search for: stereos

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

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 higher than 128kbps (I

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 traveling over

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 answer to no aval and I have asked numerious times on IRC in

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 stereo mode indirectly by choosing an appropriate bitrate?

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 % * * * stereo * /2 * kbits * of JS *

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: From mapping0.c: /* channel coupling */

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

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 thought is to

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 puzzled about the angle in the stereo couple,what does it mean?

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 streams, right? I don't know if any of the used codecs

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

My apologies for the confusion, I think I have the 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

christophe.guerin@etud.univ-pau.fr wrote: > > * Enough bandwidth to run the server. If you want to broadcast to 100 > 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

On 21 November 2014 at 18:06, Timothy B. Terriberry <tterribe at xiph.org> wrote: > > Viswanath Puttagunta wrote: >> >> a. Simplest use case to validate this optimization for correctness. >> b. Simplest use case to validate this optimization for performance. >> >> Would prefer something like opusdec that can be executed on command >> line. > >

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 with m3u files and

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

Hi Jean, Thank you for your reply. Ok... I'm gonna use 'samples per channel' everywhere I see 'samples'... but what about the 'speex_echo_playback' function ? 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