similar to: opus 1.2.1 errors

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);

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

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 xiph, I'm trying to compile libopus from git.xiph.org/opus.git with emscripten. The regular: $ emconfigure ./configure --enable-fixed-point $ emmake make gets me quite far, but it hangs on the use of __malloc_hook in the tests: tests/test_opus_api.c:1776:16: error: use of undeclared identifier '__malloc_hook'; did you mean 'malloc_hook'? orig_malloc=__malloc_hook;

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

Created corresponding unit test, and the optimization is bit exact with C function. This optimization speeds up SILK encoder on NEON as following. Fixed-point: Complexity 0-5: 0% Complexity 6-7: 6% Complexity 8-9: 10% Complexity 10: 8% Got similar results on floating-point. --- silk/NSQ_del_dec.c | 6 +- silk/SigProc_FIX.h | 4

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

Hi Lang. Thank you. I added pseudo-instructions for spill/reloads and expanded it in expandPostRAPseudo. Regards, Dmitriy. 04.03.2013 8:24, Lang Hames wrote: > Hi Dmitriy, > > As you've seen our current spill code assumes that spill/reloads are > single instructions. I think the best way to work around this is to > introduce load/store pseudo-instructions and expand these

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

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,

Hi Dmitriy, As you've seen our current spill code assumes that spill/reloads are single instructions. I think the best way to work around this is to introduce load/store pseudo-instructions and expand these after register allocation. Cheers, Lang. On Sat, Feb 23, 2013 at 12:15 AM, Dmitriy Limonov <earl at excluzive.ws> wrote: > Hi All. > > I'm writing

The problem is that the drop is only applied (or not) after the subsetting, so what R does is: - Getting the subset, which means a 1 x 1 matrix. - Only then It either returns that as is (when drop=FALSE), or removes ALL dimensions of extent 1, regardless of whether these are rows or columns (or higher dimensions). And it can't keep any names, because what name should be returned? The name

Create the fixed-point intrinsics optimization celt_fir_neon() for ARM NEON. Create test tests/test_unit_optimization to unit test the optimization. --- .gitignore | 1 + Makefile.am | 39 ++++- celt/arm/arm_celt_map.c | 17 +++ celt/arm/celt_lpc_arm.h | 65 ++++++++ celt/arm/celt_lpc_neon_intr.c

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

I read about 4.3.0 and we test this version in polygon. In production we have more than 80 sites and about 200 domain controllers. In test environment we have a lot warning KCC (samba 4.3.), and we suggest in production we have the same picture. Will you make change in KCC for nearest Samba release? Best regards, DMITRIY LUCHKO -----Original Message----- From: samba [mailto:samba-bounces

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

On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > 07.03.2015 0:24, Kevin Harwell ?????: > > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> > wrote: > >> Hello. >> >> Asterisk 13.2. >> I transfer configs from chan_sip to res_pjsip. >> In chan_sip i have