similar to: Need to set OPUS_SIGNAL_VOICE for DTX ?

Thank you Mark. I agree and have now updated the pull request with a new commit, addressing your comments. Please take a look. /Gustaf On Fri, 5 Apr 2019 at 11:41, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-01 3:37, Gustaf Ullberg wrote: > > Hi everyone, > > > > Some time ago, I sent a pull request > > <https://github.com/xiph/opus/pull/107>

Hi everyone, Some time ago, I sent a pull request <https://github.com/xiph/opus/pull/107> to the Opus github page. Jean-Marc asked me to post it to the mailing list so everyone can have a look at it. You can find the description and code changes below. Please let me know if you have any questions or concerns. Best regards Gustaf Ullberg In WebRTC, we would like to be able to

Hi. I found that DTX is always unavailable when DISABLE_FLAOT_API is not defined in OPUS_AUTO mode. As you know analysis_info.valid is alway true except NaN case and is_silence is alway true except digital zero signal case. In general, following condition will be alway true except exceptional case. (analysis_info.valid || is_silence) But in a code, there is a NOT expression in front of above

Yes, good point. I added the checking of prev_mode for Silk DTX to avoid using stale data from the Silk state. The PR is updated, and I'm attaching an updated patch. /Gustaf On Tue, 9 Apr 2019 at 12:42, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-08 4:55, Gustaf Ullberg wrote: > > Thank you Mark. > > > > I agree and have now updated the pull request

I'm running a: for band in n w u do for quality in 0 1 2 3 4 5 6 7 8 9 10 do for complexity in 3 do for vad in "" "--vad" do for dtx in "" "--dtx" do echo speexenc -${band} --quality ${quality} --comp ${complexity} ${vad} ${dtx} input.${band}.wav

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: We noticed that opus reconstructed noise is pulsing with a 400ms pattern when dtx is enabled in silk mode. This is independent of the background noise level and is found with speech + non-speech period test files as well as variable level noise-only test files. This issue can be reproduced with opus v1.1 using this command: ./opus_demo voip 16000 1 25000 -dtx input.bin

Hello, I am trying to use the DTX feature of speex to tell if the user is speaking. I have VBR, DTX and VAD turned on, but the function speex_encode always returns a non-zero value. It is my understanding that it should return a zero value when the encoder detects silence? Could anyone give me any clues to help figure out why this isn't working for me? I am using the 1.13 version of speex on

Hello, I'm try to run speex on some ARM processor. I'd like to cut away some speex features including VBR, ABR and preprocessor while still supporting VAD/DTX. But I've found some puzzles in the source code regarding VAD as below, 1) there are VAD related codes in the source file preprocess.c. Will VAD still work If I don't use the source file preprocess.c? 2)The speex manual

Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last

Could a method be added to use the VAD from the preprocessor to send the packet with the 'dtx_enable' flag in it, so the decoder on the other end knows it's in dtx mode? (and SPEEX_GET_DTX_STATUS returns 1) At the moment, I can hack around it by doing something like: if (VAD for prev frame && ! VAD for this frame) { iArg = 0;

Hi there, I enable DTX and VAD in speex_encode to detect whether there is silence in voice data. But the same programme got different result on my computers. The function speex_encode always return 38 on one of my computers although it is silence. On all other computers the function speex_encode return 1 when it is silence. It means DTX and VAD doesn't work on particular OS or PC. Do anyone

On 2019-04-08 4:55, Gustaf Ullberg wrote: > Thank you Mark. > > I agree and have now updated the pull request with a new commit, > addressing your comments. > Please take a look. > > /Gustaf I think you will also need to check the mode of the previous frame (st->prev_mode) before using internal SILK encoder state. It could have been in SILK DTX some time ago, but then

On 2019-04-01 3:37, Gustaf Ullberg wrote: > Hi everyone, > > Some time ago, I sent a pull request > <https://github.com/xiph/opus/pull/107> to the Opus github page. > Jean-Marc asked me to post it to the mailing list so everyone can have a > look at it. > > You can find the description and code changes below. Please let me know > if you have any questions or

Hi Marc, Thanks for your quick response. So if VAD is enabled then VBR will be enable although it's a special VBR. How about take out the VAD code from the VBR and remove the code else? Lianghu On 10/24/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > 1) there are VAD related codes in the source file preprocess.c. > > Will VAD still work If I

Dear Jean, Thank you so much for reply my message. I am sorry the function return 38 is speex_bits_write(). This function doesn't work on one of my computers. But after I add the function speex_preprocess(), it works well. I am wondering what is the possible reason? What the function speex_preprocess() does? Another problem I have now is when I enable dtx and vad with function

> Hi, > > It could (and should) definitely be done, but it's somewhere > (unfortunately not on top) on my TODO list. I think the simplest way > (until I really integrate the preprocessor with the codec, which will > take a while) to do it is to add a speex_encoder_ctl() call as you > proposed. I'd call it SPEEX_SET_EXT_VAD_STATUS and I'm willing to merge > a

Hi everyone, I've just released Opus 1.2-alpha. It comes with many improvements, including: - Speech quality improvements especially in the 12-20 kbit/s range - Improved VBR encoding for hybrid mode - More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s - Music quality improvements in the 32-48 kb/s range - Generic and SSE CELT optimizations - Support

Xiph.Org is pleased to announce that we've released Opus 1.2. The 1.2 release includes: . Speech quality improvements especially in the 12-20 kbit/s range . Improved VBR encoding for hybrid mode . More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s . Music quality improvements in the 32-48 kb/s range . Generic and SSE CELT optimizations .

Hello Opus community, I'd like to ask you for advice and recommendations. WebRTC uses Opus, and I noticed https://webrtc-codereview.appspot.com/5549004 started referring to currently internal Opus headers. This is possible because for Chromium the Opus sources are just checked in, so any header can be #included. I detected this when trying to package Chromium for Linux distributions with