similar to: Some Speex AGC Questions

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'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 think you'll have to modify Speex to get the functionality you're looking for. I've made a few simple modifications to the AGC to prevent it from 1) exceeding a specified level of amplification and 2) enable and disable adaptation, so I can freeze it at a certain level while speech is not detected. It's mostly just a matter of doing this at the end of speex_compute_agc():

I'm having trouble with the preprocessor's noise reduction feature. The basic issue is that it simply doesn't work very well. With my laptop (whose microphone is otherwise quite capable) I routinely hear transient background noise, typing, and other "quiet" sounds leaking through to the speex stream. Even worse, the AGC feature is blowing these things up into just awful

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

Andy Ross a ?crit : > I'm having trouble with the preprocessor's noise reduction feature. > The basic issue is that it simply doesn't work very well. > > With my laptop (whose microphone is otherwise quite capable) I > routinely hear transient background noise, typing, and other "quiet" > sounds leaking through to the speex stream. Even worse, the AGC >

hi, 1. I usr speex wb ,and find voice volume too small , my setting is : st->agc_enabled = 1; st->agc_level = 8000; I feel the agc function not work,how can i enhance the volume. 2. I test encode with 16.8K WB mode and decode with 23.8 WB mode, and find voice not good! with continues electric current sound . why? please help me. thanks a lot. tanger.

Hi Jean-Marc and all, I was playing with the AGC with different sampling rate and I experienced smaller volume level when I increase the sample rate. Should I use a different value for AGC_LEVEL when using different sample rate? I'm using speex-1.2rc1.tar.gz Tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 -

Hi, The attached patch fixes an incistency in my earlier patch. Whereas the rest of the AGC ctls are in dB, GET_AGC_GAIN was linear. This patch fixes that. It also changes the API for _GET and _SET_AGC_LEVEL to use a int32 instead of a float, meaning we don't need to do a API change when we get a fixed point AGC. Best regards, Thorvald -------------- next part -------------- ---

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); }

Hi Everyone, I am still unable to start Asterisk 1.4 that I installed using ATrpms. I was initially suspecting some permissions issues but it seems to me that its more to do with a speex codec not loading properly. Here is the message I get if I run asterisk -cvvvvvvvvvvvvvv app_userevent.so => (Custom User Event Application) == Parsing '/etc/asterisk/codecs.conf': Found --

Hello, I'm using Speex1.1.11.1 source code. I enabled dereverb by speex_preprocess_ctl(). I try different dereverb_level and dereverb_decay values and it seems to not work. I generate some test files in CoolEdit using reverb or echo and then preprocess it by speex_preprocess(preprocess, input, NULL); and output is the same as input:(. Denoising and VAD works very good. I tried dereverb with

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

Jean-Marc Valin wrote: >> gcc version 3.4.5 (mingw special) > > Can you try 4.0 or 4.1 just to be sure? mingw (native gcc for win32) doesn't officially support using gcc 4.0 and 4.1 yet (apparantly there have been some issues), so there are no binary packages. But if you think it helps, I can compile gcc 4.1 and give it a shot. >> Compile flags: >> DEFINES +=

Hi all, as in subject, speex_preprocess inject noise in my data. Someone can help ? Here's the way that i'm using: #define NN 160 /* 20msec di audio */ #define AUDIO_SAMPLERATE 8000 spx_int16_t TEMP_Buffer[NN]; speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); c = denoise; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_DENOISE,&c); c = agc;

I'm running 1.6.1.18 on an older ubuntu machine. I upgraded to dahi-linux-2.3.0. That went fine, and it works. But I decided to use the opportunity to upgrade to 1.6.2.6. That didn't work. configure, make menuselect, make, make install all went fine, or at least seemed to. But it hangs starting up here: [Apr 13 20:15:28] VERBOSE[1612] codec_speex.c: -- CODEC SPEEX: Setting

Hi all, I''m currently using ipsec with Shorewall 3.0.7 on a patched 2.6.10 kernel. Having heard that ipsec support was in the standard kernel starting from 2.6.16, I tried to upgrade to the last kernel. My problem is that shorewall won''t start anymore. I get this output in /var/log/shorewall-init.log: Starting Shorewall... Initializing... Shorewall has detected the

Hi, I've been looking a bit at the VAD/AGC situation. I noticed a leftover variable declaration. The AGC used to use nb_loudness_adapt to count the number of frames so far, but it now uses nb_adapt which is shared by all the functions. However, it still allocates and updates nb_loudness_adapt. This patch saves 4 bytes of memory and a partial cycle of CPU power ;) -------------- next part

Hi, I''m asking my question here, because I could not find any answer to my problem, but I''m affraid shorewall is not the one to blame. First of all I''m using shorewall version 2.0.15 on two linux box. I set up an ipsec tunnel beetween those 2 boxes to be ables to connect 2 not routable subnetworks. Here is my network topology: 10.66.17.0/24 - 10.66.17.1 = eth0

Is it possible to have virt-viewer come up when a domain is started with virsh start, the same way that it does with virt-install? Thanks. --- Scott Lerman