similar to: concealment

Hi all, I am developing a VoIP application and have used SPEEX for that purpose; I successfully managed to port the codec in fixed-point to an ADSP-21364 processor, and the codec works fine in narrowband, wideband and ultra-wideband mode. Currently, I have chosen, in accordance with the speex manual, a 20 ms frame size in each of these modes: in other words, the input buffer for speex was set to

Narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) compression in the same bitstream What is the meaning of this? can i find more information about this somewhere? And what is packel loss concealment, an you enable this somewhere? thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I cover some VoIP issues. I was at VoIP developer conference and asked an embedded manufacturer (we're talking Wi-Fi phones) about supporting Speex in their embedded products. He said that Speex was good but it's too many things to too many people and that he couldn't supported it in his embedded products for the following reasons. * Code size must be extremely small * Data

Can we get a response from Verisilicon on this? We spoke at the VoIP developer conference and you seemed to indicate that Speex was too difficult for you guys to implement in your embedded products. Mr. Valin would like to know specifically what your criteria is to run on your silicon. He needs to know the following. What is the maximum code size that you can use? What is the maximum data

Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

Hi everyone, I'm happy to announce that Opus 1.5 has just been released. Opus 1.5 is the first release to make extended use of ML in the encoder and decoder. You can read all the details in our demo page at: https://opus-codec.org/demo/opus-1.5/ Major changes since 1.4 include: - Significant improvement to packet loss robustness using Deep Redundancy (DRED) - Improved packet loss

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out

> Err, unless I'm totally wrong, there are a few race conditions. > > Assume the buffer is full of packets newer than the current pointer, and > one that is at the current pointer. > > get and put start at the same time. > > get will find the correct buffer index. Now, just after it finds it's > index, assume we switch to the put thread. > > Put needs

And it turns out there was a missing meson file in the original release so here's 1.5.1 with the file added: https://downloads.xiph.org/releases/opus/opus-1.5.1.tar.gz Cheers, Jean-Marc On 2024-03-04 11:26, Jean-Marc Valin wrote: > Hi everyone, > > I'm happy to announce that Opus 1.5 has just been released. Opus 1.5 is > the first release to make extended use of ML in

Hi, I am an engineering student in IIT Guwahati, India. Currently, as a summer project I am working on 'packet loss concealment in voip scenario'. I am eager to know what PLC methods are used at receiver end of speex decoder. And if I could develop some more efficient method I could able to contribute to the speex. Actually, I searched the whole speex manual but failed to find what

FYI, the README file section on building for Windows says to see the README.md files in the cmake or meson subdirectories, but there are no README files in either of those subdirectories. -----Original Message----- From: opus <opus-bounces at xiph.org> On Behalf Of Jean-Marc Valin Sent: Monday, March 4, 2024 12:52 PM To: opus at xiph.org Subject: Re: [opus] Opus 1.5 is out -- now with ML

Hi, Speex is getting close to beta4, which I'd like to be feature-complete (or as close as possible). That's why I'd like to ask if anyone here has needs for a feature that hasn't been implemented yet. If so, please let me know. For those interested, here's what's going to be in beta4. First, the VBR code has been greatly improved and now works good with wideband too.

Hi, I would like to know which method uses Speex to conceal the loss of packets. I've reviewed the code and haven't found anything. Thanks in advance. Iago Soto. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://westfish.xiph.org/pipermail/speex-dev/attachments/20040614/bfa135e0/attachment.htm

> Obviously this is Jean-Marc's decision and I'm not telling > him not to support this compiler. I am however pointing > out that this compiler is yet more work for very little > payoff. In the case of my project, the proponent of C++ > Builder sent me a huge, monsterously ugly and totally > unmaintainable patch to add C++ Builder support. Needless > to say, that

Somewhere in the new packet-loss concealment code, there is a bug. I tried encoding a 44100-hz stereo file and it decodes just fine, but if I try it with packet loss, I get a segmentation fault. If you want to see the file, I can put it up somewhere publicly accessible. If someone here can't work on it, I'll be back on my home computer in a few days, so I'll work on it then.

> 1) May I know how Speex compared with GIPS codec? It seems that Google, > Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking > or fair comparisons? I think these two emails sum up my opinion about Speex vs. iLBC: http://lists.xiph.org/pipermail/speex-dev/2005-June/003410.html http://lists.xiph.org/pipermail/speex-dev/2005-September/003652.html > 2) In

Hi Does anyone know where one can get hold of a G729 codec for asterisk which effectively can do packet loss concealment using Steve Kann's wonderful new Jitter buffer. The 2 versions that I know of, (digium's and the IPP one) do not perform great at PLC, especially with 4 or 5% loss. Thanks in advance. Clive

Le jeudi 22 septembre 2005 ? 08:19 +0800, Joe Anny a ?crit : > Thanks, Jean for the clarification. My first name's Jean-Marc, not Jean... > I look forward to delving into your tutorial codes. What we hope is to use > Speex as the primary voice codec. We also hope to get a quick start into > the programming aspects with buffer management, denoising, etc. taken care > of (or

> I cover some VoIP issues. I was at VoIP developer conference and asked an > embedded manufacturer (we're talking Wi-Fi phones) about supporting Speex in > their embedded products. He said that Speex was good but it's too many > things to too many people and that he couldn't supported it in his embedded > products for the following reasons. Speex has lots of features

Jean-Marc, I was wondering if you could add a check to ensure that memory is actually allocated during the nb_encoder_init and sb_encoder_init functions. We have been looking at using Speex on a DSP and noticed that if we didn't allocate a large enough heap space memory segment that the DSP would crash. I would recommend something like: if (!st->stack) fprintf(stderr,"ERROR