similar to: win32 demo

? ? Hi Jean-Marc, When speech is played in speaker, we are feeding that speech to echo canceller for cancelling its echo from captured data and the echo canceller work very well. But when there is no speech, we are feeding samples of value zero to speaker and the same to echo canceller. But microphone is always capturing something. But when speech is played again after these zeros(silent

Hi Jean-Marc, I am developing a voice chat for win32 using speex,when I do echo cancellation, after 2 seconds adaptation, the echo canceller work very well,but after process silent(zero) echo signal(when enabled VAD,DTX),it work very bad,I have to reset it, it seems adapt faster then do not rest it, so I always here echo produce by my fisrt two second speech,how can I solve this problem?

Hi all, speex is great,echo canceller in speex is great,when I test echo canceller with testecho.c, it is much better then the echo canceller demo in intel IPP 5.1,but when I use the echo canceller in a real voice chat program for win32(16khz,20ms,winmm or portaudio or directsound), it is not work. Who can help me? I will pay money for a demo using dsound with source code.Thanks a lot.

speex-dev How can I user echo canceller with real time net stream,when I test it with file,it's very good,but when I use it in my voice chat,it is not work,is there some working demo like GIPS? best regards Yong Chen szcj23@tom.com

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets

I've tried some further debugging to see what mdf is actually doing. Instead of sending: tmp_out = (float)ref[i] - st->y[i+st->frame_size] to the output, I just sent st->y[i+st->frame_size] to see what was being subtracted from the microphone input. When I open this in Audacity, I see a very small signal at about -40dBm. The actual echo in my sample has a power closer to -20dBm.

Hi, I wanted to play around with the echo canceller so I tried the testecho program. As speaker sound I fed it with a 440 Hz sine wave, and as mic with a 550 Hz + 440 Hz sine waves sound (but the 440 Hz component half the amplitude). I kind of expected to get a somewhat clean 550 Hz sine out, but it doesn't seem to work that way. Are there better samples I can use with testecho to see

Thanks for alerting me to the new changes. I just tried the latest code from SVN, but unfortunately I still have just about the same results. The estimated echo that gets subtracted from the actual echo is such a small signal that it doesn't really result in any noticeable echo attenuation. I currently have my filter size set to 2 seconds even though the echo in my microphone file is only

Hi Jim, I've just been made aware of these problems (look for the thread "speex echo cancellation limitations"). It's on my short-term TODO list. > If fftwrap.c, I ifdefed out the spx_fft_float and spx_ifft_float routines, > because there were not used and required smallft.c (which is not so small at > all) to be added to the build. Right, need to cleanup that

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

Just tried your files and I'm not running into any infinite loops and the cancellation works fine. Unless the C6x has the same problem, I suspect a 16-bit problem. I'll check and see if I find something. About the r=0 problem, I can't find where it ends up in a denominator, so I suspect is not (directly) the problem. Jean-Marc Le lundi 08 mai 2006 ? 20:05 -0400, Jim Crichton a ?crit

> - 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

I ran some further tests on mdf and here are the results: 1. reduced tail length to 100ms, aligned mic and speaker signals to within 10ms - almost no echo attenuation 2. aligned mic and speaker signals to within 5 samples - still almost no echo attenuation 3. ran testecho using the same file for mic and speaker - very good echo cancellation (of course this is expected, but I needed to do a sanity

On 2019/10/24 ??11:26, Liu, Yong wrote: > >> -----Original Message----- >> From: Jason Wang [mailto:jasowang at redhat.com] >> Sent: Tuesday, October 22, 2019 9:06 PM >> To: Liu, Yong <yong.liu at intel.com>; mst at redhat.com; Bie, Tiwei >> <tiwei.bie at intel.com> >> Cc: virtualization at lists.linux-foundation.org >> Subject: Re: [PATCH]

Jean-Marc, I have traced the second infinite loop further. When st->adapted becomes true (mdf.c line 623), the first Yf[i] value is 4, the leak_estimate is 0xd4e, the resulting r is 3. The first value in st->Rf is 0, so e is 1, and r is set to e>>1, or 0. A little later there is a divide by r, and there is the hang. It seems that the 0 in Rf[0] is the problem, but I am not

Sergei, Thank you for your attention. My target is a custom VLIW DSP. I am not sure dependency dag is correct when it gets scheduled and packetized. Months ago, I submitted a bug at http://llvm.org/bugs/show_bug.cgi?id=17894 which explained more details. I am not sure my understanding of this bug is proper, but modified my local codes this way and it works for my target when scheduling and

Hi, For my VoIP application machine A sends speex encoded audio of to machine B and vice versa at. Data is captured in PCM 8Khz, 16 bit and then encoded using speex 1.1.12 The packet A played and the packet A captured through mic are the input to speex echo canceller. So I am trying to remove traces of packet A played from the captured data. I have followed example testecho.c All I hear is some

Yes, I did an offline upgrade. 1. Stop all clients using gluster servers. 2. Stop glusterfsd and glusterd on both servers. 3. Backed up /var/lib/gluster* in all servers just to be safe. 4. Upgraded all servers from 3.6.x to 3.10.x (I did not have quotas or anything that required special steps) 5. Started gluster daemons again and confirmed everything was fine prior to letting clients connect. 5.

> I built and ran the same test on the TI C64 simulator, and the echo was > canceled nicely (about 10:1 reduction in the peak amplitude during the > second of two brief speech bursts). So, my problem must again be related to > the 16-bit processing on the C5X DSPs. Good. At least we've narrowed it down a bit. > Also, the line where it is hanging is: >

David, thanks for your immediate response. Since iPTR is a reserved type for tablegen internal use, can you make a further explanation? On the other hand, it can be simply treated as a register class assignment problem during register allocation. Assume both pointer and integet have a 32 bit width. backend handles it just as to i32. When it performs register allocation, it can retrieve from