| Commit message (Collapse) | Author | Age |
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r8444 in r8486
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
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Originally committed as revision 8486 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 8485 to svn://svn.ffmpeg.org/ffmpeg/trunk
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bring AC3 encoder output up to input volume level
patch by Bill O'Shaughnessy % bill P oshaughnessy A gmail.com %
+ reg tests update gruntwork by me
Original thread:
date: Nov 21, 2006 11:36PM
subject: [Ffmpeg-devel] Simpler Patch to bring AC3 encoder output up to input level
Originally committed as revision 8444 to svn://svn.ffmpeg.org/ffmpeg/trunk
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that and microshitty also doesnt seem to ever fragment them
Originally committed as revision 8265 to svn://svn.ffmpeg.org/ffmpeg/trunk
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====
Author: michael
Date: Mon Mar 5 03:41:49 2007
New Revision: 8240
Modified:
trunk/libavformat/asf-enc.c
Log:
create codec_comment_header which looks more like what M$ creates, sane or not ...
Originally committed as revision 8260 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7960 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7955 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7954 to svn://svn.ffmpeg.org/ffmpeg/trunk
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strange and have to be fixed
Originally committed as revision 7953 to svn://svn.ffmpeg.org/ffmpeg/trunk
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things have problems with our asfs, maybe that has a positive effect?
Originally committed as revision 7952 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7951 to svn://svn.ffmpeg.org/ffmpeg/trunk
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add a few asserts
Originally committed as revision 7903 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7879 to svn://svn.ffmpeg.org/ffmpeg/trunk
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(using .asf as our .ogg muxer depends on libogg, nut muxer depends on libnut and vorbis in avi/mpeg is not really a good idea)
Originally committed as revision 7874 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7865 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7519 to svn://svn.ffmpeg.org/ffmpeg/trunk
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this makes frames a few bytes smaller (0.1% for high bitrate but >1% for low bitrates)
Originally committed as revision 7401 to svn://svn.ffmpeg.org/ffmpeg/trunk
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in the onMetaData header. Patch by Allan Hsu <allan at counterpop dot net>.
Originally committed as revision 7382 to svn://svn.ffmpeg.org/ffmpeg/trunk
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slight PSNR/bitrate increase
Originally committed as revision 7361 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7349 to svn://svn.ffmpeg.org/ffmpeg/trunk
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some PSNR/bitrate improvement for adaptive quant
Originally committed as revision 7346 to svn://svn.ffmpeg.org/ffmpeg/trunk
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skiped (mpeg4 doesnt allow such MBs and in the past we did ME and encoding until at the end we droped them, so this should be faster though i didnt benchmark it, benchmark welcome)
Originally committed as revision 7343 to svn://svn.ffmpeg.org/ffmpeg/trunk
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some PSNR/bitrate gain if adaptive quant is used
initalize qscale_table correctly (it was pretty much random since the qp->lambda change)
this probably has not much effect as the table isnt used currently IIRC
Originally committed as revision 7342 to svn://svn.ffmpeg.org/ffmpeg/trunk
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some PSNR/bitrate gain for b frames with adaptive quant or QPRD
Originally committed as revision 7341 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7340 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7334 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7284 to svn://svn.ffmpeg.org/ffmpeg/trunk
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this has pretty much no quality or speed effect except very small random changes
Originally committed as revision 7202 to svn://svn.ffmpeg.org/ffmpeg/trunk
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+0.01 PSNR
idea from x264
Originally committed as revision 7189 to svn://svn.ffmpeg.org/ffmpeg/trunk
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patch by Bill O'Shaughnessy % bill P oshaughnessy A gmail.com %
+ reg tests update gruntwork by me
Original thread:
date: Nov 21, 2006 11:36 PM
subject: [Ffmpeg-devel] Simpler Patch to bring AC3 encoder output up to input level
Originally committed as revision 7160 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 7157 to svn://svn.ffmpeg.org/ffmpeg/trunk
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probably hasnt been noticed as everyone who tested stuff used -psnr)
Originally committed as revision 7151 to svn://svn.ffmpeg.org/ffmpeg/trunk
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[ Based on a patch by Brian Brice (bbrice at newtek dot com) ]
Originally committed as revision 6161 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 6072 to svn://svn.ffmpeg.org/ffmpeg/trunk
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the compiler should be able to optimize the shift away on 64bit archs)
Originally committed as revision 6045 to svn://svn.ffmpeg.org/ffmpeg/trunk
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umovimus at gmail dot com
Originally committed as revision 6022 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5982 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5943 to svn://svn.ffmpeg.org/ffmpeg/trunk
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>ffmpegdevel ad pylonsoft mot ch )
Originally committed as revision 5779 to svn://svn.ffmpeg.org/ffmpeg/trunk
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tabs removed and regression.sh fixed (it was missing in the patch) by me
Originally committed as revision 5572 to svn://svn.ffmpeg.org/ffmpeg/trunk
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1) search for optimal rice parameters and partition order. i also
modified the stereo method estimation to use this to calculate estimated
bit count instead of using just the pure sums.
2) search for the best fixed prediction order
3) constant subframe mode (good for encoding silence)
Note that the regression test for the decoded wav file also changed.
This is due to FFmpeg's FLAC decoder truncating the file, which it did
before anyway...just at a different cutoff point. The generated FLAC
files are still 100% lossless.
With this update, FFmpeg's FLAC encoder has speed and compression
somewhere between "flac -1" and "flac -2". On my machine, it's about
15% faster than "flac -2", and about 10% slower than "flac -1". The
encoding parameters are identical to "flac -2" (fixed predictors, 1152
blocksize, partition order 0 to 3).
Originally committed as revision 5536 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5528 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5515 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5476 to svn://svn.ffmpeg.org/ffmpeg/trunk
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iterative motion estimation.
Originally committed as revision 5427 to svn://svn.ffmpeg.org/ffmpeg/trunk
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(bitstream is not backwards compatible, even if refs aren't used)
Originally committed as revision 5417 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5405 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5372 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 5328 to svn://svn.ffmpeg.org/ffmpeg/trunk
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