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ALSA fix. Removed some win32 compile warnings. Refactored Variant class internally.

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This commit is contained in:
Julian Storer 2010-06-30 18:16:12 +01:00
parent e502d753d7
commit 366c8fb7e2
10 changed files with 633 additions and 456 deletions
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11
src/audio/audio_sources/juce_ResamplingAudioSource.cpp
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@ -110,6 +110,7 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
bufferPos %= bufferSize;
int endOfBufferPos = bufferPos + sampsInBuffer;
const int channelsToProcess = jmin (numChannels, info.buffer->getNumChannels());
while (sampsNeeded > sampsInBuffer)
{
@ -129,7 +130,7 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
{
// for down-sampling, pre-apply the filter..
for (int i = jmin (numChannels, info.buffer->getNumChannels()); --i >= 0;)
for (int i = channelsToProcess; --i >= 0;)
applyFilter (buffer.getSampleData (i, endOfBufferPos), numToDo, filterStates[i]);
}
@ -137,7 +138,7 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
endOfBufferPos += numToDo;
}
for (int channel = 0; channel < numChannels; ++channel)
for (int channel = 0; channel < channelsToProcess; ++channel)
{
destBuffers[channel] = info.buffer->getSampleData (channel, info.startSample);
srcBuffers[channel] = buffer.getSampleData (channel, 0);
@ -149,7 +150,7 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
const float alpha = (float) subSampleOffset;
const float invAlpha = 1.0f - alpha;
for (int channel = 0; channel < numChannels; ++channel)
for (int channel = 0; channel < channelsToProcess; ++channel)
*destBuffers[channel]++ = srcBuffers[channel][bufferPos] * invAlpha + srcBuffers[channel][nextPos] * alpha;
subSampleOffset += ratio;
@ -171,13 +172,13 @@ void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& inf
if (ratio < 0.9999)
{
// for up-sampling, apply the filter after transposing..
for (int i = jmin (numChannels, info.buffer->getNumChannels()); --i >= 0;)
for (int i = channelsToProcess; --i >= 0;)
applyFilter (info.buffer->getSampleData (i, info.startSample), info.numSamples, filterStates[i]);
}
else if (ratio <= 1.0001)
{
// if the filter's not currently being applied, keep it stoked with the last couple of samples to avoid discontinuities
for (int i = jmin (numChannels, info.buffer->getNumChannels()); --i >= 0;)
for (int i = channelsToProcess; --i >= 0;)
{
const float* const endOfBuffer = info.buffer->getSampleData (i, info.startSample + info.numSamples - 1);
FilterState& fs = filterStates[i];