search for: usedtx

...id is alway true except NaN case and is_silence is alway true except digital zero signal case. In general, following condition will be alway true except exceptional case. (analysis_info.valid || is_silence) But in a code, there is a NOT expression in front of above condition, so st->silk_mode.useDTX will be always disable Here is the code. #ifndef DISABLE_FLOAT_API st->silk_mode.useDTX = st->use_dtx && !(analysis_info.valid || is_silence); #else st->silk_mode.useDTX = st->use_dtx; #endif Is it a bug or are there any reason for this? In my opinion, the...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

...NERGY_MASK(value)); } break; + case OPUS_GET_IN_DTX_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = 0; + if (st->silk_mode.useDTX) { + /* DTX determined by Silk. */ + void *silk_enc = (char*)st+st->silk_enc_offset; + *value = ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.noSpeechCounter >= NB_SPEECH_FRAMES_BEFORE_DTX; + } +#ifndef DISABLE_FLOAT_API + e...

...te=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:119 speex/32000 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 a=rtpmap:96 SILK/8000 a=fmtp:96 maxaveragebitrate=10000 a=fmtp:96 usedtx=0 a=fmtp:96 useinbandfec=1 a=rtpmap:108 SILK/12000 a=fmtp:108 maxaveragebitrate=12000 a=fmtp:108 usedtx=0 a=fmtp:108 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109 maxaveragebitrate=20000 a=fmtp:109 usedtx=0 a=fmtp:109 us...

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

...UEST: > > + { > > + opus_int32 *value = va_arg(ap, opus_int32*); > > + if (!value) > > + { > > + goto bad_arg; > > + } > > + *value = 0; > > + if (st->silk_mode.useDTX) { > > + /* DTX determined by Silk. */ > > + void *silk_enc = (char*)st+st->silk_enc_offset; > > + *value = > > ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.noSpeechCounter >= > > NB_SPEECH_FRAMES_BEFORE_DTX; > &...

...; > +        case OPUS_GET_IN_DTX_REQUEST: > +        { > +            opus_int32 *value = va_arg(ap, opus_int32*); > +            if (!value) > +            { > +                goto bad_arg; > +            } > +            *value = 0; > +            if (st->silk_mode.useDTX) { > +                /* DTX determined by Silk. */ > +                void *silk_enc = (char*)st+st->silk_enc_offset; > +                *value = > ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.noSpeechCounter >= > NB_SPEECH_FRAMES_BEFORE_DTX; > +            } It looks...

...the encoder using the macro OPUS_SET_INBAND_FEC and I set the packet loss percentage to a constant value of 30%, using the macro OPUS_SET_PACKET_LOSS_PERC. Please find my encoder settings below: opus: encoder fmtp (maxplaybackrate=8000;maxaveragebitrate=24000;sprop-stereo=1;cbr=1;useinbandfec=1;usedtx=1) opus: encode bw=narrow bitrate=24000 fch=auto vbr=0 fec=1 expected loss=30 dtx=1 complex=10 At the decoder side when a packet is lost I call the decoder with the next params: opus_decode(ads->dec, NULL, 0, sampv, (int)(*sampc/ads->ch), 0); and set the flag packet_lost=true; When I r...