similar to: PLC and complexity?

Hi, I have read the mailing list archives and have a question regarding PLC (packet loss concealment).

hi, i'd like to ask for a suggestion for implementing PLC (packet lost concealment). right now, i don't have PLC applied in my application. when voice data arrive from the network, i just play it right away. does having PLC require that you play something every interval? like if im expecting a voice packet say every 20ms, then i play something at that rate. either the packet that came

Hi, I've try to enable the new loss concealment code with celt-0.6.1. I've run "configure --enable-new-plc" but I've got a missing plc.c file while compiling. It is also missing for version 0.6.0. Can you release this missing file? Thank you, Pascal CRC

On 2010-11-10 12:58, Renato Golin wrote: > In a nutshell, the design decisions should be communicated more > effectively, and a Wiki is a great place to start. Peer reviewers > should communicate via the Wiki, so patchers could learn and plan > before the next iteration and reduce the cost for everybody. You may consider using a review tool rather than (or in addition to)

In a fork of seanbright's opus patch for 13 there are further patches for Forward Error Correction and Package Loss Concealment, both of which ought to very useful in voip: https://github.com/traud/asterisk-opus Anybody used these patches ? Puzzled why they weren't committed to the main patch. sean

Thanks for the reply. I think I'm beginning to understand but can you clarify a bit more. 1) I am confused on "trade error-free quality" part. Basically the encoder is doing something to the encoded data by looking at the SPEEX_SET_PLC_TUNING parameter? 2) How does the receiver side determine whether the packet loss occured or not? It seems only way is assigning

Hello I've been using VoIP over a not so reliable net: I usually get a 5% to 10% packet loss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge

Hi,I want to know how to do to work PLC of Asterisk. Anyone plz help me. PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org or release note. And I see in codecs.conf, "genelicplc" setting. So I put codecs.conf in '/etc/asterisk' ,and wrote "genericplc => true". And I worked Asterisk and tested. I think PLC like this. When I send 3

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

I've been working on improving packet loss concealment in my VoIP client -- I hadn't actually realized speex performed some PLC until I was looking at nb_decode(). There are two things I'd like to do, and I think both will require some modification to nb_decode() and nb_decode_lost(). - I'd like to have my frames overlap by some number of samples. This is intended to reduce the

I've been working on improving packet loss concealment in my VoIP client -- I hadn't actually realized speex performed some PLC until I was looking at nb_decode(). There are two things I'd like to do, and I think both will require some modification to nb_decode() and nb_decode_lost(). - I'd like to have my frames overlap by some number of samples. This is intended to reduce the

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

Hi, I'm a developer currently using the speex codec in a VOIP application of ours.. It sounds amazing, especially at wideband.. my question is how do I force it to do a concealment? We have a low latency application, and based on the current API, I'm guessing concealment only kicks in when a packet is lost.. However, our jitter buffer knows when a packet is missing, and I'd like to

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

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

On 2011-08-09 16:48, John Criswell wrote: > On 8/9/11 6:49 AM, Bjorn Reese wrote: >> I have an optimization pass (FunctionPass) where I need to add global >> constructors. >> >> For cleaness sake I decided to add these in my own module. My module >> is created in my FunctionPass constructor: > > This is not how I would do it. A FunctionPass has

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, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

I have an optimization pass (FunctionPass) where I need to add global constructors. For cleaness sake I decided to add these in my own module. My module is created in my FunctionPass constructor: MyPass() : FunctionPass(ID), myModule("my_module", getGlobalContext()) {} I generate an llvm.global_ctor global variable in my module, and I add my global constructors

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems