similar to: FEC monitoring

Hello, Is FEC supposed to work in NARROWBAND mode ?(with maxaveragebitrate=12000; maxplaybackrate=8000 ) ?I am having some confusing results, it appears that FEC is enabled in the encoder, but the decoder cannot find any packet with FEC. I am also wondering if this piece of code is correct (webrtc): /* The following is to parse the LBRR flags. */? if (opus_packet_parse(payload,

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, 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

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

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

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

For the last couple months, Nuance has performed extensive testing on how the Opus codec performs in the speech recognition task. I'm hoping to publish a full report in the coming months, but until then all I have is a teaser. Opus performed within about 1% of the WER (Word Error Rate) of unencoded audio. This is compared to about 5% for Speex, which was the previous codec of choice. Well

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

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 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, 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

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 guys,

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

Thank you. Very helpful. > On Jan 27, 2017, at 12:40 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > > On 27/01/17 12:29 PM, Jon Lederman wrote: >> 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? > > Not

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.

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 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