summaryrefslogtreecommitdiff
path: root/tests/ref/acodec
Commit message (Collapse)AuthorAge
* 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>
* fate: move fate-acodec reference files to their own dirJanne Grunau2012-12-03
|
* 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
|
* 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
|
* movenc: Rudimentary IODs support.Alex Converse2011-12-15
|
* 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
|
* 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
|
* matroskaenc: don't write an empty Cues element.Anton Khirnov2011-04-07
|
* 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
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-01 20:22:28 +0000