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* 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.
* 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.
* FATE: add a pattern rule for generating asynth wav filesJustin Ruggles2012-04-20
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* FATE: optionally write a WAVE header in audiogenJustin Ruggles2012-04-20
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* FATE: pass the decoded output format and audio source file to enc_dec_pcmJustin Ruggles2012-04-18
| | | | | | | This will allow decoding to md5 and doing a diff comparison to a reference checksum instead of a fuzzy stddev or oneoff comparison. Signed-off-by: Mans Rullgard <mans@mansr.com>
* FATE: specify the input format when decoding in enc_dec_pcm()Justin Ruggles2012-04-18
| | | | | | The output format is not always the same as the file extension, which is sometimes required for correct probing. We can avoid probing by specifying the format since it is already known.
* avconv: get output pixel format from lavfi.Anton Khirnov2012-04-15
| | | | | | | This way we don't require a clearly defined corresponding input stream. The result for the xwd test changes because rgb24 is now chosen instead of bgra.
* 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.
* fate: add BMP tests.Ronald S. Bultje2012-04-04
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* FATE: Add RALF decoding testDerek Buitenhuis2012-04-01
| | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* FATE: add a test for vp8 with changing frame size.Anton Khirnov2012-03-28
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* fate: add kgv1 fate test.Ronald S. Bultje2012-03-27
| | | | Tested to be bit-exact across x86-64, x86-32 and ppc.
* fate: add wmalossless test.Ronald S. Bultje2012-03-23
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* FATE: Add ZeroCodec testDerek Buitenhuis2012-03-22
| | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
* mpegaudioenc: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | Update FATE references due to encoder delay.
* 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.
* ac3enc: update to AVCodec.encode2()Justin Ruggles2012-03-20
| | | | Update FATE references due to encoder delay.
* FATE: use +/- flag option syntax for vp8 emu-edge testsJustin Ruggles2012-03-20
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* avconv: pass input stream timestamps to audio encodersJustin Ruggles2012-03-20
| | | | | 5 FATE test references updated due to using demuxer-generated timestamps that are either not sample-accurate or are slightly off in the input file.
* xa: fix timestamp calculationJustin Ruggles2012-03-20
| | | | The packet duration is always 28 samples.
* FATE: change fate-maxis-xa to a normal demuxing testPaul B Mahol2012-03-19
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* FATE: add test for adpcm-ea-maxis-xaPaul B Mahol2012-03-19
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* adpcm: fix nb_samples rounding for adpcm_ima_dk3, and update reference.Ronald S. Bultje2012-03-18
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* fate: make compare() function compatible with POSIX bcMans Rullgard2012-03-18
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* fate: fix enc_dec_pcm tests with remote targetMans Rullgard2012-03-17
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* FATE: remove WMA acodec testsJustin Ruggles2012-03-17
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* FATE: add WMAv1 and WMAv2 encode/decode tests with fuzzy comparisonJustin Ruggles2012-03-17
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* FATE: add AC-3 and E-AC-3 encode/decode tests with fuzzy comparisonJustin Ruggles2012-03-17
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* FATE: add capability for audio encode/decode tests with fuzzy psnr comparisonJustin Ruggles2012-03-15
| | | | | | This allows for testing floating-point audio encoders across different platforms where exact comparisons are unreliable due to float rounding differences.
* FATE: allow a tolerance in the size comparison in do_tiny_psnr()Justin Ruggles2012-03-15
| | | | | This will allow for comparing decoded output to the original source when the decoded size is not exactly the same as the original size.
* FATE: use absolute difference from a target value in do_tiny_psnr()Justin Ruggles2012-03-15
| | | | | This will allow comparison to original pre-encoded content instead of comparing to expected decoded output.
* FATE: allow tests to set CMP_SHIFT to pass to tiny_psnrJustin Ruggles2012-03-15
| | | | | This will allow adjusting for any encoder or decoder delay when doing comparisons.
* FATE: use $fuzz directly in do_tiny_psnr() instead of passing it aroundJustin Ruggles2012-03-15
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* Add ffvhuff encoding and decoding regression testPaul B Mahol2012-03-14
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* fate: add pam image regression testPaul B Mahol2012-03-14
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* FATE: add shorthand to wavpack testPaul B Mahol2012-03-12
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* FATE: add test for cdxl demuxerPaul B Mahol2012-03-12
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* FATE: add CPUFLAGS variable, mapping to -cpuflags avconv option.Anton Khirnov2012-03-06
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* lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()Justin Ruggles2012-03-05
| | | | | | | | | | Also, do not give AVCodecContext.frame_size priority for muxing. Updated 2 FATE references: dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified by -t 2 in the FATE test wmv8-drm-nodec - durations are not needed. previously they were estimated using the packet size and average bit rate.
* 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.
* lavf: deobfuscate read_frame_internal().Anton Khirnov2012-03-05
| | | | | | | | | | | | | | Split off packet parsing into a separate function. Parse full packets at once and store them in a queue, eliminating the need for tracking parsing state in AVStream. The horrible unreadable loop in read_frame_internal() now isn't weirdly ordered and doesn't contain evil gotos, so it should be much easier to understand. compute_pkt_fields() now invents slightly different timestamps for two raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't be more wrong (or right) than previous ones.
* fate: Add sunrast regression testDerek Buitenhuis2012-03-03
| | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* 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
* tiertexseq: set audio stream start time to 0Justin Ruggles2012-03-03
| | | | | Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
* 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.
* vqf: set packet durationJustin Ruggles2012-03-03
| | | | | | Fixes timestamp calculation. The FATE reference is updated because timestamp calculations are now more accurate. Previous timestamps were based on average bit rate.
* mpegaudio_parser: do not ignore information from the first parsed frameJustin Ruggles2012-03-03
| | | | Update some demuxing and seeking fate tests.
* 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>
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