similar to: libopus API question - 120ms encoding

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?

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 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

I can't quite figure this out from looking at the repacketizer code. Let's say I have 4 separate stereo streams (say from an 8 channel converter) and want to transmit them somewhere in one stream e.g. RTP or the like. (assuming, custom format if necessary) So could I merge 4 packets with the repacketizer, TX the merged packets, on the other side RX them then split them with the

Why does opus_encode return 359, and not 360, please? This is my code: int iShortsRead = fread(m_RawBytes, sizeof(short), (5760/sizeof(short)), fin); for(int i=0;i<iShortsRead;i++) { opus_int32 s; s=m_RawBytes[2*i+1]<<8|m_RawBytes[2*i]; s=((s&0xFFFF)^0x8000)-0x8000; m_ShortsInput[i]=s; }

HI, I'm trying to simulate an audio conference where each leg can be with a different bit rate. This needs to encode the source PCM to to different bit rates back to back and store and send respective encoded frames/packet to the respective channel. For this I changed the opus_demo as below. But the output of the second encoded frames is completely garbled. Appreciate if anyone can suggest

If you encode with int iByteSizeEncoded = opus_encode(m_enc, m_ShortsInput, (1920/sizeof(short)), m_EncodedBytes, (960*6)); ... the byte size is 120. I thought that an odd value like 359 has to be a fault on my side. I thought it would be a multiple of something. So 359 is really correct??? Thank you for clearing this up. Am 06.10.2013 16:50, schrieb Jean-Marc Valin: > Why

Hello, I've been trying to understand how to use opus codec for quite some time. There is a thing that doesn't make sense to me. I will try to explain what it is below, In *opus_encode() *function, *opus_int16 pcm* is said to have a size of *frame_size * channels*, where *frame_size* is said to be one of* 20, 240, 480, 960, 1920, and 2880.* Let's say, frame_size is 960, then

Hi Jean-Marc, I looked at that but importantly these streams need to remain absolutely independent, Further they may have been encoded at some previous time. So my question stands. Thanks, Marc On Jul 26, 2013, at 9:10 PMEDT, Jean-Marc Valin wrote: > Hi Marc, > > I recommend you have a look at the multistream API and how we use it for > surround in the Ogg Opus draft. Sounds

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

Gents, My TI C55xx complier is complaining about (in opus_encoder.c, opus_encode()): st->silk_mode.payloadSize_ms = 1000 * frame_size / st->Fs; where .payloadSize_ms is opus_int, frame_size is an int, and Fs is a long. Should one of these be cast differently? Thx, MikeH -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Ralph, > Likewise opus_encode() takes 16 bit samples, so you need to extend each > sample from an 8 bit source before encoding. Two questions 1. In opusenc.c which API does the extending the 8-bit to 16-bit? 2. If that is the case then how will 24 bit PCM sample work? Regards Amit On Thu, Jan 7, 2016 at 12:21 PM, Ralph Giles <giles at thaumas.net> wrote: > On 07/01/16

Hi everybody I wrote a couple of months ago about implementing the opus codec on a DK52 from Nordic (nrf52832 MCU, Cortex M4). In the meantime I got a bit further and would again like to ask some questions. My settings: I included all the opus files into my project (opus-1.2.1) Defined: FIXED POINT, VAR_ARRAYS, DISABLE FLOAT API, OPUS_BUILD, CELT_MODE_ONLY I am able to allocate enough memory

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

Finally got it to compile and attach the MSVC debugger. It fails at the following: if ((!st->variable_duration && 400*frame_size != st->Fs && 200*frame_size != st->Fs && 100*frame_size != st->Fs && 50*frame_size != st->Fs && 25*frame_size != st->Fs && 50*frame_size != 3*st->Fs) || (400*frame_size <

hi,All, I want to know whether Opus has AEC features like Speex? Thanks 2012/10/17 <opus-request at xiph.org> > Send opus mailing list submissions to > opus at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.xiph.org/mailman/listinfo/opus > or, via email, send a message with subject or body 'help' to

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

Hi all, i am trying convert pcm (16bit pcm) stereo file to mono pcm file using opus_encode and opus_decode, i am able do this but i have doubt about TOC byte after opus encode. below is how encoder and decoder structures are used to do encode and decode file opus_encoder_create(8000, 2, OPUS_APPLICATION_AUDIO, &err); opus_decoder_create(8000, 1, &err); after opus encode bits looks like