similar to: Newbie: Get echo cancellation level

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,

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi I am using speex for cancelling echo.but i am not succeeded. I need echo cancellation in urgent.Please help m. Thanks in advance Devi

I recently stumbled upon <speex/speex_resampler.h> and made a chain of discoveries: - "http://speex.org/downloads" some links are broken ("rarewares" and git) - there is some (minor) development (whitespace and more) of the "dead" Speex codec ... will there be a release? (I don't really need such myself) - there is some (more useful?) development of the

Thanks ... > Those links seem to work here. 2 links "http://rarewares.hydrogenaudio.org" redirect to "https://hydrogenaud.io" ... no "rarewares" subdomain "http://git.xiph.org/speex.git" maybe better "http://git.xiph.org/?p=speex.git;a=summary" "http://git.xiph.org/speexdsp.git" maybe better

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Hi, These patches improve the resampler set_rate function. The first 2 patches do some cleanups in the GCD calculation and avoid an overflow when calculating the new phase. This patch could probably be simplified if we allowed 64bits operations in speexdsp. The 3rd patch avoid rounding errors in the phase calculation. The problem is that the new rate is calculated with the reduced rate, which

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

Hi all, I am attempting to test AEC behavior at various sample rates. I ran a little experiment: I recorded a 10 seconds voice clip and the resampled at 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000. I have a small applications that plays a wave file, records whatever comes in from the microphone and applies the Speex AEC and preprocessor on the input. It then saves the raw

I'm using Speex in Windows VoIP application and trying to implement echo cancellationI'm using waveInXXXX and waveOutXXXX API. I'm doing quite simple thing: After buffer is played (I get MM_WOM_DONE message) I'm just copying it's content to temporary playback buffer After buffer is recorded (I get MM_WIM_DONE message) I call speex_echo_cancellation(state, recorded_buf,

I did the test on another PDA which runs XScale 416Mhz CPU, the EAC arithmetic could be done in 1ms, but for the previous PDA (TI OMAP 168Mhz), it take more than 100ms! Though EAC could run in time on the new PDA, it did no help to cancel the echo, here is my test scenario: Init echo_state: echo_state=speex_echo_state_init(160,1120); int tmp=8000;

I've filed a bug for aarch64 https://github.com/xiph/speexdsp/issues/7 and provided the port in a fork with a pull request. We need someone to review/merge in the pull request? It provides the source code, but my testing was under Android builds, so there would be some configure changes needed to build it stand alone. On Tue, Apr 19, 2016 at 4:32 PM, Frank Barchard <fbarchard at

>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

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of

Hi all, I build successfully with speex-1.2rc2. And with speexdsp 1.2rc3, I build with i386, X86_64, armv7 and armv7s all passed. But when I build for ARM64 (for iPhone 6), it failed with: /Applications/Xcode.app/Contents/Developer/usr/bin/make all-recursive Making all in libspeexdsp CC preprocess.lo CC jitter.lo CC mdf.lo CC fftwrap.lo CC

2006/11/2, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>: > > Today I tried speex echo cancellation on my VoIP application running > > on a windows mobile based PDA (OMAP1510 168MHz). The result is > > disappointing: it takes 150-250 ms to do echo cancellation (by calling > > speex_echo_capture() ) for a 20 ms frame! > > Are you by any chance using an insane

Hi, *"That's in general not very reliable. You can get PEAQ to say all sorts of silly things." Can you provide me links for any more effective tools other than PEAQ? Which is more reliable for Speex resampler? * *"strongly suspect that it's just not compensating for the delay introduced by the resampler. Because higher quality means higher delay, you'd find that PEAQ

Hi, I want to do a careful measure of mic-to-net and net-to-spkr delays in sip media engine and to make it really precise I need to know delays, introduced by Speex algorithms. In particular, I'm interested in delays of: 1) encoder in NB, WB and UWB modes, 2) decoder (I guess it should be 0?) 3) resampler. I think it varies depending on input rate to output rate ratio. Any generic formula or

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?