RE: Simple filtering in realtime - again...

I'm assuming that Buffer is defined as an array of shorts, you are
capturing 16bit stereo, and therefore lBlockAlignment is 4, is that right?

BUFFERSIZE appears to be the size in bytes of the captured audio.  So,
BUFFERSIZE / 2 is the number of Samples, and BUFFERSIZE / 4 is the number
of samples on each channel (i.e the number of 'blocks' or 'frames').

Assuming you want to capture 16bit stereo data, and you only want the left
channel data converted to floats.  I think you need something like this ...

#define BUFFERSIZE 1024	// Bytes
ALshort Buffer[BUFFERSIZE/2];	// Samples (left + right channel samples)
ALfloat flOutputSamples[BUFFERSIZE/4]; // Output Samples (only left channel
is kept)

// capture samples here ...

for (i = 0; i < BUFFERSIZE/4; i++)
	flOutputSamples[i] = Buffer[i*2]/32768.0f;

Notice that you only have BUFFERSIZE/4 samples now.


To convert it back to shorts to pass to alBufferData ...

for (i = 0; i < BUFFERSIZE/4; i++)
	Buffer[i] = (ALshort)(flOutputSamples[i]*32767.0f);

NOTE :  You should really add a check to make sure that flOutputSample[i]
is within -1.0 to +1.0 of else you may get clamping - which will sound bad.

I would try and solve this problem one step at a time.  e.g. confirm
capturing and playback are working, then add the conversion to float and
back to short, and then finally apply your filter.

Dan
Creative Labs (UK) Ltd.

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                                                                   Subject 
                                       RE: [Openal] Simple filtering in    
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Thank you Dan, but I'm afraid that didn't work. I think there's something
wrong with de casting, but I'm not very sure what is it, or how to fix it.



Subject: Re: [Openal] Simple filtering in realtime - again...
To: [email protected]; [email protected]
From: [email protected]
Date: Tue, 20 Apr 2010 09:38:40 +0100

I think this line is wrong ...

entport[i]=float((Buffer[2*i])/32768);

because you are doing an integer division and then casting the result to a
float. It should be something like ...

entport[i]=Buffer[2*i]/32768.0f;

Dan
Creative Labs (UK) Ltd.


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Hi,

I keep working on my realtime app.
I've been trying to find the cause of the troubles I had, but I'm still
stuck.

I'm testing now a simple DC blocking filter (just a high pass filter); it
seems to work in terms of frequency, but it adds a very loud constant
noise.
Here's a piece of the code - the rest of the main code is the
capture-playback routine with queue-  :

...

alcCaptureSamples(pCaptureDevice, Buffer,BUFFERSIZE / lBlockAlignment);


// This part is for separating left-right channels
int i;
for(i=0;i<BUFFERSIZE/2;i++){
  entport[i]=float((Buffer[2*i])/32768); // entport: input signal - left
channel
  //right channel not needed right now
}

DCfilter(entport,portnodc,BUFFERSIZE/2); // portnodc: input signal no dc


for(i=0;i<BUFFERSIZE/2;i++){
BufferSalida[i]=ALshort(32767*portnodc.y[i]); // BufferSalida: Output
buffer
}


// Attaching data to buffer
alBufferData(BufferID[ulQueueCount], PlayFormat, BufferSalida,
BUFFERSIZE/2,
lFrequency);

...

----------------------------------------

void DCfilter(samplesvector &x, nodctype &y, int size){

int i;

float a=0.995;
for (i=0;i<size;i++){

y.y[i]=x[i]-(y.xdcant)+(a*y.ydcant); //xdcant, ydcant: previous signal
values.
y.xdcant=x[i];
y.ydcant=y.y[i];
}
}
----------------------------------------

where:
typedef float samplesvector[BUFFERSIZE/2];

typedef struct {
samplesvector y;
float xdcant;
float ydcant;
}nodctype;

----------------------------------------

I'm a little bit overwhelmed about not being able to make this work, so,
any
help will be appreciated.
Any ideas?

Thanks in advance.
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