1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-12 19:18:44 +02:00
Commit Graph

20 Commits

Author SHA1 Message Date
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
Piotr Bandurski
9bbfcc2675 rmenc: write correct bytes per minute
improves playback of ac3 in RealPlayer

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-23 16:08:10 +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
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
967facb695 Merge remote-tracking branch 'qatar/master'
* qatar/master: (26 commits)
  adxenc: use AVCodec.encode2()
  adxenc: Use the AVFrame in ADXContext for coded_frame
  indeo4: fix out-of-bounds function call.
  configure: Restructure help output.
  configure: Internal-only components should not be command-line selectable.
  vorbisenc: use AVCodec.encode2()
  libvorbis: use AVCodec.encode2()
  libopencore-amrnbenc: use AVCodec.encode2()
  ra144enc: use AVCodec.encode2()
  nellymoserenc: use AVCodec.encode2()
  roqaudioenc: use AVCodec.encode2()
  libspeex: use AVCodec.encode2()
  libvo_amrwbenc: use AVCodec.encode2()
  libvo_aacenc: use AVCodec.encode2()
  wmaenc: use AVCodec.encode2()
  mpegaudioenc: use AVCodec.encode2()
  libmp3lame: use AVCodec.encode2()
  libgsmenc: use AVCodec.encode2()
  libfaac: use AVCodec.encode2()
  g726enc: use AVCodec.encode2()
  ...

Conflicts:
	configure
	libavcodec/Makefile
	libavcodec/ac3enc.c
	libavcodec/adxenc.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/vorbisenc.c
	libavcodec/wmaenc.c
	tests/ref/acodec/g722
	tests/ref/lavf/asf
	tests/ref/lavf/ffm
	tests/ref/lavf/mkv
	tests/ref/lavf/mpg
	tests/ref/lavf/rm
	tests/ref/lavf/ts
	tests/ref/seek/lavf_asf
	tests/ref/seek/lavf_ffm
	tests/ref/seek/lavf_mkv
	tests/ref/seek/lavf_mpg
	tests/ref/seek/lavf_rm
	tests/ref/seek/lavf_ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-22 00:40:11 +01:00
Justin Ruggles
aa872af5e3 ac3enc: update to AVCodec.encode2()
Update FATE references due to encoder delay.
2012-03-20 18:46:56 -04:00
Anton Khirnov
1270e12e49 avconv: rework -t handling for encoding.
Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.

In several tests, one less frame is encoded, which is more correct.

In the idroq test one more frame is encoded, which is again more
correct.

Behavior with stream copy should be unchanged.
2012-02-07 20:11:11 +01:00
Reimar Döffinger
6c723f3f9d lavf-regression: minimal metadata test.
This tests writing the global "title" metadata.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
2011-12-12 20:51:29 +01:00
Justin Ruggles
79ee8977c2 ac3enc: correct the flipped sign in the ac3_fixed encoder 2011-04-26 17:19:37 -04:00
Justin Ruggles
e05a3ac713 ac3enc: select bandwidth based on bit rate, sample rate, and number of
full-bandwidth channels.

This reduces high-frequency artifacts and improves the quality of the lower
frequency audio at low bit rates.
2011-04-03 20:59:14 -04:00
Mans Rullgard
79997def65 ac3enc: use generic fixed-point mdct
This makes the AC3 encoder use the shared fixed-point MDCT rather
than its own implementation.  The checksum changes are due to
different rounding in the MDCT.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-04-03 19:01:53 +01:00
Justin Ruggles
e6e9823488 Add apply_window_int16() to DSPContext with x86-optimized versions and use it
in the ac3_fixed encoder.
2011-03-22 21:08:30 -04:00
Justin
323e6fead0 ac3enc: do not right-shift fixed-point coefficients in the final MDCT stage.
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
2011-03-14 08:45:26 -04:00
Justin Ruggles
50d7140441 ac3enc: change default floor code to 7.
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-15 21:40:42 +00:00
Justin Ruggles
c3beafa0f1 ac3enc: Change EXP_DIFF_THRESHOLD to 500.
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder.  I tested lowering in
increments of 100.  From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-02 20:00:43 +00:00
Justin Ruggles
ec44dd5fc2 Change the default dB-per-bit code from 2 to 3.
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.

Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.

Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-29 19:17:22 +00:00
Justin Ruggles
295ab2af6e Change FIX15() back to clipping to -32767..32767.
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.

Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-21 21:18:58 +00:00
Justin Ruggles
8c634b707b Update the test references for lavf-rm and seek-ac3_rm.
The references changed due to r25956.

Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 16:14:52 +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