similar to: AEC with different soundcards

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

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

Quoting John Ridges <jridges at masque.com>: > Speex tells me that the decoder is always 5 ms, but it says that the > encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is > there an extra frame of delay in the encoder that isn't otherwise > accounted for? Oh, delay = frame_size + lookahead If you have a frame size of 20 ms, then there's no choice but

Speex tells me that the decoder is always 5 ms, but it says that the encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is there an extra frame of delay in the encoder that isn't otherwise accounted for? John Ridges Jean-Marc Valin wrote: > Quoting John Ridges <jridges at masque.com>: > >> I also need to know the precise delays from Speex but I used

Quoting John Ridges <jridges at masque.com>: > I also need to know the precise delays from Speex but I used the > SPEEX_GET_LOOKAHEAD control requests to determine them (plus the > "speex_resampler_get_output_latency" function from the resampler). The > returned values from the Speex lookahead request don't seem to match > with the values you gave Alexander. Am I

I've been testing the AEC and have experienced it working on one sound card, and not working when using different cards for capture/playback so I know it's a real problem. And yes I know the documentation says it won't work "regardless of what you may think." I'm the inquisitive type; I'm looking to understand just what exactly is happening here. One discussion in

JM, I also need to know the precise delays from Speex but I used the SPEEX_GET_LOOKAHEAD control requests to determine them (plus the "speex_resampler_get_output_latency" function from the resampler). The returned values from the Speex lookahead request don't seem to match with the values you gave Alexander. Am I doing this wrong? Thanks, John Ridges speex-dev-request at

I?ve tried adding support for OPUS_FAST_INT64 to celt/arch.h, and I?ve found that this is indeed comparable in speed, if not a touch faster, than my inline assembly. I?ll submit patches for this. The inline assembly parts of my aarch64 patch set can thus be considered withdrawn. I haven?t yet tried replacing SIG2WORD16 (or silk_ADD_SAT32/silk_SUB_SAT32) with Neon intrinsics. That?s an obvious

Hi All, In order to deal with acoustic echo cancellation problems of most PCs which sound cards have different capture and play frequencies. I made a trial. At first, a 1000Hz sine wave is played for a long time via a speaker and its acoustic echo is recoreded. Seconds, get the frequency of the echo by a FFT analyser. So the difference between capture and play frequencies is obtained. Thirdly,

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,

Something looks odd without your values (or the doc) because hypergeom_gain() should really approach 1 as x goes to infinity. But in the end, an approximation is probably OK because denoising is anything but an exact science :-) Jean-Marc Quoting John Ridges <jridges at masque.com>: > By my reckoning the confluent hypergoemetric functions should have the > following values: >

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

It's been a while since I did the maths. M(-.5;1;-x) optimises something else, though it's not too far. I think it may be [M(-.25;1;-x)]^2 (or is it the sqrt?) that is supposed to be there. In any case, if there's a mismatch between the doc and the code, the code is the one likely to be correct. Jean-Marc John Ridges a ?crit : > I got the approximation from a Google book: >

> On Nov 13, 2015, at 1:51 PM, John Ridges <jridges at masque.com> wrote: > > Hi Jonathan, > > I'm sorry to bring this up again, and I don't want to beat a dead horse, but I was very surprised by your benchmarks so I took a little closer look. > > I think what's happening is that it's a little unfair to compare the ARM64 inline assembly to the C code,

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

Thanks, I look forward to seeing what you find out. BTW, I was wondering if you tried replacing the SIG2WORD16 macro using the vqmovns_s32 intrinsic? I'm sure it would be faster than the C code, but in the grand scheme of things it might not make much difference. On 11/13/2015 12:15 PM, Jonathan Lennox wrote: >> On Nov 13, 2015, at 1:51 PM, John Ridges <jridges at masque.com>

Hello! I'm attempting to implement Speex AEC support in GStreamer, using work started by Olivier Cr?te. In the Speex manual, I see this text: "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may think. The only exception to that is if the two cards can be made to have their sampling clock ``locked'' on the same clock

On Nov 23, 2015, at 12:04 PM, John Ridges <jridges at masque.com<mailto:jridges at masque.com>> wrote: Hi Jonathan. I really, really hate to bring this up this late in the game, but I just noticed that your NEON code doesn't use any of the "high" intrinsics for ARM64, e.g. instead of: int32x4_t coef1 = vmovl_s16(vget_high_s16(coef16)); you could use: int32x4_t coef1

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of

I got the approximation from a Google book: http://books.google.com/books?id=2CAqsF-RebgC&pg=PA385 Page 392, formula (10.33) Using this formula, you're right, hypergeom_gain() would *not* converge to 1 for large x, but would instead be gamma(1.25)/sqrt(sqrt(x)) which would approach zero. Now if the formula for the hypergeometric gain were instead gamma(1.5) * M(-.5;1;-x) / sqrt(x)