similar to: PLC Sounds Robotic - How to Implement FEC Wideband

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:

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,

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, 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, 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

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

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*

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, 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, 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: >

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

Hi guys,

Hi again, I tested lower cumul_gain limit values. In fact, I reduced the value to 240000, 230000, 220000, ..., 20000, 10000, 5000, 2000, 1000, 500. (quality 7, complexity 2) For input signal 2000 Hz, the values 190000, 130000, 60000, and 1000 look o.k., all others don't. For 500, the (zoomed out) waveform shows some kind of amplitude ripple -- i think, this value is definetely too low. I

I have three time series, x, y, and z, and I want to analyse the relations between them. However, they have vastly different resolutions. I am writing to ask for advice on how to handle this situation in R. x is a stimulus, and y and z are responses. x is a rectangular pulse 4 sec long. Its onset and offset are known with sub-millisecond precision. The onset varies irregularly -- it doesn't

Hello, I am trying to install R packages under linux, some of the packages can not be installed and I got the following error, could anybody give me suggestion on where the problem is and how to fix it? Thanks-e > .libPaths() [1] "/usr/lib64/R/library" [2] "/usr/share/R/library" [3]

I am puzzled by a filtering problem using fft(). I don't blame R. I have a waveform y consisting of the sum of 2 sinewaves having freqs f1 and f2. I do s = fft() of y. Remove s's spike at freq=f2 Do inverse fft on s. The resulting waveform still has a lot of f2 in it! But the filtering should have removed it all. What is going on, and how to fix?? Thanks very much for any help. Bill

Hi all! I would like to use Ogg-Vorbis to encode a non audio waveform. My waveform is in .wav format, on 16 bit mono, with frequency range from 100Hz to 100MHz. It's about 100MB lenght. I need to compact it with lossy for net transfer. Is there something like this, already done, that can help me ?? How can I measure the distortion that Vorbis introduce? I'm sorry for my bad english.

Hi Jean-Marc, do you have any other ideas what to look for? Or do you currently debug yourself? As I already wrote, I'm out of ideas... best regards, Frank Frank Lorenz <Frank_wtal at web.de> hat am 18. Januar 2010 um 16:39 geschrieben: > Yes, I did. > > As mentioned earlier, only the enhancer inside the docoder produces a lot of overflow messages (it points to lines 68