similar to: yet another stereo related question

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 everyone, I'd like a bit of information about the stereo mode settings in psych_44.h of libvorbis. Specifically, about the adj_stereo structure: typedef struct { int pre[PACKETBLOBS]; int post[PACKETBLOBS]; float kHz[PACKETBLOBS]; float lowpasskHz[PACKETBLOBS]; } adj_stereo; What I am attempting to do is bring lossless stereo coupling down to the lower quality levels in

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 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 all, My apologies for the confusion ... but I am re-naming this thread again because the proposal doesn't come from Ambisonia. As the founder of Ambisonia I have only been 'mediating' between some members of the ambisonic community ... so it would be wrong to associate the spec with Ambisonia. To clarify. This is what the spec proposes: - that Mapping = 1 means the contents are

Richard Lee wrote: > But there is a caveat. Does Vorbis coupling preserve "phase" relations? Any references explaining this in simple detail? Sebastian : >"coupling" is a rather loose term in the sense that it doesn't exactly specify HOW it's done. It can be understood as the opposite of coding channels independently. Obviously there are many ways to do

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

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

Hey, just came across this article. http://www.eetimes.com/news/latest/showArticle.jhtml;jsessionid=ZLCZZJMVGJYNYQSNDLSCKHA?articleID=201202250 . Many of you would be already aware of this. It mentions flac also. i feel it is a good segment(stereo wireless headsets over bluetooth) to push flac given very light decoder and using encoder at low compression(to reduce complexity). Though very less is

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

I really don't know if this is the same problem that was reported by Ingo Saitz (I really couldn't say which one was 128kbit and which was the original wav when blind-testing. The original had some distortion that perhaps does somehting with my cheapo soundcard) but here's a description of what i've found: When doing some sample encoding with rc2 (rebuilt rpm with latest redhat

hi, what quality is recommended to encode to for cd quality? i mean, for mp3, i used to encode to 128kb/s ( maybe it's a bit low.. ).. so what quality ( or bitrate ) do you recommend? thanks, gabor p.s: i'm sure this queston was asked here at least 1000x but it's hard to search for 'bitrate' here :-) <p><p><p>--- >8 ---- List archives:

Hello, I'm not a programmer or Opus developer but I tried to test the sound quality of a music file (Nick Warren - Devil's Elbow, freely available from the author's Soundcloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary 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?

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

I've made a block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit

Hi, I have a question regarding the stereo capabilities of Opus. I would like to establish a connection between two ends via Wi-Fi and the signals that are to be transmitted are of speech kind. It mean on both ends speech is both recorded and played back as stereo. Now would the delay and loss characteristics of the speech transmission at a certain bitrate be the same as Mono voice transmission?

Hi, Im trying to get World of Warcraft to sound the same way as in win xp - i mean in real 5.1 - no duplicating channels in alsa etc... Problem is that WoW in Wine 1.1.2 detects only stereo mode as SESound.log file in wow logs says: Code: 8/18 16:22:22.509 - Detected Control Panel Speaker Mode = STEREO 8/18 16:22:22.509 - Setting WoW Speaker Mode to STEREO but in winxp WoW detects real 5.1

I have an application I intend to use Opus for that involves creating recordings of two-party conversations where each party is saved as a separate channel. Audio may be later processed or analyzed on a per-party basis so if audio in one channel affects the other channel, it could create problems. My question is, if I use a single stereo Opus stream, how well will channel separation be maintained?

Just a quick question. What is the lowest possible bit rate for encoding in stereo in opus? Peter