similar to: Patch to make SPEEX_PREPROCESS_GET_AGC_GAIN use dB, and _SET_AGC_LEVEL use a int32

Hi, Here are the next two patches for the data export. speex_get_psd should be applied after speex_get_agc_gain (sent in previous mail). It allows applications to get the power spectrum for the signal and the noise estimate. speex_get_prob should be applied last. It allows fetching the speech probability of the current frame (the value that the _PROB_START and _PROB_CONTINUE parameters are

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); }

Jean-Marc Valin wrote: >>Reverberation suppression? >> >> > >Basically, it means that if you are in a room with lots of echo (long >decay), I can reduce it a bit. > > > >>I guess this would help reduce local source echoes? I've never >>_noticed_ that to be a problem in my use, but I would imagine that >>using a notebook's

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 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 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

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, 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, 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 -

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 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():

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

I've installed ADAM interactive anatomy 4.0 on Ubuntu using wine 1.1.6 and winetricks to install dot net 1.1. ADAM is a dot net application. After installation and offline activation (online fails), the application starts, but gives the error: Code: System.MissingMethodException: Method not found: Int32 SkySoftware.FileViewControl.FileView.get_SelectedCount(). at a.a.c.a.a.h.b() at

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

--- perl/lib/Sys/Guestfs/Lib.pm | 12 ++++++++++-- 1 files changed, 10 insertions(+), 2 deletions(-) diff --git a/perl/lib/Sys/Guestfs/Lib.pm b/perl/lib/Sys/Guestfs/Lib.pm index 5d48ba8..00a9bdb 100644 --- a/perl/lib/Sys/Guestfs/Lib.pm +++ b/perl/lib/Sys/Guestfs/Lib.pm @@ -1067,14 +1067,22 @@ sub mount_operating_system local $_; my $g = shift; my $os = shift; + my $ro =

Hi, Ref the disucussion on IRC yesterday; here's a patch which makes a bit more data from the analysis of the preprocessor and the echo canceller available. For the preprocessor: - Size of power spectrum. - Power spectrum and noise estimate of the previous frame. These are given as squared values, so sqrt() to get values in the 0->32767 range. - Current amplification level

Hi, Speex-dev: ? I have some problems about complie speex on pxa-270. I use libspeex.so.1.4.0 at my platform after complie speex and all function is ok, but the cpu loading is 99%. How can i do to reduce the cpu loading? ? note: 1. My Platform are os: linux 2.6.19 , cc: gcc-2.95.3 , cpu: pxa-270, ogg: libogg-1.1.3, speex: speex-1.2beta3 2. I configure speex by the follow command. ./configure

Ronald, I recently wrote some wrapper classes for the Speex encoder and decoder for use with my VoIP program. I think they're pretty high level and easy to use. They handle all buffer allocation internally to make life easy. Here's a simple little example that happens to use VAD: // buf is float[] or short int[], buflen is multiple of frame size // (there is a getFrameSize() method

On Mar 28, 2004, at 8:23 PM, Jean-Marc Valin wrote: >> 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. > > Oops... Thanks for letting me know. I'll change that for the next > release (in the mean time, the fix is