93e91145b4
fix indentation
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Originally committed as revision 8499 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 23:23:51 +00:00
36376fa3ba
break if eob is reached to avoid reading one too much byte
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Originally committed as revision 8498 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 23:23:05 +00:00
ccba88d764
cosmectics, mov -> mp4
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Originally committed as revision 8497 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 15:24:36 +00:00
295b7dc367
sort
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Originally committed as revision 8496 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 15:17:30 +00:00
2b4162f0ae
add some codec object type ids
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Originally committed as revision 8495 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 15:16:03 +00:00
b41899c66c
prefer MP3 id as mentioned in avcodec.h line 210
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Originally committed as revision 8494 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 15:11:47 +00:00
7555d2e633
add specifications reference comments
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Originally committed as revision 8493 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 15:10:12 +00:00
e2f3d94786
Fix H.264 8x8 transform selection
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It seems that it's opt parse bug, it can't process the flags which start by
digit. After change 8x8dct to dct8x8, I can set it without problem. I guess
nobody will use it since it can't work as expected, so the quick fix is change
the option name.
Patch by Limin Wang lance dot lmwang at gmail com
Originally committed as revision 8492 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 12:07:07 +00:00
199c6e3cf1
md5 test for the image formats
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Originally committed as revision 8491 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 02:14:34 +00:00
1e358f039f
bmp and tga image regression test
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Originally committed as revision 8490 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 02:03:20 +00:00
1c87ed3a34
1000000l for me ive misstakely reverted the regression checksum change from r8444 in r8486
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how did i succeed doing such a ridiculously silly thing? well i think it happened like:
1. verifying that the regression tests pass with old resample2.c
2. updating the regressions to the new resample2.c ... failed svn complained
3. svn up
4. updating the regressions to the new resample2.c success (r8485)
at that point everything was still ok
5. some more resample2.c work update regressions, read diff, commit (r8486)
my misstake was that the svn up at point 3 was run in tests/ -> iam an idiot
Originally committed as revision 8489 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-24 01:48:40 +00:00
c9fdab0dab
Some more BeOS cleanup: check for arpa/inet.h; declare the prototype for inet_aton if not found; remove barpainet.h as it's not longer needed.
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Originally committed as revision 8488 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-23 22:01:37 +00:00
56cc42664a
iwmmxt ffmpeg code does not implement dct_unquantize_h263_inter_iwmmxt, but
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armv5t does - use the armv5t function rather than default c code.
patch by Tim Chick chick at computergeek freeserve co uk
Originally committed as revision 8487 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-23 00:05:45 +00:00
8ec04d344a
set more reasonable cutoff for audio rate conversion
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Originally committed as revision 8486 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 14:27:16 +00:00
4ba67eeed5
update resgression tests after the resampler changes
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Originally committed as revision 8485 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 13:52:24 +00:00
47bbf8a71d
clarify codec_tag
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Originally committed as revision 8484 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 13:27:11 +00:00
89cff671f4
clarify sub_id
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Originally committed as revision 8483 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 13:17:03 +00:00
7e068841cb
replace long double in the audiophile kiddy mode with doubles, its faster (and more portable) and the audiophile placebo 120db stopband attenuation isnt affected
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Originally committed as revision 8482 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 11:23:38 +00:00
d6fead9089
change high precision mode to 30 bits again after making the code less overflow sensitive
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Originally committed as revision 8481 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 11:13:13 +00:00
303bd95cd2
optimize linear filter coeff interpolation code, this also makes the code less prone to overflows
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Originally committed as revision 8480 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 11:10:30 +00:00
d971a5fa81
cosmetics. align and condense.
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Originally committed as revision 8479 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 05:39:20 +00:00
319708da02
simplify
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Originally committed as revision 8478 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 05:38:00 +00:00
bbd16dea67
utilize multi-stage AC-3 bit allocation. speeds up encoding by 25-30%
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Originally committed as revision 8477 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 05:34:26 +00:00
447caf7941
revert r8469 (increase bits for non default highprecision mode)
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it causes overflows with linear interpolated filters coefficients
Originally committed as revision 8476 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 03:18:34 +00:00
55fc0eb174
AUDIOPHILE_KIDDY_MODE
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Originally committed as revision 8475 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 03:16:02 +00:00
50df67d3b5
optimize bessel function instead of trusting gcc to do trivial optimizations (as gcc doesnt ...)
