similar to: OGG loads as a naked file, but not if embedded in an IFF.

I have a test program that loads an AIFF or WAV file with libsndfile and plays then using libao. After running around in circles for a while, I found and fixed the problem, but I'm still confused as to why the solution works. Would someone please explain this to me so I don't trip over this again later in my project? I used sf_open_fd() to get a SNDFILE* pointer and an AF_INFO

Does anyone else here have a need to allow shell users to change their imap/pop3 passwords? I came up with a little thing that works similarly to regular passwd(1), dovepasswd, which can be found at https://github.com/DavidGriffith/dovepasswd. It does the job for me and I'd love for other people to take a look and give me some feedback. -- David Griffith dave at 661.org A: Because it

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 @@

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 --------------

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

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

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

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. To give an idea of the speed decrease, encoding at flac level 8: 24/96 wav file 1.21: 61.05 seconds. ratio=0.690 1.3: 222.48

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: > > > >

Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people "touch" the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone

Has anyone else created Vorbis files with more than 6 channels? I'm a game developer trying to use multichannel Ogg Vorbis files for a PS3 and XBOX 360 title. Using the latest release libs and tools, I've created a Mac application that weaves up to 8 stereo AIFF files into one 16 channel Ogg file. What I'm finding is that if anything more than 3 stereo files (corresponding

Hi again list, I have given the matter of supporting .cda as input format some more thought. What I wanted was an easier way to do drag 'n' drop music files from CDs onto the local system and then have those files encoded with FLAC. As of now most operating systems displays files on CDs as .cda. (Which has later been clarified not to even be a format.) I think that what I actually