similar to: about speex return

Hi SAITAMA Taro, I have been using speex-1.2beta1 dll in our voice application written with C#. In order to use Speex correctly firstly I advice to you examination speexenc and speexdec projects. I have used unsafe code and fixed statements in C#. I have added following method to library and I used that instead of speex_encoder_init simply. void *speex_encoder_init_new(int modeID) { const

Hi folks, I am new with Speex, while looking for Voice Chat in MSN/Yahoo, i cam across Speex. In case of Yahoo, i found Speex with variable bit rate. and in this case, the speex_decode function is not producing any output. Does in case of variable bit rate, something else is to be done with decoding process. Please guide me, what to do. Thanx in advance Lokesh

Hi, Everyone! I'm developing VoIP application using Speex, have some questions: 1) When ultra-wideband mode should I use 32000 Hz sampling rate only? If I use 44100 Hz the voice quality seems to be better comparing to 32 KHz. 2) When I use wideband mode with 16 KHz sampling rate bandwidth usage is greater comparing to 32 KHz, ultra-wideband. Why? Voice quality is worse when using 16KHz

Hi, I have read the message below, http://lists.xiph.org/pipermail/speex-dev/2006-October/004924.html and try to use P/Invoke to use speex in C#. This is a part of my code. [StructLayout(LayoutKind.Sequential)] public struct SpeexBits { IntPtr chars; // "raw" data int nbBits; // Total number of bits stored in the stream int charPtr; // Position of the byte

> Ok, what you suggest sound logical to me. Currently, I > have done a small trick to prevent this problem. What > I did is that whenever windows request a voice packet > from me and if I do not have the voice packet, I > repeat the previous packet. Hence, all the breaking > portion is filled with previous packet. This trick > seems to work so far. I am not sure what is the

Hello speex-dev! First at all I want to tnak you for this great software codec. It makes many things possible. I am a software developer. One of my projects is NetSpeakerphone - network communication programm (text messaging, voice communication, voice conferencing) for win32 platform. This project's age is about 2 years and many things works just great. Very small voice latency and time shift

I done it speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_START, &g.s.VADstart); speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_CONTINUE, &g.s.VADcontin); /*char *lisDebugCh = (char*) malloc(20); int decimalFcvt, signFcvt; static int firstDebug = 1; lisDebugCh = gcvt(g.s.VADstart, 20, lisDebugCh); if(firstDebug == 1) { ::MessageBoxA(NULL,

What you want is simply a loudness threshold-based detector. It's not very complicated to do, but there's nothing in Speex that currently does that (not that I think it's really useful in practice). Jean-Marc Le jeudi 02 mars 2006 ? 04:26 +0100, Lis a ?crit : > Sorry. > > I forgotten the words volume or loudness. > But it is know as microphone stroke too, i think. >

Hello Cesar I'm using Speex on a PPC2002 and PPC2003 with no problems. For tune-up remember to define the FIXED_POINT macro in Project->Settings in eVC. With this setting you ought to be able to at least decode in realtime while using debug-mode. Kind regards Bjoern D. Rasmussen >From: "cesarbremer@raseac.com.br" <cesarbremer@raseac.com.br> >To: speex-dev@xiph.org

Hi all I have a problem using the Speex voice codecs when using GnomeMeeting on one side and NetMeeting on the other side. I use GnomeMeeting under Suse Linux 9.0 to communicate with a friend working under Windows XP and using NetMeeting 3.0. Under Windows XP / NetMeeting we have installed and registered the Speex voice codec. (You can find more information how we have registered the Speex codec

Hi, I am working on a project which intends to enable voice communication over zigbee. As the handset node, I am using TI's C5505 ezDSP USB stick. I want to run speex encoder on it. After encode, I intend to send it over CC2520 radio to the backend zigbee network. The C5505 ezDSP has input and output for AIC3204 codec which I am using to give sampled input from laptop (lineout) and playback

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:

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

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

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

This works well. Thanks for the link. However I am curious about something. Speex requires a VAD for the VBR mode and the comfort noise preservation. Is the Speex portion for this different than the VAD result returned from the preprocessor call? Vas ________________________________ From: speex-dev-bounces at xiph.org [speex-dev-bounces at xiph.org] On Behalf Of Ken Smith [ken at alanta.com]

hello guys this is my code for the C# wrapper. ================================= using System; using System.Runtime.InteropServices; using System.IO; namespace Speex { ///<summary> ///the following class is the win32 SPEEX API that retrieve ///methods and structures from the libspeex file ///</summary> public unsafe class Speex_win32 { #region Speex Constants //global speex

blink : It use iLBC also. Voice over IP RTP: A Transport Protocol for Real-Time Applications RFC3550 RCTP: Real Time Control Protocol attribute in Session Description Protocol RFC3605 SRTP: The Secure Real-time Transport Protocol RFC3711 DTMF: Dual-tone Multi-frequency Signaling RFC2833 and inband MWI: Message Summary Event Package RFC3842 Speex and G722: Wide-band Internet Codecs G711, iLBC

I want to know speex compression rate. On the speex website,it says" Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps." Is the original voice at bitrates raning from 2 to 44 kbps,or the compressed voice at bits ranging from 2 to 44 kbps? what is the maxmium compression rate? Thanks a lot! Weiqin Bao -------------- next part -------------- An

Hi there, I have searched what seems like everywhere and was unable to find a .NET wrapper so that I can use speex in VB. I did find a speex.NET.dll but this didn't seem to work on what I needed. I therefore decided I could just use DllImport within VB .NET and create my own unmanaged links to libspeex.dll. With the help of this partial c# code