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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, 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

This is observed on a live call between webRTC browser client and another legacy client. Our server is there in between and transcoding from opus to another codec and this is observed while decoding the opus. Anyway, I'll try to capture/dump the packets in the server before feeding to the opus_decode and share with you. But this will not have the first 8 bytes (length+enc range) to directly

I just tried decoding with v1.1: ./opus_demo -d 48000 2 opus_encoded_crash.opus out.pcm and I see no issue (including with valgrind). Does the same command-line create problems for you? What compile flags did you use? fixed-point or float, any assembly, ...? Could be assembly here, or even a compiler bug wouldn't be unheard of. Cheers, Jean-Marc On 20/04/15 07:27 AM, Suresh Thiriveedi

Hi Jean-Marc, Could you please update if you got a chance to look into. As I mentioned, I don't see the same issue in 1.1.1, but I don't see any difference in 1.1.1 other than optimization based on the architecture. This optimization could have fixed some stack overflow issue in some specific cases? Thanks Suresh On 13 April 2015 at 12:39, Suresh Thiriveedi <sthiriveedi at

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 Jean-Marc, Thanks for your response. Please find the details as below. *Backtrace we got for this crash:* #0 0x0000000000800c54 in opus_decode_frame (st=0x38906b8f99d09c5, data=0xf0aa10b4ef1008ae <Address 0xf0aa10b4ef1008ae out of bounds>, len=-188613428, pcm=0x6e80016085efd57, frame_size=44037315, decode_fec=58716895) at src/opus_decoder.c:384 #1 0x00000000008009c0 in

Hi, We are able to reproduce the issue with the 1.1 opus_demo (sample file). We captured the frames in our server just before the opus_decode and fed the file to opus_demo (1.1) and it is crashing. Same file is tested with 1.1.1 and it is fine. So this is in line with our server testing observation and I think here we can conclude that the 1.1 library is crashing while handling a specific mode

To be decodable by opus_demo, you'll have to add the 8-byte "header". Just put in the length of the packet followed by "0" for the encoder range (0 means "not present"). That being said, from previous experience, the most likely cause of the crash is a bug in your software causing a corruption in Opus. So it's safe to assume that if you can't reproduce

Still can't reproduce. What OS and compiler version? Jean-Marc On 21/04/15 02:48 AM, Suresh Thiriveedi wrote: > Hi, > > There is no change in the compiler flags. I'm using as it is from the > original code. No change in the Makefile and I believe it is using the > floating point only by default. > > We are using 8k samples and mono so the commands is as follows.

Hi Looks like 1.1 version is sensitive to the system architecture/compiler/kernel version. Below is my observation in various linux system I have. As you mentioned, you are not observed the crash in your system, can you please share your system details. And also please comment on the below table/observations. *Machine IP* *optimization flags* *RHEL version* *kernel version* *gcc version*

Please provide the input file that produces this with opus_demo. On 16/04/15 03:24 AM, Suresh Thiriveedi wrote: > Hi Jean-Marc, > > Could you please update if you got a chance to look into. As I > mentioned, I don't see the same issue in 1.1.1, but I don't see any > difference in 1.1.1 other than optimization based on the architecture. > This optimization could have

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

Hi, There is no change in the compiler flags. I'm using as it is from the original code. No change in the Makefile and I believe it is using the floating point only by default. We are using 8k samples and mono so the commands is as follows. [root at MEDIA opus-1.1]# ./opus_demo -d 8000 1 opus_encoded_crash.opus opus_encoded_crash.pcm *And segmentation is as below..*. ............ Calling

Red Hat Enterprise Linux Server release 6.4 (Santiago) gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) We see the issue in all our Intel based Linux servers. Thanks Suresh On 21 April 2015 at 12:41, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Still can't reproduce. What OS and compiler version? > > Jean-Marc > > On 21/04/15 02:48 AM, Suresh Thiriveedi

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 All. I have an application where the sample rate of the sender and receiver can vary by a small margin and the latency needs to be maintained within bounds and can't drift significantly and the system has to be able to cope with clock mismatches up to 0.5%. For example, the sender may have a clock rate of 48.1kHz and the receiver may have a clock rate of 47.9kHz. Unfortunately the clock

Opus 1.0.2 fixes an out-of-bounds read that could be triggered by a malicious Opus packet by causing an integer wrap-around in the padding code. Considering that the packet would have to be at least 16 MB in size and that no out-of-bounds write is possible, the severity is very low. This new release also has the following changes: Quality-impacting - Changed the behaviour of the PLC to always

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:

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,