search for: reverberate

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

Hi, I've been trying to understand the documentation 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

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

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 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

David Baker a ?crit : > Hi, > > I've been trying to understand the documentation 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

> Thank you all for valuable input, > > another question: when do I actually need the echo cancellation > (hardware / on board /on module ) ? > It depends on your 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

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

On Mar 28, 2004, at 8:23 PM, Jean-Marc Valin wrote: >> 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. > > Oops... Thanks for letting me know. I'll change that for the next > release (in the mean time, the fix is

Johan Nilsson a ?crit : >> There's also a parameter to control the maximum amount of >> suppression allowed: SPEEX_PREPROCESS_SET_NOISE_SUPPRESS : noise >> suppression SPEEX_PREPROCESS_SET_ECHO_SUPPRESS : echo suppression >> when there is no local talk >> SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE: echo suppression in >> double-talk > > Yes, I am

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 this? Are there important problems that need to be worked out before it will be reliable, or does it just need more testing and bett...

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

>What happens if you make SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE less >aggressive. Does it end up with too much echo or it just doesn't realise >that it's in double-talk conditions? My impression is that it does not make much difference on the timing to set this parameter less aggressive. Depending on the how loud the near end is talking it may detect double talk but most

...o 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 necessary to do all this splicing. From reading the Ogg documentation and the vcut source I have a good understanding of the Ogg format and how I will go about rearranging Vorbis packets. However I have a few questions I...

>> 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

> - 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

> 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. 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