similar to: Stereo AEC, transition to Git

can you try git 1.5.x just in case that's the problem? Jean-Marc Alfred E. Heggestad a ?crit : > Jean-Marc Valin wrote: > <snip> >> >> git clone http://git.xiph.org/speex.git >> > > I am getting this error: > > [...] > walk 8e52b996f69c9419e3057ce0ed215fcb93a7be45 > walk 0f6a78e98ea5e7ccc4b81d1d7064a4a251d019d7 > walk

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 =

Please, I noticed in the AEC module documentation that I can set the sampling rate and the frame size (in samples), but I found nothing regarding to the number of channels, no SET and no GET... So, I 'd like to know how I setup the channel's count on AEC.. if the buffer passed to the echo cancellation function should be mono or stereo... Another question: Is there a preferred

Hi, I plan to use AEC for a live performance, storytelling for very young children (and their parents!) in a mongolian yourte . Actually the storyteller can make vocal loops, there is an omnidirectional microphone in the center of the yourte, 5 loudspeakers in a circle along the yourte's wall and Pure Data in a linux box. And now she wants to make vocal loops over music and loops over

Hi, I wanted to know how the speex echo canceller can be used for multiple channels audio. I am aware that support had been added for multiple speakers and multiple microphones. Is there any documentation on how speex_echo_state_init_mc can be used in conjunction with speex_channel_decorrelate? Thanks, Mustafa -------------- next part -------------- An HTML attachment was scrubbed... URL:

If the left and right channels are processed separately as you do, the parameter Mic-Number in speex_echo_state_init_mc should be 1, not 2 On Wed, Feb 10, 2010 at 4:00 AM, <speex-dev-request at xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

The prospect of people actually putting B-format audio (via the panner or directly input) into Ogg/Vorbis brings an interesting challenge: What do we do with the audio after decoding it? The following sane options exist: A) Simply output the B-format audio B) Produce a downmix 1) Mono. 2) Stereo blumlein crossed pairs 3) Stereo UHJ 4) binaural C) Produce speaker feeds 1) Fully

Hi Opus list. I subscribed because your discussion on the IETF draft ("Ambisonics in an Ogg Opus Container") was mentioned on the sursound list. I tried Opus for ambisonics more than a year ago. It does works with uncoupled channels (I had to patch the encoder). I don't know what else could be done to optimize support for ambisonics, as I'm not a codec expert. So I think that

May I still try to sync them using some kind of DPLL (digital phase locked loop) mechanism? Something like matching clocks and interpolating samples to try to sync them together? Do the interpolated samples get to be eligible for an aec in your opinion? Thanks, Dario > To everyone on the list: do *NOT* attempt to do echo cancellation with > signals sampled using different clocks. This will

> 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:

The problem with different sound cards is that their clocks are not usually synchronized, and therefore the clock drift adds a non-linear factor to the audio path. The AEC can only cancel linear changes to the audio path, and so the AEC never converges.One solution is to measure the clock drift and resample either the input or output signal so that they *are* synchronized, and then the AEC

speex-devDear Jean-Marc Valin I got some problems with evaluating the AEC module of speex. I wrote a test main function and compiled it with the speex lib in VC6.0, it initialized the AEC state and called the AEC main function in the same way as what was done in testecho.c. The near-end input wave file was a simple delaying and adding version of the far-end input wave, eg. y(n) =

Ok, let me be more clear on this. AFAIK, Windows OS doesn't expose speaker input as other OS ( Linux, Mac OS...). That puts you in bad spot in using Speex AEC with windows. Only way to work is to use Soundcard with ASIO functionality which does give you speaker input. But then you would have to impose that requirement on all your users. I heard Speex AEC works great from developer who

> 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent > to the client and decoded when it is recevied so the AEC is always > performed on raw PCM16 8KHZ ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the

Jean-Marc Valin <Jean-Marc.Valin <at> USherbrooke.ca> writes: > > Le mercredi 07 juin 2006 ? 10:03 +0000, shridhar desai a ?crit : > > > > hi all, > > i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" > > in my VOIP application. Is it that the frame length to be chosen > > should always be 20ms or can i have

Hello Jean-Marc Valin and all the Speex-dev team, Today I have a little question about patents and the AEC. I' am not a lawyer, but after a deep research in esp@cenet.com, I found that in August 2003, an AEC called "Double talk insensitive NLMS algorithm" (DNLMS) have been patented by Gary Q Jin, Gordon J Reesor, and others (US 6.608.897 B1). It uses a continuous adaptation step

Analog Devices sponsored a fixed point version for AEC which was posted early this year, but it seems from the source code that the fft routines are still in floating point. Is the port still not complete or am I missing something? Has anyone out there ported speex AEC on RISC architecture? Please let me know. Thanks in advance, -Deepa -------------- next part -------------- An HTML

I've looked at some partial examples and read in the pdf Manual that the Acoustic Echo Canceller can be used outside of the speex codec as a preprocessor. I currently have a java app that uses the ffmpeg api for live streaming. I'm exploring integrating echo cancellation & I'd like to use the speexdsp AEC by mapping a javacpp library to communicate between java & speex. Audio

AFAIK, that's a common point for all AECs. But some of them solve the problem by resampling on of the end to keep it in sync with the other. On Tue, Jul 7, 2009 at 5:14 PM, ggb<ggb at tid.es> wrote: > Thank you John. > > On 07/06/2009 11:03 PM, John Ridges wrote: > > ly synchronized, and therefore the clock drift adds a non-linear > factor to the audio path. The AEC

Hi, I've got an AEC Star T4110 connected with an RS232 port by a megatec cable. On the list of supported devices (http://www.networkupstools.org/stable-hcl.html) there is only one AEC device, so I tried genericups with upstype=21. The problem is, that the UPS (while being on-line) reports constantly being on battery. # upsc aec ups.status OB My question is: What is the recommended procedure