Commit Graph

18 Commits

Author SHA1 Message Date
Paul B Mahol 9e2387a6a9 fate: upate after 55d32eed8f
Signed-off-by: Paul B Mahol <onemda@gmail.com>
2013-01-08 18:47:09 +00:00
Alexander Strasser ac25b31ede lswr: Improve default resampler's default parameters
After making some blind tests on a small collection of music
samples for home usage. It turned out that the default cutoff
was too low.

The impact of filter_size was not clearly distinguishable (the
results were on the edge) with the music samples but turned out
to be clearly audible in some synthetic samples.

Thanks to Daniel for helping out with the listening tests.

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-01-04 16:47:57 +01:00
Michael Niedermayer 9e9b5159e9 mpegvideo_enc: reduce QMAT_SHIFT to avoid overflow in dnxhd
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-09-27 19:43:31 +02:00
Nicolas George 2fc354f90d ffmpeg: rework checks for the -t option.
This commit is based on libav's implementation and
makes sure to compare output timestamps together.
It also reduces the differences with avconv.

The changes to the test reference files are caused
by an additional packet at the end, the timestamp
of the frame encoded by this packet is always
strictly below the limit stated by the -t option.
2012-07-04 16:20:47 +02:00
Michael Niedermayer 6ba692f8a7 af_aresample: fix rounding that led to sample accumulation in the buffers.
This fixes a regression that apparently was missed when switching to the
in af resampler

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-18 22:57:02 +02:00
Anton Khirnov fc49f22c3b ffmpeg: add support for audio filters.
Some of the FATE changes are due to off-by-one different rounding being used
(lrintf vs av_rescale_q).
Some fate changes are due to 1 audio frame less being encoded (the new variant seems
matching what qatar does and according to ffprobe its closer to the requested duration)
the mapchan feature sadly is lost in this commit because it depends on resampling
being done in ffmpeg.c which is now moved completely into the av filter layer
-async is broken after this commit, this will be fixed in subsequent commits
the new filter reconfiguration system is flawed and will drop a frame on each
parameter change which is why the nelly moser checksums need updating.

Conflicts:

	ffmpeg.c
	tests/ref/fate/smjpeg
2012-05-17 03:29:21 +02:00
Michael Niedermayer 3194ab78a6 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  avcodec: add a cook parser to get subpacket duration
  FATE: allow lavf tests to alter input parameters
  FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
  FATE: replace the acodec-g726 test with 4 new encode/decode tests
  FATE: replace current g722 encoding tests with an encode/decode test
  FATE: add a pattern rule for generating asynth wav files
  FATE: optionally write a WAVE header in audiogen
  avutil: add audio fifo buffer

Conflicts:
	doc/APIchanges
	libavcodec/version.h
	libavutil/avutil.h
	tests/Makefile
	tests/codec-regression.sh
	tests/fate/voice.mak
	tests/lavf-regression.sh
	tests/ref/acodec/g722
	tests/ref/acodec/g726
	tests/ref/acodec/pcm_s24daud
	tests/ref/lavf/dv_fmt
	tests/ref/lavf/gxf
	tests/ref/lavf/mxf
	tests/ref/lavf/mxf_d10
	tests/ref/seek/lavf_dv

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-04-20 22:18:26 +02:00
Justin Ruggles acb1730218 FATE: allow lavf tests to alter input parameters
Change some lavf tests to avoid resampling and channel mixing.
2012-04-20 10:23:57 -04:00
Anton Khirnov cd1ad18a65 rawenc: switch to encode2().
This changes a number of FATE results, since before this commit, the
timestamps in all tests using rawenc were made up by lavf.

In most cases, the previous timestamps were completely bogus.

In some other cases -- raw formats, mostly h264 -- the new timestamps
are bogus as well. The only difference is that timestamps invented by
the muxer are replaced by timestamps invented by the demuxer.

cscd     -- avconv sets output codec timebase from r_frame_rate
and r_frame_rate is in this case some guessed number 31.42 (377/12),
which is not accurate enough to represent all timestamps. This results
in some frames having duplicate pts. Therefore, vsync 0 needs to be
changed to vsync 2 and avconv drops two frames. A proper fix in the
future would be to set output timebase to something saner in avconv.

nuv      -- previous timestamps for video were wrong AND the cscd
comment applies, one frame is dropped.

vp8-signbias -- the file contains two frames with identical timestamps,
so -vsync 0 needs to be removed/changed to -vsync 2 and avconv drops one
frame.

vc1-ism -- apparrently either the demuxer lies about timestamps or the
file is broken, since dts == pts on all packets, but reordering clearly
takes place.
2012-02-08 21:51:24 +01:00
Clément Bœsch b18ebcbe83 timecode: add write regressions tests. 2012-02-02 14:16:34 +01:00
Matthieu Bouron 070a40f1b7 gxfenc: support timecode option
Reviewed-by: Baptiste Coudurier
2011-12-07 15:29:26 +01:00
Michael Niedermayer 957867ab13 ffmpeg: rewrite vsync / notimestamps handling
The qatar implementation makes no sense.
a muxer without timestamps is constant fps thus needs vsync.
the crc/mp5 are special cases that have timestamps yet allow any
nonsensical timestamps.
raw (yuv/rgb) video is constant fps thus needs vsync too.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-24 04:57:38 +01:00
Vitor Sessak 96573c0d76 lavf/utils.c: Order packets with identical PTS by stream index.
This allows for more reproducible results when using multi-threading.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-04-12 19:06:26 -04:00
Baptiste Coudurier a7ba165a0c Update gxf regression tests because of r25399
Originally committed as revision 25400 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-07 19:37:37 +00:00
Reuben Martin ad4c3c6840 In gxf muxer, fix flt entry offset, patch by Reuben Martin, reuben dot m at gmail dot com
Originally committed as revision 25395 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-07 19:15:35 +00:00
Thierry Foucu 591db22dff gxfenc: Fix ES name in the UMF media description, by using strlen instead of sizeof
Patch by Thierry Foucu, tfoucu at gmail

Originally committed as revision 24379 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-21 07:46:02 +00:00
Måns Rullgård cc3e2472f3 Place regression test output files in subdirs per family
Originally committed as revision 22155 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-02 21:41:52 +00:00
Måns Rullgård eca478c317 regtest: split reference files allowing tests to run individually
With this change, the output is checked immediately after each test
has run.  This means commands like "make regtest-mpeg2" can now be
used to run a single test and get meaningful results.

By default, make will abort if any test fails.  To run all tests
regardless, use make -k.

Originally committed as revision 21254 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-01-16 20:18:13 +00:00