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Author SHA1 Message Date
Justin Ruggles
32173df3d2 roqaudioenc: use AVCodec.encode2() 
The first frame pts must be saved until we have 8 frames since RoQ audio
requires 8 frames in the first packet.
 2012-03-21 12:49:35 -04:00
Justin Ruggles
b03dcf07f6 libspeex: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles
57a52f258e libvo_amrwbenc: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles
db440fa12d libvo_aacenc: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles
27bacfeb57 wmaenc: use AVCodec.encode2() 2012-03-21 12:49:32 -04:00
Nicolas George
3493390d47 lavfi: add tile video filter. 2012-03-21 15:52:45 +01:00
Nicolas George
7084985173 vsrc_color: port to new drawutils API. 2012-03-21 15:52:45 +01:00
Nicolas George
53b7a3fe08 vf_pad: port to new drawutils API. 2012-03-21 15:52:45 +01:00
Nicolas George
8ec0832743 drawutils: new API. 
This new API stores useful data in a dedicated structure
and has clearly delimited init functions.

Hopefully, uses of the old API can be replaced quickly.
 2012-03-21 15:52:45 +01:00
Michael Niedermayer
87a72b9122 swscale: Merge a hunk from qatar that seems to have been forgotten or lost. 
Author of the code from qatar is Ronald S. Bultje
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 15:30:52 +01:00
Stefano Sabatini
8e0d3c0369 lavfi/ass: add dar option 
Allow to specify the display aspect ratio adopted for rendering
subtitles.
 2012-03-21 15:14:28 +01:00
Stefano Sabatini
c9399538b7 lavfi/ass: use a default DAR value of 1.0 
Previously it was using the same value of the input video DAR, which is
inconsistent with most implementations.

Fix trac ticket #1098.
 2012-03-21 15:14:28 +01:00
Stefano Sabatini
e71e65ff1d lavfi/aspect: check for a negative code from av_parse_ratio() 
Check on < 0 rather than on != 0, this is more correct as a positive
error code from av_parse_ratio() value doesn't mean an error.
 2012-03-21 15:14:28 +01:00
Nicolas George
6cf53927c4 graphdump: use av_bprintf API. 2012-03-21 13:39:28 +01:00
Nicolas George
b75c67dc01 lavu: add av_bprintf and related. 2012-03-21 13:39:28 +01:00
Paul B Mahol
594a3d6315 bink: no need to increase width twice 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 04:18:34 +01:00
Michael Niedermayer
9e69d3c6d4 zerocodec: factorize loop 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 04:12:48 +01:00
Paul B Mahol
15e07348fe ttadec: refactor ttafilter_process() 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:39:14 +01:00
Paul B Mahol
1a7a707f74 tgq: use bytestream2_get_bytes_left() 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:09:58 +01:00
Paul B Mahol
0acacd23d4 xxan: use bytestream2_size() 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:09:04 +01:00
Paul B Mahol
ff05fd6249 xxan: remove write-only variable 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:08:29 +01:00
Paul B Mahol
8a90148dfe smc: use bytestream2_size() 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:07:43 +01:00
Paul B Mahol
0ee5be4ee4 bytestream: add functions for accessing size of buffer 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:05:19 +01:00
Paul B Mahol
841e669a39 cdxl: swap CHUNKY and BYTE_PLANAR 
This is how it is defined in Amiga Developer CD from year 1992 and
is consistent with files created with ADPro.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 03:03:33 +01:00
Michael Niedermayer
3eaf712053 sgienc: fix packet size. 
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 02:54:49 +01:00
Michael Niedermayer
0ebd83617f Merge remote-tracking branch 'qatar/master' 
* qatar/master: (27 commits)
  avconv: free packet in write_frame() when discarding due to frame number limit
  FATE: use +/- flag option syntax for vp8 emu-edge tests
  lavf: make av_interleave_packet_per_dts() private.
  lavf: deprecate av_read_packet().
  oggdec: output correct timestamps for Vorbis
  avconv: pass input stream timestamps to audio encoders
  lavc: shrink encoded audio packet size after encoding.
  xa: set correct bit rate
  xa: do not set bit_rate, block_align, or bits_per_coded_sample
  xa: fix end-of-file handling
  xa: fix timestamp calculation
  bink: fix typo in FFALIGN() argument
  bink: align plane width to 8 when calculating bundle sizes
  doc: pass -Idoc texi2html and texi2pod
  doc: texi2pod: add -I flag
  movenc: Add a min_frag_duration option
  rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
  libavformat: Set the default for the max_delay option to -1
  Generate manpages for AV{Format,Codec}Context AVOptions.
  doc/avconv: remove entries for AVOptions.
  ...

