similar to: Auto level control

Hi, I've been using Icecast for live and recorded audio (MP3) streams for a fair bit of time. I need to delay a live stream by a finite period (somewhere in the order of 5 minutes or so, the delay doesn't have to be very accurate). Does anyone know of an application or a plugin which can do this readily? I would assume it will be some sort of FIFO buffering app. My preferred platform is

On Sun, 15 Aug 2010, Michael C. Cambria wrote: > I can't seen to get edcast standalone to let me do anything except live > streaming, via rear mic (or line-in or front mic). From some research > (links below), live streaming can be turned off (a big red X), but I > can't seem to turn it off. I've already seen import source being a > CD-Drive etc. vs

Hi, I have setup Icecast2 and edcast (standalone) on an XPSP3 machine. The basic setup works. I can speak into a microphone and hear my voice from another PC on a LAN. Now I want to "stream a file" such as an mp3 podcast I made or just a song (mp3, acc etc.) or play a CD and don't know what to even search for in the mail archives, forum or Google to get help. I can't

These are my results with the Speex AGC and denoiser: Over the past couple of months, I've spent probably 10-20 hours of conversation with Speex (wideband) doing AGC + denoiser on both ends of the call. I think it does an excellent job of removing most noise while keeping speech intact. Yes, it's possible to make some constant noise (like humming) and have that get removed, but

this is the link http://users.tpg.com.au/rioradio/ > Date: Mon, 5 Sep 2011 11:24:26 -0400 > From: dave.mehler at gmail.com > To: geoff at quitelikely.com > CC: icecast at xiph.org > Subject: Re: [Icecast] Winamp streaming to icecast2 > > Hello, > > Thanks, I'll check out edcast. I don't need to play music, just stream > the event. From what I understand

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 >

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

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets

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

First, I'd just like to thank the Speex community, and Jean-Marc especially, for their great work. I'm developing a VoIP library (which uses IAX, the asterisk protocol) as the network protocol. I've been putting off integrating Speex for a while, as things have been working pretty well so far with GSM. (for those interested, the code is at iaxclient.sourceforge.net). However,

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

> 1) AGC: This seems to work pretty well in all cases. I had previously > hacked-in the "compander" filter from sox for a similar effect. What > I've noticed is that speex_preprocess's AGC has no "knobs", and it > seems to use an attack/decay that is a lot faster than what I had > chosen from the sox compander, but it works pretty well nonetheless.

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

Icecast2 Source Tutorial Poor man's basic Icecast source setup instructions. Everything needed to set up a basic streaming system without the mess. This tutorial assumes you've successfully setup icecast2 for this. This tutorial also assumes you've got audio files to use to stream to iceca= st2. You will also need a method seperate from your source computer to tune-in t= o the stream

I am using the denoiser and AGC, but the problem persists if I turn them off (stop calling speex_preprocess() altogether). Also, I have reproduced it on both a noisy system and a quiet system that hardly needs the denoiser. I know what you mean though, I have heard similar sounds when pushing the denoiser too hard. For example, doing 100x AGC, or trying to use an extension cord with my

Ok all, I have deleted the edcast folder on my machine and reinstalled edcast. However, I can't get the mp3 encoder to install. Tim said that it should ask if you want the encoder, and I thought I selected it under the select components to install, but it isn't installing. Two additional things are happening, 1. When I open edcast, I don't see anything in the dropdown lists.

> Then I pulled the microphone out. Our system still records noise. To > isolate the problem, I wrote a small app just to open the device and > record raw samples, calls speex_denoise() and outputs both sample > sets. > The noise is still there, with level fluctuating with gain level, > unless > "All mute" is chosen. > In the case when NO microphone is

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, I am just starting up on SPEEX AEC algorithm and have couple of questions around echo cancelation algorithm used in SPEEX. 1) Usually Echo Cancelation Algorithm has support for number of components ? - Non-Linear Processor (NLP) - Automatic Microphone Gain Control (AGC) - Transducer Equalization (EQ) - Dynamic Range Compression (DRC) - Ambient