4x faster than c (somehow, even though doubles only allow 2x simd).
overal flac encoding: 15-50% faster on core2, 4-11% on k8, 3-13% on p4.
Originally committed as revision 10621 to svn://svn.ffmpeg.org/ffmpeg/trunk
overall flac encoding: 4-15% faster.
output is not identical to the previous algorithm due to occasional rounding
errors, but the differece is less than .0005% bitrate.
Originally committed as revision 10612 to svn://svn.ffmpeg.org/ffmpeg/trunk
overall flac encoding: 15-50% faster on core2, 8-30% on k8, 2-20% on p4 (depending on compression_level)
Originally committed as revision 10606 to svn://svn.ffmpeg.org/ffmpeg/trunk
37%/45%/90% faster on core2/k8/p4, making flac encoding overall 15%/17%/40% faster at compression_level>=8 (less at low levels).
Originally committed as revision 10585 to svn://svn.ffmpeg.org/ffmpeg/trunk
1. Add a PUT_UTF8 macro to common.h; code borrowed from libavcodec/flacenc.c.
2. Make use of the macro in flacenc.c
Patch by Zuxy Meng % zuxy P meng A gmail P com %
Original thread:
Date: Nov 5, 2006 9:56 AM
Subject: [Ffmpeg-devel] PUT_UTF8 & asf format enhancement
Originally committed as revision 6911 to svn://svn.ffmpeg.org/ffmpeg/trunk
this will find the coefficients which minimize the sum of the squared errors,
levinson-durbin recursion OTOH is only strictly correct if the autocorrelation matrix is a
toeplitz matrix which it is only if the blocksize is infinite, this is also why applying
a window (like the welch winodw we currently use) improves the lpc coefficients generated
by levinson-durbin recursion ...
optionally (use_lpc>2) support iterative linear least abs() solver using cholesky
factorization with adjusted weights in each iteration
compression gain for both is small, and multiple passes are of course dead slow
Originally committed as revision 5747 to svn://svn.ffmpeg.org/ffmpeg/trunk
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
