similar to: trouble getting speex_echo_cancel() to work

> - set sampling rate to 8 kHz (at least for now) > - make sure the far end signal in the playback signal is always a bit in > advance (never late) compared to the mic signal. > - Set the tail length to something around 100 ms. > > Also, if you're using two different soundcards (as I understand) for the > playback and the capture, you're *never* going to get echo

Hi, A couple things you may want to check: - set sampling rate to 8 kHz (at least for now) - make sure the far end signal in the playback signal is always a bit in advance (never late) compared to the mic signal. - Set the tail length to something around 100 ms. Also, if you're using two different soundcards (as I understand) for the playback and the capture, you're *never* going to get

> I did a bit more testing, and ended up creating a set of testsamples, as > follows: > > ideal.pcm => me saying "This is what I'd like to hear" > junk.pcm => me saying "With a bit of luck, this is gone" > mixdown.pcm => ideal.pcm, with junk.pcm started 7 ms later and 3 dB lower. The problem with that is that it's nowhere near real

Hi! I want to use speex for echo cancellation in my program, but I have bad results. I will explain what my program does. it is a client-server application. I run a server in room A and a client in room B. the client sends some voice to the server and the server plays it on loudspeakers. I run another server in room B and connects to it from room A using the same application that runs

Hi. I've implemented "echo cancel" to voip project. But it doesn't work well . (I cannot feel the enhancement with activated AGC) My encode settings : 1. frame size : 320 bytes. 2. sampling rate : 16000 Hz (WB) 3. mono echo state initialized like this : echo_state = speex_echo_state_init(framesize, framesize*16); and befor encode : speex_echo_cancel(echo_state,

Hi, Indeed I too have troubles implementing echo removal, I like ask kindly to Jean-Marc (or any other) if him can put a source code demo to show us how to use effectively echo removal API and parameters in real case scenario. A big thank you! Roberto -----Messaggio originale----- Da: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] Per conto di hs Inviato: marted? 11 ottobre

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

Hello, Need some help using the speex_echo_cancel. I've read the documentation about the speex_echo_cancellation function: speex_echo_cancellation(echo_state, input_frame, echo_frame, output_frame); (in) echo_state => speex internal state. (in) input_frame => audio captured by mic. (in) echo_frame => the signal that was played in the speaker. (out) output_frame => the

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of

Hello Jean-Marc and all, I?ve got a problem with speex_echo_cancel usage. I only managed to get an attenuation of 5-8 db using testecho.c and it depends heavily on the playback/capture delay. I used clear speech as far-end input and same speech with a bit of reverb, attenuation and delay as mic nput. I also tried using played and recorded speech for mic input. I tried different constant delays

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of

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

Simply to say, in a quiet room, you can play a impulse signal and then find it's impulse response signal from the microphone. For example, if the delay between the impulse signal and its response signal range from 500 to 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the filter length to 4000. It is also called to align far-end signal and near-end signal. BTW: Speex

Hello, I would like to test the echo cancel algorithm available in unstable version 1.1.4. This echo canceller can be used with other codecs like G711? Somebody could send me some documentation or sample, or explain the next functions: SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); void speex_echo_state_destroy(SpeexEchoState *st); void

I'm going to test the echo cancellation... but before doing something wrong, i want to know what is the right way to use it... ------------------------------------------------------------------------------------- short * speakerSample; //will contain what the speakers are going to play short * microphoneSample; //what the microphone is going to register short* cancelledSample; //the ouput

hello jean-marc and everybody, I keep getting no results when trying to use speex_echo_capture, speex_echo_playback and speex_echo_cancel in a multi-threaded application, as suggested in the manual. Though, the cancellation works properly when i use a file with human voice for far-end input and the same file with echo added in SoundForge for mic input. When i try to insert and remove silence

Terrific! The patch fixes my problem quite decisively. I will want to use this with a 16K sample rate. Is it expected to work well with just another slight patch, or will we need to modify it more deeply? Also, are there any tunable parameters besides tail length? I am wondering about the aggressiveness of adaptation and I know you are an expert in that area. Thanks so much, Michael

Hi there. I am having a little trouble understanding the frame sizes chosen by the codec. testenc_uwb.c from the speex-1.0 source distribution has a framesize of 640 hardcoded and makes use of this value exclusively. However, a mode query on the actual codec returns 320 as a framesize for this mode. int tmp; speex_mode_query(&speex_uwb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);

Hi, Can someone please tell me how should I go about changing the frame size which is hardcoded to 160 for NB and WB and 320 for UWB. For NB speech(8KHz) the framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear