similar to: multi channel Ogg Vorbis file corruption

I was doing a search for Mac OS cd audio extraction with error-correction, and stumbled upon an awe-inspiring collection of misinformation from last year: http://www.macslash.org/AskMacSlash/02/09/14/0257246.shtml My favorite bit is the part about Audio CDs being composed of AIFF files that can be easily copied bit-for-bit perfect with no error-correction algorithm. -- agreenbu @ nyx . net

On Jan 12, 2013, at 14:28, Martijn van Beurden wrote: > On 12-01-13 22:46, Brian Willoughby wrote: >> I would suggest that everyone keep in mind the vast installed base of >> hardware FLAC recorders and players, and not senselessly make them >> obsolete without extremely compelling reasons. > > This can be done for the same reason the change from 1.1 to 1.2 > added

Ok, so here's the deal... I've been having a lot of trouble sleeping, so I've done some research and discovered this program called "Hemi-Sync". I decided to torrent it from the Pirate Bay to try it out before spending my money. What I torrented worked incredibly, so bought it, then torrented another Hemi-Sync program: Lucid Dreaming. It's a DVD-Audio program. What I

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

On 3/22/07, Brian Willoughby <brianw@sounds.wa.com> wrote: ... > But to return to your question, exactly what kind of "support" are > you looking for? I wasn't looking for any particular support, but just to see what support was there (if any). ... > I do not believe that there is any need for code specifically > supporting Ambisonics. FLAC supports conversion to

I vote for documenting the --channel-map option in the --help I don't like the idea of rejecting a multichannel file merely for mapping, so there should be a documented option plus an error message pointing to the option. This should compare to the WAVE and AIFF errors where the utility suggests to the user how to get the file converted safely. Brian Willoughby Sound Consulting On

Hi all, Siliva contacted me about this OggPCM proposal and asked me to join in. For those who don't know me, I am the main author and maintainer of libsndfile and therefore know quite a bit about how uncompressed audio is stored in sound files. However even I would not consider myself an expert; there are areas to do with channel assignments that I know I am ignorant of. I am also quite

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 ---

On May 13, 2007, at 05:45, Harry Sack wrote: > If I encode 192 kHz sound @ 24 bit for some days (WAV file) and I > encode it to FLAC, I think you can have a very big file and 1.5 TB > is reached very quickly. > And in the future audio will even get bigger, when used for HD-DVD > en Blu-ray media and 5.1 channels is considered the 'minimum' > setting for surround

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

Hi, I have posted this three times to the flac-dev, vorbis-dev, and ogg-dev mailing lists. I wanted to see what code there was currently to support Ambisonics. So I downloaded the code from the xiph download page for libogg-1.1.3, libvorbis-1.1.2, vorbis-tools-1.1.1 and flac-1.1.4, but wasn't able to find anything. If it exists then I missed it, so could somebody please point me to it.

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

On Dec 14, 2014, at 10:02 AM, lvqcl <lvqcl.mail at gmail.com> wrote: > Currently the header of a decoded WAV file can be different to the > original WAV file because FLAC doesn't preserve 'fmt ' chunk. > > For example: create a 24-bit stereo .wav file with WAVEFORMATEXTENSIBLE > header with channel mask == 3. Encode it to .flac then decode back to .wav. > FLAC

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