search for: stereo

Hello, I've been using Opus on an STM32 M4 platform for speech coding in mono mode. I thought I'd try stereo for grins to see if I can handle the CPU load, and I'm getting a return code of -1 from opus_decode_float (using CBR and 40ms frames). I decided to try the opusenc and opusdec tools to just see how the command line apps would behave. I am getting decoded audio, but I am losing the stereo sepa...

Hello, I've been postponing some of my encoding for when joint stereo gets implemented. The reason I've been doing this, is that I am under the impression this is the largest step in the quality/bitrate ratio that's left. Now I'm wondering if I am correct in thinking this. lame's documentation seems to imply it doesn't make much of a difference at...

Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including trav...

Hello all, I can't get stereo with wineasio, even the new verison of wineasio which I have compiled > 1.7.4. when I open an app with wine, I do see the connections in Qjack connections, but It is not in stereo Ardour can verify this, and both left and right connection are connected from wine to jack, I have googled for this...

Hi! >From http://www.vorbis.com/stereo.psp : "Oggenc's default choice varies by bitrate and each mode is selectable by the user" So, how am I supposed to do that? E.g. what do I have to do when I want an 128kbps .OGG using lossless stereo, or does this "selectable by the user" mean only that I can choose the ste...

Hello all, I created today a fake stereo piece of music (the left and right channels were completely equal), and encoded it with the all the modes the RC2 encoder supports, and found that: * * dual * waste * waste * * mode * joint * stereo * in * in % * *...

Hi, I've read the Vorbis stereo documentation on square polar mapping and currently reading the source code to understand it. But there are some things which I don't quite understand and hope I can get some guidance on. I understand the decoding/decoupling part as it is the same as the one described in the stereo docs:...

Hi, I've just read this on Vorbis Xtreme site: "11. YOU SAY THAT OGG VORBIS IS PATENT-FREE, BUT I SAW A PATENT NAMED 'JOINT-STEREO' ON FRAUNHOFER'S PATENT LIST? SO OGG VORBIS ACTUALLY ISN'T PATENT-FREE SINCE IT ALSO USES JOINT-STEREO? No. You can't judge on a patent just by looking at its name - what's 'inside' is what matters. So if the name of the patent says 'joint-stereo', that doesn&...

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one though...

Dear All: I studied the stereo couple documents and now confused, I wonder why it can be done like that, and what is the principle behind that. I think i am confused perhaps because i have no background knowledge, can you help me out? Also, when doing the T/F transform,we used MDCT, so it means no imaginary part. And i am much...

Hey Kevin! Once you start dropping under (I think) 32kbps, the codec naturally starts losing stereo separation for the benefit of quality. If you really want perfect stereo at 16kbps CBR (this is horrifically low in my opinion) you might be better off splitting the audio and encoding each part at 8kbps CBR. Or, there may be a magic "enforce 100% stereo" one could pass? (doesn't look...

Hi, If I want to encode two audio channels that are not related, audio from two different sources not stereo, is it best to create two instances of a single channel encoder or use one encoder with two channels? thank you, Paul.

If call recordings were stored in stereo and the callers were evenly distributed along the stereo spectrum. BAM. Just a cool idea I thought up, but probably completely impossible, and even if not, likely too much work for too little reward. Even less likely would be live stereo conference calling. But hey, RTP certainly supports stereo s...

On 27/01/17 12:29 PM, Jon Lederman wrote: > Thank you. Yes, we do need both channels independent. So, if we > encode each channel separately, we will be sacrificing the > compression ratio we would achieve with stereo encoding, correct? Not necessarily. Stereo makes two assumptions: 1) It assumes the two channels are somehow correlated 2) It assumes the two channels are meant to be listened to using headphones or speakers setup for stereo listening If instead you have (e.g.) two different people talking in each...

...mapping layout correct in this patch. Cheers! On Tue, Apr 25, 2017 at 10:07 AM Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 25/04/17 10:12 AM, Drew Allen wrote: > > We assume that the input file is ordered first by ACN ambisonic channels > > followed by a (possible) stereo track, and we want to swap the order for > > the API in order to couple the stereo for coding. > > Well, if you look at section 5.1.1 of RFC7845: > > The 'coupled stream > count' field indicates that the decoders for the first M Opus > streams ar...

...> listeners at 24kbps, you'll need about 24kbps*100 = 2,400kbps = 2.4Mbps > of bandwidth. That's about 2 T1 lines worth of bandwidth. Trying to > push 100 128kbps listeners down your 768kbps cable modem isn't going > to work :) It doesn't say if it's using stereo or mono quality. You should read the MP3-HOWTO. In the bandwidth considerations section you'll find: "Consider this scenario. A T1 link has a capacity of approx. 1.55 Mb/Sec. If you stream your MP3's at 128K/Bps stereo, each connecting player will use 256K/Bps, so only 6 users could c...

...s a bunch of extra things, plus for interactive use we care about both the encoder and decoder, and celt_pitch_xcorr gets used vastly more by the encoder than the decoder... I think the decoder only uses it for PLC). > > Something like > ./opus_demo restricted-lowdelay 48000 2 96000 comp48-stereo.sw /dev/null > > comp48-stereo.sw can be found here: https://people.xiph.org/~tterribe/opus/comp48-stereo.sw > > celt_pitch_xcorr also gets used by the SILK encoder (more in fixed-point than float, but the float one uses it, too). So it may be worth doing a run with the application set...

hello, a very short while ago there was a brief discussion when to use stereo and when to use mono. as you know, Radio France kept its word and now is webcasting in ogg. In total Radio France (through tv-radio.fr/ycast.com) is webcasting eight different stations, using the ogg format. on http://cuba.calyx.nl/~oink/oggstreams/rfoggstreams.html you can find all the streams...

Hi there ! I checked Vorbis' performace for mono files at approx. 64-80 kbps (it does a good job) and I'm wondering why the current OggEnc still uses Point-Stereo (>10kHz) for -q4 and -q5 I know, we're usually unable to percieve those phase correlations above 10 kHz, But a Dolby Prologic Decoder isn't. I think a future version of OggEnc which is able to use a user-selectable stereo mode would be graet. I think back to the time, where everyone j...

...tion ? it does the following loop: ... for (i=0;i<st->frame_size;i++) st->play_buf[st->play_buf_pos+i] = play[i]; ... So... if frame size is 'samples per channel' it will copy only half the play buffer since there are two channels interleaved (supposing the play buffer is stereo of course) I think it should take into account the number of channels, resulting in something like: ... for (i=0;i<st->frame_size * play_channels;i++) st->play_buf[st->play_buf_pos+i] = play[i]; ... Sorry if what I said doesn't make any sense... I'm just trying to unders...