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Originally committed as revision 8474 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 00:52:49 +00:00
14f887efc6
select more sensible default windows (= attenuation beyond the dynamic range of your input is silly if it negatively affects other parameters)
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Originally committed as revision 8473 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-22 00:16:41 +00:00
20cf58c32b
make kaiser windows with other beta than 16 available
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Originally committed as revision 8472 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 23:44:49 +00:00
f88e7c833b
enable iwmmxt support, removing mm_support() stub
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patch by Tim Chick, chick at computergeek freeserve co uk
Originally committed as revision 8471 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 23:07:24 +00:00
c252b26d55
make high precision mode accessible at compile time
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Originally committed as revision 8470 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 22:39:00 +00:00
ca6940f871
increase bits for non default highprecision mode
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Originally committed as revision 8469 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 22:38:07 +00:00
ae232dd740
code for testing the accuracy of the resampler
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Originally committed as revision 8468 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 22:37:05 +00:00
66a148a19a
remove dithering of filter coefficients, improves precision by 1-2 bits and
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improves subjective sound quality on artificial sample (udial.wav resampling
to 32khz)
Originally committed as revision 8467 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 22:02:52 +00:00
8076609a40
The NSV demuxer assumes that a video frame's timestamp increases by one on each
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frame, but some low-bitrate NSV files omit video frames for some NSV frames,
and expect the timestamp to increase by one every NSV frame. This is noticeable
in 64vp3.nsv where the video runs several times faster than the audio. Fix this
by unconditionally incrementing the video's timestamp with each NSV frame.
patch by David Conrad, umovimus gmail com
Originally committed as revision 8466 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 11:05:35 +00:00
df7625ed62
av_estimate_timings_from_pts() flushes the packet queue but doesn't
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reset the streams' cur_dts values. This can lead to a fatal "error,
non monotone timestamps ..." message later, because the out-of-date
cur_dts values are used to compute some packet's dts.
Fix this by calling av_read_frame_flush() and eliminate code
duplication in the process.
The additional hunk gives more detailed error messages.
patch by Wolfram Gloger, wmglo dent.med.uni-muenchen de
Originally committed as revision 8465 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 10:48:10 +00:00
e14f79ed37
cosmetics, sample_size_v1 -> bytes_per_frame / samples_per_frame
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Originally committed as revision 8464 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 10:28:03 +00:00
638fd2fce7
cosmetics
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Originally committed as revision 8463 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 10:19:37 +00:00
2d2432b738
set audio frame size based on stts
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Originally committed as revision 8462 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 10:17:52 +00:00
4278662d45
treat frame_size > 1 as compressed audio
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Originally committed as revision 8461 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 10:03:27 +00:00
c72e05ba0f
Smacker Audio stereo fix
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Originally committed as revision 8460 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-21 05:56:29 +00:00
5d171758c0
update links to old probably obsolete bttv patches
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Originally committed as revision 8459 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 21:17:21 +00:00
afe28f7ce9
link to ffv1, msmpeg4, asv1, 4xm docs
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Originally committed as revision 8458 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 21:12:06 +00:00
04704a7881
libav* headers for pascal
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Originally committed as revision 8457 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 21:09:02 +00:00
058ee0cf95
Small simplifications to subband coefficient handling and use av_random().
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Patch by Ian Braithwaite ian .. braithwaite ... dk
Originally committed as revision 8456 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 18:08:51 +00:00
0ec4bc6a27
oops, encoder was under ifdef decoders.
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Originally committed as revision 8455 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 16:51:30 +00:00
028d6f3e71
revert r8428, the check was not that useless
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should fix constant pts=0 with broken demuxers
Originally committed as revision 8454 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 13:59:59 +00:00
1424a50772
add myself as AC-3 maintainer
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Originally committed as revision 8453 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 13:29:33 +00:00
a843d1ff32
i think this is more correct
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Originally committed as revision 8452 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 13:01:39 +00:00
b314cfe709
Correctly handle data_size on decoding
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Originally committed as revision 8451 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 05:44:42 +00:00
b5a05cc295
fix handling of special case for lowest snroffset. regressions are unaffected.
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Originally committed as revision 8450 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-03-20 02:12:13 +00:00
Baptiste Coudurier
Limin Wang
Michael Niedermayer
François Revol
Tim Chick
Justin Ruggles
David Conrad
Wolfram Gloger
Kostya Shishkov
Ian Braithwaite
Loren Merritt