Commit Graph

34 Commits

Author SHA1 Message Date
Michael Niedermayer
8593b743a8 rematrix: dont use floats for int16 code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 09:25:50 +02:00
Michael Niedermayer
b5875b9111 Add libswresample.
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 07:04:17 +02:00
Michael Niedermayer
9a9ceb8776 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  lavfi: add select filter
  oggdec: fix out of bound write in the ogg demuxer
  movenc: create an alternate group for each media type
  lavd: add libcdio-paranoia input device for audio CD grabbing
  rawdec: refactor private option for raw video demuxers
  pcmdec: use unique classes for all pcm demuxers.
  rawdec: g722 is always 1 channel/16kHz

Conflicts:
	Changelog
	configure
	doc/filters.texi
	libavdevice/avdevice.h
	libavfilter/avfilter.h
	libavfilter/vf_select.c
	tests/ref/lavf/mov

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-17 22:36:43 +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
Michael Niedermayer
3c54e7ed4f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
  ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
  ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
  wavpack: fix wrong return value in wavpack_decode_block()
  avconv: fix parsing metadata specifiers.
  fate: use +frame+slice named constants instead of '3'
  mpeg12: propagate more real return values through chunk decode error return and fix some indentation
  wavpack: use context reset in appropriate places
  avconv: move mux_preload and mux_max_delay to options context
  avconv: move bitstream filters to options context.
  avconv: move rate_emu to options context.
  avconv: move max_frames to options context.
  avconv: move metadata to options context.
  avconv: move ts scale to options context.
  avconv: move chapter maps to options context.
  avconv: move metadata maps to options context.
  avconv: move codec_names to options context.

Conflicts:
	avconv.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-05 22:29:16 +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
Michael Niedermayer
75a37b57a5 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  APIchanges: fill in date and commit for request_sample_fmt
  Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
  Add support for request_sample_format in ffmpeg and ffplay.
  Add APIchanges entry for request_sample_fmt.
  Add request_sample_fmt field to AVCodecContext.
  Add float_interleave() to FmtConvertContext with x86-optimized versions.
  Remove unused make variable SEEK_REFFILE
  fate: remove redundant aref and vref references
  fate: remove do_ffmpeg_nocheck function
  fate: do not collect -benchmark output
  mpegaudiodec: remove decode_end() function
  fate: run aref and vref as regular tests
  mpegaudio: sanitise compute_antialias_* names
  mpeg12: add slice-threading checks to slice-threading initializers.
  h264: copy pixel_shift between slice threading contexts.
  mdec: enable frame-level multithreading.
  mdec.c: fix overread.

Conflicts:
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/dca.c
	libavcodec/h264.c
	libavcodec/mdec.c
	libavcodec/mpeg12.c
	libavcodec/options.c
	libavcodec/version.h
	libavcodec/vorbisdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-19 06:00:31 +02:00
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
Baptiste Coudurier
35d3d44a84 adpcmenc: fix QT IMA ADPCM encoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-08 13:15:48 +02:00
Baptiste Coudurier
b3d5a4b06e adpcmdec: Fix QT IMA ADPCM decoder
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-08 13:13:17 +02:00
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
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
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
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
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
6fd96d1a85 Change the AC-3 encoder to use floating-point.
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.

Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-04 11:53:44 +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
918cd2255c Simplify fix15().
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.

Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 14:51:02 +00:00
Justin Ruggles
c7d89948a3 Set a constant frame size for encoding G.726 audio.
Originally committed as revision 25107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-11 19:52:09 +00:00
Måns Rullgård
c43d77c163 tiny_psnr: skip wav headers on input files
The byte count printed excludes the header, and offsets are applied
after the the headers are skipped.

Reference files updated to reflect new output.  Some stddev/psnr values
have changed slightly due to headers no longer being compared.

Originally committed as revision 24143 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-09 16:06:05 +00:00
Vitor Sessak
cb0067ec25 tiny_psnr: print max absolute difference between files
Regression test reference updates are due to the extra output
from tiny_psnr.

Patch by Vitor Sessak

Originally committed as revision 24132 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-09 00:40:37 +00:00
David Conrad
e7ddafd515 matroskaenc: Don't write a second seekhead for the clusters; mkvalidate agrees
with me that it's unnecessary.

Originally committed as revision 23478 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-04 22:41:11 +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
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
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
David Conrad
30f06a58a0 Simplify starting and ending clusters
Originally committed as revision 22199 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-04 08:53:08 +00:00
Måns Rullgård
cc3e2472f3 Place regression test output files in subdirs per family
Originally committed as revision 22155 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-02 21:41:52 +00:00
Måns Rullgård
c676895fd9 Separate audio-only tests so they are only run once
Originally committed as revision 21556 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-01-30 21:47:13 +00:00