similar to: Streaming buffers/ov_read question

I am getting the following error from MSVC when calling ov_open() from my app: Unhandled exception in Progname.exe (KERNEL32.DLL): 0xC0000005: Access Violation The call stack shows: KERNEL32! bff7b9a6() VORBISFILE! 00471d1d() I am running Win98 and v1.0 off the vorbis libs. The app is single- threaded. The following code works just fine in a test app I built, but when I add it to another

The sound file is played correctly for the first time, then when rewind to the initial position. then copy PCM to buffer, the OpenAL report an error. It seems like the OpenAL doesn't recognize the PCM data. The OpenAL error number : AL_INVALID_VALUE 0xA003 void Buffer::PCMData (ALuint id, ALenum eFormat, ALvoid *data, ALsizei size, ALsizei freq) { // Copy

looks fine, I would suspect how the PCM sample are formatted and sent to process(), could you show that part of the code? Josh --- James Smith <jsmith@landmarkdigital.com> wrote: > > I'm using libFLACC++ and libFLAC and I think that I'm using the calls > in the > typical order (see code below). But every monoe or stereo file that > I send > thru I get files

I'm using libFLACC++ and libFLAC and I think that I'm using the calls in the typical order (see code below). But every monoe or stereo file that I send thru I get files that are the same sze as the orginal wave files. Doing a flac -a on the flac files I see that I get: frame=9 blocksize=4608 sample_rate=8000 channels=1 channel_assignment=INDEPENDENT subframe=0

Here's how I set up the data for processing: // For moving data into 32 bit shape uint8_t *buffer8 = NULL; uint16_t *buffer16 = NULL; uint32_t *buffer32 = NULL; unsigned sample32; unsigned sample, channel; uint32_t bitsPerSample = this->get_bits_per_sample(); numFrames = inData.GetSize();

Hello, Am I missing something? float** pcm_channels; int bitstream; int where = ov_pcm_tell(&vf); -> The Result is "0"; ov_read_float(&vf, &pcm_channels, 1, &bitstream); -> pcm_channels[0][0] is "0.000338580" ov_pcm_seek(&vf, 0); char* buffer = new char[2]; ov_read(&vf, buffer, sizeof(buffer), 0, 2, 1, &bitstream); short temp =

I am trying, for the first time, to decode and play an ogg file with two buffers. My audio is choppy and I can't see a problem on the wavOutxxx end of things. So I am wondering about how I call ov_read. From my TRACE statements, the read seems to work perfectly, eventually returning zero. Is it a simple matter of making the same call to ov_read, aside from changing the buffer and

Hi! I solved the problem using ov_read_float instead ov_read. Still do not know why ov_read returned garbage, but ov_read_float works fine! Thanks! El 17 de julio de 2011 10:41, Juan Miguel Mart?n Mu?oz <jmcubo at gmail.com>escribi?: > Hi, thanks for your replies! > > ogg.k, I'm sure MACOSX is defined. When I'm debugging line by line the > program runs into ov_read.

Hi, I've ditched the idea of calling the oggvorbis .dlls, and I'm compiling the 1.0beta3 source into my own dll. After opening as "rb" instead of "r", the ov_open() call works. However, ov_read() seems to consistently read less data than I ask for. My buffer is enough to hold two seconds of data, 16 bit 44k mono, that's 176400 bytes. The first call to ov_read()

When I call ov_read ( ) I cannot get more then 1024 bytes decoded even when the buffer size is 4K. The function only decodes 1024 bytes max even when I tell it to decode 4096 bytes. Does anyone has an idea what is going on. <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

Hi, I'm new to mailing lists, so don't argue if I do sth wrong... I'm coding on a C++ game with Vorbis- and 5.1-Surround-compatibily. For two channels (stereo), everything's fine, but when I try to read a 6-channel-ogg, ov_read returns -131 (OV_EINVAL). That's strange, because as the specification says, it only returns OV_HOLE or OV_EBADLINK as errors [1]. Have I missed

Ok here is the breakdown. I am trying to impliment streaming into our game using openAL and ogg-vorbis. In the non-streaming function when I call ov_read on the file I should be streaming the buffer gets filled with the correct data. However, when I run ov_read on the same OggVorbis_File when I read the buffer gets filled with 0's. The buffer has not been zero'd and starts filled

When I am using vorbisfile on a disk file, ov_read does not always (in fact - most of the time) read the number of bytes I request. It doesn't seem to be dependent on the number I ask for, and the ratio between what I request and what is read varies. Can somebody tell me what's going on here? TIA Lance --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

How does one deal with the case when an fread callback doesn't have any more data but is going to get more data in the future? In particular, this is an issue when reading an incoming Ogg stream from a socket (icecast2, in this case). The issue is that the fread can't return 0 (that signals EOF and hoses everything), and it doesn't have a negative code that says come back later

Folks: Granted, this slipped past my review too, but do *not* submit patches that have not been tested. This bug was a no brainer; running the code only even once would have found it. The mainline of Vorbis is our face to the world. A simple mistake like this costs us credibility when we're already going to be fighting hard against some of the most powerful media organizations in the

Hi, I have this code to decode ogg data: unsigned long PSS_OggStream::DecodeOggVorbis(OggVorbis_File *psOggVorbisFile, char *pDecodeBuffer, unsigned long ulBufferSize, unsigned long ulChannels) { int current_section; long lDecodeSize; unsigned long ulSamples; short *pSamples; unsigned long ulBytesDone = 0; while (true) { #ifdef WIN32 lDecodeSize = ov_read(psOggVorbisFile,

I have a wav file which has been converted to an ogg file with libvorbis via ffmpeg. However I've tested a few files just to make sure. I run a standard while loop to using ov_read to pull all the data out of ogg file. Using 4096 as the size to read each call. Watching my debug logs it is reading 2048 bytes each call. When I load the wav file the data chunk for the pcm data is 167680

This is a rebase/resend of earlier patches. I skipped the pure cosmetic patches for now. Mostly this is consolidation earlier changes, removing dead code etc. The important part is the change for allowing a vmbus channel to get callback directly in interrupt mode; this is necessary for NAPI support. Stephen Hemminger (14): vmbus: use kernel bitops for traversing interrupt mask vmbus: drop

This is a rebase/resend of earlier patches. I skipped the pure cosmetic patches for now. Mostly this is consolidation earlier changes, removing dead code etc. The important part is the change for allowing a vmbus channel to get callback directly in interrupt mode; this is necessary for NAPI support. Stephen Hemminger (14): vmbus: use kernel bitops for traversing interrupt mask vmbus: drop