similar to: speex preprocess

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

I'm not sure if this is due to the same problem, but I get the same clipping behavior when trying to use VAD+denoiser on a particular mic+speaker setup that has significant background noise (enough to really foul up the codec should the denoiser be turned off.) Leaving VAD off and transmitting continuously seems to be the only way to avoid this clipping. Another person I regularly

Jean Marc, Thanks for the advice. The estimate update may come in handy. I have had some time to play with the adaptation time. I tried making the adaptation time 80, 160 and 320. It seemed like the smaller value was actually the best in my initial testing, but I need to test this against a more noisy setup than I have here. I am guessing that for vad a shorter adaptation time is probably

Hi all and Jean-Marc, Firstly thanks a lot to you for helps, at last I accomplished to use speex-1.2beta1 in my speech application. I try to understand speex source code and improve speech quality. The problem in my application (like many people in the list says) is noise. So I've added VAD in preprocessor as following. VAD in preprocessor is very good, there is not noise outside speech. I

Jesus, Unfortunately, FFT and magic algorithms don't work (yet?). You might want to try this if you're not satisfied with Speex VAD: http://lists.xiph.org/pipermail/speex-dev/2008-August/006860.html It won't perform any miracles, but I think it works pretty well and is easy to tweak. Tom >---- Original Message ---- >From: jmorion at toomeeting.com >To: speex-dev at

Hi all and Jean-Marc, Firstly thanks a lot to you for helps, at last I accomplished to use speex-1.2beta1 in my speech application. I try to understand speex source code and improve speech quality. The problem in my application (like many people in the list says) is noise. So I've added VAD in preprocessor as following: int param_enabled = 1; int param_disabled = 0; int sampling_rate

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

Steve, The main problem I am having with the system is clipping off the start of someone's speech when they first start talking- the ends of the sentences seem to be handled properly. I am wondering whether this is the fault of the audio playback system or whether this is a speex issue- I also get the musical artifacts problem with the denoiser. This seems to be more of a problem on open

Tom Harper wrote: > Hi All & Jean Marc, > > Once again I find myself delving into the pre-processing code to fiddle > with the VAD, AGC and denoising code. > > Where i am at is that I have implemented all of Steve Kann's mods, and > they are 90% of the way there in terms of working, except that I am still > having issues denoising open air mics. But that is

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

hi guys, i am trying to develop an application having a capability to detect voice activity in voice stream and then record only portion of this stream which contains the activity. i went through these steps. 1. Captured voice through Waveform Functions of Windows Multimedia API 2. Started Speex Preprocessing liberary 3. Turned on preprocessor by calling speex_preprocess_state_init and

For what it's worth, I've had significantly better luck with the WebRtc VAD. It's pretty good. http://code.google.com/p/webrtc/source/browse/#svn%2Ftrunk%2Fsrc%2Fcommon_audio%2Fvad%2Fmain%2Fsource In general, the WebRTC voice engine seems to be more sophisticated and mature than the Speex preprocessor (as opposed to the Speex codec, which is pretty good). Ken Smith Cell:

Lis, I suggest you try tweaking Speex's VAD probabilities as Steve suggested. But consider a simple threshold-based approach as a backup option. Personally, I struggled with Speex's VAD algorithms (both encoder and preprocessor) for a long time, tweaked the probabilities, wrote special case code to work around the mistakes, and was still never satisfied with the results. In times

Hi...Tom, How to use the code you written? Can you show me some example? Thanks, -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On = Behalf Of Tom Grandgent Sent: Friday, March 03, 2006 12:57 AM To: Steve Kann; Lis Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Voice Activation Level (speex 1.1.11.1) Lis, I suggest you try tweaking Speex's

From 1.2rc1 source code: preprocess.c: /* FIXME: This VAD is a kludge */ st->speech_prob = Pframe; if (st->vad_enabled) { if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue)) { st->was_speech=1; return 1; } else { st->was_speech=0; return 0; } } else { return 1; } AND case

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 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

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

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 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