similar to: looking for consulting assistance for opus

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

hi folks apologies for sending a commercial mail. I am the CEO of http://talk.to - a unified messaging product. We are still in early stages of launch. We intend to build in voip capabilities in our product. Our product is intended to work in browsers and on all mobile platforms. We are looking for someone who can provide us an ARM-Neon optimized version of Opus for use on mobile platforms so

Hi Jean-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but I hear

What's the impact on encoded speech quality (per given bitrate) when the encoder cpu complexity is reduced all the way down for Opus? Rather, how big is the impact? Secondly, can someone comment on wideband speech quality comparison between Opus and iSAC with and without the cpu complexity of Opus turned all the way down? Thanks! -------------- next part -------------- An HTML attachment was

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

OK. Different mailing lists are set up differently. This list is unusual because your answers only go to the person who replied to you. So if you want the other people on the listserv to see your answer, you should make sure that Speex-dev at xiph.org<mailto:Speex-dev at xiph.org> is added to the TO: field of your outgoing message. Hopefully someone else will also attempt to answer your

Ben Greear wrote: > Francois Menard (Mailing List Account) wrote: > >> >> did you try speex in wideband mode ... what bitrate? >> >> f. > > > 15kbps mode. It does significantly better with a different > speaker (the male does better than the female), as well. Most CELP based codecs do a better job with lower pitched voices, so male voices tend to sound

Francois Menard (Mailing List Account) wrote: > > did you try speex in wideband mode ... what bitrate? > > f. 15kbps mode. It does significantly better with a different speaker (the male does better than the female), as well. I'm considering purchasing a commercial PESQ license and wrap the PESQ software in a server application. I could then allow other folks to run PESQ

Hello, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it "registers" incoming calls and works through the asterisk

Wang, Chris wrote: > Benjamin, > > Thanks for the prompt response. Are there other recommended methods to > verify encoder implementations? There are few if any requirements placed on the encoder by the actual RFC, however there are a number of things you can do which provide various amounts of assurance your encoder is working as expected. The first test I would run is to write

Hi Opus list, I'm noticing a discontinuity in the quality between use of 10ms and 20ms framesize for mode=AUDIO at a bitrate of about 28000. Quality drops audibly for voice signals when encoded at 20ms framesize, versus quality at 10ms. This effect is mode=AUDIO only. Using mode==VOIP shows no sig. difference between 10 and 20ms framing at this bitrate. Pesq totally

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Can anyone work out why my callerid doesn't work on my isdn4Linux with asterisk (or without asterisk for that matter)... This used to work fine, and I am quite confident that the telco is sending callerid information (because they always do on all ISDN lines standard, only extra cost on POTS lines). This is the information from dmesg, whether asterisk is running or not: isdn_net: Incoming

Hi Everyone. I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer >From the sdp can anyone suggest why secure audio cannot be provided ????v=0 ????o=- 6611325078116277019 2 IN IP4 127.0.0.1 ????s=- ????t=0 0 ????a=group:BUNDLE audio ????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l ????m=audio

Jean-Marc Valin wrote: >> What commercial VOIP test products can I turn to in order to objectively >> evaluate your CODEC; in order to analyze audio streams for MOS scores and >> performance parameters? >> > > The MOS evaluation is subjective, not objective and actually involves > getting (many) real people to listen to the audio, not just buying a >

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

Hello everyone, I have been using an AVM Fritz! card with chan_capi and mISDN for quite a while in PTM mode and it was working finely. Now, I needed more DID/MSN, so I switched to PTP. But now nothing works anymore :( I am using Asterisk on Debian Sarge stable and installed Asterisk along with chan_capi from apt-get. I installed mISDN from the CVS of isdn4linux.de. It is : - Asterisk

Hi, There is a lot of speex quality estimations. One of this comparative estimation is even available on the official site <http://speex.org/comparison/>. I'd like to present yet another one. And I thought that the best place for this presentation would be Speex-dev mailing list. I want to get feedbacks and criticisms please. If Speex authors consider to make some parts of this work

Hi Jim, First of all - thanks, turning the highpass filter off was what I needed, and the waveforms match now. But, when i did the PESQ tests again I found an interesting result : version 1.0.5 still got a slightly better average score, but the standard deviation on version 1.2 beta1 was much smaller. The cause for that is this - on some samples versions 1.0.5 and 1.2beta2 produced a single