similar to: Opus 1.3-rc released

Thanks for all the amazing work with ambisonics Drew et al. We're looking forward to the 1.3 final release and have already been successfully using the ambisonic work in production code. Varun -- Engineering Manager Facebook Audio ---------------------------------------------------------------------- Message: 1 Date: Sun, 3 Jun 2018 13:02:18 +0100 From: Peter Robinson

Hi Felicia, A few comments: > - /* CELT can only support up to 20 ms */ > subframe_size = st->Fs/50; > - nb_subframes = frame_size > st->Fs/25 ? 3 : 2; > + nb_subframes = frame_size/subframe_size; This will use six 20ms frames to make a 120ms packet, even for SILK-only mode where frames can be up to 60ms. For SILK, two 60ms frames would be a more

Attached is the amended second patch. It now extends the multistream API as well to 80/100/120 ms and incorporates changes based on Mark's comments. Thanks, Felicia On Mon, Jun 13, 2016 at 4:21 PM Felicia Lim <flim at google.com> wrote: > Hi Mark, Jean-Marc, > > Thanks for your comments. > > On Sun, Jun 12, 2016 at 6:34 AM Mark Harris <mark.hsj at gmail.com>

Hi everyone, I'm happy to announce that Opus 1.3 was just released. This major release again brings quality improvements, new features, and bug fixes. You can read all the details in this release demo page: https://people.xiph.org/~jm/opus/opus-1.3/ Changes since 1.2.x include: - Improvements to the VAD and speech/music classification using an RNN - Support for ambisonics coding using

Hi Ulrich, thanks for the suggestion. My concern is that one of the valid inputs is "2.5", which would require conversion to an int, e.g. x10, but doing something like this would start to affect the code readability. On Mon, Jun 27, 2016 at 3:02 PM Ulrich Windl < Ulrich.Windl at rz.uni-regensburg.de> wrote: > Hi! > > A note on style: Looking at this chunk of the patch

Hi, I wondered if are there any further thoughts on these patches? Thanks, Felicia On Thu, Jun 2, 2016 at 2:13 PM Felicia Lim <flim at google.com> wrote: > OK, I've amended the second patch and also added 80 and 100 ms. > > Thanks, > Felicia > > > On Thu, Jun 2, 2016 at 7:20 AM Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > >> On 06/01/2016 02:06

On 06/01/2016 02:06 PM, Felicia Lim wrote: > That was my intention with refactoring out the subframe encoding and > repacketizing bit. Or do you mean I should merge the explicit check for > 120 ms frame and the existing checks for 40/60 ms wideband? What I mean is that this line in opus_encoder.c: if (frame_size > st->Fs/50 && (st->mode == MODE_CELT_ONLY ||

Hi, I just released Opus 1.3-beta. This is a beta release towards the upcoming Opus 1.3. Changes include: - The spec fixes in RFC 8251 are now enabled by default - Improvements to the VAD and speech/music classification using an RNN - Improvements to stereo speech coding at low bitrate - Added support for ambisonics projection using mapping 3 (disabled by default) - Fixes to the CELT PLC

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 Jean-Marc, gdb out is "Program terminated with signal 8, Arithmetic exception." most likely this division by zero. you're right, this crash is reproduce on seq number 4294967265 (20ms rtp packet). This is about 994 days. "Jean-Marc Valin" <jmvalin at jmvalin.ca> писал(а):Hi Dmitry, > >So it's not explicitly in your report, but it looks like the crash

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

Stack: (gdb) bt #0 0x0000000000aaf38a in silk_VAD_GetNoiseLevels (pX=pX at entry=0x7f26740297a0, psSilk_VAD=psSilk_VAD at entry=0x15897c38) at silk/VAD.c:315 #1 0x0000000000aa4a9d in silk_VAD_GetSA_Q8_sse4_1 (psEncC=0x15897c18, pIn=<optimized out>) at silk/x86/VAD_sse.c:177 #2 0x0000000000a9f92b in silk_encode_do_VAD_FLP (psEnc=psEnc at entry=0x15897c18) at

Just to make sure, what's the goal here? Is the goal 1) to have a fast Opus implementation or are you 2) looking for an interesting FPGA implementation project? If 1), then an FPGA is most likely not necessary since Opus is not computationally expensive. If 2), then it depends on the desired size of the project and the desired quality. The simplest encoder possible is indeed simpler than the

Hello Jean-Marc et all, I've attached an updated patch which I feel addresses all 3 cases. PTAL, Thanks! Cheers, Drew On Wed, Mar 7, 2018 at 8:39 PM Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Make sure the modified libraries work in all of these three cases: > 1) libopus 1.2.1 > 2) libopus master (or 1.3-beta) with --enable-ambisonics > 3) libopus master (or

Hi guys, as MP3 and Opus have very similar objectives, I think the original poster's question was a valid one: Why does Opus have more artefacts in the lower frequency ranges than MP3 has? The spontaneous suspect that lower frequency artefacts may be more noticeably than higher frequency artefacts seems plausible, also. Is it a matter of energy (which is higher for higher frequencies)? When

Hi, I am working on trying to compile opus to be as small as possible. I have passed the FIXED_POINT and the DISABLE_FLOAT_API but I am looking into making this lib even smaller in size. Where can I find out about how to best strip this lib to only compile in what I need?

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

Here is the modified patch. I added a flag to configure.ac which is set to 0 to disable ambisonics, and 1 to enable it. Right now the implementation simply creates a surround encoder with N uncoupled streams. Thanks, Michael Graczyk -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello: We noticed that opus reconstructed noise is pulsing with a 400ms pattern when dtx is enabled in silk mode. This is independent of the background noise level and is found with speech + non-speech period test files as well as variable level noise-only test files. This issue can be reproduced with opus v1.1 using this command: ./opus_demo voip 16000 1 25000 -dtx input.bin

Hello Daniele It would be worthwhile to attach an external serial flash or USB thumb drive, if the intent is store data. This allows for far more flexibility in storage Regards Amit On Fri, Aug 26, 2016 at 11:02 AM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 26/08/16 11:40 AM, Daniele Barzotti wrote: > > You're right! I forgot to say that I need only the encoder