similar to: Last call for Opus specification

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).

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

Hi! Useful advice, thanks! Actually I had been using foobar2000 to recode, because it just makes it so easy to convert multiple files while keeping the metadata (I confess, I'm a "tagger"). But it's easy to miss some encoder option when being presented some default suggestions in a dialog form... Apart form that I always had the impression that Opus could be quite smart

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

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

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

Hi Jean-Marc, I looked at that but importantly these streams need to remain absolutely independent, Further they may have been encoded at some previous time. So my question stands. Thanks, Marc On Jul 26, 2013, at 9:10 PMEDT, Jean-Marc Valin wrote: > Hi Marc, > > I recommend you have a look at the multistream API and how we use it for > surround in the Ogg Opus draft. Sounds

I can't quite figure this out from looking at the repacketizer code. Let's say I have 4 separate stereo streams (say from an 8 channel converter) and want to transmit them somewhere in one stream e.g. RTP or the like. (assuming, custom format if necessary) So could I merge 4 packets with the repacketizer, TX the merged packets, on the other side RX them then split them with the

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

This patch adds a new option "channel-mapping" to opusenc which sets the channel mapping family used by the multistream encoder. Please let me know whether adding this option is worthwhile and whether the help string is okay. I tried to keep it short but accurate. The error message for an unimplemented channel mapping is "Error cannot create encoder: request not implemented".

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

Hi all, We (WebRTC/Google) would like to extend Opus to natively support 120 ms encoding instead of relying on repacketization as a post processing step. This is to ensure that a valid 120 ms packet is always available. I've attached a couple of patches to add this to opus_encoder(), based on the internal repacketization process carried out by 60 ms CELT. We intend to extend this later for

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

Hi, I don't understand how works the multi stream api in opus. I need to send two mono streams over network with RTP. I think I'm right when I create an OpusMsDecoder with opus_multistream_decoder_create (48000, 2, 2 ,0 ,mapping, NULL) where mapping is: unsigned char mapping[2] = {0,1} isn't it ? Next, i need to encode data which I get from jack (float) so I use

Hey everyone, I've added Ogg Opus support to my Adobe Premiere plug-in here: http://github.com/fnordware/AdobeOgg Now I'll add Opus support to me WebM plug-in too. I've got this Opus stuff mostly figured out, but I have a few questions. Here's one: What do the numbers in mapping mean? I see that opus.h refers to the Vorbis channel mapping order, so does mapping in Opus take

This patch adds top level functions to create an ambisonic multistream encoder. The implementation currently just calls the analogous surround sound functions with channel mapping 255 to create an encoder that bundles uncoupled streams. Forthcoming patches will actually set channel bitrate and other configuration. My main concern is that adding additional

Hi Jean-Marc, A simpler question. How does opus_encode() generate packets of 20ms (SILK-only or Hybrid)? Concatenating two 10ms frames or doing it straight with just one 20ms frame?

So, once all those things are in place what Andrew says WILL happen? I think plans as important as that deserves an announcement. Roadmap for the project? Future directions for the main committers? Are there patent issues? Date: Mon, 11 Jul 2011 20:45:14 -0400 > From: Gregory Maxwell <gmaxwell at gmail.com> > Subject: Re: [CELT-dev] Mailing list and project shutdown >

I use the Opus multistream encoder for both mono and stereo encodings and after updating from 1.1.3 to 1.3 I noticed the size of the produced Opus files had doubled for 1-channel encodings whereas switching to the standand Opus encoder gave me roughly the same sizes as before. In these tests, I'm encoding an 8kHz mono stream containing only speech with the following options set: frame length