similar to: Confusion on CELT or Silk

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

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

I'm using opus to encode some music (classical lute, if it makes a difference). How do I use these encoder ctl's ? They seem to be doing the same thing. How are they different ? Are they different ? Should they both be set ? What happens if just one or the other is set ? Thanks for any help. sean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Since our device can only handle the Celt part of Opus (due to a MIPS limitation), we have two questions: 1. Is it possible to via SDP force the remote party to send a specific mode of the 32 different possible modes or to force the remote side to use CELT only? 2. In the reference implementation of Opus it looks like the only way to force the encoder to use CELT only

Hey, I am having trouble with the size of the encoded bytes by Opus. I am also having issue with the Bandwidth ctl. Here is the scenario. If I encode 16khz sampled audio: opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND)) opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&x)) = 1102 opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&x)) = 1103 average encoded size = 120 bytes if I

Thank you Mark. I agree and have now updated the pull request with a new commit, addressing your comments. Please take a look. /Gustaf On Fri, 5 Apr 2019 at 11:41, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-01 3:37, Gustaf Ullberg wrote: > > Hi everyone, > > > > Some time ago, I sent a pull request > > <https://github.com/xiph/opus/pull/107>

Hi all, Please guide me How to change frame_size of opus codec at run-time (20ms, 40ms, 60ms) I'm stucking in this case: 1. init codec width default config (frame_size =20ms, bandwidth=48KHz, bitrate = 48kbps...), then in runtime changing: - bitrate = 24, 16, 6kbps: sound is OK - frame_size = 40ms, 60ms: Not OK, sound is distort so bad 2. init codec with frame_size = 40ms , others is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 I believe what you want is called Opus custom (OPUS_CUSTOM in the code). With that you can compile only the contents of the celt/ directory and use it like the old CELT. Jean-Marc On 07/06/2015 06:48 PM, Andrew Lentvorski wrote: > Is there a configuration or compile flag that lets me disable the > SILK portion of the codec and use CELT

Hello All, Based on reading the documentation and some online resources, it seems that there are 2 codecs CELT and SILK. However when calling opus_encode* API, it is not clear which of the two are used? Is there a specific mode to be selected to use one or the other? Regards Amit -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I saw the custom API, but nothing explicitly says "CELT-only" just "custom sample rate and frame size". I'll dig further now that you've pointed me in a direction. Thanks, -a On 7/6/15, 6:18 PM, Jean-Marc Valin wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > I believe what you want is called Opus custom (OPUS_CUSTOM in the > code).

Is there a configuration or compile flag that lets me disable the SILK portion of the codec and use CELT only? I could have sworn that there is something, but I can't seem to find it in the mailing list archives. The application here is that I am attempting to update from the old CELT codec to OPUS. Unfortunately, the CELT codec was running *very* close to the CPU (MIPS32--80MHz) limit

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

Hi Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth

Hi There, I have two OPUS handset clients say A & B A is 8 KHz, 12.2kbps cvbr supported OPUS client B is 8 KHz, 16kbps cvbr supported OPUS client When i try to encode a same voice frame(20ms sample frame) at different time intervals(not parallel encoding) for both A & B using same encoder handle by changing only bit rate. Issue here is, some noise is heard for B OPUS client, Ex:

Hi, I’m wondering how can I get the speech/music classification result when encoding the audio in opus 1.3.1? I found in the file opus_encoder.c, there is a opus_encoder_ctl request as OPUS_GET_VOICE_RATIO_REQUEST, so I wrote in my program the below code: #define OPUS_GET_VOICE_RATIO(x) 11019, __opus_check_int_ptr(x) int32_t voiceRatio; opus_encoder_ctl(encoder,

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

On 2019-04-08 4:55, Gustaf Ullberg wrote: > Thank you Mark. > > I agree and have now updated the pull request with a new commit, > addressing your comments. > Please take a look. > > /Gustaf I think you will also need to check the mode of the previous frame (st->prev_mode) before using internal SILK encoder state. It could have been in SILK DTX some time ago, but then

Hi Is it possible to cross-compile opus-tools with mingw and Ubuntu? So far I have done this:- # prepare $ mkdir $HOME/source $ mkdir $HOME/builds $ export PATH="$PATH:$HOME/mingw-w64-i686/bin" $ PKG_CONFIG_PATH="$HOME/builds/lib/pkgconfig" # Install ogg $ cd $HOME/source $ svn co http://svn.xiph.org/trunk/ogg $ cd ogg $ ./autogen.sh && ./configure

Yes, good point. I added the checking of prev_mode for Silk DTX to avoid using stale data from the Silk state. The PR is updated, and I'm attaching an updated patch. /Gustaf On Tue, 9 Apr 2019 at 12:42, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-08 4:55, Gustaf Ullberg wrote: > > Thank you Mark. > > > > I agree and have now updated the pull request