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* adpcm: fix clipping for yamahaPaul B Mahol2017-02-15
| | | | | | | According to specification max value allowed is 0x6000. Fixes #5862. Signed-off-by: Paul B Mahol <onemda@gmail.com>
* avformat: add a TTA MuxerJames Almer2016-08-04
| | | | | Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
* Merge commit '393596f9d51134d6e45d81ae129223f4faea1232'Clément Bœsch2016-06-23
|\ | | | | | | | | | | | | | | | | | | | | * commit '393596f9d51134d6e45d81ae129223f4faea1232': mpegtsenc: stop impersonating ses in sdt This commit also includes the needed FATE updates later spotted by Martin Storsjö and fixed in 34effe816f9f3df2e6b8bc738e2b5a86a24fd0d7 on Libav side. Merged-by: Clément Bœsch <u@pkh.me>
* | Merge commit 'dc6527ed908e4d330738f139074455ffbe56a2de'Derek Buitenhuis2016-02-29
|\| | | | | | | | | | | | | | | | | | | FATE tests have been updated to patch. They do not differ in any meaningful way. * commit 'dc6527ed908e4d330738f139074455ffbe56a2de': nutenc: do not use AVCodecContext.frame_size Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* | libavformat/matroska: Write stream durations in metadata, in the format of ↵Sasi Inguva2015-08-05
| | | | | | | | | | | | | | | | | | mkvmerge. Compute individual stream durations in matroska muxer. Write them as string tags in the same format as mkvmerge tool does. Signed-off-by: Sasi Inguva <isasi@google.com>
* | avformat/rawenc: Store sample number for ADXMichael Niedermayer2015-06-21
| | | | | | | | | | | | | | Fixes Ticket4540 Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: Add test for -exact_rice_parameters 1Michael Niedermayer2015-05-19
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* | fate: add tta encoder testJames Almer2015-04-13
| | | | | | | | | | Signed-off-by: James Almer <jamrial@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | tests/fate: Add S302M testMichael Niedermayer2015-03-02
| | | | | | | | | | Reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: add wavpack encoderPaul B Mahol2015-02-13
| | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com>
* | avcodec/adxenc: fix roundingMichael Niedermayer2014-11-30
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | avcodec/adxenc: match prediction used in the decoderMichael Niedermayer2014-11-30
| | | | | | | | | | | | The prediction used in the encoder was not correct Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | adpcm: Write the proper predictor in trellis mode in IMA QTMartin Storsjö2014-06-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The actual predictor value, set by the trellis code, never was written back into the variable that was written into the block header. This was accidentally removed in b304244b. This significantly improves the audio quality of the trellis case, which was plain broken since b304244b. Encoding IMA QT with trellis still actually gives a slightly worse quality than without trellis, since the trellis encoder doesn't use the exact same way of rounding as in adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble. Fixes part of Ticket3701 Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: enable adpcm-ima_qt-trellisMichael Niedermayer2014-06-06
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | tests: add adpcm trellis testsTimothy Gu2014-06-05
| | | | | | | | | | | | | | | | adpcm_ima_qt does not produce reproducible results, so it is temporarily disabled (see #3701). Signed-off-by: Timothy Gu <timothygu99@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | ffmpeg: prefix encoder with "Lavc " in bitexact modeMichael Niedermayer2014-05-18
| | | | | | | | | | | | This avoids misleading encoder names like "encoder = prores" Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge commit '6656370b858329ca07a60a2de954d5e90daa0206'Michael Niedermayer2014-05-18
|\| | | | | | | | | | | | | | | | | | | | | | | * commit '6656370b858329ca07a60a2de954d5e90daa0206': avconv: set the "encoder" tag when transcoding Conflicts: ffmpeg.c tests/ref/lavf/mkv tests/ref/seek/lavf-mkv Merged-by: Michael Niedermayer <michaelni@gmx.at>
* | ff_put_wav_header: add flag to force WAVEFORMATEXDaniel Verkamp2014-04-30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Partially undoes commit 2c4e08d89327595f7f4be57dda4b3775e1198d5e: riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs. This flag is used in the Matroska muxer (the cause of the original change) and in the ASF muxer, because the specifications for these formats indicate explicitly that WAVEFORMATEX should be used. Muxers for other formats will return to the original behavior of writing PCMWAVEFORMAT when writing a header for raw PCM. In particular, this causes raw PCM in WAV to generate the canonical 44-byte header expected by some tools. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: force 128kb/sec for mp2 testMichael Niedermayer2014-04-15
| | | | | | | | | | | | This fixes rounding differences between platforms Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | avcodec/mpegaudioenc_template: default to 384k bitrate as defaultMichael Niedermayer2014-04-11
