similar to: [patch] speex AEC state save & restore

On 04/04/2011 06:58 PM, Simon Morlat wrote: > 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

> Today's Topics: > > 1. AEC with WB mode (Jean-Christophe.Berge@etu.enseeiht.fr) > 2. Multiple frame encode and decode (Reza Fatahillah) > 3. cant link speex_echo.h (jesus) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 20 Sep 2006 08:46:03 +0200 (CEST) > From:

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 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

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

Hi Is it possible to use only the speex echo cancellation module w/o using the speex codec? Here's the scenario: 1. I have my voice recorded in PCM audio file format 2. I want to cleanup the recorded voice by removing any echo included in the audio file/buffer 3. can I just use the ff APIs? - SpeexEchoState* speex_echo_state_init() - int speex_echo_ctl() - void speex_echo_capture()

Hi Anton Thanks for the info... What if I use this function do I still need to sync the playback and capture? Assuming that the captured voice (*rec) already contains the echo which I need to cancel out... -------------------------------- void speex_echo_capture (SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out) Parameters: st Echo canceller state rec signal

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 ?

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

Hi: Can anyone give some insight on: 1) As per the Speex manual, there is a mention of the relationship between input frame and echo frame (Pg 19, Para 2). I was wondering of what would be the ideal delay between these two frames, so that the echo is already fed to the canceller and the echo cancellation filter is not inefficient. 2) Corresponding to the above mentioned delay, what filter

Hello, We've integrated the speex echo canceller into PulseAudio, and coupled with the AGC/denoiser, it works reasonably well and has been a breeze to plug in, so thank you! One thing we're seeing, though, is that the canceller seems to take some time (a few seconds) to "learn", so initially the echo is clearly audible and it slowly fades out. This sometimes occurs in the middle

Aymeric Moizard a ?crit : > > I'm working on the echo canceller with the SVN, with Windows Mobile 5 > PocketPC. > > Obviously, I can't make it run correctly. What happens/doesn't happen? > My question: If I use the SPEEX_PREPROCESS_SET_ECHO_STATE, I have > changes in the sound: noise seems to be removed. Noise should be removed even without

Try using SPEEX_ECHO_SET_SAMPLING_RATE to specify your sampling rate. Also, don't forget that the tail need to be longer (proportional to the sampling rate). Last thing, if you use resampling, make sure you use a decent resampler (the Speex one is fine) because otherwise, any aliasing left will not be cancelled. Jean-Marc Mihai Balea a ?crit : > Hi all, > > I am attempting to test

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

> No, the 1.2beta1 preprocessor works rather well. I have a different opinion, but glad you like it anyway :-) > In fact, I very much welcomed the change that the preprocessor in > 1.2beta1 now starts out with the assumption of there being no speech, > and only activating and adapting AGC when where there is new activity > detected above that. As a result, with 1.2beta1, there is

Hi: I have been trying to test the speex AEC by playing a voice sample (pre-recorded) as a echo and capturing a new voice sample and playing the pre-recorded one, for reference. [Sampling rate 8 KHz, frame size 64] For some reason, I dont see any removal of echo. Does anybody have a idea of * Testing the AEC in some other mechanism or * Getting some sample files for testing AEC Thanks Su ---

Hi: By 'no success', I mean result gets worse in terms that, there comes in a shrill noise with high gain and also *no* attenuation of the echo. I am trying to validate the DC offset part. But can you please feed some light on how does the DC offset matter for AEC. As I have already mentioned, I tried adding silence (from 2 ms to 10 ms) in the start of the reference signal. This is for

Itay, >I am trying these things, but the main problem that has been bothering >me recently is that the fixed-point algorithm works "sometimes". Meaning >that sometimes it will work well, and other times it will not work at >all. > >I think I've found the source of the problem. The speex_alloc() >function, called by speex_echo_state_init(), calls calloc() and

I understand that the capture/playback signals need to be sync'd for an AEC to adapt. I'm a little bit confused on the requirements of synchronous sampling between the near end (mic/speaker) and the far end (phone line). I have an embedded DSP system with mic and speaker getting 1msec packets containing 8 samples. We can watch the DSP and ISDN clock frames drift and every few minutes we

Hi: I'm using speex to perform echo cancellation in Windows. I'm aware of the problem about out of sync clocks in record and play sample rates in usual sound cards . In order to have an idea of how good is my echo cancelation working I would like to know if there is any #define thing i can pass to speex_echo_ctl to get the actual level of echo cancellation. If not, how can i extract that