similar to: Convert frame Ultrawideband to narrowband

Hi List, Now I will send to you more specific what I am trying to do. I have one Asterisk Channel where receives Midia Frames in the codecs format: Speex UltraWideBand and Speex NarrowBand. When I use Speex NarrowBand the Asterisk is able to convert this frame to G711. But when I use Speex UltraWideBand the Asterisk don't convert it. Then I need in my Asterisk Channel Source include the Speex

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

The resampler will convert raw audio from one sample rate to another. You are starting with Speex-encoded audio frames. So, what you need to do is: 1. Run the media frames through the Speex decoder (look at Speexdec or testenc_uwb). This will give you raw audio with a 32 KHz sample rate. 2. Use the resampler to convert the audio from 32 KHz to 8 KHz sample rate. 3. Run the 8KHz audio

Hi List, Would someone can to explain me the main difference in SWITCHES or ESWITCHES in AEL. context default { switches { DUNDi/e164; IAX2/box5; }; eswitches { IAX2/context@${CURSERVER}; }; }; All the best, Thiago -- ---------------------------------------------------------------- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malufrj at

Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to

Acrescentando o report do Dell, os equipamentos da Khomp s?o homologados pela Anatel - funcionamento normalmente nas implementa??es de Asterisk puro, FreePBX ou Elastix. Caso desejem mais informa??es sobre equipamentos da Khomp, consultem a CAM Tecnologia. A CAM Tecnologia atua com revenda ou venda direta da khomp para o cliente final. Contato: Rubens Duarte de Andrade Tel: (21) 3189-1050

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

----- Original Message ----- From: "Conrad Parker" <conrad at metadecks.org> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca>; <speex-dev at xiph.org> Sent: Thursday, November 13, 2008 1:18 AM Subject: Re: [Speex-dev] SPEEX on iPhone ? > 2008/11/13 Vincent Burel

Hi Peter, Have you considered resampling the raw 44.1kHz stereo source files using a program such as http://audacity.sourceforge.net/, say to 16kHz mono or 32kHz mono, and then using the wideband or ultrawideband speex modes to encode the result? Alternatively, if you want to programmatically do the resampling yourself, you could try the new resampling API in the svn head of speex, or the GPL

My wish for 1.1 is to stream my station with VBR at bitrates averaging 28-32kb/s with a minimum 32khz sample rate. Hopefully this will be achievable with the new low bitrate tuning that is planned. This could put it on a par with Real Audio at 32kb/s. Currently a quality of -1, 32khz sample rate produces an average around 34-38kb/s which is too high for modems. I could currently use a managed

I've been playing with speex for use in a VoIP application between PC's. One thing I've found (correlating to the documentation) it that speex runs much faster and produced much better output when it's fed a 32kHz signal instead of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate and feed it 44.1kHz or tell it 32kHz and feed it 44.1kHz. What part of the

hi on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. does anyone provide me a reference to some code that does that? are there any trick to do that? i tried to add to subsequent sample but the result was very bad. what are the requrment from a decimation filter for audio? thanks, nir -------------- next part

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call

Hello Mailing List, I am a Speex supporter and user that would really like to know how much memory Speex uses to decode a 8kHz, 16kHz and 32kHz (primarily the 8kHz) and is it possible to use a 1kBytes of RAM to decode a 8kHz stream? (I was thinking of the possibility of using a ATmega168 to decode Speex) //P?r, Sweden

At this bitrate the encoder likely decides that it's better to put all the bits in the normal packet than use FEC. When you enable FEC it steals a lot of bits from the non-FEC content. Also, the use of FEC depends on the reported percentage of packet loss. The more loss there is, the lower the threshold for enabling FEC. Overall, the encoder attempt to make the best decision on a

Hello, playing a little around with speexenc/speexdec, I discovered that if I pass the "-n" or "-w" command line option to speexenc (version 1.1.10), I get a segmentation fault. Looking closely at the code, I found that the line: fprintf (stderr, "Warning: Trying to encode in %s at %d Hz. I'll do it but I suggest you try ultra-wideband instead\n",

Hi all, When encoding narrowband with speexenc I notice the bitrate is the same for qualities 3 / 4, 5 / 6, and 7 / 8 (8000, 11000, and 15000 respectively). I didn't test this on prior versions of the encoder so I'm not sure if this is something new, but am curious if there is a difference between the above respective settings (does quality 4 yield better results on narrowband than

Hi, What is the MIPS count for the narrowband decoder for 24.6 kbps mode??? Also what is the expected MIPS count on say C55x after optimization?? Thanks in advance Regards, Naveen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070330/634df5de/attachment.htm