similar to: ov_read, callbacks and 0 bytes read

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to

Is there a particular reason why CELT grabs 100KB of stack immediately? Is that really required or can that be trimmed down some/a lot? -a

I saw the custom API, but nothing explicitly says "CELT-only" just "custom sample rate and frame size". I'll dig further now that you've pointed me in a direction. Thanks, -a On 7/6/15, 6:18 PM, Jean-Marc Valin wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > I believe what you want is called Opus custom (OPUS_CUSTOM in the > code).

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:

If everything has moved and this mailing list has gone dormant, how about an official announcement as well as official pointers to the new stuff? It would also be good to have some sort of "migration document" given that you have to do specific things to OPUS in order to get the old CELT behavior. Thanks, -a

I need to test the latency of a system that I ported CELT to. I'd like to set a flag when the stream "changes". In order to see the stream "change", I'd like to be able to scan the compressed CELT directly to set the flag. ie. I'd like to be able to find a byte or two that change reliably between two values when the stream "changes". I don't

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful for

All, Just a heads up that the IETF codec working group has issued a last call on the draft Opus specification. This means we think the draft adequately documents the format, and that we're willing to live with whatever bugs are present in the reference implementation, if they can't be fixed without breaking decodes. In turn, this means we could really use some feedback from implementers.

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

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

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

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'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()

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 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;

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

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, 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

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