similar to: SPEEX_PREPROCESS_SET_AGC_LEVEL

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, 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 -------------- ---

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

As I understand it, there are two separate ways to get VAD information from Speex: 1) Using the encoder. 2) Using speex_preprocess(). I present the following observations from an application developer's perspective. They may be wrong, in which case I would appreciate corrections. - The two VAD systems are implemented differently. - speex_preprocess()'s VAD provides more accurate

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

Bjoern, As far as I understand it, AGC keeps the volume level of the speech elements (i.e. gain) stable no matter how quiet or loud the input becomes- Tom At 10:53 AM 3/28/2004, Bjoern Rasmussen wrote: >Hello > >The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() >function of the preprocess.c file (alloced in line 167). It's in Speex >1.1.4 by

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 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;

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,

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

Hello The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() function of the preprocess.c file (alloced in line 167). It's in Speex 1.1.4 by the way. I'm trying to make the denoiser work with my application and has got reasonable noise reduction after applying the denoiser. I, however, haven't been able to find any information of what the purpose of

Yes, you can use it independently by using speex_preprocess(). This function can do noise removal, AGC, and VAD. I've been using it to do all three and it usually works very well. There can be a train going by outside, producing lots of sound that makes it through the noise filter, and yet the VAD knows it's not speech. However, sometimes the VAD seems to get into a bad state and

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 All I am using speex in one of my WinCE project. I have been trying to use speex to perform denoise on the captured audio packet and echo cancellation. Following behavior I have observed while using various options. I would really appreciate if you could help me with the issues I am facing. 1. Denoise: I have written the following to code to perform denoise. //To initialize speex

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

> That is true for distinguishing it by waveform, but not by amplitude. > As I mentioned, these transients are objectively tiny. *Your* transients may be "tiny" and in any case, it doesn't help if you don't know the level you're recording at. I guess I'd be > curious as to which voice codec applications require no squelch (other > than trivial examples

Dear all, I need to use echo cancellation in a VOIP application using speex narrowband codec.I am facing some problems with echo cancellation. It will be very helpfull if someone can put some light on the following issues. 1: =================== -- When call is started EC works fine for first few seconds. Bur after a certain period [usually 20 sec or more] it does not work or works very bad

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

I have been tyring to use the preprocessor standalone by feeding it pcm data from a headerless pcm file (mono, 22050hz) and then write the result to another file. Unfortunately, all I get back is noise (first frame is set to 0, rest is noise). Frame size is set to 440. I just modified testdenoise.c slightly to read/write from a file (see below) but I can't get useful results. What am I