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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: avoid channel mixing in lavf-dv_fmtJustin Ruggles2012-04-24
| | | | | | | This partially reverts acb1730218f1c614dc8ca3ba45d9de1e05059515 which would only have needed to change the checksums if channel mixing had been properly avoided. This changes the output file size reference and the seek test reference back to the previous values.
* matroska: Add incremental parsing of clusters.Dale Curtis2012-04-22
| | | | | | | | | | | | | | | | | | | | | | Reduces the amount of upfront data required for cluster parsing thus decreasing latency on seek and startup. The change in the seek-lavf_mkv FATE test is due to incremental parsing no longer reading as much data as the old parser and thus not having that additional data to generate index entries based on keyframes. Index entries are added correctly as the file is parsed. All FATE tests pass and Chrome has been using this patch for ~6 months without issue. Currently incremental parsing is not supported for files with SSA tracks since they require merging packets between clusters. In this case the code falls back to non-incremental parsing. Signed-off-by: Aaron Colwell <acolwell@chromium.org> Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* FATE: allow lavf tests to alter input parametersJustin Ruggles2012-04-20
| | | | Change some lavf tests to avoid resampling and channel mixing.
* 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.
* mpegaudioenc: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | Update FATE references due to encoder delay.
* ac3enc: update to AVCodec.encode2()Justin Ruggles2012-03-20
| | | | Update FATE references due to encoder delay.
* FATE: remove WMA acodec testsJustin Ruggles2012-03-17
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* aiffdec: do not set AVCodecContext.frame_sizeJustin Ruggles2012-03-05
| | | | | | | | | It is unnecessary. Also, for some codecs we're reading more than 1 frame per packet. Instead we use a private context variable to calculate the bit rate, stream duration, and packet durations. Updated FATE seek test, which has slightly different timestamps due to a more accurate bit rate calculation.
* voc/avs: Do not change the sample rate mid-stream.Justin Ruggles2012-03-03
| | | | | Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.
* mpegaudio_parser: do not ignore information from the first parsed frameJustin Ruggles2012-03-03
| | | | Update some demuxing and seeking fate tests.
* rawdec: set timebase to 1/fps.Anton Khirnov2012-02-26
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* FATE: update reference for seek-alac_mp4Justin Ruggles2012-02-11
| | | | This should have been updated in b590f3a7bf9103ac7a7a61c48568676201d6824b.
* avconv: rework -t handling for encoding.Anton Khirnov2012-02-07
| | | | | | | | | | | | | Current code compares the desired recording time with InputStream.pts, which has a very unclear meaning. Change the code to use actual timestamps of the frames passed to the encoder. In several tests, one less frame is encoded, which is more correct. In the idroq test one more frame is encoded, which is again more correct. Behavior with stream copy should be unchanged.
* 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>
* mxfdec: hybrid demuxing/seeking solutionJanne Grunau2012-01-22
| | | | | | | | | | | | | | | | | | This uses the old demuxing code for OP1a and separate demuxing code for OPAtom. Timestamp output is added to the old demuxing code. The seeking code is made to seek to the start of the desired EditUnit only, from which the normal demuxing code takes over (if OP1a). This means we do not use delta entries or slices, only StreamOffsets. OPAtom seeking basically works like before. This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other words, we allow seeking before the start or past the end for D-10 too. Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and Reimar Döffinger <Reimar.Doeffinger@gmx.de>. Changed av_calloc to av_mallocz, added overflow checks.
* fate: update asf seektestLuca Barbato2011-12-02
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* 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.
* regtest: split video encode/decode tests into individual targetsMans Rullgard2011-11-22
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* Replace vendor string in Ogg and FLAC muxers.Diego Biurrun2011-11-02
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* remove the zork pcm seek testJustin Ruggles2011-10-26
| | | | this was forgotten when the encoder was removed
* 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>
* fate: remove seek-mpeg2reuse testMans Rullgard2011-09-28
| | | | | | The input file for this test is no longer generated. Signed-off-by: Mans Rullgard <mans@mansr.com>
* ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.Justin Ruggles2011-09-05
| | | | Update FATE references accordingly.
* ffmpeg: deprecate loop_input and loop_output optionsAnton Khirnov2011-07-08
| | | | They were replaced by (de)muxer private options.
* lavf/utils: fix ff_interleave_compare_dts corner case.Vitor Sessak2011-05-10
| | | | | | | | | | This should fix behavior introduced by commit 96573c0d7605672d69b42ae1dcf18764ce47c71a. Av_rescale_rnd() is not lossless so if two timestamps are equal after being rescaled they are not always actually identical. This patch use av_compare_ts() to get always a correct result. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* ac3enc: correct the flipped sign in the ac3_fixed encoderJustin Ruggles2011-04-26
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* lavf/utils.c: Order packets with identical PTS by stream index.Vitor Sessak2011-04-12
| | | | | | This allows for more reproducible results when using multi-threading. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* 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.
* Add apply_window_int16() to DSPContext with x86-optimized versions and use itJustin Ruggles2011-03-22
| | | | in the ac3_fixed encoder.
* asf: update seek test referenceMans Rullgard2011-03-17
| | | | | | | | | | | This updates the seek test reference to match de11ee9. Before this change, most of the seeks requested positions before the supposed start of the file (the preroll time), resulting in the first packet being returned. With the preroll subtracted, some of these seeks will land within the file and some beyond the end, thus returning a different set of packets. Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
* 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 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
* Update the test references for lavf-rm and seek-ac3_rm.Justin Ruggles2010-12-14
| | | | | | The references changed due to r25956. Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Add a FLAC parser.Michael Chinen2010-12-07
| | | | | | | | | Seek test reference updated because FLAC seeking now works properly. Fixes roundup issue 1150. Patch by Michael Chinen [mchinen at gmail] Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Avoid negative SCR in mpeg ps muxer.Michael Niedermayer2010-10-16
| | | | | | Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor) Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Update rv20 seek test referenceMåns Rullgård2010-09-26
| | | | Originally committed as revision 25204 to svn://svn.ffmpeg.org/ffmpeg/trunk
* regtest: rename seektest ref files using alphanumeric chars onlyMåns Rullgård2010-07-20
| | | | Originally committed as revision 24345 to svn://svn.ffmpeg.org/ffmpeg/trunk
* regtest: add seektest reference files for rgb/yuv in aviMåns Rullgård2010-06-06
| | | | Originally committed as revision 23506 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
* matroskaenc: Write codec time base as default duration for video tracks.David Conrad2010-05-24
| | | | | | | | This isn't exactly semantically equivalent, but the field has already been long abused to mean this, and writing it helps in determining a decent cfr time base when transcoding from a mkv where the video codec stores none (VP8). Originally committed as revision 23284 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
* In ogg muxer, pack multiple frames into one page, much lower overheadBaptiste Coudurier2010-05-21
| | | | Originally committed as revision 23231 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Change default for bidir_refine to 1.Michael Niedermayer2010-04-02
| | | | Originally committed as revision 22778 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
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