Commit Graph

79 Commits

Author SHA1 Message Date
Justin Ruggles
11dcddb97b ffm: do not write or read the audio sample format 2012-10-06 12:21:54 -04:00
Anton Khirnov
3b4bb19e63 lavf: flush the output AVIOContext in av_write_trailer().
This is consistent with stdio and is what we want to do in all cases.

Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
2012-09-15 18:25:07 +02:00
Alex Converse
41e9682af2 movenc: Write chan atom for all audio tracks in mov mode movies. 2012-06-04 10:08:31 -07:00
Mans Rullgard
11e33402ca fate: use standard diff options
diff -w is not a standard option.  This fixes the reference files
to match what the tests actually output and switches to using the
standard diff -b which is sufficient to handle different line ending
styles.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-15 19:47:18 +01:00
Justin Ruggles
c5671aeb77 FATE: avoid channel mixing in lavf-dv_fmt
This partially reverts acb1730218
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.
2012-04-24 15:55:45 -04:00
Justin Ruggles
acb1730218 FATE: allow lavf tests to alter input parameters
Change some lavf tests to avoid resampling and channel mixing.
2012-04-20 10:23:57 -04:00
Anton Khirnov
b7327887ea avconv: get output pixel format from lavfi.
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.
2012-04-15 20:22:36 +02:00
Justin Ruggles
b0f75ba272 mpegaudioenc: use AVCodec.encode2()
Update FATE references due to encoder delay.
2012-03-20 18:56:22 -04:00
Justin Ruggles
aa872af5e3 ac3enc: update to AVCodec.encode2()
Update FATE references due to encoder delay.
2012-03-20 18:46:56 -04:00
Paul B Mahol
05e0061ef6 fate: add pam image regression test
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
2012-03-14 15:34:50 +01:00
Derek Buitenhuis
6aa6e3e814 fate: Add sunrast regression test
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
2012-03-03 20:57:03 -05:00
Martin Storsjö
85b221e4d3 dpxenc: Don't include the libavcodec ident if bitexact mode is enabled
This avoids breaking fate every time the lavc version is bumped.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-29 20:08:09 +02:00
Anton Khirnov
63efd83ae1 mpegvideo_enc: add chroma/luma_elim_threshold private options.
Deprecate corresponding AVCodecContext fields.
2012-02-29 07:23:31 +01:00
Reimar Döffinger
a86ca94615 Enable already existing rso regression test.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
2012-02-21 14:42:30 -05:00
Reimar Döffinger
ecdb31caf2 Add regression test for "sox" format muxer/demuxer.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
2012-02-21 14:42:30 -05:00
Carl Eugen Hoyos
8ee2ddcb2a Add dpx encoding regression test.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
2012-02-21 14:42:30 -05:00
Anton Khirnov
cd1ad18a65 rawenc: switch to encode2().
This changes a number of FATE results, since before this commit, the
timestamps in all tests using rawenc were made up by lavf.

In most cases, the previous timestamps were completely bogus.

In some other cases -- raw formats, mostly h264 -- the new timestamps
are bogus as well. The only difference is that timestamps invented by
the muxer are replaced by timestamps invented by the demuxer.

cscd     -- avconv sets output codec timebase from r_frame_rate
and r_frame_rate is in this case some guessed number 31.42 (377/12),
which is not accurate enough to represent all timestamps. This results
in some frames having duplicate pts. Therefore, vsync 0 needs to be
changed to vsync 2 and avconv drops two frames. A proper fix in the
future would be to set output timebase to something saner in avconv.

nuv      -- previous timestamps for video were wrong AND the cscd
comment applies, one frame is dropped.

vp8-signbias -- the file contains two frames with identical timestamps,
so -vsync 0 needs to be removed/changed to -vsync 2 and avconv drops one
frame.

