similar to: VOIP: FEC and NARROWBAND

At this bitrate the encoder likely decides that it's better to put all the bits in the normal packet than use FEC. When you enable FEC it steals a lot of bits from the non-FEC content. Also, the use of FEC depends on the reported percentage of packet loss. The more loss there is, the lower the threshold for enabling FEC. Overall, the encoder attempt to make the best decision on a

Hi all, I try to enable FEC in 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

Hello, I understand the encoder provides an option for FEC to provide some protection against packet loss, but I don't understand the details of this arrangement. I'd appreciate answers to the following: * Adding FEC seems to change the encoded audio bit-stream itself, i.e., it doesn't just add additional protection bits, but also changes the encoded bits. This is easy to show by

Hi Dragos, The issue is that not all packet loss maybe congestion related. Often, reducing bitrate seems to have no impact on improving packet loss. Thanks, Manpreet. On Tue, Dec 16, 2014 at 2:09 AM, Dragos Oancea <droancea at yahoo.com> wrote: > > Hi > > You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if > there is packet loss. You know if

Hi, I am looking at the conformance test using opus_compare and the test vectors and I am a bit confused about FEC. Is there a way to validate that a decoder processes LBRR frames correctly? Thanks, Alexis

Hi, I would like to monitor FEC usage in order to include it in RTCP EX or calculate MOS estimation, etc. However the Opus codec library does not seem to expose such information. "Was LBRR found and used or was it PLC ?" I saw in WebRTC that they are using a technique to parse the "frame header" WebRtcOpus_PacketHasFec() It this something that is supported ? What would you

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

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, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks, Manpreet. -------------- next part -------------- An

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

Hi, I'm trying to use the FEC feature. I have a service which does the encoding with OPUS_SET_INBAND_FEC(1) and OPUS_SET_PACKET_LOSS_PERC(20) with 10ms packets. I'm not clear on the decoding process though. 1- When a packet is lost, do I need to call decode with fec=1 ONLY or do I need to call decode with fec=0 after as well? 2- How do I know up front the size of the pcm that I send to

HI All, We are trying to use Opus in a VoIP environment for sending stereo audio. We have noticed a phenomenon where when FEC is enabled and packet_loss_percentage>0, that there is a mixing of audio from the left channel into the right channel and vice versa. That is, rather than hearing each channel in its pristine form as it was in the file, there is a combination of right and left

Hello, We have a use case where we want to record an opus RTP stream to a .opus file. We want to fill in any gaps in the stream, and we also want to take advantage of inband FEC whenever possible. The ogg/opus draft describes[1] how to fill in gaps by generating zero-byte frames, but I do not understand how (and if) FEC can be used. Is this possible, and if so, what is the recommended way of

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Thanks. When you say that with fec enabled, the threshold is increased, do you mean the bitrate - i.e., you need higher bitrate with fec enabled to suppress crosstalk? Also, can you make any recommendations to decide whether to use fec or not? We are trying to tune the parameters appropriately. Audio quality and reduction of crosstalk are critical for us. Thanks again. Jon Sent from my

Dear Opus community, I have a question about the SILK inband FEC. I am using the opus_demo (opus-1.2-alpha) for testing opus codec, and when I choose the "-inbandfec" option I get the exactly the same results when I don't add it to the command line. Are these results logical? Thank you in advance. Best regards, Ayoub. *--------------------BOUZIANE AyoubPhone : (+212) 633 092 157*

Hi, When we pass around 9K samples of only ambient noise (no voice), the encoder which is enabled FEC+DTX is detecting only some 140 frames as non-voice (returning only TOC, no frame content). We were expecting all or more to be identified as non-voice. Our idea was to check how the decoder re-generates the original ambient noise during the silence duration (when we feed NULL to decoder) when

Hi guys,

On 27/01/17 12:16 PM, Jon Lederman wrote: > When you say that with fec enabled, the threshold is increased, do > you mean the bitrate - i.e., you need higher bitrate with fec enabled > to suppress crosstalk? Correct. Another effect I forgot to mention is that Using FEC will actually force SILK/hybrid rather than CELT, so it's possible that disabling FEC makes you use CELT, which

Hi, One other question I was wondering about. Is the reason that we hear the crosstalk with fec and packet loss percentage>0 is that Opus uses information from the left channel to try to error correct the right channel and vice versa? I am trying to understand the origin of the crosstalk. Thanks. -Jon > On Jan 27, 2017, at 12:29 PM, Jon Lederman <jon at soniccloud.com> wrote: >