similar to: opus-codec.org/comparison: Mono or Stereo?

Looking at the Opus comparison page[1], I can't figure out whether the Opus/AAC/Vorbis/MP3 lines are meant to imply a mono or stereo encoding. Could someone please update the caption to clarify this? The single dot for G.711 is clearly mono, but for stereo music, are the codecs at the top meant to converge near 128 kbps, or 256 kbps? [1] http://opus-codec.org/comparison/

Paul, you may find the following page has more useful/accurate numbers than that diagram: https://wiki.xiph.org/Opus_Recommended_Settings ________________________________ From: opus <opus-bounces at xiph.org> on behalf of Paul Marks <pmarks at google.com> Sent: Thursday, February 13, 2020 6:31:39 PM To: opus at xiph.org <opus at xiph.org> Subject: [opus]

Hi, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

Hello, I am trying to encode an Opus file with Ambisonics including Head-Locked (non-diegetic) Stereo sound for a Virtual Reality 360° video. YouTube describes the spatial audio requirements here: https://support.google.com/youtube/answer/6395969 It's the last list item 5. > 5. Supported First Order Ambisonics (FOA) with Head-Locked Stereo format: > W, Y, Z, X, L, R as a 6-channel

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello For some reasons, I need to encode in mono a stereo wave file, as with the --downmix option from oggenc. I searched in speexenc man page and in the on line documentation, unsuccessfully. Of course I could first use something like sox to down mix my file, but it would be more practical to have an option directly in speexenc. - -- RIQUER

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

Hi, Up until now, I've been providing narrowband streams at 24kbps MP3. We've reluctantly been using MP3 instead of say Ogg Vorbis due to wider support. But it's getting to the point where low bitrate MP3 doesn't really cut it anymore. With technologies like AACPlus out there in common usage, people have come to expect better and one can hardly blame them. So I thought

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

When I encode a monophonic wav file, I would expect the resulting ogg file to be at about half the bit rate specified on the command line, as stated in the "oggenc -h" help text: "The 6 modes are approximately 112, 128, 160, 192, 256, and 350 kbps (for stereo 44.1kHz input. Halve these numbers for mono input).". This doesn't seem to be happening, though. I took a 16-bit,

Hi, Does Celt (in particular version 0.7.1) exploits correlation between the two channels in stereo mode? In practice, is it possible to use the two channels as two mono signals without affecting quality? Thanks Best Regards Riccardo Riccardo Micci Senior DSP Engineer, Wireless Group Cambridge Consultants Science Park, Milton Road Cambridge, CB4 0DW, England Switchboard: +44 (0)1223

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 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

Sorry for the questions out of the mail list's subject. 1) Could you help me with getting information on how can I convert multichannecl (e.g. 5.1) non-encoded (PCM) data into mono\stereo? I suppose this performed in the manner like: Left = x1*(Front Left)+x2*(Back Left).... But is there exist commonly used standards to perform this task? 2) It seems FLAC pjoject have been frozen for some

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?

Hello Jean-Marc, yes, it works for both 32 and 48 kb/s. Also this option marginally increases the size of the file which is understandable. I tried a few other music files and found that this stereo narrowing happens not only in the case of track I submitted (although in smaller scale) so it would be great if the final libopus 1.2 had officialy a parameter to force music "mode" of

Hi Matěj, I had a look at your files and it seems like the issue is that the speech/music detector thinks that some segments are speech rather than music and at low bitrates SILK tends to significantly narrow the stereo image. We're planning on adding an option to opusenc to force it to treat the input as speech/music. Just to confirm that's the issue, can you try encoding with the

On Fri, Nov 16, 2018 at 7:40 PM ongaku zettai <sergeinakamoto at gmail.com> wrote: > Hello. > i have over 30GB of Opus music and noticed that > solo instrumentals and solo vocals uses more bitrate > than full-mixes. > Here's example where Opus 1.3 used 190 kbps for > piano solo and 159 kbps for full-mix. > (--bitrate 160 --music) > Download example piano solo

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

I halved the volume of the sample before encoding with `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. So the artifact I heard was just clipping when encoding or decoding. Opus remains unbeaten for me at 133 kbps. That's totally awesome. Cheers On 10 April 2017 at 17:14, Agustín Dall'Alba <agustin at dallalba.com.ar> wrote: > Hello! I found a