similar to: Quality degradation with 1.3.1 when using FEC

Hi, I noticed that in some scenarios, Opus 1.2.1 produces better quality than 1.3.1 does. In the use case here, I'm enabling FEC and "transcode" signals from telephony networks (PCMU, 8kHz sampling) to VoIP (48kHz here). In this case, Opus always produced some leakage/ringing above 4kHz but for 1.3.1, these artifacts became worse. The small script below can be used to demonstrate

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, When running opus_demo with FEC enabled, there is a bug related to the length of FEC frames. The FEC frames will be decoded with a length of max_frame_size and the resulting output file will be much longer than the input file.

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

Hi guys,

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

having a hard time communicating on IRC, thank you gmaxwell, very informative. anyway, we were discussing the proper implementation of FEC on the decoder side. well, encoder side is just a boolean thing so that's alright. i gave an example where the receiver lost 5 rtp packets, 1 2 3 4 and 5 during which, we call opus_decode with a null pointer and fec=0 for every packet lost. now, when it

Hi all, Newbie to the group. Just started using Opus as part of a WebRTC project and amazed by the versatility of the codec. Great stuff!!!! I have been trying to understand the performance of Opus inband FEC in the presence of bursty loss. Although I do not have exact characterization of the loss profile, we are seeing issues over WiFi. RTCP reports about 33% loss, but I am guessing a lot of it

What do you mean by "implement"? You're just using the Opus built-in PLC (passing NULL), right? The PLC generally attempts to find periodicity and replicate it. I guess if your signal isn't periodic it can lead to a repetition that isn't great. It's something that could probably be improved in the PLC. Cheers, Jean-Marc On 08/25/2015 01:21 PM, Scott Boekweg wrote:

Hi, I changed the VAD sensitivity by varying VAD_SNR_FACTOR_Q16 and now it shows the difference in silence frames. Thanks Suresh On 4 October 2016 at 17:57, Suresh Thiriveedi <sthiriveedi at gmail.com> wrote: > 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

Hi Jon, On 27/01/17 08:31 AM, Jon Lederman wrote: > 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

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*

The -inbandfec option means that the encoder is *allowed* to use FEC, not that it has to. The final decision is based on the bitrate and the percentage of packet loss (and a few minor factors). If you're not seeing a difference, it means that given the options you're using, the encoder thinks actually using FEC would result in worse quality. Cheers, Jean-Marc On 08/05/17 07:11 PM,

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

In a fork of seanbright's opus patch for 13 there are further patches for Forward Error Correction and Package Loss Concealment, both of which ought to very useful in voip: https://github.com/traud/asterisk-opus Anybody used these patches ? Puzzled why they weren't committed to the main patch. sean

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 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

Hi Jean-Marc, Thank you. Yes, we do need both channels independent. So, if we encode each channel separately, we will be sacrificing the compression ratio we would achieve with stereo encoding, correct? So, based on what you say here is my understanding. Please confirm this is correct or not: 1) If we use fec, we can reduce cross-talk but increasing bitrate. However, that should result in

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