similar to: question about coding

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

I'm sorry, but I've found in the paper "Speex: A Free Codec For Free Speech" at page 5 the phrase "If more than one channel is used, then an instance is required for each channel (Speex is not stateless)". What does it mean? Could it be that is the problem that I've met? Thanks Andrea ---------- Original Message ---------------------------------- From:

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

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

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

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

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

Hi, I'm trying to encode a raw audio file with speex. No problem if I normally decode it (as done in the samplenc.c, for example). But if I separate odd and even samples and I try to encode separately these two files (one is [odd odd odd odd..], the other [even even even even..], then when I decode and I join the samples in one file [odd even odd even odd even..] I heard a terrible white

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

Hi, sndfile-convert already converts from all these formats to FLAC, but the flac tool itself has more flac-specific options. Is it possible to use sndfile-convert to provide the input data? In any case Erik is maintaining both libsndfile and libflac, and it's unlikely he'd want to duplicate the code. If anything it'd make more sense to remove code for reading other formats from the

Hi all, i'd like to ask what the best options are for safe 24bit 96kHz archiving. Currently i'm only using -8 but there are also some other options like block size etc. Can anyone suggest? Also i'd like to ask whether a no-lax 192kHz mode is planned in flac, seems like 192kHz isn't directly supported although flac can compress such rate in lax mode. My last question - is there

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

Hello, I'm using flac for an application. The fact is that i'm trying to encode a file that is not necessary a wav file. So, at first, i would like to know if it's possible to encode a file different from the wav file with the flac encoder ? When i put my data into : ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need); after some rounds in my boucl i get : encoding:

On Sun, 14 Dec 2014 01:14:16 -0800 Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: > > -------------------------- > > Also, I have a question. > > > > Currently flac complains about 24-bit .wav files if they have > > old WAVEFORMATEX header and not 'proper' WAVEFORMATEXTENSIBLE > > header. However it writes such files itself. > > >

The commit http://git.xiph.org/?p=flac.git;a=commitdiff;h=99257e177eac96fa41a107b423080232f59ebe45 also requires some changes in write_iff_headers() function: currently flac don't write WAVEFORMATEXTENSIBLE header if decoder_session->channel_mask is equal to 0, 1 or 3. After the commit 99257e17 flac should do this for channel_mask equal to 0, 4 or 3. The patch fixes this.

Ian Malone wrote: > This is all well and good but OggPCM is in an Ogg transport > stream, so that needs to be unpacked anyway. Fair enough. Since the ogg pages (which I beleive are 4k) need to be unpacked anyway, there is little harm in having to (possibly) do endswapping as well. Erik -- ----------------------------------------------------------------- Erik de Castro Lopo

Hallo, yesterday I was puzzled when I discovered that I probabliy miss something in the interepretation of intercept in two-way lm models. I thought that the intercept, using the default contr.treatment contrasts, represents the mean of the group of observations having zero in all column of the model.matrix. It turns out not to be case To be more more clear I am attaching a short example:

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 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

On 5/2/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Well, I don't quite understand why AC_C_BIGENDIAN and the solution > you're proposing is likely to break other big endian machines (the ones > that don't have __BIG_ENDIAN__). Can you send a patch that addresses > that (i.e. still uses AC_C_BIGENDIAN when it works)? It is not that AC_C_BIGENDIAN