similar to: Contents of Flac-dev Digest, Vol 65, Issue 5

Djeizon Barros wrote: > > Another possibility could be different line endings in some > > metadata file (ie '\r\n' on windows and '\n' on Linux). > > > > The produced files are tagless. So I don't know about this... but > it is a good guess Its a good guess and its wrong :-). > I have uploaded a small file example for you to check. > I am

Hello dudes, I had given up by the first one who answered, but I will go through this since two of you were interested about this... I actually don't care about the filesize, I just wanted to know what exactly causes this. (for the record, wavpack always produces the same filesize in win and lin). Here you go: djzn at orion:~$ flac --version flac 1.2.1 djzn at orion:~$ wine

Hi everyone, I was fiddling with FLAC executable on Linux and also making FLAC files with foobar2000 under WINE. I was using the same exact options for both linux native executable and flac.exe through foobar2000. But I have noticed something: The FLAC files themselves created from flac.exe and flac(ELF) are not bit identical. They differ in something (Not audio data). Looking at this, I used a

Hi, I am using vorbis-tools 1.4.0 on Fedora 14 to convert Ogg Vorbis files to MS WAV format. It seems that ogg123 is encoding the wrong AudioFormat value. For Uncompressed PCM, the AudioFormat should be 1 but for some reason, it is being encoded in the WAV file as -1. This used to work on some prior version of either Fedora or vorbis-tools. I thought I'd ask in the email list in case

So some research and experimentation I think I found the problem, but I'm not sure how to go about fixing it. If I import one of the wav files into Audacity then export them as "16 bit PCM" then they encode to flac and play fine. The files bitrates are currently at 24. If I export them from audacity as a 24 or 32 (anything higher than 16), flac fails on me. With a bitrate of 24, the

Hello, I am currently learning the FLAC C API and had the code working with FLAC__stream_decoder_init_file. However, since I'd need the Unicode filename support, I tried _wfopen_s in combination with FLAC__stream_decoder_init_FILE, however I get a runtime crash as sonn as I call FLAC__stream_decoder_process_until_end_of_stream. The same code (partially taken from the examples) is working

I managed to get around it. I used the stream functions and provided my own callbacks for reading and writing. What's strange is that what I've done is just copied the contents of read/write/seek/tell/eof callbacks from the sources to my application and it works just fine, no glitches. When I use the build-in implementation, it just crashes without any reason. It's not a problem to

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

I have Windows XP on my computer. I am trying to access the audio on a file that my client has sent me. The message says: Codec is missig. Speex Voice A109 is not installed on your computer. When I try to install it from the internet, it tells me it cannot be installed. Can you give me any help on installing Speex Voice A109 so I can access the audio? Ann Felkins 308-235-4416 e-mail:

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've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

Dear experts, I am trying to use an example I found here: http://svn.xiph.org/trunk/vorbis/examples/encoder_example.c Could you please tell me what float** buffer is needed for (I don't see it used anywhere). Also, could you please tell me what this code is doing: for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)|

I am making a lightweight opus audio player. This is my portaudio callback function. But i hear a stammering song when i try to play a .opus song file. Can anybody say what is wrong or provide me link to any code which can play opus file written in c/c++(light). I tried to see some open source(vlc opus play) but was unable to follow. int MyCallback(const void* input, void* output, unsigned

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

And more info... Screen capture of the WAV file's hex. This tells me little, but maybe someone out there knows more about this than me. http://img139.imageshack.us/my.php?image=screenshot251awavghexnd1.png Thanks again, Matthew On Feb 6, 2008 12:18 PM, Matthew Davis <zasdarq@gmail.com> wrote: > Came across another error that might help! Using flac -t I get: > > 251_A.wav:

On Oct 12, 2008, at 22:32, Paul Davis wrote: > On Sun, 2008-10-12 at 19:26 -0700, Brian Willoughby wrote: >> 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

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

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

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 May 26, 2017, at 9:52 PM, Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: > Mark H. David wrote: >> Notice the warning was given only for the WAVE file produced by the >> flac reference executable. > > I agree. The flac decoder should not generate WAV files which it > warns about when encoding. > >> Should I file a bug for this? > > No need.