similar to: 1.17 Release

Hi, I use speex 1.1.12 in my program, but I found a strange thing about Echo canceller. That is, I'll hear an explosion in a certain frequency inputing. And I use the echo canceller in 44100Hz sample rate, disable AGC and Denoise. Can you tell me what's wrong with me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

We are using Speex as our major codec for voice application. We like the Speex solution so far. Currently, We have tried to compare voice quality between different public available VOIP solutions such as Skype and others. We notice Skype use higher CPU for signal processing. Is it because of this extra work the audio quality sounds clear? (Sometimes, it sounds like the audio signal is not real.

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

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, 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

> 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

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

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

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

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'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",

Hi -- I'm looking at adding Speex decoder and encoder support to the Miles Sound System, a commercial library used by game developers on various platforms including PCs, the Mac, and game consoles. As part of my initial evaluation, I've gone ahead and downloaded the 1.0.4 and 1.1.1 Win32 binary distributions and played around encoding and decoding various files with the speexenc.exe and

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

>> 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

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

My data is speech sampled as pcm mono 32khz. I want to use vad for regulating the agc gain mechanism. So... if voice is not detected I just reduce the agc value in order to not amplify the 'silence' otherwise we just increase the agc value if it is below the minimum. Thanks, Daniel ----- Original Message ----- De: Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> Para: danflu at

Hi, I have been looking at the speex library and I like it a lot. Good job! I have a couple of questions regarding the preprocessor: 1) I looked at the source code. Even though I am an experienced programmer it looks like Chinese to me ;-) What would I have to do to modify / tweak the audio processing code for my needs? Are there documents describing the algorithms applied for AGC, NS and VAD or

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 all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); }