- 23 Aug, 2013 1 commit
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John Stebbins authored
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>
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- 02 Dec, 2012 1 commit
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Janne Grunau authored
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- 29 May, 2012 1 commit
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Mans Rullgard authored
Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 17 May, 2012 1 commit
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Mans Rullgard authored
This fixes a bogus bitrate value in the header of WAV files with alaw/ulaw audio. Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 20 Apr, 2012 3 commits
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Justin Ruggles authored
This avoids resampling and channel mixing by using a source with the correct channel layout and sample rate.
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Justin Ruggles authored
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.
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Justin Ruggles authored
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.
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- 10 Apr, 2012 1 commit
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Justin Ruggles authored
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.
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- 20 Mar, 2012 1 commit
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Justin Ruggles authored
FATE reference updated due timestamp rounding because of resampling from 44100 Hz to 16000 Hz in avconv.
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- 17 Mar, 2012 1 commit
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Justin Ruggles authored
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- 03 Mar, 2012 1 commit
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Justin Ruggles authored
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
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- 02 Mar, 2012 1 commit
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Martin Storsjö authored
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>
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- 20 Feb, 2012 1 commit
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Justin Ruggles authored
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
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- 11 Feb, 2012 1 commit
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Justin Ruggles authored
Otherwise it is not needed because it matches the frame size as encoded in the extradata.
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- 02 Feb, 2012 1 commit
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Mans Rullgard authored
There is no point in this test using the RM format. Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 10 Jan, 2012 1 commit
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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- 07 Jan, 2012 1 commit
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Justin Ruggles authored
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.
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- 03 Jan, 2012 1 commit
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Justin Ruggles authored
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- 15 Dec, 2011 1 commit
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Alex Converse authored
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- 09 Dec, 2011 1 commit
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Justin Ruggles authored
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.
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- 05 Dec, 2011 1 commit
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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- 01 Dec, 2011 1 commit
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Justin Ruggles authored
this removes 2 redundant tests for pcm in mkv. we can add the coverage back in later as fate-lavf tests if needed.
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- 02 Nov, 2011 1 commit
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Diego Biurrun authored
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- 26 Oct, 2011 1 commit
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Justin Ruggles authored
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.
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- 14 Oct, 2011 1 commit
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John Brooks authored
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>
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- 24 Sep, 2011 2 commits
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Baptiste Coudurier authored
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Baptiste Coudurier authored
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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- 17 Sep, 2011 1 commit
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Anton Khirnov authored
Partially fixes bug 44.
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- 05 Sep, 2011 1 commit
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Justin Ruggles authored
Update FATE references accordingly.
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- 18 May, 2011 1 commit
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Mans Rullgard authored
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>
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- 26 Apr, 2011 1 commit
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Justin Ruggles authored
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- 07 Apr, 2011 1 commit
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Anton Khirnov authored
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- 04 Apr, 2011 1 commit
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Justin Ruggles authored
full-bandwidth channels. This reduces high-frequency artifacts and improves the quality of the lower frequency audio at low bit rates.
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- 03 Apr, 2011 1 commit
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Mans Rullgard authored
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>
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- 23 Mar, 2011 1 commit
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Justin Ruggles authored
in the ac3_fixed encoder.
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- 14 Mar, 2011 1 commit
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Justin authored
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.
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- 16 Feb, 2011 1 commit
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Justin Ruggles authored
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>
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- 15 Feb, 2011 1 commit
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Justin Ruggles authored
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>
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- 02 Feb, 2011 1 commit
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Justin Ruggles authored
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>
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- 08 Jan, 2011 1 commit
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Justin Ruggles authored
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
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