similar to: Obtaining loudness information in 1.2beta2

What would be a good parameter to return that would better represent the loudness of the signal? As for the your other comments, I'll make the necessary changes and resubmit. Thanks, Mihai On Oct 25, 2007, at 7:45 PM, Jean-Marc Valin wrote: > I'm in favor of the idea, but not of the current implementation. There > are two problems: > 1) The st->loudness parameter

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Mihai Balea wrote: > What would be a good parameter to return that would better represent the > loudness of the signal? > As for the your other comments, I'll make the necessary changes and > resubmit. Could just be to first take the 5th root, a bit like in the code. Jean-Marc > Thanks, > Mihai > > On Oct 25, 2007, at 7:45 PM, Jean-Marc Valin wrote: >

I'm in favor of the idea, but not of the current implementation. There are two problems: 1) The st->loudness parameter doesn't represent the amplitude of the signal, but the amplitude^5 (fifth power). That makes it hard to use 2) The call returns a float, but it should really be an int (spx_int32_t) so that it can work easily with fixed-point code There's a last minor thing:

Did you check it against the trunk in SVN? If it's not applied, and you can hook Jean-Marc up with an email address like yours, I'm sure he will get right on it. :) Tom Mihai Balea <mihai@hates.ms> wrote: > > Hi all, > > Did anything happen to this patch? > It seems to me that it fixes a valid issue, but I'm not an expert. > Anyways, I didn't see

Hi, Does someone know if I could use SpeexPreprocessState::loudness or SpeexPreprocessState::loudness2 to create a VU meter. And if so, how? <p>Thanks, Tom Moers --- >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

> As you can tell, the AAGC integration with speex was really a classic > hack. Instead of re-creating the hack, what's probably best here is to > integrate AAGC back into speex, and have a proper API. Agreed here. If you can come up with a clean patch to add that feature, it's something I'd like to see in Speex. > For those of you just tuning in, what I call

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

> I hate to be a talker and not a do-er, but I won't be able to write this > myself, probably someone on the iaxclient team could do it. Anyway, let me know if/when someone's working on that. >> Hmm, or does that mean the analogue AGC is actually completely >> independent from the "real" AGC. Any thoughts? >> > > It's actually a bit more

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

I just wondered whether anyone actually got this working and produced a how-to ? I recently had a customer ask about embedding it into their web pages for there customers to call them with ?? To be honest I have no idea how etc.... Gary .

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol. It is distributed as part of Steve Kann's iaxclient library. I've just posted new Windows, Linux and Mac OSX binaries to sourceforge. The Windows binary was compiled on WinXP. The Linux binary was compiled on RedHat 9. The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4 (Tiger) beta.

Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. The IaxClient seems to use silence suppression, and not sure if this can be disabled. The client works fine through most routers, but for some it disconnects the client after about 5 minutes and gives this error in the asterisk logs: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to

I was curious if you had ever peeked at a Teamspeak voice packet? I already have large chunks of the protocol torn apart. I believe the voice packets are the last big hurdle. As I know nothing about speex encoding, it is proving difficult to figure out the start of the actual voice data and/or any voice specific state data contained within the packet. If you have any information and are

On 5/3/06, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> wrote: > > I must say I really like the generalized jitter buffer though :) It's a > > cleaner and more flexible implementation and can more easily be adjusted > > to contain additional information with each packet. This looks interesting to tie into asterisk's jb and plc code as well.

Hi all, Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved. For the interested people, you can download the new DLL (just the IAX2 version) from the following location: http://www.laser.com/dante/diax/wiax2.zip Replace the wiax2.dll file in the app directory with the new one and this is all. Please test it and send me your feedback. I intend to release a new DIAX version this

Hi, I seem to be having problems with IAX clients based on the iaxClient library. I have been working on my own client (an augmentation to the Call Manager I released last week) and it seems to regularly miss incoming calls entirely. It also occasionally misses the drop signal when the remote end drops a call. Has anybody else seen this kind of behavior? I have tested with my client, with