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Did you check it against the trunk in SVN? If it's not applied, and you can hook Jean-Marc up with an email address like yours, I'm sure he will get right on it. :) Tom Mihai Balea <mihai@hates.ms> wrote: > > Hi all, > > Did anything happen to this patch? > It seems to me that it fixes a valid issue, but I'm not an expert. > Anyways, I didn't see and follow-up or reply... > > Cheers, > Mihai > > On Oct 29, 2007, at 7:14 PM...

...fixed-point code > > There's a last minor thing: instead of > SPEEX_PREPROCESS_GET_LOUDNESS, it > should be called SPEEX_PREPROCESS_GET_AGC_LOUDNESS. If you can send > me a > patch that addresses these issues, I'll gladly apply it. > > Jean-Marc > > Mihai Balea wrote: >> Hi all, >> >> I'm working on a project called iaxclient >> (http://sourceforge.net/projects/iaxclient). >> We use speex for the majority of our audio processing, including >> voice >> compression and processing. >> One of the features...

Mihai Balea wrote: > What would be a good parameter to return that would better represent the > loudness of the signal? > As for the your other comments, I'll make the necessary changes and > resubmit. Could just be to first take the 5th root, a bit like in the code. Jean-Marc > Thanks...

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...n int (spx_int32_t) so that it can work easily with fixed-point code There's a last minor thing: instead of SPEEX_PREPROCESS_GET_LOUDNESS, it should be called SPEEX_PREPROCESS_GET_AGC_LOUDNESS. If you can send me a patch that addresses these issues, I'll gladly apply it. Jean-Marc Mihai Balea wrote: > Hi all, > > I'm working on a project called iaxclient > (http://sourceforge.net/projects/iaxclient). > We use speex for the majority of our audio processing, including voice > compression and processing. > One of the features of iaxclient is Analog AGC, which driv...

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...not "a hack" :) On Nov 5, 2007, at 2:08 PM, Tom Grandgent wrote: > Did you check it against the trunk in SVN? > > If it's not applied, and you can hook Jean-Marc up with an email > address like yours, I'm sure he will get right on it. :) > > Tom > > Mihai Balea <mihai@hates.ms> wrote: >> >> Hi all, >> >> Did anything happen to this patch? >> It seems to me that it fixes a valid issue, but I'm not an expert. >> Anyways, I didn't see and follow-up or reply... >> >> Cheers, >> Mihai >>...

...E to specify your sampling rate. Also, don't forget that the tail need to be longer (proportional to the sampling rate). Last thing, if you use resampling, make sure you use a decent resampler (the Speex one is fine) because otherwise, any aliasing left will not be cancelled. Jean-Marc Mihai Balea a ?crit : > 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...

Hi all, I have a couple of questions about the speex resampler: 1. Can a resampler state be used to process frames belonging to several audio streams? I would guess that there is information about the audio streams stored in the state so the answer is no, but it would be good to have some confirmation. 2. What is the best way to reset a resampler state? speex_resampler_reset_mem appears

Hi, I'm trying to use Speex Echo Canceller for mobile device. The AEC works well so far. However, I want to know how to remove reverberation significantly. I noticed it's not implemented yet in Speex AEC. Are you going to implement it in next version? Could you provide some ideas how to handle reverberation on current version? Thanks in advance. Sincerely, Dennis

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