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* newdev/master:
avio: make udp_set_remote_url/get_local_port internal.
asfdec: also subtract preroll when reading simple index object
matroskaenc: remove a variable that's unused after bc17bd9.
avio: cosmetics - nicer vertical alignment.
Remove unnecessary icc version checks
Disable 'attribute "foo" ignored' warnings from icc
rtsp: Don't use a locale dependent format string
Add xd55 codec tag for XDCAM HD422 720p25 CBR files.
configure: get libavcodec version from new version.h header
lavc: move the version macros to a new installed header.
matroskaenc: simplify get_aac_sample_rates by using ff_mpeg4audio_get_config
Do not use format string "%0.3f" for RTSP Range field.
Add apply_window_int16() to DSPContext with x86-optimized versions and use it in the ac3_fixed encoder.
Document usage of import libraries created by dlltool
configure: Set the correct lib target for arm/wince dlltool
fate: simplify regression-funcs.sh
fate: add support for multithread testing
Conflicts:
libavformat/rtspdec.c
libavutil/attributes.h
libavutil/internal.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* newdev/master: (33 commits)
Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
Add kbdwin.o to AC3 decoder
Detect byte-swapped AC-3 and support decoding it directly.
cosmetics: indentation
Always copy input data for AC3 decoder.
ac3enc: make sym_quant() branch-free
cosmetics: indentation
Add a CPU flag for the Atom processor.
id3v2: skip broken tags with invalid size
id3v2: don't explicitly skip padding
Make sure kbhit() is in conio.h
fate: update wmv8-drm reference
vc1: make P-frame deblock filter bit-exact.
configure: Add the -D parameter to the dlltool command
amr: Set the AVFMT_GENERIC_INDEX flag
amr: Set the pkt->pos field properly to the start of the packet
amr: Set the codec->bit_rate field based on the last packet
rtsp: Specify unicast for TCP interleaved streams, too
Set the correct target for mingw64 dlltool
applehttp: Change the variable for stream position in seconds into int64_t
...
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/ac3dec.c
libavformat/avio.h
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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in the ac3_fixed encoder.
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Factor ffmpeg calls using run_ffmpeg()
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Add a THREADS variable to fate calls.
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This updates the wmv8-drm reference after c47d383.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not Pulled:
commit 42cfb3835b5dad327b7dc22740e8b6e482ecfcd2
Author: Mans Rullgard <mans@mansr.com>
Date: Mon Feb 28 18:06:58 2011 +0000
Remove Sonic experimental audio codec
commit 2912e87a6c9264d556734e2bf94a99c64cf9b102
Author: Mans Rullgard <mans@mansr.com>
Date: Fri Mar 18 17:35:10 2011 +0000
Replace FFmpeg with Libav in licence headers
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This script was used to import Mike's old fate database and
is no longer needed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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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>
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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.
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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There is no need to expand to 16-bits. Just use memcpy() to copy the raw data.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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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>
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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>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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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>
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The output was changed by a7827a17c6b3388322350456d445c94b3a82cd25.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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The reserved bits between PCR base and extension fields must be
set to 1.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly. This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
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Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
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Originally committed as revision 26342 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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
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Originally committed as revision 26231 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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
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it should not be necessary anymore.
Originally committed as revision 26198 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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
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Originally committed as revision 26134 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 26119 to svn://svn.ffmpeg.org/ffmpeg/trunk
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code.
Originally committed as revision 26112 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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
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x86_64 boxes.
Originally committed as revision 26064 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 26063 to svn://svn.ffmpeg.org/ffmpeg/trunk
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due to VC-1 decoder overreads resulting in different output.
Originally committed as revision 26055 to svn://svn.ffmpeg.org/ffmpeg/trunk
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to complement the existing video-only decode test.
Originally committed as revision 26054 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Whether the behaviour for streams using scrambling makes sense
is unclear.
Originally committed as revision 26053 to svn://svn.ffmpeg.org/ffmpeg/trunk
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The references changed due to r25956.
Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
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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
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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
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This includes a revert of r25840
Originally committed as revision 25842 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 25840 to svn://svn.ffmpeg.org/ffmpeg/trunk
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