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Reverting r12591 as it was not complete and we're too close to

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freeze to risk fixing it now.  Will readdress after 2.0.5 is 
released.

Basically, RingBuffer is ill equiped to handle an input stride
other than 1.

Thanks to Peter for testing this for me.
This commit is contained in:
lllucius
2013-10-07 12:37:15 +00:00
parent 8cb1681e47
commit e5a4eecb25
3 changed files with 40 additions and 6 deletions
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40
src/AudioIO.cpp
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@@ -3613,10 +3613,44 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
if (len > 0) {
for( t = 0; t < numCaptureChannels; t++) {
gAudioIO->mCaptureBuffers[t]->Put(((samplePtr)inputBuffer) + (t * SAMPLE_SIZE(gAudioIO->mCaptureFormat)),
// dmazzoni:
// Un-interleave. Ugly special-case code required because the
// capture channels could be in three different sample formats;
// it'd be nice to be able to call CopySamples, but it can't
// handle multiplying by the gain and then clipping. Bummer.
switch(gAudioIO->mCaptureFormat) {
case floatSample: {
float *inputFloats = (float *)inputBuffer;
for( i = 0; i < len; i++)
tempFloats[i] =
inputFloats[numCaptureChannels*i+t];
} break;
case int24Sample:
// We should never get here. Audacity's int24Sample format
// is different from PortAudio's sample format and so we
// make PortAudio return float samples when recording in
// 24-bit samples.
wxASSERT(false);
break;
case int16Sample: {
short *inputShorts = (short *)inputBuffer;
short *tempShorts = (short *)tempBuffer;
for( i = 0; i < len; i++) {
float tmp = inputShorts[numCaptureChannels*i+t];
if (tmp > 32767)
tmp = 32767;
if (tmp < -32768)
tmp = -32768;
tempShorts[i] = (short)(tmp);
}
} break;
} // switch
gAudioIO->mCaptureBuffers[t]->Put((samplePtr)tempBuffer,
gAudioIO->mCaptureFormat,
len,
numCaptureChannels);
len);
}
}
}