similar to: FLAC C API / Visual Studio 2008 FILE* Issue

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

Ivailo Karamanolev wrote: > 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 >

Ivailo - FILE objects are internal to the C runtime library, they are not system objects like HANDLEs (windows) or file descriptors (unix). This means that if libFLAC has linked against a different C runtime library than your application, then the two FILE objects are incompatible. This isn't just a Windows specific issue either - if libFLAC was compiled against libc and your application

I thought about this, and the MSVCRT mismatch also. What annoys me is that I even tried compiling the library myself (with exactly the same Visual Studio 2008 as my application) and the bug didn't change the least. Thank you for your ideas, but unless someone can confirm this, there still remains the possibility that I've made an error somewhere. Also someone with more experience with

else. Stuart =20 ----- Original Message -----=20 From: Ivailo Karamanolev=20 To: flac-dev at xiph.org=20 Sent: Wednesday, January 06, 2010 7:49 AM Subject: Re: [Flac-dev] FLAC C API / Visual Studio 2008 FILE* Issue I thought about this, and the MSVCRT mismatch also. What annoys me is = that I even tried compiling the library myself (with exactly the same = Visual Studio 2008 as my

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 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

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

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

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

Thanks for helping Erik, I have uploaded a small file example for you to check. I am curious about what is exactly producing this mismatch filesize! I know it's ridiculous and probably too paranoid, but knowledge is always welcome. -rw-r--r-- 1 djzn djzn 341418 2010-02-11 20:34 bandstand-linux.flac -rw-r--r-- 1 djzn djzn 556308 2010-02-11 20:34 bandstand.wav -rw-r--r-- 1 djzn djzn 341429

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:

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 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

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

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

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:

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

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