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* 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>
* adpcmenc: fix QT IMA ADPCM encoderBaptiste Coudurier2011-09-23
| | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* adpcmdec: Fix QT IMA ADPCM decoderBaptiste Coudurier2011-09-23
| | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* movenc: create an alternate group for each media typeAnton Khirnov2011-09-17
| | | | Partially fixes bug 44.
* ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.Justin Ruggles2011-09-05
| | | | Update FATE references accordingly.
* fate: run aref and vref as regular testsMans Rullgard2011-05-18
| | | | | | | | | 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>
* ac3enc: correct the flipped sign in the ac3_fixed encoderJustin Ruggles2011-04-26
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* matroskaenc: don't write an empty Cues element.Anton Khirnov2011-04-07
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* ac3enc: select bandwidth based on bit rate, sample rate, and number ofJustin Ruggles2011-04-03
| | | | | | | full-bandwidth channels. This reduces high-frequency artifacts and improves the quality of the lower frequency audio at low bit rates.
* ac3enc: use generic fixed-point mdctMans Rullgard2011-04-03
| | | | | | | | 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>
* Add apply_window_int16() to DSPContext with x86-optimized versions and use itJustin Ruggles2011-03-22
| | | | in the ac3_fixed encoder.
* ac3enc: do not right-shift fixed-point coefficients in the final MDCT stage.Justin2011-03-14
| | | | | | | | | 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.
* ac3enc: fix bug in stereo rematrixing decision.Justin Ruggles2011-02-16
| | | | | | | 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>
* ac3enc: change default floor code to 7.Justin Ruggles2011-02-15
| | | | | | | 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>
* ac3enc: Change EXP_DIFF_THRESHOLD to 500.Justin Ruggles2011-02-02
| | | | | | | | | 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>
* Add stereo rematrixing support to the AC-3 encoders.Justin Ruggles2011-01-08
| | | | | | | | 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
* Change the AC-3 encoder to use floating-point.Justin Ruggles2011-01-04
| | | | | | | | 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
* Change the default dB-per-bit code from 2 to 3.Justin Ruggles2010-12-29
| | | | | | | | | | | | 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
* Change FIX15() back to clipping to -32767..32767.Justin Ruggles2010-12-21
| | | | | | | | 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
* Simplify fix15().Justin Ruggles2010-12-14
| | | | | | | | 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
* Set a constant frame size for encoding G.726 audio.Justin Ruggles2010-09-11
| | | | Originally committed as revision 25107 to svn://svn.ffmpeg.org/ffmpeg/trunk
* tiny_psnr: skip wav headers on input filesMåns Rullgård2010-07-09
| | | | | | | | | | 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
* tiny_psnr: print max absolute difference between filesVitor Sessak2010-07-09
| | | | | | | | | Regression test reference updates are due to the extra output from tiny_psnr. Patch by Vitor Sessak Originally committed as revision 24132 to svn://svn.ffmpeg.org/ffmpeg/trunk
* matroskaenc: Don't write a second seekhead for the clusters; mkvalidate agreesDavid Conrad2010-06-04
| | | | | | with me that it's unnecessary. Originally committed as revision 23478 to svn://svn.ffmpeg.org/ffmpeg/trunk
* matroskaenc: Mux clusters betterJames Zern2010-06-04
| | | | | | | | | | Start them on keyframes when reasonable, and delay writing audio packets to help ensure that there's audio samples available for the first frame in clusters. Patch by James Zern <jzern at google> Originally committed as revision 23473 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Update regression tests after removing track timecode scale from mkvencDavid Conrad2010-05-22
| | | | Originally committed as revision 23248 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Add VorbisComment writing to FLAC files.James Darnley2010-03-20
| | | | | | Patch by James Darnley <james darnley at gmail>. Originally committed as revision 22605 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Simplify starting and ending clustersDavid Conrad2010-03-04
| | | | Originally committed as revision 22199 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Place regression test output files in subdirs per familyMåns Rullgård2010-03-02
| | | | Originally committed as revision 22155 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Separate audio-only tests so they are only run onceMåns Rullgård2010-01-30
Originally committed as revision 21556 to svn://svn.ffmpeg.org/ffmpeg/trunk
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