similar to: flac error with some aiff files?

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

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

Brian Willoughby <brianw at sounds.wa.com> wrote: > Hello all, > > Is anyone here potentially up to the task of adding support for CAF > (the CoreAudio Format) into the flac command-line? ... > I've already made some recordings > that are so long that they cannot be uncompressed from FLAC to WAV or > AIFF because they would exceed 4 GB - the maximum file size for

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.

--- 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: > > > >

--- 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 > >

Josh, I applied this patch and all the "make check" tests passed (but I do not have valgrind and I did not use --enable-exhaustive-tests during configure, so it isn't all the tests). Do you see anything suspicious about it? --- flac-1.1.2.oldBuild/src/flac/decode.c 2005-01-24 20:13:46.000000000 -0800 +++ flac-1.1.2/src/flac/decode.c 2006-09-06 23:36:26.000000000 -0700

Hello, I have written a libsndfile (http://www.mega-nerd.com/libsndfile/) file output plugin for libao. For the moment I haven't done any changes to the core of libao. However some changes could make this more useful. Before I embark on anything like this I'd like to have your opinion on a few things. --- First, what do you prefer ? 1a. Keeping the AU/WAV/RAW plugins and adding

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

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

I use Sony (previously Sonic Foundry) Sound Forge, which allows me to save audio files in .w64 (Wave 64) format to get around the 2GB .wav file limitation. W64 was invented by Sonic Foundry, and is an open format as far as I know. The only programs I know about using the .w64 format at the moment are Sound Forge and Steinberg Nuendo, although there may be others out there. With increasing

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: >

Hello all, Is anyone here potentially up to the task of adding support for CAF (the CoreAudio Format) into the flac command-line? This would present minimal difficulty under OSX, due to the presence of the CoreAudio API, but the real challenge would be to support CAF on Unix and Windows - everywhere that flac is now available. Although the format is rather unknown, there are some very

On 5 June 2016 at 21:34, Frank Esposito <fpesposito at gmail.com> wrote: > Hello -- > > I have a flac file encode 92/24bit -- what tools are available to > convert these files to wave 44.1/16 bit so then I can convert to > mp3? Thanks > > -- > Frank Esposito Don't you mean 96/24? Also, which platform (*nix, Mac, Windows)? Off the top of my head: Audacity,

Hi Martijn, thank you for your answer. So the problem would be suboptimal compression due to suboptimal assumption about the input signal, right? What I do not understand is how the format of a FLAC format would be affected by supporting A-law and mu-law files as input (and thus output). Despite of suboptimal performance, is it possible to treat 8bit *-law samples as 8bit linear PCM files and

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

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 guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable. Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection. 1) Is it

Thanks for the answer. So data are stored in little-endian format on the WAVE file. But have I to encode in big endian format or little endian is ok?! Thanks ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Tue, 23 May 2006 12:15:05 -0700 >You are probably reading little-endian samples on a big-endian platform, and