similar to: OPUS encoded data size and bandwidth of encoder

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

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

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 am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

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 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

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Jean-Marc, Thanks for the response and the helpful info. I am trying to get the library to build without using the pseudostack define, and use either VAR_ARRAYS or ALLOC, but it seems the global stack is not defined. Where do can I define this in my example? VR -----Original Message----- From: Jean-Marc Valin [mailto:jmvalin at jmvalin.ca] Sent: Tuesday, February 20, 2018 5:40 PM To:

Hi All. I am seeing the following unexpected behavior with OPUS_SET_MAX_BANDWIDTH. I expect that setting this to OPUS_BANDWIDTH_NARROWBAND would give similar results to passing an 8Khz sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl

Hi everyone, Some time ago, I sent a pull request <https://github.com/xiph/opus/pull/107> to the Opus github page. Jean-Marc asked me to post it to the mailing list so everyone can have a look at it. You can find the description and code changes below. Please let me know if you have any questions or concerns. Best regards Gustaf Ullberg In WebRTC, we would like to be able to

Hi! Short version of my question: How to produce Opus frames which can be safely concatenated and how to embed them into a WebM file? Long version: I'm currently implementing a web-based audio player which streams audio as opus/WebM using the HTML5 media source extensions. Currently, the server decodes a set of input files to a fixed RAW audio format (stereo, 48000 kHz) and encodes the

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

Here is a simple patch to fix a bug in opusenc's set-ctl-int code -- Thanks, Michael Graczyk -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fix-potential-uninitialized-access-for-set-ctl-int.patch Type: text/x-patch Size: 992 bytes Desc: not available URL: <http://lists.xiph.org/pipermail/opus/attachments/20160425/22994ffa/attachment.bin>

Hello Joshua, I understand. My main aim to first understand the processing overhead when decisions have to be made for a fixed audio in signal, reduce it so that I can then begin making an update to reduce the memory (on chip flash and RAM) footprint. Thanks Amit On Thu, Dec 31, 2015 at 7:58 PM, Joshua Bowman <silverbacknet at gmail.com> wrote: > Opus will always use whatever mode it

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,

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

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 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