vc1-ism -- apparrently either the demuxer lies about timestamps or the
file is broken, since dts == pts on all packets, but reordering clearly
takes place.
2012-02-08 21:51:24 +01:00
Anton Khirnov
1270e12e49 avconv: rework -t handling for encoding.
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.
2012-02-07 20:11:11 +01:00
Anton Khirnov
7063b6eaee lavc: increase major version to 54.
The lavf-ffm test results change because ffmenc writes
AVCodecContext.flags/flags2 and the defaults for those change.
2012-01-27 10:38:30 +01:00
Paul B Mahol
7de9af65c7 fate: add XWD image regression test
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2012-01-26 01:51:26 +01:00
Martin Storsjö
5c7c9a9f33 fate: Update file checksums after the mov muxer change in a78dbada55
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-10 16:54:23 +02:00
Justin Ruggles
8e8c51318c movenc: simplify handling of pcm vs. adpcm vs. other compressed codecs
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.
2011-12-09 16:12:58 -05:00
Anton Khirnov
81ac4cda0b fate-lavf-ts: use -mpegts_transport_stream_id option.
Serves as a test of muxer private options.
2011-11-18 11:01:46 +01:00
Diego Biurrun
c6cd0e17f3 Replace vendor string in Ogg and FLAC muxers. 2011-11-02 10:43:39 +01:00
John Brooks
2c4e08d893 riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
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>
2011-10-14 13:28:58 +02:00
Anton Khirnov
7574cacbd5 movenc: create an alternate group for each media type
Partially fixes bug 44.
2011-09-17 08:42:30 +02:00
Anton Khirnov
9c684feadc libx264: add 'direct-pred' private option
Deprecate AVCodecContext.directpred
2011-09-07 07:27:55 +02:00
Anton Khirnov
0635a8aa21 libx264: add 'partitions' private option
Deprecate AVCodecContext.partitions.
2011-09-07 07:27:18 +02:00
Mans Rullgard
0218808d49 fate: separate lavf-mxf_d10 test from lavf-mxf
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-07-17 16:30:49 +01:00
Jindrich Makovicka
575c38d76c mpegtsenc: set Random Access indicator on keyframe start packets
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2011-07-13 20:49:26 +02:00
Anton Khirnov
f5302e5dcf ffmpeg: deprecate loop_input and loop_output options
They were replaced by (de)muxer private options.
2011-07-08 19:58:19 +02:00
Anton Khirnov
5e8d2e337e lavf: deprecate AVStream.quality.
AVStream is no place for it and it's unused outside of ffmpeg anyway.
2011-07-06 20:10:41 +02:00
Michael Niedermayer
0af8a71d66 swscale: fix JPEG-range YUV scaling artifacts.
YUV planes were marked as uint16_t, but they contained signed data.
Fixes issue 1108 and 675.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-06-14 09:46:49 -04:00
Baptiste Coudurier
7e19a6e868 movenc: always write esds descriptor length using 4 bytes.
ipod shuffle doesn't support anything else.

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2011-05-13 07:38:54 +02:00
Vitor Sessak
ecc297308f lavf/utils: fix ff_interleave_compare_dts corner case.
This should fix behavior introduced by commit
96573c0d76. 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>
2011-05-10 07:53:19 -04:00
Anton Khirnov
f8fec05052 mpegtsenc: make PMT PID really start on pmt_start_pid 2011-04-28 07:26:40 +02:00
Justin Ruggles
79ee8977c2 ac3enc: correct the flipped sign in the ac3_fixed encoder 2011-04-26 17:19:37 -04:00
Diego Biurrun
fd0c3403f6 Update regtest checksums after revision 6001dad.
The string "FFmpeg" was replaced by "Libav" in metadata that
got encoded in file headers.
2011-04-17 22:46:42 +02:00
Vitor Sessak
96573c0d76 lavf/utils.c: Order packets with identical PTS by stream index.
This allows for more reproducible results when using multi-threading.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-04-12 19:06:26 -04:00
Justin Ruggles
e05a3ac713 ac3enc: select bandwidth based on bit rate, sample rate, and number of
full-bandwidth channels.

This reduces high-frequency artifacts and improves the quality of the lower
frequency audio at low bit rates.
2011-04-03 20:59:14 -04:00
Mans Rullgard
79997def65 ac3enc: use generic fixed-point mdct
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>
2011-04-03 19:01:53 +01:00
Justin Ruggles
e6e9823488 Add apply_window_int16() to DSPContext with x86-optimized versions and use it
in the ac3_fixed encoder.
2011-03-22 21:08:30 -04:00
Justin
323e6fead0 ac3enc: do not right-shift fixed-point coefficients in the final MDCT stage.
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.
2011-03-14 08:45:26 -04:00
Justin Ruggles
50d7140441 ac3enc: change default floor code to 7.
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>
2011-02-15 21:40:42 +00:00
Justin Ruggles
c3beafa0f1 ac3enc: Change EXP_DIFF_THRESHOLD to 500.
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>
2011-02-02 20:00:43 +00:00
Mans Rullgard
79dca23dc2 Update mpegts test reference
The output was changed by a7827a17c6.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-28 17:02:54 +00:00
Georgi Chorbadzhiyski
535638b55f mpegtsenc: set reserved bits to 1 in PCR field
The reserved bits between PCR base and extension fields must be
set to 1.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-26 00:02:42 +00:00
Justin Ruggles
a4f5af13fb Add regression test for stereo s16le in voc.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-19 12:51:42 +00:00
Justin Ruggles
ec44dd5fc2 Change the default dB-per-bit code from 2 to 3.
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
2010-12-29 19:17:22 +00:00
Justin Ruggles
295ab2af6e Change FIX15() back to clipping to -32767..32767.
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
2010-12-21 21:18:58 +00:00