similar to: Why most AC97 soundcard has different sample rates of of capturing and rendering?

Hi all, We all know that mismatch between clocks of ADCs of far-end voice and near-end voice is not allowed in a time-domain or frequency-domain LMS based AEC system. It means that capture and render audio streams must be synchronized to a same sample rate. However, I found that this restriction is removed in microsoft AEC from Windows XP SP1. Anyone knows how microsoft AEC do it? This technology

I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio

Hello, I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2 compiled and installed. I have modules alsa 0.9.8 compiled and installed My PC have and audio card ac97 chipset intel i810 in motherboard. The list of the modules loaded is: namor:/etc/asterisk# lsmod Module ? ? ? ? ? ? ? ? ?Size ?Used by ? ?Not tainted snd-pcm-oss ? ? ? ? ? ?35652 ? 0 snd-mixer-oss ? ? ? ?

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

Hello, I am using Wine 0.9 under Linux Slackware 10.2 (official binary package) with kernel 2.6.14. I have a GeForce 2 Ti (using NVidia's 3d accelerated driver) and a VIA AC97 as a sound card (using kernel alsa drivers). I have tried several games : none would start until I switch sound acceleration from "Full" to "Emulation" (as suggested by wine's text output).

I've got Asterisk up and running on a VIA mini-itx M10K with a TDM-400 card but I'm having a terrible time with the sound setup. I tried RH9 and Debian distributions, with 2.4.18, 20, 22, 24 and 26 kernels. I've used the VIA drivers, ALSA drivers, OSS drivers and Kernel drivers with varying results. I can setup the audio to work with a few different applications, 2 channels and 6

Hi Underwood, Thank you for your help. I agree with your opinion. But it is almost impossible to further reduce the frequent difference between play and capture. 1. I used a 2^18 step FFT to analyse the echo frequency. So the freq resolution is 8000HZ/(2^17)=0.0625Hz. The analyser need at least 2^18/8000=32 seconds acoustic echo record signal from the microphone. Better freq resolution relies

Hi Steve, Have you read this paper? (Heping Ding, David I. Havelock, Drift-Compensated Adaptive Filtering for Improving Speech Intelligibility in Cases with Asynchronous Inputs. EURASIP J. Adv. Sig. Proc. 2010:) Let me call is paper-Drift. It provided a method to evaluate Relative Sample Offset (RSO, d[i]) which is omitted in the microsoft paper (Challenges and Solutions for Designing Software

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most

On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > Hi All, > Many Thanks to Underwood for her excellent review of our big trouble > which prevent LMS-based AEC algorithms to be used in most computer. > Maybe it can be summaried as follows: > 1. Different sample rate of sampling and rendering does exists in most > low-cost soundcards (In my experiments over more than 20 soundcards,

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

Hi Steve, > I don't know if this has only recently been put on line, but I never > noticed it until today - > www.iwaenc.org/proceedings/*2008*/contents/papers/9044.pdf > > That paper is from people at MS describing, in some detail, what the > Windows kernel echo canceller does to handle synchronisation issues. It > tracks both time varying sample clock drift and hiccups

Hi, What does an't locate module sound-service-0-3 mean in the context below? I don't use curl as far as I know. Should I "noload" it in modules.conf? [app_curl.so]Jun 19 08:41:25 b2 kernel: Registered tone zone 0 (United States / North America) Jun 19 08:41:34 b2 kernel: Intel 810 + AC97 Audio, version 1.01, 02:25:38 Dec 24 2004 Jun 19 08:41:34 b2 kernel: PCI: Found IRQ

Hi, I can't find any LINT file in the /usr/src/sys/i386/conf at my new 5.1 FreeBSD. Can u help me?

> > JR Richardson wrote: > > > I?ve got Asterisk up and running on a VIA mini-itx M10K with a TDM-400 > > card but I?m having a terrible time with the sound setup. I tried RH9 > > and Debian distributions, with 2.4.18, 20, 22, 24 and 26 kernels. I?ve > > used the VIA drivers, ALSA drivers, OSS drivers and Kernel drivers > > with varying results. I can

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

When I right click on desktop and bring up "sound". I have two items. Digital output (S/PDIF) buildin audio HDMI/Displayport 2 - Gk208 HDMI audio controller It seems I always default to S/PDIF.... How can I get this to default to HDMI always ? If I select HDMI and reboot it goes back to S/PDIF. I am running pulse audio. Thanks, Jerry

2011/4/20 Li Maoquan <limaoquan2000 at 126.com> > 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