similar to: Encoding and decoding problem in speex 1.0.4

I'm getting garbled playback with decoded fragments and I'm hoping someone here can point me in the right direction to correcting the problem. Essentially I'm capturing audio from the microphone. I stream it over the net, but for testing purposes with this API I'm just grabbing the whole chunk and encoding / decoding it right away and then updating the sound buffer for

Hi, I am back. Thanks everybody for their suggestions. Finally I could get something out of the decoder that makes sense. But the problem is it is highly noisy and the voice is not very clear. Normally without using the codec I get good audible sound through the Microphone. I'm using Speex version 1.1.6. I've also used 1.0.4 beforehand and experienced the same problem with it. 1.

Whoops, left this message in my outbox. I managed to fix the problem. Apparently I was only copying 160 bytes (Frame Size) back into the audio stream when I should have been copying 320 (chars <-> shorts confused me there). Hence why I could hear myself yet it was distorted. Half the wav was missing =) To answer some of the other questions here, for any insight into what I'm doing:

Hi, Can someone please help me with my problem below. Any suggestions is appreciated. thanks, Carine ----- Original Message ---- From: Carine Liang <carineliang@yahoo.com.sg> To: speex-dev@xiph.org; speex-dev@xiph.org Sent: Tuesday, 31 October 2006 1:05:49 PM Subject: [Speex-dev] Integrating speex with VideoNet application: Constant background noise Hi, I am developing a peer-to-peer

Hi John, Thanks for your reply. Yes, my output device is in 8-bit mode, same as my input mode. I'm actually working in char (8 bit) arrays. Is it necessary to work in signed short integers only? I changed all the short arrays in the example to char. Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To: carineliang@yahoo.com.sg; speex-dev@xiph.org Sent: Thursday,

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles

Hi, I am developing a peer-to-peer video conference application which uses speex as a codec for the voice. I am new to speex, so please bear with me if I asked the obvious. After I added the encode and decode function to my MFC app, I heard a constant background noise, even when no one is speaking into the microphone. #define FRAME_SIZE 160 The application is coded in MFC C++. The record

Hi Jean-Marc, I finally make it work! Thanks for telling me that the algo will not work for 8-bit PCM. In fact I've tried using 16 bit PCM last time, but apparently there's a bug. The bug lies with the RecBuffer. The RecBuffer must be set to 320bytes instead of 160 bytes since every sample is now 16 bit. The compression is amazing, from 320 bytes to 38bytes! Thanks a million, Carine

Hi, I'm still unable to get rid of the background noise though I've changed the PCM samples to 16 bit. I'm began to wonder if I can just take the PCM samples and encode them. What my application tries to do is to record the voice from a microphone, encode it, send it via the socket to a peer, decode at the peer's side and play the voice. My application works fine without the

Hi, I read in the documentation something about how to realize packing more than one encoded frame into a packet. Before I read this, I always encoded every frame seperate and packet them as following together: [encoded length][encoded frame][encoded length][encoded frame]... But if I understood the docu correct, it should be easier: I call speex_encode_int() for every frame until the whole

Hi John, I noticed the values after decode is very much different from the values of the original PCM samples. For cases when the PCM sample values are constant (because no one is speaking), the decoded values fluctuate too. These fluatuations are causing the background noise. Is my observation expected? Thanks, Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To:

Is this a common warning? The decoder doesn't return an error on it, but I see it a lot in my test application on windows. It is non existent on my linux box. I haven't tried mingw yet. please note that I'm using visual studio 2008 w/the vcproj that Bjoern Rasmussen made for 0.5.2 (w/some file references removed) at the moment and it is giving a lot of C4554 warnings

Vinod Vijayan wrote: >Hi, > I am using the speex 1.0.4 library from Windows. > I have posted my problem before but didn't get a solution. I am doing an >VOIP project > in which i am recording sound and streaming it to the peer. I wanted to >encode and decode > wav files that brought me to this site. > > I am recording sound in the following format:- > >

hi sirs, i am only recently studying speex and i'm looking for a C++ wrapper for speex that has higer-level implementations for the Encode and Decode functions, to sort of base my initial understaning of how the speex API work. i have in fact created my own, following on the instructions in the speex codec manual, but i havn't really made any positive progress. (a lot of people have

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

Ronald, I suggest we take this discussion off the list after this post. I don't see any problems with your code. However, I can make a few suggestions: 1) If you replace the lines: pSpeexHdr->lpData = (char*)pSound; pSpeexHdr->dwBufferLength = decodedSamples * sizeof(short); with: pSpeexHdr->lpData = lpHdr->lpData; pSpeexHdr->dwBufferLength = lpHdr->dwBufferLength;

hi Tom, i downloaded your speex wrappers and they're really what i'm looking for! i really cannot worry about container support at the moment because i'm working with real-time audio, and hopefully toward VoIP when everything else is in order. im using waveform to capture data from the microphone. the raw PCM sample is stored in the WAVEHDR structure's member lpData, and this

Hi, I don't understand how works the multi stream api in opus. I need to send two mono streams over network with RTP. I think I'm right when I create an OpusMsDecoder with opus_multistream_decoder_create (48000, 2, 2 ,0 ,mapping, NULL) where mapping is: unsigned char mapping[2] = {0,1} isn't it ? Next, i need to encode data which I get from jack (float) so I use

The absolute values may be different for a variety of reasons; you can hardwire them all to 0 to see if your noise goes away. Make sure the values on the decompressed (PCM) side are little-endian (low byte first in memory), assuming you're working in Windows on x86. -- john RAD Game Tools -----Original Message----- From: Carine Liang [mailto:carineliang@yahoo.com.sg] Sent: Sunday,

Hi, I would like to create a movie out of a lot of png files. Using mencode I would do have something similar to mencoder "mf://*.png" -mf type=png:fps=5 -vf spp,scale -ovc lavc -o output.avi but I want to have an ogg format. I.e. an ogg movie How to do it? convert can convert the pngs to an mpg but not to an ogg! convert: unable to open module file