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Commit Graph

71 Commits

Author SHA1 Message Date
Michael Niedermayer
f27d5bd3d2 tests/fate: Add S302M test
Reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-02 11:09:08 +01:00
Paul B Mahol
8bb489fab5 fate: add wavpack encoder
Signed-off-by: Paul B Mahol <onemda@gmail.com>
2015-02-13 09:22:18 +00:00
Michael Niedermayer
0b5adc3520 avcodec/adxenc: fix rounding
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-11-30 12:56:02 +01:00
Michael Niedermayer
6b0ab561d0 avcodec/adxenc: match prediction used in the decoder
The prediction used in the encoder was not correct

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-11-30 12:52:56 +01:00
Martin Storsjö
fa8f060b75 adpcm: Write the proper predictor in trellis mode in IMA QT
The actual predictor value, set by the trellis code, never
was written back into the variable that was written into
the block header. This was accidentally removed in b304244b.

This significantly improves the audio quality of the trellis
case, which was plain broken since b304244b.

Encoding IMA QT with trellis still actually gives a slightly
worse quality than without trellis, since the trellis encoder
doesn't use the exact same way of rounding as in
adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble.

Fixes part of Ticket3701

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-06 17:08:21 +02:00
Michael Niedermayer
fb7646d92c fate: enable adpcm-ima_qt-trellis
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-06 17:08:21 +02:00
Timothy Gu
da53de0730 tests: add adpcm trellis tests
adpcm_ima_qt does not produce reproducible results, so it is temporarily
disabled (see #3701).

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-05 12:20:49 +02:00
Michael Niedermayer
0c152fe916 ffmpeg: prefix encoder with "Lavc " in bitexact mode
This avoids misleading encoder names like "encoder = prores"

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-05-18 22:57:20 +02:00
Michael Niedermayer
1e49439f04 Merge commit '6656370b858329ca07a60a2de954d5e90daa0206'
* commit '6656370b858329ca07a60a2de954d5e90daa0206':
  avconv: set the "encoder" tag when transcoding

Conflicts:
	ffmpeg.c
	tests/ref/lavf/mkv
	tests/ref/seek/lavf-mkv

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2014-05-18 21:31:51 +02:00
Daniel Verkamp
5e7d21c7ad ff_put_wav_header: add flag to force WAVEFORMATEX
Partially undoes commit 2c4e08d893:

    riff: always generate a proper WAVEFORMATEX structure in
    ff_put_wav_header

A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the
use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs.

This flag is used in the Matroska muxer (the cause of the original
change) and in the ASF muxer, because the specifications for
these formats indicate explicitly that WAVEFORMATEX should be used.

Muxers for other formats will return to the original behavior of writing
PCMWAVEFORMAT when writing a header for raw PCM.

In particular, this causes raw PCM in WAV to generate the canonical
44-byte header expected by some tools.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-04-30 16:41:35 +02:00
Michael Niedermayer
37f3f32d51 fate: force 128kb/sec for mp2 test
This fixes rounding differences between platforms

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-04-15 16:43:38 +02:00
Michael Niedermayer
268b1eae22 avcodec/mpegaudioenc_template: default to 384k bitrate as default
If 384k is too high for the samplerate, choose the closest
possible

Idea to increase the bitrate from: 46439e1562
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-04-11 01:13:16 +02:00
Michael Niedermayer
04e06cdf7d avcodec: split mp2 encoder into float and fixed
This makes the USE_FLOATS == 0 available to the end user
More float optimizations can easily be added as well now
common code should be factored out into a common file once all
fixed point & floating point optimizations are done, this is to
avoid having to move code back and forth between files.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-03 21:12:00 +01:00
Michael Niedermayer
800ea20cad Merge remote-tracking branch 'qatar/master'
* qatar/master:
  movenc: Make tkhd "enabled" flag QuickTime compatible

Conflicts:
	libavformat/movenc.c
	tests/ref/acodec/alac
	tests/ref/acodec/pcm-s16be
	tests/ref/acodec/pcm-s24be
	tests/ref/acodec/pcm-s32be
	tests/ref/acodec/pcm-s8
	tests/ref/lavf/mov
	tests/ref/vsynth/vsynth1-dnxhd-1080i
	tests/ref/vsynth/vsynth1-mpeg4
	tests/ref/vsynth/vsynth1-prores
	tests/ref/vsynth/vsynth1-qtrle
	tests/ref/vsynth/vsynth1-svq1
	tests/ref/vsynth/vsynth2-dnxhd-1080i
	tests/ref/vsynth/vsynth2-mpeg4
	tests/ref/vsynth/vsynth2-prores
	tests/ref/vsynth/vsynth2-qtrle
	tests/ref/vsynth/vsynth2-svq1

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-23 13:49:24 +02:00
John Stebbins
30ce289074 movenc: Make tkhd "enabled" flag QuickTime compatible
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>
2013-08-23 09:55:42 +02:00
Alexander Strasser
ac25b31ede lswr: Improve default resampler's default parameters
After making some blind tests on a small collection of music
samples for home usage. It turned out that the default cutoff
was too low.

The impact of filter_size was not clearly distinguishable (the
results were on the edge) with the music samples but turned out
to be clearly audible in some synthetic samples.

Thanks to Daniel for helping out with the listening tests.

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-01-04 16:47:57 +01:00
Piotr Bandurski
52f2176366 aiffenc: set correct number of bits foru8 in aiff
with this change QuickTime is able to play u8 aiff file generated by FFmpeg

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-20 16:05:30 +01:00
Michael Niedermayer
7711f19eda Merge commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6'
* commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6':
  fate-seek: remove use of gnu make 3.82 only private modifier
  fate: move vsynth reference files to their own directory
  fate: move fate-acodec reference files to their own dir
  configure: avplay now depends on avresample
  fate: split dependencies for fate-seek tests

Conflicts:
	configure
	tests/fate/seek.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-03 02:33:27 +01:00
Janne Grunau
337dbe2adb fate: move fate-acodec reference files to their own dir 2012-12-03 00:29:35 +01:00
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
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
a6c8cca2a8 FATE: replace current g722 encoding tests with an encode/decode test
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
d3c59d5003 avconv: use default channel layouts when they are unknown
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.
2012-04-10 11:30:01 -04:00
Justin Ruggles
bb03b6f7b1 g722enc: use AVCodec.encode2()
FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.
2012-03-20 18:47:23 -04:00
Justin Ruggles
85cf49fab7 FATE: remove WMA acodec tests 2012-03-17 11:46:15 -04:00
Justin Ruggles
51ddf35c90 wmaenc: fix m/s stereo encoding for the first frame
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
2012-03-03 18:20:10 -05:00
Martin Storsjö
b087ce2bee g722: Fix the QMF scaling
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>
2012-03-02 18:58:19 +02:00
Justin Ruggles
770a5c6d02 adpcmenc: Use correct frame_size for Yamaha ADPCM.
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
2012-02-20 15:52:32 -05:00
Justin Ruggles
b590f3a7bf alacenc: only encode frame size in header for a final smaller frame
Otherwise it is not needed because it matches the frame size as encoded in
the extradata.
2012-02-11 12:49:22 -05:00
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
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
77c5b66cbe g722enc: set frame_size, and also handle an odd number of input samples
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.
2012-01-07 13:38:23 -05:00
Justin Ruggles
3e57573fce fate: add ADX encoding/decoding test 2012-01-03 18:47:42 -05:00
Alex Converse
d3b8bde2f1 movenc: Rudimentary IODs support. 2011-12-15 14:06:13 -08:00
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
Martin Storsjö
714cd7e758 g722: Add a regression test for muxing/demuxing in wav
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-12-05 12:41:46 +02:00
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
Diego Biurrun
c6cd0e17f3 Replace vendor string in Ogg and FLAC muxers. 2011-11-02 10:43:39 +01:00
Justin Ruggles
85579b6381 avcodec: remove the Zork PCM encoder.
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.
2011-10-26 12:01:07 -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
Baptiste Coudurier
b304244b54 adpcmenc: fix QT IMA ADPCM encoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-23 20:54:29 -04:00
Baptiste Coudurier
bf334535b4 adpcmdec: Fix QT IMA ADPCM decoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-23 20:54:28 -04:00
Anton Khirnov
7574cacbd5 movenc: create an alternate group for each media type
Partially fixes bug 44.
2011-09-17 08:42:30 +02:00
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
Mans Rullgard
70378ea190 fate: run aref and vref as regular tests
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>
2011-05-18 14:45:46 +01:00
Justin Ruggles
79ee8977c2 ac3enc: correct the flipped sign in the ac3_fixed encoder 2011-04-26 17:19:37 -04:00
Anton Khirnov
9181976348 matroskaenc: don't write an empty Cues element. 2011-04-07 18:11:24 +02:00
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