similar to: Streaming from memory

Hi! I'd like to make a stereo 8-bit wav file from a stereo (16 bit) ogg vorbis file (in delphi, with the vorbisfile). I use: ret := ov_open_callbacks(filein,vf,nil,0,ops_callbacks); if ret = 0 Then begin fileOut := TFileStream.Create(savedlg.FileName,fmCreate); repeat ret := ov_read(VF, pcmout, BufferSize, 0, 1, 1, nil); //'cause of the 8-bit

Why ov_open_callbacks(fd_, &vf, NULL, 0, OV_CALLBACKS_NOCLOSE) takes so long to open OGG file? The OGG file has just 210 MB. It takes a lot of time open the file. I am working on OGG audio recorder and player. Thanks in advance! With kind regards, Pavel

Hi all, I'm very new to Ogg, and am learning about it in an attempt to stream it from a CF card through a microcontroller. This particular setup requires me to move parts of the file from the CF card to the microcontroller buffer, effectively making the file like a data stream. Also, should this question be directed at the Tremor group instead since that is the codec I'm using? It

hi, I downloaded the vorbis-sdk-1.0rc3 and wrote an application that plays ogg files using DirectSound(win32 application). It uses streaming technique - I'm implementing and passing IStream interface to the DirectX. The interface is used by DirectX to fill his buffers during playing. DirectX is assuming that it's readind from a wav file, so the interface decodes on-the-fly so that

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

More newbie questions, please bear with me =) I've got my high level code pretty well structured, and I'm now trying to decode vorbis streams on the fly using the ov_open_callbacks() routines. I can't seem to figure out how to tell it what output format I want -- is this possible? For example, I would like to be able to specify number of channels output (e.g. stereo vs. sum-to-mono

Hello, I'm currently trying to implement streaming off ogg-files out of an archive file using SDL and SDL_mixer for playback. Therefore i create a FILE * filepointer, place it to the beginning of an ogg file inside the archive and hand over the filepointer to ov_open. ( command: ov_open(fp, &music->vf, NULL, 0) ). This works fine so far. The problem is: if i put many oggs together in

I'm in the final throes of debugging (hopefully) and the last thing I'm trying to do is figure out why I'm getting static instead of usable data. I have a verified Ogg file (it plays fine in WMA); I've visibly inspected the info struct and it matches my output format; I've verified my audio playback code by using a generated sine wave (sounds correct). So that leads to a

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'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

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,

I'm new to Speex and was wondering if there is a nice streaming API similar to Vorbis' "ov_open_callbacks", "ov_read", "ov_time_tell", etc. If not, is there an easy way to get Speex streaming from files? My appologies if this has already been answered a million times, but the archives aren't currently accessable! Thanks, Kevin -------------- next part

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

There was a bug in the new ov_open_callbacks where it would not zero out some member variables of the file info struct, and if a file was not successfully opened this would cause a crash at destruct time. Fixed. I've also made some changes so that vorbis will compile cleanly under win32 without one having to muck around with it. I have added a "vorbis_vc6" directory which contains

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

I want to use Vorbis to stream music off the HD in my DirectX app. Currently, we have a generic streamer that writes raw digital sound data into the primary sound buffer. We plug sound sources (e.g. wav files, PCM files) into this streamer by having the sound sources have an interface something like this: class SourceDataSourceInterface{ virtual bool Init(char* pSource) = 0;

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

Jack, I got the asm we were working last night committed ont he beta 3 branch. It's in use in both vorbisfile:ov_read() and the lsp lookup (when using float lookups) I'd like to have a patch of the other things you did in the past day or so, even if you don't want to commit yet. Monty --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: