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Commit Graph

58 Commits

Author SHA1 Message Date
Michael Niedermayer
800ea20cad Merge remote-tracking branch 'qatar/master'
* qatar/master:
  movenc: Make tkhd "enabled" flag QuickTime compatible

Conflicts:
	libavformat/movenc.c
	tests/ref/acodec/alac
	tests/ref/acodec/pcm-s16be
	tests/ref/acodec/pcm-s24be
	tests/ref/acodec/pcm-s32be
	tests/ref/acodec/pcm-s8
	tests/ref/lavf/mov
	tests/ref/vsynth/vsynth1-dnxhd-1080i
	tests/ref/vsynth/vsynth1-mpeg4
	tests/ref/vsynth/vsynth1-prores
	tests/ref/vsynth/vsynth1-qtrle
	tests/ref/vsynth/vsynth1-svq1
	tests/ref/vsynth/vsynth2-dnxhd-1080i
	tests/ref/vsynth/vsynth2-mpeg4
	tests/ref/vsynth/vsynth2-prores
	tests/ref/vsynth/vsynth2-qtrle
	tests/ref/vsynth/vsynth2-svq1

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-23 13:49:24 +02:00
John Stebbins
30ce289074 movenc: Make tkhd "enabled" flag QuickTime compatible
QuickTime will play multiple audio tracks concurrently if this flag is
set for multiple audio tracks.  And if no subtitle track has this flag
set, QuickTime will show no subtitles in the subtitle menu.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2013-08-23 09:55:42 +02:00
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
Piotr Bandurski
52f2176366 aiffenc: set correct number of bits foru8 in aiff
with this change QuickTime is able to play u8 aiff file generated by FFmpeg

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-20 16:05:30 +01:00
Michael Niedermayer
7711f19eda Merge commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6'
* commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6':
  fate-seek: remove use of gnu make 3.82 only private modifier
  fate: move vsynth reference files to their own directory
  fate: move fate-acodec reference files to their own dir
  configure: avplay now depends on avresample
  fate: split dependencies for fate-seek tests

Conflicts:
	configure
	tests/fate/seek.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-03 02:33:27 +01:00
Janne Grunau
337dbe2adb fate: move fate-acodec reference files to their own dir 2012-12-03 00:29:35 +01:00
Mans Rullgard
7263cd5544 fate: convert codec-regression.sh to makefile rules
Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-29 08:35:41 +01:00
Mans Rullgard
7d7b40f48a pcmenc: set correct bitrate value
This fixes a bogus bitrate value in the header of WAV files with
alaw/ulaw audio.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-17 02:34:57 +01:00
Justin Ruggles
5052980400 FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.
2012-04-20 10:23:57 -04:00
Justin Ruggles
03caef1bed FATE: replace the acodec-g726 test with 4 new encode/decode tests
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
2012-04-20 10:23:57 -04:00
Justin Ruggles
a6c8cca2a8 FATE: replace current g722 encoding tests with an encode/decode test
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
2012-04-20 10:23:57 -04:00
Justin Ruggles
d3c59d5003 avconv: use default channel layouts when they are unknown
If either input or output layout is known and the channel counts match,
use the known layout for both. Otherwise choose the default layout based on
av_get_default_channel_layout().

Changed some FATE references due to some WAVE files now having a non-zero
channel mask.
2012-04-10 11:30:01 -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
85cf49fab7 FATE: remove WMA acodec tests 2012-03-17 11:46:15 -04:00
Justin Ruggles
51ddf35c90 wmaenc: fix m/s stereo encoding for the first frame
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.

CC:libav-stable@libav.org
2012-03-03 18:20:10 -05:00
Martin Storsjö
b087ce2bee g722: Fix the QMF scaling
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.

This makes the decoder output have double the magnitude
compared to before.

