similar to: Sound card problem in acoustic echo cancellation

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

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

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I'm after the cheapest way to decode a flac stream with following criteria; - Transport from UPnP (DNLA) NAS using either WiFi or Cat5 networking - Toslink digital out for using existing HiFi DAC. - Low power consuption (eg just few Watts - it's just a decoder anyway). There seems a real lack of Flac players that are cheap and HiFi separates integratable like current-day CD players. All

Well, I am not the person to be asked about how Speex AEC works :) And I was referring speaker input as having direct access into what has been played thru speaker. (ASIO Soundcard provides interface for that) So if I am remebering correctly, with data from Mic and Speaker in sync, speex AEC should do the job. Tim _____ From: Guilherme Balena Versiani [mailto:guibv at

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, >>> >>> I have a scenario in a mobile VoIP app that requires echo cancellation but >>> is somewhat different from what's described in the docs. >>> >>> Audio is received from and sent to the network at 8000Hz. Each packet >>> contains 160 samples worth a playback of 20ms. >>> >>> But the hardware

I'm being PHBed into a VOIP project, and Speex sprang to mind. Bandwidth is going to be a fairly serious issue for us. With regards to a Speex enc/decoder, I was wondering: Rick Kane and David Siebert have already asked about this, but seem to have gotten very different responses - the former a call to arms, and the latter a "well, if you do it, it'll get done." What's the

Hi again, I got a new dmesg log with the debug parameters, but I guess I could not isolate VESA. I blacklisted it at /etc/modprobe.d/blacklist.conf but comparing those logs I see no difference... Is it possible to isolate VESA somehow without removing it? The new logs are here, I'm posting full content because I'm not sure what information is more relevant...

Hi, all, I am porting speex on ARM7TDMI, I have done some optimization, the result is that the encoder and decoder need about 60 MCPS for 5.96kbps bitrate and complexity 0. Can someone give me informtion about Speex optimization on ARM7? Another question, my ADC and DAC are 12 bits, but Speex codec is 16bits, Did someone try to modify speex to 12 bits? I think if I modify speex to 12 bits, the

Hi Andrey, May be that we don´t understand. I need to transform speex to pocket pc platform. The Windows API is different to Pocket Pc API. Some IDEs that help us to create code for pocket pc, for example, are Visual C++ 3.0 embedded or Visual Studio.Net. Christian will develop this work with Visual C embedded. If you can develop this work quickly or you can guide me step by step, i would be very

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 to align far-end signal and near-end signal. BTW: Speex

Hi?Jean-Marc? Thanks for your suggestions very much! > > I am porting speex on ARM7TDMI, I have done some optimization, the > > result is that the encoder and decoder need about 60 MCPS for 5.96kbps > > bitrate and complexity 0. Can someone give me informtion about Speex > > optimization on ARM7? > > That's quite good. A few suggestions here: > 1) Don't

On 3/10/2012 at 11:18 PM yahoo2 at rcn.com wrote: |FLAC uncompressed is just rather a dumb idea (IMHO) because all you end |up wityh is a file that is rather large and will be larger then the |original WAV because of the FLAC container and metadata. Why would you |want FLAC uncompressed when compressed FLAC will not sound any different |because it isn't different. It's exactly the same

On Fri, 26 Aug 2016, Peter Griffin wrote: > This patch adds the DT nodes for the uniperif player > IP blocks found on STiH407 family silicon. > > Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen at st.com> > Signed-off-by: Peter Griffin <peter.griffin at linaro.org> > --- > arch/arm/boot/dts/stih407-family.dtsi | 76 +++++++++++++++++++++++++++++++++++ > 1

On Fri, 26 Aug 2016, Peter Griffin wrote: > This patch adds the DT nodes for the uniperif player > IP blocks found on STiH407 family silicon. > > Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen at st.com> > Signed-off-by: Peter Griffin <peter.griffin at linaro.org> > --- > arch/arm/boot/dts/stih407-family.dtsi | 76 +++++++++++++++++++++++++++++++++++ > 1

Hi Thierry et al. I noticed that since commit 3dde5a2342cd ("ARM: tegra: Add VIC on Tegra124") graphics on Apalis TK1 is broken. During boot it fails loading the vic firmware: [ 1.595824] tegra-vic 54340000.vic: Direct firmware load for nvidia/tegra124/vic03_ucode.bin failed with error -2 [ 1.606140] tegra-vic: probe of 54340000.vic failed with error -2 Subsequently Tegra HDMI

Wow! Ogg is sounding _really_ fine these days. Waaaay past my ability to tell from CD on my crappy sound card, even at pretty low bitrates. Which leads to my question: there's a huge difference between what my sound card puts out and what my CD player can do (Rotel RCD-950 to Classe' Audio Twenty preamp to Acurus A80 amp to Epos ES-12 speakers, in case anyone cares). If my computer is

On Fri, Aug 14, 2009 at 5:05 PM, Josh Coalson<xflac at yahoo.com> wrote: > it's unlikely flac will ever support floating-point samples natively. ?the main application for it is audio engineering, which demands easy editing and very high speed for both encoding and decoding above everything else. thats not why floating point is used. the highest current feasible bit resolution for

This series adds support for GM20B PLL's Maxwell features, namely glitchless switch and (more importantly) DFS support. DFS lets the PLL lower its output speed according to input current variations, making the clock more stable and allowing it to run safely at lower voltage. All GM20B additions are done in the last patch, which consequently ends up being considerably big ; fortunately, it