search for: ov_read

...m::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, pDecodeBuffer + ulBytesDone, ulBufferSize - ulBytesDone, 0, 2, 1, &current_section); #elif MACOSX // El cuarto parametro es lo del big endian de los cojones lDecodeSize = ov_read(psOggVorbisFile, pDecodeBuffer + ulBytesDone, ulBufferSize - ulBytesDone, 1, 2, 1, &current_sec...

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 bytes. Running the file in audacity proves this size to be accurate. When I load the...

I've been experimenting with the ideas of Replay Gain[1] and find that ogg123 doesn't have a way of specifying the scaling applied to replayed samples (like -f in mpg123). Looking at libvorbisfile, I see no function exactly matching this possibly desirable behaviour. ov_read() scales by either 128 (byte output) or 32768 (word output), but there's nothing in between. ov_read_float() outputs floats, which is OK, but means more work in the frontend program. My suspicion is that floats are not used internally by ov_read(), and there would be a speed penalty for using...

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 with chode. I even went as far as ignoring the OggVorbis...

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() asks for that much but only gets 52,608 bytes. The next call asks for 3,652 bytes but only gets 2048. Subsequent calls as...

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 hom...

...#39;t have a negative code that says come back later when you have more data. I can't allow this to block as the system I am using does not have threads (embedded system). A good example of this is to put too much information in the info/comment field that overflows the initial decode. If ov_read can't get enough data to completely decode a packet, things go badly wrong. What do other people do for handling streaming like this? -a

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...

Has anyone had experience using ov_read in a thread with streaming directsound buffers? i have tried the following, but it just produces a repeating garbage noise. i would appreciate some help. notes: test.ogg in my code was a song that i converted to the vorbis format (Gorillaz - Cling Eastwood) pcmData is defined like this: char p...

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 datalength? This is what I am doing everytime I fill one of the buffers: long bytes_read = ov_read(&m_OggVorb...

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 sth? Reguards, Matthias Brandt [1] http://xiph.org/vorbis/doc/vorbisfile/ov_read.html

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...

...s to stay that way. I'm not pointing any fingers: I just want all of us to be properly careful. Testing new code is not suggested, it's required. Monty ------- Begin Forwarded Message [...] From: "Mike Nordell" <tamlin@algonet.se> To: <webmaster@xiph.org> Subject: ov_read bugfix Date: Sat, 13 May 2000 12:18:48 +0100 Sorry for possibly posting to the wrong address. In vorbisfile.c, ov_read around line 1045 there's short *dest; that's redefined inside the loop. The statement buffer=(char *)dest; at around line 1057 and 1071 doesn't initialize buffer...

The documentation for ov_read states: "ov_read() will decode at most one vorbis packet per invocation". Can I increase the packet size in my encoder so my decoder makes fewer calls to ov_read? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/at...

...39;t succeed in remove the three (yes, three!!) while loops around the actual retrieval of audio samples, without breaking the code. Aghr - while loops, terrible. What a weird way of writing decoding - doesn't everybody just want a plain function with retrieveBuffer(float**inhere) just like ov_read() is? Thanks in advance for any help (either a non-looped version of the decoding example, or a piece of code how to read the ov_read() buffer would do), Niels _________________________________________________________________ Add photos to your e-mail with MSN 8. Get 2 months FREE*. http://joi...

Is there any particular reason why ov_read() packs floats to integer PCM inline, rather than being implemented in terms of ov_read_float() and a separate packing fucntion? There are obviously many advantages doing audio manipulation on the floats before packing, but right now you have to reinvent the packing stage yourself - in a repl...

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 = *((short*)buffer); float test = temp / 32768.0f; -> test...

...| | | | +-------------+-+ | | buffer 2 | | +-------------+-+ | | | +-------------+-+ +-| buffer 3 | +-------------+ First i fill all buffest with ov_read. Then I start to play firts buffer. When i stops I'll start to play second buffer and fill firts buffer with ov_read. Buffer setting are: p_oggB->wfx.wFormatTag = WAVE_FORMAT_PCM; p_oggB->wfx.nChannels = oggstruct.vi->channels; p_oggB->wfx.nSamplesPerSec =...

...the main() in to four tasks using if(!eof) statements. In this modification, each if() decodes one packet of data at-a-time, sequentially. Then, I ran the code on a single Processor and the decoding was successful. The changes in the ivorbisfile_example.c are as follows: while(!eof){ long ret=ov_read(&vf,pcmout,sizeof(pcmout),&current_section); if (ret == 0) { /* EOF */ eof=1; } else if (ret < 0) { /* error in the stream. Not a problem, just reporting it in case we (the app) cares. In this case, we don't. */ } else { /* we don't both...

Hi, Please excuse my ignorance but I'm having trouble with a very basic matter: I'm using vorbisfile to stream audio to the waveout device in Win32 (using waveOutWrite). I'm basically reading packets from an ogg file and streaming them using a simple buffering scheme. The thing is, this works great when the bitrate is more or less constant but the audio sounds garbled if there is a