similar to: Stereo dropping to mono with libopus 1.2 RC

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

OK, so at the link : https://uloz.to/!yyVrCY2Y8sn1/devil-s-elbow-opus-7z (https://uloz.to/%21yyVrCY2Y8sn1/devil-s-elbow-opus-7z) (change the language to English by clicking at the flag at the right upper side of the web page or just simply click at "Stáhnout pomalu" - it may take some time as the file is 55 MB+ and the download service is free) there is 7zip archive with 5 music

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

Hi All, We are using libopus 1.0.2 lib version currently and observing one crash issue in opus_encode() function some times. ------------------------------------------- (gdb) bt #0 0x00007fcdf7c90ea2 in opus_encode () from /usr/lib64/libopus.so.0 -------------------------------------------- Anybody faced this problem earlier and have any idea whether its solved in the latest version libopus

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?

The libopus encoder's opus_encode() method is documented as "Encodes an Opus frame". Does that mean that it always produces a single Opus frame (i.e. the number of frames in the TOC byte will always be 0)? It's not clear from the documentation, but the fact that it cannot produce a 120ms Opus packet makes me wonder if that was the intention and any multi-frame Opus packets must

I was digging around and saw a bunch of different people have created ways of building libopus for android (even I did one back in the CELT days). Is there an official one somewhere that I missed? I even notice this is the Google sources, but it seems to be a touch long in tooth at this point: https://android.googlesource.com/platform/external/libopus/+/android-wear-5.1.1_r1 I am happy to build

Hi, I'm looking for the description of possible combinations of libopus parameters. For instance in case of low bitrate encoding (voip application, SILK part) it does not have sense to set 2 channels. Audio applications (CELT part) widely use 2 channels. What about HYBRID mode? Is it closer to voip or audio applications? What is the typical number of channels? Can constant bitrate be used in

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Hi all, I''m having issues with Unicorn and connecting to RabbitMQ using the tmm1-amqp gem. I''ve tried lots of approaches. The classic initializer with Thread.new { EM.run } for the Rails app and even tried using the Qusion library. (https://github.com/danielsdeleo/qusion) I''ve made a simple mod to Qusion for it to monkey patch unicorn too. The code looks

Hello everyone, For a private project I would like to make a new Internet Intercom device (for my kids and me). At the moment I use a separate DSP with Ogg/Vorbis (VS1053) but I like a challenge ;-) Is a eg. ST32F10x (cortex m3) or a new ST32F40x (cortex m4) powerful enough to do encoding and decoding ? Any existing projects to learn from? Thanks for your time, Edwin van den Oetelaar

>On 11/20/2013 03:37 PM, O'Connor, Kevin wrote: >> 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.

Hello, I am a new user of R. I am trying to use the packages fBasics and fExtremes when i am running the examples I get few error. Could someone tell me what is happenig? Thank you beforehand. from Fbasics packages: xmpfBasics() Error in file(file, "r") : unable to open connection In addition: Warning message: cannot open file '/usr/lib/R/library/fBasics/demoIndex'

>>> Agustín Dall'Alba <agustin at dallalba.com.ar> schrieb am 14.04.2017 um 22:53 in Nachricht <CAHBqS-w3v44WM5x+_4XdFMkD42A2iYTbEWKEBmvJc2P3Y-LJGA at mail.gmail.com>: > 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

Hello! I found a sample I can ABX successfully when encoded at 133.333 kbps. I was targetting 1 MB/min. https://drive.google.com/drive/folders/0B8KWShoIrA1kQzR1Z0FFRUlfcEU floex.wav is 4:54–5:04 of a lossless copy of 'Forget-me-not' by Floex, downloaded from http://store.floex.cz/album/zorya floex-133.opus was created with `opusenc --bitrate 133.333333 floex.wav floex-133.opus`,

> > This is as expected. The build system is trying to get a git revision > string out of the source tree to compile in for reference. If you're > building from a downloaded source .zip, there won't be any git > information available and it will fall back to the hard-coded release > string. That's why the error isn't fatal to the build. Ok, this seems

Hello all! I'm having trouble with a cross compiled libopus for an embedded application and would like some help. preamble: embedded details: processor: bf537 kernel: Linux release 3.0.8-ADI-2011R1-svn4693, build #2834 Fri Apr 22 12:35:34 ADT 2016 toolchain: bfin-linux-uclibc-gcc release gcc version 4.3.5 (ADI-2011R1-RC4) audio interface: ALSA

Crew, I've run into something a little odd with the vorbis encoder and would like some input. I use SpacialAudio's SAM3 Broadcaster and am having problems with the ogg encoder when I create a mono output stream. It seems as if the signal level of the left and right sides are being combined before the encoding process, leaving the sound muddy and overdriven. As such, if I reduce

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

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/