similar to: Using FEC

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

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 Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

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,

For some reason the attachment did not go through. Re-attaching. From: Mukund Raman Sent: Wednesday, April 01, 2015 6:12 PM To: 'opus at xiph.org' Subject: Opus multi-stream/surround: Audio corruption on decoded content Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I

Hi. I am currently trying to run some Spearman correlations, and have encountered two issues. 1) When using cor.test() with a variable that includes ties, I get the "Cannot compute exact p-values with ties" error. I have read that this function now uses an asymptotic formula that allows for ties, so do not understand why I am getting this error. (I am running version 2.4.0.) I

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

Jean-Marc, Thanks for the response and the helpful info. I am trying to get the library to build without using the pseudostack define, and use either VAR_ARRAYS or ALLOC, but it seems the global stack is not defined. Where do can I define this in my example? VR -----Original Message----- From: Jean-Marc Valin [mailto:jmvalin at jmvalin.ca] Sent: Tuesday, February 20, 2018 5:40 PM To:

Hi everyone, I have a question on selecting and grouping elements of a data frame. For example: A.df<- [ a c 0.9 b x 0.8 b z 0.5 c y 0.9 c x 0.7 c z 0.6] I want to create a list of a data frame that gives me the unique values of column 1 of A.df so that i can create intersects. That is: B[a]<- [ c 0.9] B[b]<- [ x 0.8

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I am a new R user, and am looking to call Postgres through R. I am not so technical and need help in setting up. I have downloaded the tar file but cannot get passed this step. Is there documentation for dummies to get working with this? I don't even know where I should save this file? [[alternative HTML version deleted]]

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 guys,

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

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

I am specifically using Celt Wideband (48kHz) over WiFi multicast that naturally leads to lost packets and am trying to minimize the impact to the audio. I implemented PLC but the audio it produces is robotic. Have I implemented PLC correctly? Checking the waveform it is using the previous received waveform to fill in a missing packet but not the full waveform so it has to repeat. Basically,

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, 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: >