similar to: speex documentation

Hi, Tank you for your reply. I have tried the sample programs given the manual and done encoding and decoding a wave file. That is working fine. I have already read the manual and API reference. But this problem did not occured with the wave file encoding/decoding. While capturing the data from MIC and encoding/decoding is giving this problem. Thanks and Regards, Vijender Reddy, -----

The reasons I have posted these questions are: 1) To find out if Speex can take advantage of SIMD extensions. 2) To maybe learn from someone with previous experience in optimizing Speex for moderns x86 architectures before I set off trying all kinds of things on my own. See answers inline: 2009/6/15 Tom Grandgent <tom at grandgent.com> > Why haven't you tried using release build

This must be a simple issue, but I cannot figure it out. I want to use VAD, but I don't know how to check if the actual frame has voice in it or not. So, in my code, I do: int tmp = 1; speex_preprocess_ctl(preprocess_state, SPEEX_PREPROCESS_SET_VAD, &tmp); speex_preprocess_ctl(preprocess_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp); then later, for each frame

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

Thanks for your reply. I changed my code to: if (speex_preprocess_run(preprocess_state, shortPointer) == 1) { speex_encode_int(enc_state, shortPointer, &enc_bits); } In the mobile version of the software, compiled against the mobile build of Speech, I get 1 and 0 based on whether the speech is detected. In the version of the software compiled against the Win32 version of Speex,

If more than one speaker receives the *same* signal, it doesn't matter the number of speakers. It only gets tricky when the speakers are playing slightly different signals (e.g. from a stereo song). Jean-Marc Quoting Greger Burman <greger at mobile-robotics.com>: > One thing I can think of is if you are using two or more speakers. If the > speakers are not at the exact same

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

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

I have a question about the overall performance of Speex and what I can do to improve it. I'm running Speex Windows x86, Visual C++ EE compiler. I will say right away that I've only compiled debug so far and used no compiler optimizations at all. I use the uwb-mode, preprocessing, denoising and echo cancellation. I've noticed that speex consumes a lot of cpu resources. When I run this

Anton A. Shpakovsky <saa <at> tomsksoft.com> writes: > > 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

Hi, I got the following problem with speex. Could you help me to resolve the following issue. Description: I am taking speech input from MIC storing in a buffer after that i will encode by using speex then decoding and sending to output(speakers). it is small demo project of speech encoding/decoding in VC++. I am reading the data from buffer in the following way where data is short[] type and

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

On 2010-03-16 14:22, Josh Gargus wrote: >> If more than one speaker receives the *same* signal, it doesn't >> matter the number of speakers. It only gets tricky when the >> speakers are playing slightly different signals (e.g. from a stereo >> song). > > Does "tricky" mean that the Speex AEC won't handle such situations > well? Or just that you

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally,

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Hi all, as in subject, speex_preprocess inject noise in my data. Someone can help ? Here's the way that i'm using: #define NN 160 /* 20msec di audio */ #define AUDIO_SAMPLERATE 8000 spx_int16_t TEMP_Buffer[NN]; speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); c = denoise; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_DENOISE,&c); c = agc;

Hi all and Jean-Marc, Firstly thanks a lot to you for helps, at last I accomplished to use speex-1.2beta1 in my speech application. I try to understand speex source code and improve speech quality. The problem in my application (like many people in the list says) is noise. So I've added VAD in preprocessor as following. VAD in preprocessor is very good, there is not noise outside speech. I

hi guys, i am trying to develop an application having a capability to detect voice activity in voice stream and then record only portion of this stream which contains the activity. i went through these steps. 1. Captured voice through Waveform Functions of Windows Multimedia API 2. Started Speex Preprocessing liberary 3. Turned on preprocessor by calling speex_preprocess_state_init and

Hello, all. 1) ACE does not work for me I am in a voip project using Speex, failed to have hte Speex ACE work. here is how I initialize it: /** * Configurations : * #define BITS_PER_SAMPLE (16) * #define SAMPLE_RATE (8000) * #define CHANNEL_NB (1) * #define DURATION (20) * SPEEX_MODEID_NB */ _eco_state = speex_echo_state_init(_encframe_size, 10*_encframe_size); speex_echo_ctl(_eco_state,

I'm trying to understand a bit more about the inner workings of the speex resampler. I want to do resampling on a voip stream. High quality and low latency is important. 1) What does speex_resampler_magic() do? 2) I find that resampling has quite an impact on performance and I'm trying to figure out where the cpu cycles are going and see what can be done to reduce the cost. I ran a profile