similar to: 1.21 vs 1.3 encoding speed

Please cc: the results from dev list back here though I would run some tests on a diff platform but don't have access to my PC for a few weeks Would also be interesting to see what decoding stats you get > On Jul 25, 2014, at 7:38 AM, Scott Brown <scottcbrown at gmail.com> wrote: > > I will post to the dev list, sorry about that. > > To give an idea, though, at flac

I will post to the dev list, sorry about that. To give an idea, though, at flac level 8: 24/96 wav file 1.21: 61.05 seconds. ratio=0.690 1.3: 222.48 seconds. ratio=0.690 smaller 16/44.1 wav file 1.21: 14.28 seconds. ratio=0.487 1.3: 51.21 seconds. ratio=0.487 Scott On Fri, Jul 25, 2014 at 2:39 AM, <dathead2 at gmail.com> wrote: > Are you sure they didn't change the default

Why are you using such an old SDK? Your Macbook came with at least Mountain Lion, right? I have a Retina Macbook mid 2012, and it came with ML... On Fri, Jul 25, 2014 at 3:00 PM, <flac-dev-request at xiph.org> wrote: > Send flac-dev mailing list submissions to > flac-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

Are you sure they didn't change the default encoding level ? I would include some example timings to give a better idea of what "significantly slower" is Wonder if you see same for 44/16 files? > On Jul 25, 2014, at 2:29 AM, Martijn van Beurden <mvanb1 at gmail.com> wrote: > > Hi, > > You might want to report this to the flac-dev mailinglist instead of the

a. Intel 2.8 Ghz Core I7 (dual core, I7-4558U) in late 2013 Macbook Pro with Retina Display b. I compiled it the same way I compiled 1.2.1: ./configure -enable-static -disable-shared CFLAGS=" -isysroot /Developer/SDKs/MacOSX10.6.sdk -mmacosx-version-min=10.6" make Am i doing something wrong? Thanks, Scott On Fri, Jul 25, 2014 at 12:54 PM, lvqcl <lvqcl.mail at gmail.com>

Hello, I'm on a Mac and I'm noticing that encoding via the flac command line is significantly slower with version 1.3.0 than 1.2.1. I'm encoding a 24/96 file to flac, both from wav and aiff and both formats are showing the same speed decrease when using 1.3.0. Also, 1.2.1 will not encode an aiff-c file, but 1.3.0 will. Is this change documented anywhere? Thanks, Scott --------------

Does anybody plan to post binary rem for SL? Id''love to rely on its availability in the installation script... As wx included in installation package makes it huge... Sergey Chernov real.sergeych@gmail.com

The patch I submitted only reads AIFF files. I'm about to start the patch to write AIFF files. To do so, we need a command-line option to specify AIFF. My inclination is to add an option: -ff { raw | wav | aif } In some sense, "-ff" is silly since it probably stands for "format format". Still, I think it's better than just "-f", since the first

Is anyone working on making oggenc use some nice library that reads lots of different audio file formats so that people can encode from files other than WAV? For example, the 'AudioFile' library (just found it via a search on freshmeat) reads wav, aiff, aiff-c, .au, and .snd. Maybe there are other better libraries (OpenAL, maybe), but this library does what I want (read AIFF). -tim ---

I know this is probably blindingly obvious, but I can't find it anywhere. Is there a way to tell the Vorbis codec what endian your samples are or do I have to do my own byte reversal. AIFF is Be's 'standard' audio file format, and its encoded big endian, not little endian even on Intel boxes. I know for decoding ov_read has an endian option, but I can't see an equaivalent for

Thought this might be useful to at least consider being included in 1.0.1. It adds, among other things (maybe), an endian variant of AIFF-C. Nathan -------------- next part -------------- diff -urN vorbis-tools-1.0.orig/oggenc/audio.c vorbis-tools-1.0/oggenc/audio.c --- vorbis-tools-1.0.orig/oggenc/audio.c Thu Jul 11 16:20:33 2002 +++ vorbis-tools-1.0/oggenc/audio.c Mon Jan 20 09:42:57 2003 @@

flac.exe is choking on some aiff files as input with the following: "ERROR: SSND chunk size inconsistent with sample frame count" According to Daren on the dBpoweramp forum "The flac.exe program ignores the amount of data bytes quoted in the SSND chunk and tries to read to the end of the file, which would be fine if the SSND data was the last data in the file. The problem is

John Miles wrote: > You'll definitely have to encode in little-endian form. PCM audio is always > little-endian, to the best of my knowledge. I have never seen any > big-endian audio data files. Try AIFF, AU and IFF just for starters. They are definietly not the only ones. Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo

just recently I have finished implementing support for saving non-audio chunks in RIFF WAVE and AIFF with the FLAC file in application metadata blocks, and restoring them when decoding. you can read about more about it here: http://www.hydrogenaudio.org/forums/index.php?showtopic=56968 the zip file there has windows binaries; CVS HEAD has everything checked in if you need a different platform.

Hello, in fact : this error(FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DAT) occurs when the sample size is bigger than the exepected size in the encoder. Now, can someone tell me how to model the signal, for example : *using a Fixed linear predictor for the input signal ? * 2012/4/13 Brian Willoughby <brianw at sounds.wa.com> > > On Apr 12, 2012, at 07:12, Rafael Velasquez

I built ogg vorbis from the rc1 cvs source on Irix 6.5.12 with gcc 2.95.2. Using oggenc I encoded about 8,000 aiff files but found about a dozen where oggenc would go into an infinite loop. I tracked the problem with Laguerre_With_Deflation() as far back as logmask being Inf in floor0_forward. I'm now building gcc 3.0 with the expectation this is a compiler issue. If not, I'll back

Josh, Automake 1.10 on OSX is giving me warnings like the following: test/Makefile.am:50: wildcard *.raw: non-POSIX variable name test/Makefile.am:50: (probably a GNU make extension) test/Makefile.am:50: wildcard *.flac: non-POSIX variable name test/Makefile.am:50: (probably a GNU make extension) test/Makefile.am:50: wildcard *.oga: non-POSIX variable name

I can now load and play AIFF, WAV, and OGG files using libsndfile and libao. Now I'm moving on to loading and playing files embedded in an IFF file. This is presenting a problem. Playing an AIFF embedded in an IFF works fine. Playing an OGG embedded in the same IFF fails. Specifically, it appears that sf_open_fd() can't figure out what it's being told to load. Except for

Would somebody please help me break this row: "Main Group\t1000\tMP Test\tMP Test, 1\tAudio (1, f1-qaddara.aiff)\tl (target is right word)\tl\tPressed\tl (target is right word)\tC\t3111\t\t\t\t\t" into multiple columns along the \t separator? When I try the strsplit (x,"\t") command I get: [[1]] [1] "Main Group" "1000"

--- Matt Zimmerman <mdz@debian.org> wrote: > On Tue, Jul 30, 2002 at 09:04:38PM -0500, Brady Patterson wrote: > > > The patch I submitted only reads AIFF files. I'm about to start > the patch to > > write AIFF files. > > > > To do so, we need a command-line option to specify AIFF. My > inclination is to > > add an option: > > > >