012-05-19 15:30:51 +02:00
Mans Rullgard
63dcd16d56 fate: fix dependencies on SAMPLES being set 
This allows fate to run without errors with or without SAMPLES
being set.

Signed-off-by: Mans Rullgard <mans@mansr.com>
 2012-05-19 14:30:41 +01:00
Mans Rullgard
8ccc1c5975 fate: remove unnecessary libavfilter dependency 
avconv now requires libavfilter so this dependency is redundant.

Signed-off-by: Mans Rullgard <mans@mansr.com>
 2012-05-19 14:25:00 +01:00
Stefano Sabatini
0e82d317eb lavfi: rename vf_split.c -> f_split.c 
The file now contains both an audio and a video split filter.
 2012-05-19 13:24:57 +02:00
Stefano Sabatini
150227e8ed lavfi/asplit: move asplit code to vf_split.c, and make it support N outputs 
The move allows to share the init code already used by split.
 2012-05-19 13:24:53 +02:00