similar to: Bitrate in encoder

> speex_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: Did you make sure it really *sets* and not just return an incorrect result? > speex_bits_init(&globals->bits); > globals->enc_state = speex_encoder_init(&speex_nb_mode);

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 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:

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

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

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>

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

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

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

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,I'm having performance issues encoding with speex on an iphone4 + iOS5. The same code yields excellent results on iPad2 + iOS5, Mac OS X 10.6 as well as Windows 7. The call to speex_encode_int is takes 0.0065 to complete. Because encoding is so latent the receiver incurs underruns. Here is my speex setup: ? ? unsigned val = 0; ? ? /**? ? ?* Initialize the encoder in narrow band mode.? ?

I've targeted Speex to run under eCos with Thumb interworking using arm-elf-gcc 3.3.3 I'd written a small test case to verifiy the operation: #include <speex/speex.h> void speex_test(void) { SpeexBits bits; unsigned int frame_size; void *enc_state; float *speex_frame; cyg_uint64 t1, t2; int i; speex_frame = (float *)0x00030000; /* Address of Speex data

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

Hello, I'm trying to use the speexlib but I just can't do it. I've made just like the example (on the site) says. Some times I get SIGFAULT. Also, I'm reading the input_frames from /dev/dsp (directly). What is missing here? speex_bits_init(&bits); enc_state = speex_encoder_init(&speex_nb_mode); speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frame_size);

int frameSize; speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frameSize); 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 buffer. Also, do I need to call speex_encoder_ctl( enc_state, SPEEX_SET_SAMPLING_RATE, &sampleRate ); Depending on sample rate I record my audio in? If not, is there any benefit to

Hi, in my application i quite often (not allways) get those two warning: "Invalid mode encountered: corrupted stream?" "Invalid wideband mode encountered: corrupted stream?" The stream is not corrupted but i wish to know what those warning mean to correct my code. My encode and decode routines are those (libspeex 1.1.4): SpeexBits enc_bits, dec_bits; void *enc_state,

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

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,

I bought a ADSP-BF533-EZkit-LITE V2.1 to develop a speech codec application using SPEEX. I have tested SpeexEcho project from both BlackfinSDK-Rel201 and BlackfinSDK-Rel310. When disabling SPEEX processing, i.e. using PASS-THROUGH mode, the output voice is ok, which is exactly as the input voice. However, when enabling SPEEX processing, the output is bad, no matter how to set the parameters in

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"