Conflicts:
	doc/Makefile
	doc/ffmpeg.texi
	doc/muxers.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/options.c
	libavcodec/vp8.c
	libavformat/options.c
	tests/fate/demux.mak
	tests/ref/fate/truemotion1-15
	tests/ref/fate/truemotion1-24

Merged-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-21 01:33:53 +01:00
Justin Ruggles
b0f75ba272 mpegaudioenc: use AVCodec.encode2() 
Update FATE references due to encoder delay.
 2012-03-20 18:56:22 -04:00
Justin Ruggles
3d853d7ab3 libmp3lame: use AVCodec.encode2() 2012-03-20 18:56:18 -04:00
Justin Ruggles
1987a940b7 libgsmenc: use AVCodec.encode2() 2012-03-20 18:55:39 -04:00
Justin Ruggles
d1afb2f94e libfaac: use AVCodec.encode2() 
Encoder output is delayed by several frames, so we keep a queue of input
frame timing info to match up with corresponding output packets.
 2012-03-20 18:55:36 -04:00
Justin Ruggles
59041fd053 g726enc: use AVCodec.encode2() 2012-03-20 18:47:23 -04:00
Justin Ruggles
bb03b6f7b1 g722enc: use AVCodec.encode2() 
FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.
 2012-03-20 18:47:23 -04:00
Justin Ruggles
910bdb9a42 flacenc: use AVCodec.encode2() 2012-03-20 18:47:19 -04:00
Justin Ruggles
24e74f0a0f adpcmenc: update to AVCodec.encode2() 2012-03-20 18:46:57 -04:00
Justin Ruggles
aa872af5e3 ac3enc: update to AVCodec.encode2() 
Update FATE references due to encoder delay.
 2012-03-20 18:46:56 -04:00
Justin Ruggles
ad95307f92 aacenc: use AVCodec.encode2() 2012-03-20 18:46:49 -04:00
Michael Niedermayer
745a33a443 fate/zerocodec: fix permissions 
Reported-by: Deamon404
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-20 21:21:14 +01:00
Justin Ruggles
4bf64961a9 avcodec: add code for a frame queue for use by audio encoders with delay 
This simplifies matching of timestamps between input frames and output
packets.
 2012-03-20 16:04:21 -04:00
Justin Ruggles
c9594fe0fb avconv: free packet in write_frame() when discarding due to frame number limit 
Fixes a memleak when using the -frames option with audio.
 2012-03-20 15:51:58 -04:00
Justin Ruggles
e056f8d37d FATE: use +/- flag option syntax for vp8 emu-edge tests 2012-03-20 15:51:58 -04:00
Michael Niedermayer
15db6a9590 pngenc: Fix incorrect mask used for interlaced mode. 
Fixes Ticket1109

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
 2012-03-20 20:39:32 +01:00
Anton Khirnov
a6733202cc lavf: make av_interleave_packet_per_dts() private. 
There is no reason for it to be public, it's only meant to be used
internally.
 2012-03-20 20:12:16 +01:00
Anton Khirnov
3c90cc2ef2 lavf: deprecate av_read_packet(). 
The caller can achieve the same effect (i.e. getting raw unparsed/mangled
packets) with av_read_frame() and AVFMT_FLAG_NOPARSE |
AVFMT_FLAG_NOFILLIN
 2012-03-20 20:12:16 +01:00
Justin Ruggles
f63412fc74 oggdec: output correct timestamps for Vorbis 
Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
 2012-03-20 14:39:57 -04:00
Justin Ruggles
9b9fc9ba32 avconv: pass input stream timestamps to audio encoders 
5 FATE test references updated due to using demuxer-generated timestamps that
are either not sample-accurate or are slightly off in the input file.
 2012-03-20 14:12:54 -04:00
Justin Ruggles
a1977e0103 lavc: shrink encoded audio packet size after encoding. 2012-03-20 14:12:54 -04:00
Justin Ruggles
777365fe86 xa: set correct bit rate 
Also fixes stream duration calculation.
 2012-03-20 14:12:54 -04:00
Justin Ruggles
a54bc52265 xa: do not set bit_rate, block_align, or bits_per_coded_sample 
The values in the header refer to decoded data, not compressed data.
 2012-03-20 14:12:53 -04:00
Justin Ruggles
64de57f645 xa: fix end-of-file handling 
Do not output an extra packet when out_size is reached.
Also return AVERROR_EOF instead of AVERROR(EIO).
 2012-03-20 14:12:53 -04:00
Justin Ruggles
cd2ffb67ad xa: fix timestamp calculation 
The packet duration is always 28 samples.
 2012-03-20 14:12:53 -04:00