| | | | | | | | | | | | | | | | If 384k is too high for the samplerate, choose the closest possible Idea to increase the bitrate from: 46439e156219d27f059cf687743ba5aacf238b87 Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | avcodec: split mp2 encoder into float and fixedMichael Niedermayer2013-12-03
| | | | | | | | | | | | | | | | | | | | This makes the USE_FLOATS == 0 available to the end user More float optimizations can easily be added as well now common code should be factored out into a common file once all fixed point & floating point optimizations are done, this is to avoid having to move code back and forth between files. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2013-08-23
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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>
| * movenc: Make tkhd "enabled" flag QuickTime compatibleJohn Stebbins2013-08-23
| | | | | | | | | | | | | | | | 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>
* | lswr: Improve default resampler's default parametersAlexander Strasser2013-01-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | aiffenc: set correct number of bits foru8 in aiffPiotr Bandurski2012-12-20
| | | | | | | | | | | | with this change QuickTime is able to play u8 aiff file generated by FFmpeg Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6'Michael Niedermayer2012-12-03
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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>
| * fate: move fate-acodec reference files to their own dirJanne Grunau2012-12-03
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* fate: convert codec-regression.sh to makefile rulesMans Rullgard2012-05-29
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* pcmenc: set correct bitrate valueMans Rullgard2012-05-17
| | | | | | | This fixes a bogus bitrate value in the header of WAV files with alaw/ulaw audio. Signed-off-by: Mans Rullgard <mans@mansr.com>
* FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum testJustin Ruggles2012-04-20
| | | | | This avoids resampling and channel mixing by using a source with the correct channel layout and sample rate.
* FATE: replace the acodec-g726 test with 4 new encode/decode testsJustin Ruggles2012-04-20
| | | | | | 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.
* FATE: replace current g722 encoding tests with an encode/decode testJustin Ruggles2012-04-20
| | | | | | 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.
* avconv: use default channel layouts when they are unknownJustin Ruggles2012-04-10
| | | | | | | | | 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.
* g722enc: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | | FATE reference updated due timestamp rounding because of resampling from 44100 Hz to 16000 Hz in avconv.
* FATE: remove WMA acodec testsJustin Ruggles2012-03-17
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* wmaenc: fix m/s stereo encoding for the first frameJustin Ruggles2012-03-03
| | | | | | | | 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
* g722: Fix the QMF scalingMartin Storsjö2012-03-02
| | | | | | | | | | | | | | | 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>
* adpcmenc: Use correct frame_size for Yamaha ADPCM.Justin Ruggles2012-02-20
| | | | | | | | | | | 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
* alacenc: only encode frame size in header for a final smaller frameJustin Ruggles2012-02-11
| | | | | Otherwise it is not needed because it matches the frame size as encoded in the extradata.
* fate: make acodec-ac3_fixed test output raw AC3Mans Rullgard2012-02-02
| | | | | | There is no point in this test using the RM format. Signed-off-by: Mans Rullgard <mans@mansr.com>
* fate: Update file checksums after the mov muxer change in a78dbada55d6Martin Storsjö2012-01-10
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* g722enc: set frame_size, and also handle an odd number of input samplesJustin Ruggles2012-01-07
| | | | | 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.
* fate: add ADX encoding/decoding testJustin Ruggles2012-01-03
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* movenc: Rudimentary IODs support.Alex Converse2011-12-15
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* movenc: simplify handling of pcm vs. adpcm vs. other compressed codecsJustin Ruggles2011-12-09
| | | | | | 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.
* g722: Add a regression test for muxing/demuxing in wavMartin Storsjö2011-12-05
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* fate: split acodec-pcm into individual testsJustin Ruggles2011-12-01
| | | | | this removes 2 redundant tests for pcm in mkv. we can add the coverage back in later as fate-lavf tests if needed.
* Replace vendor string in Ogg and FLAC muxers.Diego Biurrun2011-11-02
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* avcodec: remove the Zork PCM encoder.Justin Ruggles2011-10-26
| | | | | | 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.
* riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_headerJohn Brooks2011-10-14
| | | | | | | | | | | | | | 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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