similar to: Integrating speex with VideoNet application: Constant background noise

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'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 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 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:

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:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM;

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 Jean-Marc, Thanks. I've got the code for sampleenc and sampledec. But I didn't know how to verify whether the decoded file is correct. So I tried to analyse the decoded output and compile with my input. Test case: PCM samples, 8 bit, 8000 samples per sec. When I pulled all my input to 0s, after encode and decode, the output fluctuates around -0.12 to 0.2 (in float), so when finally

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

hello everybody in this great mailing list , i have some difficulties to follow my code . i solved some problems thanks to Carine Liang , but i still have one problem and i think it is fatal one. when i encode the voice data in a wav file it is decoded without any errors it gives me 84 bytes wav file size for 139 kbytes wav audio data .Naturally i wanted to return my file back to its normal

I'm only doing one frame using speex_encode_int greatly simplifies my code I'm not sure why the sample I was working off of was converting the shorts to floats then calling the other encode/decode methods. Based off of your suggestions I tried the following but I get the same result. virtual Enigma::u8* Encode(Enigma::u8* inputBuffer,size_t inputSize, size_t& outputSize) {

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

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:- > >

Alright noted, I changed me code so that the state is created in the constructor and destroyed in the destructor of the object. However I'm still getting the same issue although I'm sure that would have bit me sooner or later. The new code is as follows. virtual Enigma::u8* Encode(Enigma::u8* inputBuffer,size_t inputSize, size_t& outputSize) { short *in=(short*)inputBuffer;

Hi! I'm totally new to audio programming. I managed to create a VoIP with waveform (PCM). The problem was the really high traffic, so I desided to use speex for compressing the data. I'm recording with mono, 16bit/sample, 8000 samples per second. I tried to add the speex compression, but I always get a crash when I try to decode the data. Here is my encode and decode function: int

I write a sample code like speex provide , as follows: SpeeXCodec::SpeeXCodec() { rate = 8000; speex_bits_init(&enbits); speex_bits_init(&debits); enc_state = speex_encoder_init(&speex_nb_mode); dec_state = speex_decoder_init(&speex_nb_mode); // set option for encoder speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frame_size); speex_encoder_ctl(enc_state,

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

--- src/libFLAC/include/private/bitmath.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/libFLAC/include/private/bitmath.h b/src/libFLAC/include/private/bitmath.h index 61b0e03..d32b1a7 100644 --- a/src/libFLAC/include/private/bitmath.h +++ b/src/libFLAC/include/private/bitmath.h @@ -42,7 +42,7 @@ #endif /* Will never be emitted for MSVC, GCC, Intel compilers */

Hi, Sorry! Marc Gimpel. I wrote it in a false thread before. It was my first time. :-) I have to write a small programm which should be able to record speeches and in addition synchronize the recorded speeches with measuring data. My idea is to record the speech in the normal way with speex and without ogg. If a sync message was received, i write a custom in-band messages using mode 13