1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-18 03:19:31 +02:00
Commit Graph

57 Commits

Author SHA1 Message Date
Mans Rullgard
7263cd5544 fate: convert codec-regression.sh to makefile rules
Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-29 08:35:41 +01:00
Mans Rullgard
7d7b40f48a pcmenc: set correct bitrate value
This fixes a bogus bitrate value in the header of WAV files with
alaw/ulaw audio.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-17 02:34:57 +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
Dale Curtis
8336eb6f85 matroska: Add incremental parsing of clusters.
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>
2012-04-22 17:23:50 -07:00
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
Justin Ruggles
5052980400 FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.
2012-04-20 10:23:57 -04:00
Justin Ruggles
03caef1bed FATE: replace the acodec-g726 test with 4 new encode/decode tests
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.
2012-04-20 10:23:57 -04:00
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
Justin Ruggles
85cf49fab7 FATE: remove WMA acodec tests 2012-03-17 11:46:15 -04:00
Justin Ruggles
8d1a20aa7c aiffdec: do not set AVCodecContext.frame_size
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.
2012-03-05 13:08:17 -05:00
Justin Ruggles
0883109b27 voc/avs: Do not change the sample rate mid-stream.
Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.
2012-03-03 17:03:27 -05:00
Justin Ruggles
0b8b7db01b mpegaudio_parser: do not ignore information from the first parsed frame
Update some demuxing and seeking fate tests.
2012-03-03 17:03:26 -05:00
Anton Khirnov
87d7a92b62 rawdec: set timebase to 1/fps. 2012-02-26 07:30:21 +01:00
Justin Ruggles
b498867d66 FATE: update reference for seek-alac_mp4
This should have been updated in b590f3a7bf.
2012-02-11 16:41:01 -05:00
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
Mans Rullgard
2c98f407c8 fate: make acodec-ac3_fixed test output raw AC3
There is no point in this test using the RM format.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-02-02 14:31:54 +00:00
Janne Grunau
f028d4d1c3 mxfdec: hybrid demuxing/seeking solution
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.
2012-01-22 14:40:53 +01:00
Luca Barbato
5a2e251645 fate: update asf seektest 2011-12-02 16:43:05 +01:00
Justin Ruggles
ca12401376 fate: split acodec-pcm into individual tests
this removes 2 redundant tests for pcm in mkv.
we can add the coverage back in later as fate-lavf tests if needed.
2011-12-01 13:27:56 -05:00
Mans Rullgard
3fe5fc9325 regtest: split video encode/decode tests into individual targets
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-11-22 12:13:04 +00:00
Diego Biurrun
c6cd0e17f3 Replace vendor string in Ogg and FLAC muxers. 2011-11-02 10:43:39 +01:00
Justin Ruggles
82ed4f1ed8 remove the zork pcm seek test
this was forgotten when the encoder was removed
2011-10-26 18:48:02 -04:00
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
Mans Rullgard
bc3a741fa0 fate: remove seek-mpeg2reuse test
The input file for this test is no longer generated.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-09-28 02:30:30 +01:00
Justin Ruggles
ae264bb29b ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
Update FATE references accordingly.
2011-09-05 10:09:44 -04:00
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
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
Justin Ruggles
79ee8977c2 ac3enc: correct the flipped sign in the ac3_fixed encoder 2011-04-26 17:19:37 -04:00
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
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
Mans Rullgard
487fef2dcc asf: update seek test reference
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>
2011-03-17 19:51:28 +00:00
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
5b54d4b376 ac3enc: fix bug in stereo rematrixing decision.
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>
2011-02-16 23:39:57 +00:00
Justin Ruggles
dc7e07ac1f Add stereo rematrixing support to the AC-3 encoders.
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
2011-01-08 23:21:17 +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
Justin Ruggles
8c634b707b Update the test references for lavf-rm and seek-ac3_rm.
The references changed due to r25956.

Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 16:14:52 +00:00
Michael Chinen
475ae04a27 Add a FLAC parser.
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
2010-12-07 14:50:50 +00:00
Michael Niedermayer
94bdb1f80c Avoid negative SCR in mpeg ps muxer.
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
2010-10-16 21:31:31 +00:00
Måns Rullgård
8c067b5dfd Update rv20 seek test reference
Originally committed as revision 25204 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-26 09:02:51 +00:00
Måns Rullgård
f729c4aea8 regtest: rename seektest ref files using alphanumeric chars only
Originally committed as revision 24345 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-20 10:09:54 +00:00
Måns Rullgård
48ede394d5 regtest: add seektest reference files for rgb/yuv in avi
Originally committed as revision 23506 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-06 16:50:18 +00:00
James Zern
ac9baa716b matroskaenc: Mux clusters better
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
2010-06-04 22:40:50 +00:00
David Conrad
7bb83d8ddf matroskaenc: Write codec time base as default duration for video tracks.
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
2010-05-24 08:58:19 +00:00
David Conrad
85e86b6810 Update regression tests after removing track timecode scale from mkvenc
Originally committed as revision 23248 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-05-22 02:13:01 +00:00
Baptiste Coudurier
95ca3b1e20 In ogg muxer, pack multiple frames into one page, much lower overhead
Originally committed as revision 23231 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-05-21 21:38:54 +00:00
Michael Niedermayer
008593be52 Change default for bidir_refine to 1.
Originally committed as revision 22778 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-02 14:08:51 +00:00
James Darnley
66061a1220 Add VorbisComment writing to FLAC files.
Patch by James Darnley <james darnley at gmail>.

Originally committed as revision 22605 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-20 13:36:43 +00:00