similar to: Size of each block in a circular buffer and the sample rate

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

Hi, I have a module sampling raw PCM data on Win CE as 10ms time slice (160 bytes), mono, 8000HZ, 16 bits per sample. Does anyone know what is the mflops for using fixed point on a Win CE compared to using floating point? Looking at the manual, "In practice, frame_size will correspond to 20 ms when using 8, 16, or 32 kHz sampling rate." for a 8 kHz sampling, the framesize should be

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

Hi I apologise if the following message is inappropriate for this list. However, I feel this is the only place where I can get some help with my problem. BOOL CALLBACK RecordProc(HRECORD hRecord, void *buffer, DWORD length, DWORD user) { try { long sent = 0; long inputOffset = 0; long outputOffset = 0; long nbBytes = 0; char *outputBuffer = NULL;

Thanks! I hope this isn't a stupid question but if I want to play two or more voices at the same time, I just add the values of the samples right? If I'm using an unsigned short, should I then clamp the values to 65535? Jean-Marc Valin wrote: >> Is frameSize in bytes? If not, what unit is it in? I need to know so I >> know how big to make each element in my circular

Changing from using floats to shorts did fix the high pitched tone problem. I'm having other problems but I'll look into it more first. SteveK wrote: > > On May 21, 2006, at 6:33 PM, Kevin Jenkins wrote: > >> When I just copy the microphone input buffer to the output buffer the >> sound plays OK. But if I encode and decode the buffer through Speex I >>

Hi Tom, Thanks for your reply, I'm having a few difficulty's following you advise. You mention that I should be calling 'speex_encoder_int' & 'speex_decoder_int', I'm having problems finding these functions in the speex header. I am currenlty calling speex_encoder_init & speex_decoder_init in my test code, I have repeated the relevent bits of the example

Thank you for the quick response Thorvald, but I think that's not the problem here :[ I know how to capture the buffer and how to play it in the output buffer of the DS. The problem is (probably) with same kind of short/floats/bytes error in conversion/copying that the coder doesn't get. I can have my buffer locked during the compression, it not the problem at the moment. What I really

Hi, I'm using the latest version of libspeex (checked out today) and have some problems with checking the target bitrate in the encoder. Calling: peex_encoder_ctl(globals->enc_state,SPEEX_GET_BITRATE, &bitrate); Seems to set bitrate to 24600 no matter what the qualitysetting is.. The functions used previous to this call is: peex_bits_init(&globals->bits);

In short try calling speex_encode_int and speex_decode_int. The calls to speex_encode and speex_decode expects the data to be a floating point values. The data is passed by pointer and the compiler does not do the conversion. This is why you are getting the segfault on decode as the short is 16-bits and a standard float is 32-bits. Tom -----Original Message----- From:

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

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,

When I just copy the microphone input buffer to the output buffer the sound plays OK. But if I encode and decode the buffer through Speex I get a high pitched constant tone in the background. I actually do hear my voice speaking when I talk, but it's faint and much quieter than the tone. Here's what my data looks like: Input is the first 5 floats of each input buffer. Output is

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any

Hi All, Almost all my question is in the subject. I have written the following small program and I get an error message when I try to use SPEEX_GET_QUALITY. This functionality seems not to be used in speexenc.c or speexdec.c. I would appreciate any hints. best regards, Alain Aubord ---------------------------test.c--------------------------------------- #include <stdio.h> #include

hy, here is my speex encoder/decoder .. the sampleRate i use is 16000 and quality,complexity are at 5. can someone take a look in to the code and see if there is something that is making that robot voice here is a link to colored and numbered code, same as below http://barvanjekode.gama.us/temp/1257361243.html thanx. code ----------------------- #include "codec.h"

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

Hello everyone I've a following problem with the speex codec. I'm developing application that is supposed to read sounds from the microphone with DirectSoundCapture. Unfortunately when I want to compress and decompress the sound buffer from DirectSound I only have jitter. Maybe you have some code samples how to compress standard DS buffers: LPDIRECTSOUND8 lpDS = NULL;

Hi All, i need your help in determining the problem in the following sample code (taken from speex manual) for fixed point encoding. i tested encoding a 160 sample frame and then decoding it back But The problem is that the decoded frame is totally different from the original frame(see output below).What is the possible reason for this?.thanks for any help #include<stdio.h>

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us