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#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "platform/audio/DownSampler.h"
#include "wtf/MathExtras.h"
namespace WebCore {
DownSampler::DownSampler(size_t inputBlockSize)
: m_inputBlockSize(inputBlockSize)
, m_reducedKernel(DefaultKernelSize / 2)
, m_convolver(inputBlockSize / 2) // runs at 1/2 source sample-rate
, m_tempBuffer(inputBlockSize / 2)
, m_inputBuffer(inputBlockSize * 2)
{
initializeKernel();
}
void DownSampler::initializeKernel()
{
// Blackman window parameters.
double alpha = 0.16;
double a0 = 0.5 * (1.0 - alpha);
double a1 = 0.5;
double a2 = 0.5 * alpha;
int n = DefaultKernelSize;
int halfSize = n / 2;
// Half-band filter.
double sincScaleFactor = 0.5;
// Compute only the odd terms because the even ones are zero, except
// right in the middle at halfSize, which is 0.5 and we'll handle specially during processing
// after doing the main convolution using m_reducedKernel.
for (int i = 1; i < n; i += 2) {
// Compute the sinc() with offset.
double s = sincScaleFactor * piDouble * (i - halfSize);
double sinc = !s ? 1.0 : sin(s) / s;
sinc *= sincScaleFactor;
// Compute Blackman window, matching the offset of the sinc().
double x = static_cast<double>(i) / n;
double window = a0 - a1 * cos(twoPiDouble * x) + a2 * cos(twoPiDouble * 2.0 * x);
// Window the sinc() function.
// Then store only the odd terms in the kernel.
// In a sense, this is shifting forward in time by one sample-frame at the destination sample-rate.
m_reducedKernel[(i - 1) / 2] = sinc * window;
}
}
void DownSampler::process(const float* sourceP, float* destP, size_t sourceFramesToProcess)
{
bool isInputBlockSizeGood = sourceFramesToProcess == m_inputBlockSize;
ASSERT(isInputBlockSizeGood);
if (!isInputBlockSizeGood)
return;
size_t destFramesToProcess = sourceFramesToProcess / 2;
bool isTempBufferGood = destFramesToProcess == m_tempBuffer.size();
ASSERT(isTempBufferGood);
if (!isTempBufferGood)
return;
bool isReducedKernelGood = m_reducedKernel.size() == DefaultKernelSize / 2;
ASSERT(isReducedKernelGood);
if (!isReducedKernelGood)
return;
size_t halfSize = DefaultKernelSize / 2;
// Copy source samples to 2nd half of input buffer.
bool isInputBufferGood = m_inputBuffer.size() == sourceFramesToProcess * 2 && halfSize <= sourceFramesToProcess;
ASSERT(isInputBufferGood);
if (!isInputBufferGood)
return;
float* inputP = m_inputBuffer.data() + sourceFramesToProcess;
memcpy(inputP, sourceP, sizeof(float) * sourceFramesToProcess);
// Copy the odd sample-frames from sourceP, delayed by one sample-frame (destination sample-rate)
// to match shifting forward in time in m_reducedKernel.
float* oddSamplesP = m_tempBuffer.data();
for (unsigned i = 0; i < destFramesToProcess; ++i)
oddSamplesP[i] = *((inputP - 1) + i * 2);
// Actually process oddSamplesP with m_reducedKernel for efficiency.
// The theoretical kernel is double this size with 0 values for even terms (except center).
m_convolver.process(&m_reducedKernel, oddSamplesP, destP, destFramesToProcess);
// Now, account for the 0.5 term right in the middle of the kernel.
// This amounts to a delay-line of length halfSize (at the source sample-rate),
// scaled by 0.5.
// Sum into the destination.
for (unsigned i = 0; i < destFramesToProcess; ++i)
destP[i] += 0.5 * *((inputP - halfSize) + i * 2);
// Copy 2nd half of input buffer to 1st half.
memcpy(m_inputBuffer.data(), inputP, sizeof(float) * sourceFramesToProcess);
}
void DownSampler::reset()
{
m_convolver.reset();
m_inputBuffer.zero();
}
size_t DownSampler::latencyFrames() const
{
// Divide by two since this is a linear phase kernel and the delay is at the center of the kernel.
return m_reducedKernel.size() / 2;
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)