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I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure why I always get this error "warning: > > Invalid mode encountered: corrupted stream?"....

...our client is using a compatible packing, > you'll need to have asterisk do something which requires asterisk to > decode the frames: Like bridge the call to a channel using another > codec, or record the call to a non-speex format (i.e. write to a PCM > WAV file), etc. > > -SteveK > > > > > Jonathan > > > > 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > > > Hi, > > > > > > Does anyone ever used Speex with app_conference in > >...

...In order to determine if your client is using a compatible packing, you'll need to have asterisk do something which requires asterisk to decode the frames: Like bridge the call to a channel using another codec, or record the call to a non-speex format (i.e. write to a PCM WAV file), etc. -SteveK > > Jonathan > > 2006/1/31, Steve Kann <stevek@stevek.com <mailto:stevek@stevek.com>>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard ti...

...on is: *static* *int* *speex_get_samples*(*unsigned* *char* *data, *int* len), and you just point it at some speex data, it it returns the number of samples that are there. Look for it at: http://cvs.sourceforge.net/viewcvs.py/iaxclient/iaxclient/lib/libiax2/src/iax.c?rev=1.71&view=markup -SteveK > Jean-Marc > >Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > > >>jonathan blais wrote: >> >> >>>I'm using Linphone. I tested with Asterisk and Speex only, I created >>>a channel with echo and it worked. It seems to...

Hi folks, Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is

...the bits out, and then you'll know how many you have. If you're trying to stream an existing stream of speex-encoded bits, then it's pretty trivial to parse the stream. I wrote something to do that a long time ago (google speex_get_bits), though it may not do exactly what you want. -SteveK On 11/1/10 10:44 AM, "Jeff Ramin" <jeff.ramin at singlewire.com> wrote: > >Thanks Steve. > >Is there a document anywhere that shows how many bytes/bits of data >are produced by the speex encoding process for a given amount of time >sampling rate and quality...

Have you tried typing "speex rtp" into google code search? It gives lots of examples of real applications which do exactly that. http://www.google.com/codesearch?as_q=speex+rtp -SteveK On 11/1/10 1:13 PM, "Jeff Ramin" <jeff.ramin at singlewire.com> wrote: > >Thanks again Steve. I'll search for the term you mention below. > >What I really want is to take the output of the speex encoder and spit >it out on the network via RTP. I haven't bee...

Hi, Does anyone ever used Speex with app_conference in Asterisk ? I'm having a hard time to figure why I always get this error "warning: Invalid mode encountered: corrupted stream?". Jonathan Blais -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060131/386141a8/attachment.htm

On 08/06/07, Steve Kann <stevek@stevek.com> wrote: > > I'd look at the speech-to-text implementations for this -- I think CMU > Sphinx has done something like this. > Thanks. I had a look at their web pages, and the Sphinx software looks interesting, but I was unable to determine if there is a "hook"...

...ler, etc), their transport > layer, hardware access (audio/video capture), etc. It's all wrapped up > to be part of a javascript API in the browser, but it seems like the > individual components are useable without the rest. > > https://sites.google.com/site/webrtc/ > > -SteveK > As I said last week on this very list.... yes. Steve

...#39;s other options -- you could encode stereo if you wanted to, but it won't be simple, because JSpeex probably doesn't support that, and with 8-bit resolution, there's probably nothing interesting. A better idea would be to try to get 16-bit samples instead of 8-bit samples. -SteveK Vipin Das wrote: > > > > > Hi > I am using Jspeex for my project which requires compression of audio > in realtime..so far i managed to capture sound using java's sound > api.The capturing format i use is 8 bit 8khz ,stereo pcm.The captured > sound is buffere...

...> Jean-Marc Valin wrote: > > >If you set the denoiser to "on" and the VAD to "off", what difference > >does it make in CPU time? > > > > > Same program, running on Athlon XP 1700+: > > Test 1, using VAD, but AGC, denoise off: > > stevek@canarsie:~/work/hms/app_conference $ time ./vad_test > /tmp/demo-instruct.sw 5 > reading from /tmp/demo-instruct.sw, repeating 5 times > read 537760 samples > beginning pass > beginning pass > beginning pass > beginning pass > beginning pass > done. >...

...efore another reason why it's good to be able to set the max AGC gain. One of my users reminded me of it tonight: he wanted to be able to limit the max AGC gain so that when he's not speaking, the AGC doesn't pick up on other sounds from the room. Tom ---- Original Message ---- From: stevek@stevek.com To: tgrand@canvaslink.com Subject: Re: [Speex-dev] Proposed AGC additions Date: Fri Jul 2 20:11:34 2004 >Tom, > > Are you also using the VAD from the preprocessor? Does the VAD >detect this noise as speech? > >I've got this same thing out through a lot of test...

Hello Jean-Marc: On 08/06/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > Either one. The question is: If we treat the software like a black > > box, and we feed in PCM audio, we get Speex encoded data out. Where is > > the information that indicates whether the encoded data contains > > speech or not? The API has a "get VAD status", but it

Changing from using floats to shorts did fix the high pitched tone problem. I'm having other problems but I'll look into it more first. SteveK wrote: > > On May 21, 2006, at 6:33 PM, Kevin Jenkins wrote: > >> When I just copy the microphone input buffer to the output buffer the >> sound plays OK. But if I encode and decode the buffer through Speex I >> get a high pitched constant tone in the background. I...

...but I'm restricted to Java. On 11/01/2010 12:21 PM, Steve Kann wrote: > Have you tried typing "speex rtp" into google code search? It gives lots > of examples of real applications which do exactly that. > > http://www.google.com/codesearch?as_q=speex+rtp > > > -SteveK > > > On 11/1/10 1:13 PM, "Jeff Ramin"<jeff.ramin at singlewire.com> wrote: > >> Thanks again Steve. I'll search for the term you mention below. >> >> What I really want is to take the output of the speex encoder and spit >> it out on the net...

...derstand why. It seems to look like that i can get throught the whole speex code. I now try Toms solution. After that i will try to find out how the speech_calculate_vad() function is workin. Tanks for your help, thanks for your time. ----- Original Message ----- From: "Steve Kann" <stevek@stevek.com> To: "Lis" <lis@1234567890qwertzuiopasdfghjklyxcvbnm.de> Cc: <speex-dev@xiph.org> Sent: Thursday, March 02, 2006 3:33 PM Subject: Re: [Speex-dev] Voice Activation Level (speex 1.1.11.1) > > Lis, > > The Voice Activity Detection (VAD) algorithm in...

...n approved standard, it's more standardized than a lot of interoperable traffic on the internets these days. The RFC specifies packetization guidelines, which is basically that you put one or more frames in a packet, and then pad the rest with 0 bits until you have a while number of octets. -SteveK On 11/1/10 9:55 AM, "Jeff Ramin" <jeff.ramin at singlewire.com> wrote: > >I need to stream speex-encoded audio over RTP, which doesn't seem >to be standardized yet, so I'm gonna roll my own code. I control both the >sending and receiving sides, so I can prett...

...ning pass beginning pass beginning pass beginning pass done. real 0m5.359s user 0m4.301s sys 0m0.024s <p>==================== So, it doesn't seem to make much difference. I also ran the code, unoptimized, with oprofile. I'll send results from that to you separately. -SteveK --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages...

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.