Commit Graph

17 Commits

Author SHA1 Message Date
Christophe Gisquet
1086f09da3 dv: more precise weight table for 8x8
It is derived from the actual equations of the specs. In
particular, it is closer to the inverse of what the encoder uses.

fate tests accordingly updated.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-10-25 19:01:45 +02:00
Nicolas George
dd9555e94b ffmpeg: remove obsolete workaround in trim insertion.
The bug it was working seems to have been fixed.
This change causes ffmpeg to use the trim filter to implement
the -t option.
FATE tests are updated due to the more accurate handling of
the last packets.
2013-08-07 16:20:41 +02:00
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
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
3ead79eaa3 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  FATE: use updated reference for aac-latm_stereo_to_51
  avconv: use libavresample
  Add libavresample
  FATE: avoid channel mixing in lavf-dv_fmt

Conflicts:
	Changelog
	Makefile
	cmdutils.c
	configure
	doc/APIchanges
	ffmpeg.c
	tests/lavf-regression.sh
	tests/ref/lavf/dv_fmt
	tests/ref/seek/lavf_dv

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-04-25 23:17:41 +02:00
Justin Ruggles
c5671aeb77 FATE: avoid channel mixing in lavf-dv_fmt
This partially reverts acb1730218
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
2012-04-24 15:55:45 -04:00
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
Clément Bœsch
b18ebcbe83 timecode: add write regressions tests. 2012-02-02 14:16:34 +01:00
Michael Niedermayer
8593b743a8 rematrix: dont use floats for int16 code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 09:25:50 +02:00
Michael Niedermayer
b5875b9111 Add libswresample.
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 07:04:17 +02:00
Clément Bœsch
75af0e6a16 dv: honor timecode in DV muxer.
This is based on the original work by Baptiste Coudurier.
2011-08-13 19:13:03 +02:00
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