similar to: FEC-related bug in opus_demo.c

--- src/opus_demo.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/src/opus_demo.c b/src/opus_demo.c index 72506b5..5a75679 100644 --- a/src/opus_demo.c +++ b/src/opus_demo.c @@ -245,14 +245,14 @@ int main(int argc, char *argv[]) double bits=0.0, bits_max=0.0, bits_act=0.0, bits2=0.0, nrg; double tot_samples=0; opus_uint64 tot_in, tot_out; - int

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,

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*

I didn't expect this, is it normal? When you use inbandfec, and have packet losses, you end up with more audio samples than you started with. With short frame (2.5), the FEC isn't supposed to do anything but it shouldn't do that should it? and 20mS frame, should do 'something' but? add audio? I'm using opus_demo for this. Here's the steps (I have my test audio in

Hello! I encoded a voice file (48kHz) with opusbin\opusenc.exe with the standard settings and decoded it. The output was amazing. I could not hear any loss at all. Then i encoded the same file with opus_demo.exe and standard settings and then decoded it. The output had a sizzling noise, even when I used full bandwidth. I think I have played around with any of the settings in opus_demo.exe,

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 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

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, I am interested in this part of the answer: "Make sure both are also linked to the same libopus (opusenc could be using the system libopus 1.0.x while opus_demo is from git)" I dowloaded a tarball distrib of libopus (1.1-beta),build and installed it,and afterwards did the same with opus-tools (0.1.7),using the former installed libopus library (also using libogg 1.3.1). When I execute

I suspect the main difference is due to time alignment of the samples and the exact trimming at the beginning and at the end of the encoded files. I also note that there's a missing "--" in front of "bitrate" and only one "-" instead of two for cvbr in your opusenc command line. Make sure both are also linked to the same libopus (opusenc could be using the system

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,

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

Hi I used opus_demo to clarify how opus works, I try this command: opus_demo -e" audio 8000 1 8000 -cbr /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus And I get the file test.opus, then comes the first problem, this file hasn't header so i can't reproduce it using an external tool, how can i add a header to this file Then when i trie to run opus_demo -d 8000 1

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

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

On 2013-10-21 5:08 AM, Antonio Juan wrote: > I used opus_demo to clarify how opus works, I try this command: > > opus_demo -e" audio 8000 1 8000 -cbr > /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus opus_demo doesn't support .opus files. It uses a special raw format just for testing. You want opusenc and opusdec from the opus-tools package. -r

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

This patch lets you pass paths for opus_demo and opus_compare to the run_vectors.sh script. The motivation for this was for my aarch64 patches (forthcoming), which I've been testing under qemu. I want to run an emulated opus_demo, but a native opus_compare, so the script completes in a reasonable amount of time. --- tests/run_vectors.sh | 4 ++-- 1 file changed, 2 insertions(+), 2

Hello OPUS, It appears to me that multi-frame packets are not supported by opus_demo.c (e.g. 30ms packest containing 3 10ms frames), can you confirm. However RF6716 does support such packets. Thanks Cliff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20131121/56bab79a/attachment.htm