similar to: Compile failure current SVN

This small patch will make st->Zlast = Zframe, to allow applications access to an estimate of the signal-to-noise level. This used to be in there earlier, but was removed when Zlast was no longer used to compute Pframe. -------------- next part -------------- Index: preprocess.c =================================================================== --- preprocess.c (revision 10007) +++

Hi I just upgraded to http://svn.xiph.org/trunk/speex r10357 and got this build warning: alfredh@io:$ make -s mdf.o libspeex/mdf.c: In function 'speex_echo_cancel': libspeex/mdf.c:321: warning: statement with no effect libspeex/mdf.c:317: warning: `adapt_rate' might be used uninitialized in this function Is this intentional? In any case here is a simple fix: Index:

>> Current SVN compilation fails on Win32 (Intel C++ compiler 9.0): >> >> ..\..\libspeex\mdf.c(317): error: declaration may not appear after >> executable statement in block >> float adapt_rate; >> >> Just moved it up to the variable declaration block to solve it locally, but >> as it's not valid C, I thought I'd mention it ;) > >

I think you'll have to modify Speex to get the functionality you're looking for. I've made a few simple modifications to the AGC to prevent it from 1) exceeding a specified level of amplification and 2) enable and disable adaptation, so I can freeze it at a certain level while speech is not detected. It's mostly just a matter of doing this at the end of speex_compute_agc():

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

Jean-Marc, I have traced the second infinite loop further. When st->adapted becomes true (mdf.c line 623), the first Yf[i] value is 4, the leak_estimate is 0xd4e, the resulting r is 3. The first value in st->Rf is 0, so e is 1, and r is set to e>>1, or 0. A little later there is a divide by r, and there is the hang. It seems that the 0 in Rf[0] is the problem, but I am not

Hi, I made a small error in the impulse reponse function; when doing the inverse fft in fixed point, it will overflow unless it's properly scaled. This patch uses the same scaling as that used when updating the filters, and the outputs now have the same shape and the same scale in fixed and floating point. Best regards, Thorvald -------------- next part -------------- diff -ubBwr

Jean Marc & list, Following up on this- there seems to be an issue with this calculation in mdf.c (Smooth echo energy estimate over time) Pey += Eh*Yh; Pyy += Yh*Yh; it goes too large for floating point, which seems to be caused by extremely large values in st->Rf[j], st->Yf[j] and smaller values in st->Eh[j], st->Yh[j]- things were relatively quiet

Hi, I am currently working on a thin-client "low-cpu" usage telephone application. The current setup is that you can make phonecalls using an alsa supported usb headset and a smartlink based usb modem (a sweex usb modem, which -in large quantities- goes for 17.95 euro's) for analog lines. Unfortunately, it is really made as a modem and not for audio purposes, so, there is a pretty

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Hi Ted, Thanks a lot for this analysis. > In FLOAT_DIVU() it hangs at the following: > while (a.m >= b.m) > { > e++; > a.m >>= 1; > } > for the case where a and b are both zero (yes, division by zero). > This happens from mdf.c: True, that needs to be fixed even after I fix the rest. > leak_estimate =

And here's a patch for Intel Math Kernel Library. This allows commercial users of Speex to use a high-speed FFT library that isn't GPL'd. (You do need to pay for it though). This is 3 times faster than the default FFT in speex, and also faster than FFTW3 since MKL has native support for the complex packing Speex uses. Since Intel hasn't supplied any pkg-config files, and which

Hi, Attached is a patch which enables FFTW3 to be used by libspeexdsp. Using SSE, the FFT code is more than twice as fast, and it's 50% faster even without SSE. To enable the library, you'll need to call configure with '--with-gpl-fftw3'. Based on the benchmarking we did, this also changes the default FFT to smallft for floating point implementations when not using FFTW3.

Attached is a snapshot of work-in-progress of a FFT based resampler. At the moment it works in floating point only, and only basic quality inspection has been done. Some benchmarks comparing the filter-based resampler at Q3 with the FFT resampler with overlap = in_len / 2, using 20ms chunks of data. (-O3 -ffast-math, FFTW3, gcc 4.3.0 on x86_64) 16=>48: 59us vs 19us 16=>44.1: 204us vs

Alexander Chemeris wrote: > Hi, > > On 5/27/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> And here's a patch for Intel Math Kernel Library. This allows commercial >> users of Speex to use a high-speed FFT library that isn't GPL'd. (You do >> need to pay for it though). This is 3 times faster than the default FFT in >> speex, and

Alexander Chemeris wrote: > Hi, > > Here are some questions from user point of view. :) > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, >> and there aren't any obvious artifacts. I also did a 16=>16 test, and the >> results are delayed by 10ms

Just tried your files and I'm not running into any infinite loops and the cancellation works fine. Unless the C6x has the same problem, I suspect a 16-bit problem. I'll check and see if I find something. About the r=0 problem, I can't find where it ends up in a denominator, so I suspect is not (directly) the problem. Jean-Marc Le lundi 08 mai 2006 ? 20:05 -0400, Jim Crichton a ?crit

Hi, I'm moving my "metadata" which accompanies some of the speech packets into the speex bits. However, as usual, I'm a bit confused. The manual (PDF from current SVN), section 5.10 (bottom of page 21) states that: "Finally, applications may define custom in-band messages using mode 13. The size of the message in bytes is encoded with 5 bits, so that the decoder can