similar to: Recall: AGC

Hi Thank you for your work. We are trying to use the speex SW with a C6000 TI DSP. I am working with an HW codec that does AGC by HW inside. Will doing the AGC before the preprocessor make a problem? Does VAD work well in a noisy eviroment? In SW there are some remarks on things to be fixed, can I count on SW as working or are some part still to be improved. I also tried to understand SW but I

Steve, You're right. The AGC gain does not max out when using VAD (via the preprocessor). So instead of not transmitting when the AGC max gain is reached, I now do this instead: Start the call with VAD enabled and AGC disabled. When speech is detected, disable VAD (if 100% continuous transmission is desired for the call) and enable AGC. This seems to be working reliably so far. However,

I'm getting confused, what's the problem exactly? The AGC doesn't remove the DC or it adds one? Jean-Marc Aymeric Moizard a ?crit : > Hi Jean-Marc & all, > > I'm currently working on a wav file with very low volume. > This wav also contains a DC component. > > Using testdenoise with modification (AGC turned on with > 20000.0 value) I'm able to

Hi, The leakage problem you describe is very, very common and you will need to do something to address it. I modified the version of Speex I use to implement an adjustable max gain. If you look at speex_compute_agc in preprocess.c, you will see: if (agc_gain>200) agc_gain = 200; This max of 200 is usually more than enough to amplify leakage which occurs either in the sound

I've been using Speex's AGC in my VoIP program with mixed results. One of the problems I had was: At the beginning of the call, if the user hadn't spoken yet and there weren't any background sounds for the AGC to lock on to, most of the time the AGC would almost immediately jump way up to 200x gain (the default max). This is undesirable and usually results in nasty feedback. If

Hi, I'm trying to use AGC in combination with Asterisk 1.8 and an odd telephone which is very loud when used with a headset and more quiet when used "normal". Regarding to the documentation, AGC should be available since * 1.6 - but every time I want to set it, the CLI tells me: -- Executing [0160xxxxxxx at intern:2] Set("SIP/intern-xxx-000000d2",

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

Hi, I've had a small case with noise suppression and AGC. I have a fairly noisy environment here, and with the default parameters, noise suppression works fairly well while I talk. However, when I shut up, AGC starts slowly increasing the gain until it has amplified whatever noise is left to levels about equal to having no filtering at all. As soon as I talk, AGC backs down fairly quick

Jean-Marc Valin wrote: >> I've had a small case with noise suppression and AGC. I have a fairly >> noisy environment here, and with the default parameters, noise >> suppression works fairly well while I talk. However, when I shut up, AGC >> starts slowly increasing the gain until it has amplified whatever noise >> is left to levels about equal to having no

Hi Jean Marc, I found that the AGC API and algorithm has changed in svn head trunk. Is it safe to use it? Or is it for testing purpose only? You also said that VAD in svn is broken in a previous post, is it related to the AGC change? or can we mix the old VAD algorithm with the new AGC safely? kind regards, fredo

I have a recorded conversation from an analog trunk. As usual one side is stronger that the other one. In my case, the gap between signal levels are even bigger. How does speex AGC preprocessor will perform on this type of audio recording? Maybe I am wrong and AGC is not really what I need to equalize the two persons in my phone conversation? As I Understand, AGC will perform better if each

>> Yes, after I stop speaking, the noise slowly starts climbing again, and >> if I peek at st->agc_gain, that's slowly climbing too. I think part of >> the trouble is that the noise in here isn't uniform white noise; there's >> traffic outside the window and people walking in the hallway outside my >> door. Each little event is enough to cause the AGC

Hi, where I can found that patch? Thanks Seif 2008/7/16 Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> > Hi Michael, > > Thorval Natvig posted an experimental patch to get the AGC working in > fixed-point. Can you try that and see if it works well? > > Jean-Marc > > Michael Evseev a ?crit : > > Hello All, > > > > > > > >

Hi, I applied the following minor diff to preserve the value of the extern variable across calls to process_sign_request2() in ssh-agent.c. I've no idea if the attachment of the attachment will come out correctly so a cvs diff is also attached. Regards, Alistair -------------- next part -------------- An embedded message was scrubbed... From: "Alistair G. Crooks" <agc at

2011.08.23. 15:38 keltez?ssel, Yanick Bourbeau ?rta: > Since I don't have access to different channels as I record a phone call > using a man in the middle approach, there is something else I can use > to equalize the sound ? What I would do then probably is try to manually separate the two channels/talkers; say channel 1 goes from 0 seconds to 13 seconds, channel 2 goes from 13

Hello All, I'm using speex codec in ARM based embedded project and really satisfied by its performances and quality (thanks a lot for speex development team for great work!). I'd like to add AGC functionality and as I actually understood it is not supported in FPU-less platforms. Is it correct? If it is, do you have any updates of preprocessor or an experience to run AGC in

Hello All, I use AGC as one of speex preprocess options in VOIP application. In most cases it works very fine. However, at the very beginning of a call when no one has started talking [ and some times in the middle of calls], AGC makes large noise and it gradually increases. Interestingly, when someone starts talking, from that point it works very well. Is it the normal behaviour? Since

> Yes, after I stop speaking, the noise slowly starts climbing again, and > if I peek at st->agc_gain, that's slowly climbing too. I think part of > the trouble is that the noise in here isn't uniform white noise; there's > traffic outside the window and people walking in the hallway outside my > door. Each little event is enough to cause the AGC to increase a little

Hi, I am working on a VOIP implementation were one of the key design goals is zero user configuration. Similar to Skype. What I've come to notice is that my soundcard (NForce4 based) leaks audio from the playback path to the recording path. This is probably not unique for my hardware at all and will happen on some percentage of all users of my software. (All of this is Win32) What happens is