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Hi Andras, others, Andras Kadinger <bandit <at> surfnonstop.com> writes: > > > Daniel, > I recommend you to start from a simple case and gradually progress > towards your goal. > Can you make things work with the "Speex in a Disco" (Example 6) > testcase at http://ns.surfnonstop.com/~bandit/speex/echoc...

The echo suppression in SVN is shaping up quite nicely! I have also observed some - occasional, apparently data-dependent - misbehaviour in AGC. http://www.surfnonstop.com/~bandit/speex/echocard1/

Hi, I am using speex-1.2 beta1 version. When I am using sample files from http://www.surfnonstop.com/~bandit/speex/echocard1/ ,example 6, as input to mdf through testecho the output is fine However when I am recording the signals in echocancel.cxx in OpalEchoCanceler::ReceivedPacket taking speaker signal from echo_buf and mike signal from ref_buf, echo cancellation is not affective. Can...

2011.06.21. 19:22 keltez?ssel, Arun Raghavan ?rta: > Hi Andras, others, > > Andras Kadinger<bandit<at> surfnonstop.com> writes: >> >> Daniel, >> I recommend you to start from a simple case and gradually progress >> towards your goal. >> Can you make things work with the "Speex in a Disco" (Example 6) >> testcase at http://ns.surfnonstop....

...ct/definition file is out > of date. That or you're simply using an old (1.0.x or early 1.1.x) > version of Speex. > > Jean-Marc > > > ------------------------------ > > Message: 3 > Date: Fri, 09 Feb 2007 15:05:01 +0100 > From: Andras Kadinger <bandit@surfnonstop.com> > Subject: Re: [speex-dev] Speex Options > To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> > Cc: speex-dev@xiph.org > Message-ID: <45CC7F8D.8070500@surfnonstop.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > When you say inverted,...

I'll try to make some shorter samples later, but for now here are the ones I have tried with: http://www.surfnonstop.com/~bandit/speex/1.2beta1_AEC_garble/ The original recordings are in mic.raw and spk.raw. Jean-Marc Valin wrote: > You may have triggered an instability problem. Can you upload your files > somewhere so I can have a look at them? > > Jean-Marc > > Andras Kadinger a ?crit : &gt...

On Tue, 2011-06-21 at 20:18 +0200, Kadinger Andr?s wrote: > 2011.06.21. 19:22 keltez?ssel, Arun Raghavan ?rta: > > Hi Andras, others, > > > > Andras Kadinger<bandit<at> surfnonstop.com> writes: > >> > >> Daniel, > >> I recommend you to start from a simple case and gradually progress > >> towards your goal. > >> Can you make things work with the "Speex in a Disco" (Example 6) > >> test...

> I am using speex-1.2 beta1 version. When I am using sample files from > http://www.surfnonstop.com/~bandit/speex/echocard1/ ,example 6, as input > to mdf through testecho the output is fine That's a good start indeed. > However when I am recording the signals in echocancel.cxx in > OpalEchoCanceler::ReceivedPacket taking speaker signal from echo_buf and > mike signal from...

Hi, Arun and me are working on a echo-canceler module for pulseaudio build around speex. When using the preprocessor's echo filter, specifically in combination with denoise and AGC, it's unclear whether these have to run before or after the AEC. Right now we've got one instance of speex_preprocessor doing AGC and denoise running before the echo canceler; and a second for echo

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

Greetings everyone, I was about to compare AEC performance between 1.1.12 and 1.2beta1 when I noticed something. If I configure (and compile) speex-1.1.12 with ./configure --enable-shared=no --enable-static=yes it compiles and works as expected: I can run a mic and speaker signal through testecho, it runs in a reasonable amount of time (about 23 secs for 3 minutes of audio) and I get back

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

Jean-Marc Valin ?rta: > Andras Kadinger wrote: >> I have now found a testcase where SVN AGC starts out good, but then it >> keeps increasing the gain but never decreasing it, and eventually goes >> out through the roof, during the same single recording. > > I've been doing massive changes in the svn preprocessor, so I'm not too > surprised I broke the AGC,

2011.05.25. 4:52 keltez?ssel, Arun Raghavan ?rta: > On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: >> The fact that the AEC takes a few seconds to converge is normal. The > Do you think there might be a way to reduce this? The shorter you make the tail length the faster it will adapt. But this will make it more important to reduce HW/SW latency so that you don't waste

2011.06.21. 22:57 keltez?ssel, Arun Raghavan ?rta: > On Tue, 2011-06-21 at 11:39 -0700, Arun Raghavan wrote: > [...] >> I'm also running this on x86 (x86_64, technically), and it's all >> floating-point, so I guess this is a regression somewhere. Will try to >> see if I can run it without any optimisations if possible, which I >> assume should serve as an

2011.08.23. 15:38 keltez?ssel, Yanick Bourbeau ?rta: > Since I don't have access to different channels as I record a phone call > using a man in the middle approach, there is something else I can use > to equalize the sound ? What I would do then probably is try to manually separate the two channels/talkers; say channel 1 goes from 0 seconds to 13 seconds, channel 2 goes from 13

[At the risk of educating you about something you might already know] Natural speech in most human languages gradually changes from one phoneme to the next. Concatenating phonemes together from a fixed, prerecorded, unflexible set would give rise to abrupt changes between them (both in phoneme quality and in pitch), and thus make the resulting speech hard to understand and/or uncomfortable

When you say inverted, do you mean a) input and output exchanged with each other, or b) input or output having changed sign numerically (e.g. multiplied by -1)? Andras Jean-Marc Valin wrote: > Make sure that: > > 1) You're using svn head, which has many improvements > 2) The AEC inputs aren't inverted > 3) The delay between the near end and far end signals isn't too