similar to: Speech detection in preprocessor with echo

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 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

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

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'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

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

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_prob to detect when > there is enough speech to consider AAGC and then loudness2 to

Hi, Attached is a patch which enables FFTW3 to be used by libspeexdsp. Using SSE, the FFT code is more than twice as fast, and it's 50% faster even without SSE. To enable the library, you'll need to call configure with '--with-gpl-fftw3'. Based on the benchmarking we did, this also changes the default FFT to smallft for floating point implementations when not using FFTW3.

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

>> 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, Does someone know if I could use SpeexPreprocessState::loudness or SpeexPreprocessState::loudness2 to create a VU meter. And if so, how? <p>Thanks, Tom Moers --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word

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'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

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'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

Current SVN compilation fails on Win32 (Intel C++ compiler 9.0): ..\..\libspeex\mdf.c(317): error: declaration may not appear after executable statement in block float adapt_rate; Just moved it up to the variable declaration block to solve it locally, but as it's not valid C, I thought I'd mention it ;) PS: With the same compiler, AGC fails horribly (Zlast goes to several

> 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

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); }

This small patch will make st->Zlast = Zframe, to allow applications access to an estimate of the signal-to-noise level. This used to be in there earlier, but was removed when Zlast was no longer used to compute Pframe. -------------- next part -------------- Index: preprocess.c =================================================================== --- preprocess.c (revision 10007) +++