similar to: Increase packet size created by an encoder?

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

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'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()

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

Hi all, Perhaps a stupid question, but I just need pieces of audio from a file. I know I have to use vorbisfile, and the example compiles, but I don't know how to read the buffer. In fact, it's not even documented (or I'm looking at the wrong place). I assumed it was a interleaved buffer containing unsigned shorts, but that didn't work. I also tried the normal decode example,

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 replaygain backend that I'm working on, I ended up copying

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()

Hi all! I've problem with playing ogg files. I'm triing to use tripple buffer method: +-------------+ +-| buffer 1 |-+ | +-------------+ | | | | +-------------+-+ | | buffer 2 | | +-------------+-+ | | | +-------------+-+

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 pcmData[4096]; the ogg file and the directsound buffer

I'm working on an application where I need to record the current playing position and return to it later. and I need this to be done the most efficient way, so I used ov_raw_tell() and ov_raw_seek() because the documentation says they are the best when speed is a concern. but the problem is that sometimes ov_raw_tell returns the same value before and after calling ov_read; here's an

Some months ago I tried to get some answers (and didn't) for a problem I've had with ov_pcm_total for files 8kb or smaller. I have no idea whether it should make a difference, but I operate inside Visual Studio, with C++ 6.0 and MFC. The value returned by ov_pcm_total doesn't agree with ov_read. My solution had been to add 0.2 to the buffer size, so that ov_read would make it to

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 folks, i would like to know if it is valid to load a whole .ogg file into memory and call ov_open like this: ov_open(NULL, &OggVorbisFile, pRawData, RawDataLen); An other question is, is it valid to copy some ogg files into one (resource) file, position the filehandle and use ov_read to read only the ogg file which starts at the current position of the file handle? (In other words, does

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

Should I ov_clear a failed ov_open/test/test_open call? The xmms plugin does an fclose on failure, and that seems to work, but I thought I ought to know for certain. (An observation: I think the quantity of questions and bugs raised along the lines of "I did open file, ov_open, ov_read, close file, open another file, ov_read, and it exploded", or "I tried to make vorbisfile go

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:

I suspect the stopping behaviour is my mishandling of either OV_EBADLINK or OV_HOLE. What should I be doing if either of these values are returned by ov_read? Having experimented more Winamp has no trouble with the "jerky" songs. But the songs where my player stops prematurely pause for a second or two in Winamp. And with WMP they continue to play properly, but from the point where they

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

Hi guys, I'm working doing the Java->JNI->OggVorbis thing. As a test to get me going, I've just written a quick routine that dumps info about the file test.ogg in the current directory. The problem arises when I call ov_clear. I get a segfault everytime. Note that I am *not* doing any decoding (ov_read) at all, just ov_comment and ov_info. Should I only call ov_clear if I have

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