search for: reverberation

Hi, I'm trying to use Speex Echo Canceller for mobile device. The AEC works well so far. However, I want to know how to remove reverberation significantly. I noticed it's not implemented yet in Speex AEC. Are you going to implement it in next version? Could you provide some ideas how to handle reverberation on current version? Thanks in advance. Sincerely, Dennis -------------- next part -------------- An HTML attachmen...

...mentation on the Speex echo canceller, and unless I'm confused, I think there's a mistake somewhere. The documentation on http://speex.org/docs/manual/speex-manual/node7.html#SECTION00740000000000000000 says, " The recommended tail length is approximately the third of the room reverberation time". However, the Octave script echo_diagnostic.m has the code: if (b > .3*tail_length) printf ('This is too much delay, try delaying the far-end signal a bit\n'); This seems to mandate that the tail length should be at least 3 (or 3.33...) times the delay, which co...

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

...; 2) Corresponding to the above mentioned delay, > what > > filter length should we define while creating the > echo > > canceller state so that we can expect the AEC to > > remove or reduce the echo. > > So far, my rule of thumb is to use about half of the > 60 dB reverberation > time. For example, in a small room, the > reverberation time is in the > order of 250 ms, so a tail length of 125 ms would be > good. > > Jean-Marc > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protectio...

Hi Jean-Marc > Can you explain what you mean here by "speech switching" By speech switching I mean the adaption of "gain2" when near-end or far-end is talking. What is important is that the timing is good and that the gain is low/high while far/near-end is talking. By timing I mean that the "gain2" should remain low until all far-end talk is final and that the

...ex echo > canceller, and unless I'm confused, I think there's a mistake > somewhere. The documentation on http://speex.org/docs/manual/speex-manual/node7.html#SECTION00740000000000000000 > says, " The recommended tail length is approximately the third of > the room reverberation time". However, the Octave script > echo_diagnostic.m has the code: > > if (b > .3*tail_length) > printf ('This is too much delay, try delaying the far-end > signal a bit\n'); > > This seems to mandate that the tail length should be at least 3 (or...

...r environment. If there are still analog devices in addition to VoIP, I'd say always, but Asterisk has a rudimentary echo canceller already on board. The Telcos use echo cancellers themselves, but it cannot hurt to have a hardware canceller on your BRI card. Nowadays I see more problems with reverberation in connection with cheap speakerphones or simple mics and speakers on PCs, but that's a different story. jh

hi list this is as probably more of networking problem than an icecast problem but it is related - hope people don't mind. I've been running tests for a project that starts 20 August. I'm sending stream from my audio software to a local icecast server, then relaying this to a commercial shoutcast service. Initially i was using 7000 for the relay port. Worked fine. Then one morning

hello Another beginners question..... I've been googling through the icecast archives looking for information on how icecast handles multiple clients. Haven't been able to find what I'm looking for. Could someone please explain (or direct me to links) what happens to the upload bandwidth of a box running icecast, when more than one client connects? For example, if a box on a 128kbps

...the preprocess.c file (alloced in line 167). It's in >> Speex 1.1.4 >> by the way. > > Oops... Thanks for letting me know. I'll change that for the next > release (in the mean time, the fix is obvious). In case you're > interested, I'm currently working on a reverberation suppression > algorithm. I'll put it in CVS. Reverberation suppression? I guess this would help reduce local source echoes? I've never _noticed_ that to be a problem in my use, but I would imagine that using a notebook's built-in microphone, you'd get some echo off of the s...

...g plus a fairly reverberant room. We > have been able to solve this by adding a weighting factor and some > accumulation on the residual_echo and echo_noise. This modification > works perfect on the far-end-problem but worsen the main problem even > more. This is probably the effect of reverberation and can probably be solved by tuning/improving the current recursive averaging of the echo estimate. Jean-Marc

Hi Iain, It's a simple answer: > For example, if a box on a 128kbps upload connection is > serving a 64Kbps stream (and not audio-on-demand) - is it limited to two > connections Yes it is. > or is icecast somehow more efficient? No - icecast can't work miracles! The underlying technology of the internet (the way it is at present, at least) only allows a packet of

Unlivable. I fix it. I forgot the speex_preprocess(m_pPreprocess, buf, NULL) need to do enough samples too. Tks a lot! And speex_preprocess_ctl(preprocess, SPEEX_PREPROCESS_SET_DEREVERB, &tmp); f=.4; speex_preprocess_ctl(preprocess, SPEEX_PREPROCESS_SET_DEREVERB_DECAY, &f); f=.3; speex_preprocess_ctl(preprocess,

This is a heads up to let you know of my plans to work on a senior thesis project dealing with editing Vorbis losslessly in Audacity. You can find more information about my plans and what I'm up to here: http://www.reverberate.org/computers/thesis/ "vcut" from vorbis-tools will be an excellent starting point for me. I notice that it's marked "experimental"; why is

Hello The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() function of the preprocess.c file (alloced in line 167). It's in Speex 1.1.4 by the way. I'm trying to make the denoiser work with my application and has got reasonable noise reduction after applying the denoiser. I, however, haven't been able to find any information of what the purpose of

...ant room. We >> have been able to solve this by adding a weighting factor and some >> accumulation on the residual_echo and echo_noise. This modification >> works perfect on the far-end-problem but worsen the main problem even >> more. > >This is probably the effect of reverberation and can probably be solved >by tuning/improving the current recursive averaging of the echo estimate. Yes, I have basically solved this. Our main problem with the poor reliability during near end talk is what we need to find a solution to now. Best Regards Johan

Hello again. If you recall, I am the guy working on enabling Audacity to losslessly rearrange Ogg Vorbis files for my senior thesis. I have more or less completed the required modifications to Audacity to support this. You can read more about the work this entailed by reading my journal: http://www.reverberate.org/computers/thesis/journal Now my attention moves to the Ogg Vorbis code

...rom the flows or insert zeros into them when AEC is started. 3. The delay (let's call it D) between samples at the same position of near-end signal and far-end signal is constant if acoustic echo path is fixed, and ADC and DAC of the soundcard have exactly the same sample rate. 4. If the echo reverberation time is A cycles, We know the length (L) of AEC filter must be longer then A, and delay between near-end signal and far-end signal cound be set to (L-A)/2 in order to deal with drift of echo path. For example, echo reverberation time of a meeting room is 2500 cycles, you can set the length of the...

> - 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

...ate_destroy() > function of the preprocess.c file (alloced in line 167). It's in Speex 1.1.4 > by the way. Oops... Thanks for letting me know. I'll change that for the next release (in the mean time, the fix is obvious). In case you're interested, I'm currently working on a reverberation suppression algorithm. I'll put it in CVS soon. Jean-Marc -- Jean-Marc Valin http://www.xiph.org/~jm/ LABORIUS Université de Sherbrooke, Québec, Canada -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature...