similar to: libopus API question - 120ms encoding

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

Thanks Jean-Marc and Benjamin for the answers. One follow-up question. If I use a repacketizer as Jean-Marc suggested by combining two 60ms frames to form a 120ms frame, without extracting individual frames and using a new TOC, I would need to have a "de-packetizer" that does the exact opposite of repacketizer. De-packetizer would need to separate this 120ms frame into two 60ms frames

On 10/26/2013 01:11 PM, Wang, Chris wrote: > 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? Just one 20 ms frame. It always returns a single frame except when it just can't (e.g. 60 ms CELT). > From your explanations below, opus_encode() will concatenate

ly or Hybrid frames for 40 or 60ms packet, respectively. That is based on = concatenating 20ms frames, right? Is 60ms the largest packet opus_encode() can generate? In order to get pac= kets of up to 120 ms by combining multiple frames as described in RFC6716 c= lause 2.1.4 one would need to use the "repacketizer". That is if I want to= have a 120 ms packet, I would need to take

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

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>

>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.

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 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

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

Christian, I will give 64kbit/s a try and post the figures. My own project is voice only and requires low bitrate so was hoping that it was just the way I was compiling and not an actual regression in speed for SILK. The raspberry PI is quite a cheap and handy reference platform though the ARM side is fairly underpowered but has a great GPU. It also has no audio in which is a pain for playing

I have just started trying Opus with a view to using it in a project. I am interested in embedded hardware and tried it on the Raspberry Pi using the raspbian distro. The version of libopus in the repos is 0.9.14. I installed this and tried encoding 2 minutes of speech from a librevox recording. It managed this at a respectable pace for complexity 10: Skipping chunk of type "LIST",

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

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

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 had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

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

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 ||