The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-02 18:58:19 +02:00
Justin Ruggles
770a5c6d02 adpcmenc: Use correct frame_size for Yamaha ADPCM.
Output packet size should match avctx->block_align. The target output packet
size is 1024 bytes.
Before:
mono   - 1024 samples -> 512 bytes
stereo - 2048 samples -> 2048 bytes
After:
mono   - 2048 samples -> 1024 bytes
stereo - 1024 samples -> 1024 bytes
2012-02-20 15:52:32 -05:00
Justin Ruggles
b590f3a7bf alacenc: only encode frame size in header for a final smaller frame
Otherwise it is not needed because it matches the frame size as encoded in
the extradata.
2012-02-11 12:49:22 -05:00
Mans Rullgard
2c98f407c8 fate: make acodec-ac3_fixed test output raw AC3
There is no point in this test using the RM format.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-02-02 14:31:54 +00:00
Martin Storsjö
5c7c9a9f33 fate: Update file checksums after the mov muxer change in a78dbada55
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-10 16:54:23 +02:00
Justin Ruggles
77c5b66cbe g722enc: set frame_size, and also handle an odd number of input samples
The fate reference is updated because the previous test skipped a sample in
each encode() call due each input frame having an odd number of samples.
2012-01-07 13:38:23 -05:00
Justin Ruggles
3e57573fce fate: add ADX encoding/decoding test 2012-01-03 18:47:42 -05:00
Alex Converse
d3b8bde2f1 movenc: Rudimentary IODs support. 2011-12-15 14:06:13 -08:00
Justin Ruggles
8e8c51318c movenc: simplify handling of pcm vs. adpcm vs. other compressed codecs
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
2011-12-09 16:12:58 -05:00
Martin Storsjö
714cd7e758 g722: Add a regression test for muxing/demuxing in wav
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-12-05 12:41:46 +02:00
Justin Ruggles
ca12401376 fate: split acodec-pcm into individual tests
this removes 2 redundant tests for pcm in mkv.
we can add the coverage back in later as fate-lavf tests if needed.
2011-12-01 13:27:56 -05:00
Diego Biurrun
c6cd0e17f3 Replace vendor string in Ogg and FLAC muxers. 2011-11-02 10:43:39 +01:00
Justin Ruggles
85579b6381 avcodec: remove the Zork PCM encoder.
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore
the encoder based on the decoder is also incorrect. There is no good reason to
keep the encoder.
2011-10-26 12:01:07 -04:00
John Brooks
2c4e08d893 riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
The cbSize field should be included in all cases, even with PCM where
its value is ignored.

Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.

Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
2011-10-14 13:28:58 +02:00
Baptiste Coudurier
b304244b54 adpcmenc: fix QT IMA ADPCM encoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-23 20:54:29 -04:00
Baptiste Coudurier
bf334535b4 adpcmdec: Fix QT IMA ADPCM decoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-23 20:54:28 -04:00
Anton Khirnov
7574cacbd5 movenc: create an alternate group for each media type
Partially fixes bug 44.
2011-09-17 08:42:30 +02:00
Justin Ruggles
ae264bb29b ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
Update FATE references accordingly.
2011-09-05 10:09:44 -04:00
Mans Rullgard
70378ea190 fate: run aref and vref as regular tests
These tests create reference files used for psnr calculation in
the other codec tests.  Treating them as (mostly) regular tests
simplifies the makefile and makes them visible in the fate reports.
The latter makes errors in these runs easier to identify.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-05-18 14:45:46 +01:00
Justin Ruggles
79ee8977c2 ac3enc: correct the flipped sign in the ac3_fixed encoder 2011-04-26 17:19:37 -04:00
Anton Khirnov
9181976348 matroskaenc: don't write an empty Cues element. 2011-04-07 18:11:24 +02:00
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
5b54d4b376 ac3enc: fix bug in stereo rematrixing decision.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-16 23:39:57 +00:00
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
dc7e07ac1f Add stereo rematrixing support to the AC-3 encoders.
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.

Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-08 23:21:17 +00:00
Justin Ruggles
6fd96d1a85 Change the AC-3 encoder to use floating-point.
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.

Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-04 11:53:44 +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
918cd2255c Simplify fix15().
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.

Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 14:51:02 +00:00
Justin Ruggles
c7d89948a3 Set a constant frame size for encoding G.726 audio.
Originally committed as revision 25107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-11 19:52:09 +00:00
Måns Rullgård
c43d77c163 tiny_psnr: skip wav headers on input files
The byte count printed excludes the header, and offsets are applied
after the the headers are skipped.

Reference files updated to reflect new output.  Some stddev/psnr values
have changed slightly due to headers no longer being compared.

Originally committed as revision 24143 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-09 16:06:05 +00:00