similar to: FEC

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

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 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

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

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, 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

1 is the way to use FEC. But: How do I know if this *previous* packet contains FEC information? If not, I want to fallback to invoking PLC for obtaining samples. Or does the decode_fec call automatically use PLC internally as a fallback if FEC is not available? Thanks! --001a113aba02fce1df050d475c7b Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: quoted-printable <div

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 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

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, 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

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, 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: >

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

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 necessarily. Stereo makes two assumptions: 1) It assumes the two channels are somehow correlated 2) It assumes the two channels are meant to be

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

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

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,

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