similar to: AEC in WB mode fixed yet ?

I did the test on another PDA which runs XScale 416Mhz CPU, the EAC arithmetic could be done in 1ms, but for the previous PDA (TI OMAP 168Mhz), it take more than 100ms! Though EAC could run in time on the new PDA, it did no help to cancel the echo, here is my test scenario: Init echo_state: echo_state=speex_echo_state_init(160,1120); int tmp=8000;

2006/11/2, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>: > > Today I tried speex echo cancellation on my VoIP application running > > on a windows mobile based PDA (OMAP1510 168MHz). The result is > > disappointing: it takes 150-250 ms to do echo cancellation (by calling > > speex_echo_capture() ) for a 20 ms frame! > > Are you by any chance using an insane

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 all, I am attempting to test AEC behavior at various sample rates. I ran a little experiment: I recorded a 10 seconds voice clip and the resampled at 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000. I have a small applications that plays a wave file, records whatever comes in from the microphone and applies the Speex AEC and preprocessor on the input. It then saves the raw

Hello, We are experiencing a few problems with Speex AEC. We are using it to process audio data on a real time stream over IP with Speex codec (frame size = 320). We initialize the echo state like this : SpeexEchoState * echo_state; echo_state = speex_echo_state_init(320,512); int sr = 16000; speex_echo_ctl(echo_state, SPEEX_ECHO_SET_SAMPLING_RATE,&sr); We are not really sure about

Hi, Can anybody show me how to enable stereo AEC, I tried with the following code and the result is bad, degraded output. Init: ec_state = speex_echo_state_init_mc(frame_size, aec_tail, 2, 2); speex_echo_ctl(ec_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sample_rate); preprocess_state_left = speex_preprocess_state_init(frame_size, sample_rate); preprocess_state_right =

Hi, I've set up speex AEC in our application. The echo's seem to be canceling, but the captured voice is distorted somehow. It sounds to me like low and/or high frequencies are removed. This happens even if I set my playback(echo) data to only zero's. Is this "normal" for the AEC? As far as I understand, if I send 0's as playback/echo data, the resulting data should be

I am using 'portaudio' and 'speex' to implement a speech audio capture/playback. Preprocess works fine but not the echo canceller (associated with preprocessor). I have read 'testecho.c' but it didn't help me. What I am doing bad? This is my implementation: One thread to capture and playback, both are different functions called in different times directly from

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,

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?

Hi Jean Marc, I found that the AGC API and algorithm has changed in svn head trunk. Is it safe to use it? Or is it for testing purpose only? You also said that VAD in svn is broken in a previous post, is it related to the AGC change? or can we mix the old VAD algorithm with the new AGC safely? kind regards, fredo

HA! Just when I was getting ready to drop myself from this list, along comes a ham radio question! Shoving digital voice down a 100 Hz pipe would be mighty darned tough/impossible. N0YMV has a good suggestion. Try those WSJT modes. They use narrow bandwidth and are fun to play with. There are documented instances where moon-bounce QSOs have been completed with these modes. No, it's

? hi all, i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" in my VOIP application. Is it that the frame length to be chosen should always be 20ms or can i have flexibility in chosing the frame lenght? on what parameters does the frame length choise depends? thank you all, Shri. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 implemented a small patch that allows the internal convergence state of the echo canceller to be saved in a file for later use, especially after a process restart or machine reboot. This enables immediate echo cancellation the second time the AEC is run. Of course this works only if the acoustic environment of the device doesn't change and if the soundcard latency is constant. To use

Hello, I've just caught a strange crash in speex_echo_state_init routine. It happened only on one WinXP machine, while on others using XP, Vista and 7 everything is fine. Crash occures in mdf.c line 434: st->spec_average = DIV32_16(SHL32(EXTEND32(st->frame_size), 15), st->sampling_rate); Got any ideas or should I provide more information of the OS? Thanks a lot!

Hi, my VoIP system uses speex with framesize = 160 samples(20 ms) at 8khz, sending and receiving paquets of 1600 samples(200 ms). When I receive a packet, I buffered it (I have also tried with speex_echo_playback) before sending to the speaker. When I capture from microphone, I fist remove DC offset (I saw in OPAL sources) and then I call speex_echo_cancellation for every one of the 10

Hello, When my Samba is the local master browser, it doesn't appear in the Network Neighborhood of the client. There is only the client itself. When the client (Windows NT 4 Workstation) is the local master, the Network Neighborhood shows the only two nodes that I have on my test network, ie the Samba server and the NT 4 Workstation. For the browsing, I use the default

Hi, I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE as in speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, &m_SampleRate ); I get 2 The encoder is initialized with m_pEncoderState = speex_encoder_init ( &speex_wb_mode ); On the decoder side, the correct sampling rate is returned: m_pDecoderState = speex_decoder_init ( &speex_wb_mode );

Hi, I still have the problem of "Out of connection structures" with Samba 2.2.6 or 2.2.7 and TSE clients. This problem was encountered by Richard van Beers : > From: richard van beers > To: jra@samba.org (Jeremy Allison), Gunther Hess > Subject: Re: Samba and Out Of Connection Structures from Terminal Servers (was on Solaris) > Cc: