search for: aiffs

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

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

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

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: > > -ff { raw | wav | aif } > > In some sense, "-ff" is silly since it probably stands

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.

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 @@

How about the opposite? What I would like to see is FLAC support in libaudiofile, so applications written with libaudiofile could transparently take advantage on FLAC's compression. Ironically I guess it would still make sense to use libaudiofile in FLAC for getting input. Mmmm, cyclic dependancies... Or is that just a stupid idea? Dave On Wed, 2002-07-31 at 13:10, Matt Zimmerman wrote: >

--- 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: > > > >

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

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 ---

Hi, I am attempting to use flac to encode 96k/24-bit broadcast wav (BWF) files. BWFs are wav files with some extra meta-data chunks, and is the favoured archival format for many institutions around the world. These files are encoded successfully by flac, however the resulting flac file is not playable on all flac players - it plays successfully in foobar2000 but is silent in winamp, and when

--- man/flac.1 | 25 +++++++++++++++++++++---- man/flac.sgml | 2 ++ 2 files changed, 23 insertions(+), 4 deletions(-) diff --git a/man/flac.1 b/man/flac.1 index fef4ded..3d7bd50 100644 --- a/man/flac.1 +++ b/man/flac.1 @@ -68,6 +68,9 @@ Prefix each output file name with the given string. This can be useful for enco \fB--delete-input-file \fR Automatically delete the input file after

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

For those that are interested in beating it up and complaining that it doesn't work on particular input files, cvs oggenc now supports AIFF input. I've tested it with 1 (one) file, which was a conversion using sox. So I don't expect it to work with everything. If others have files that it doesn't work properly with, let me know (preferably with a short description of what was

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

--- Brady Patterson <brady@spaceship.com> wrote: > > Brady, I would say for now, your proposal is fine. I am going > > to move flac to getopt soon... > > > > Matt, that would be cool if you wanted to take on the audiofile > > support. I have actually been waiting to ask you for that, > > waiting until I got the getopt support done, and the new > >

Hi all, Currently I'm looking for users of the --keep-foreign-metadata feature of FLAC. There has been some improvement of this feature in FLAC 1.4.0. Since 2007 there has been a warning in FLAC that --keep-foreign-metadata is a new feature. I think removal of this warning is long overdue, but there are still some issues surrounding it. So, if there are users of this feature on the mailing

On Tue, 30 Jul 2002, Josh Coalson wrote: (Matt wrote): > > A more flexible approach might be to use libaudiofile or a similar facility, > > so that multiple input formats can be automatically recognized, and future > > formats supported in a generalized way. Agree completely. I did this because I wanted to get something working with as small a change as possible. > Brady, I

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