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Bug 865246 - Part 2: Add the Biquad processing implementation to the build system; r=padenot
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@ -26,20 +26,11 @@
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* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "config.h"
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#if ENABLE(WEB_AUDIO)
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#include "DenormalDisabler.h"
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#include "Biquad.h"
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#include "DenormalDisabler.h"
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#include <algorithm>
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#include <stdio.h>
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#include <wtf/MathExtras.h>
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#if OS(DARWIN)
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#include <Accelerate/Accelerate.h>
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#endif
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namespace WebCore {
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@ -47,18 +38,6 @@ const int kBufferSize = 1024;
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Biquad::Biquad()
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{
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#if OS(DARWIN)
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// Allocate two samples more for filter history
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m_inputBuffer.allocate(kBufferSize + 2);
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m_outputBuffer.allocate(kBufferSize + 2);
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#endif
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#if USE(WEBAUDIO_IPP)
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int bufferSize;
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ippsIIRGetStateSize64f_BiQuad_32f(1, &bufferSize);
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m_ippInternalBuffer = ippsMalloc_8u(bufferSize);
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#endif // USE(WEBAUDIO_IPP)
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// Initialize as pass-thru (straight-wire, no filter effect)
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setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
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@ -67,21 +46,10 @@ Biquad::Biquad()
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Biquad::~Biquad()
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{
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#if USE(WEBAUDIO_IPP)
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ippsFree(m_ippInternalBuffer);
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#endif // USE(WEBAUDIO_IPP)
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}
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void Biquad::process(const float* sourceP, float* destP, size_t framesToProcess)
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{
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#if OS(DARWIN)
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// Use vecLib if available
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processFast(sourceP, destP, framesToProcess);
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#elif USE(WEBAUDIO_IPP)
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ippsIIR64f_32f(sourceP, destP, static_cast<int>(framesToProcess), m_biquadState);
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#else // USE(WEBAUDIO_IPP)
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int n = framesToProcess;
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// Create local copies of member variables
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@ -122,93 +90,18 @@ void Biquad::process(const float* sourceP, float* destP, size_t framesToProcess)
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m_b2 = b2;
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m_a1 = a1;
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m_a2 = a2;
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#endif
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}
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#if OS(DARWIN)
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// Here we have optimized version using Accelerate.framework
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void Biquad::processFast(const float* sourceP, float* destP, size_t framesToProcess)
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{
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double filterCoefficients[5];
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filterCoefficients[0] = m_b0;
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filterCoefficients[1] = m_b1;
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filterCoefficients[2] = m_b2;
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filterCoefficients[3] = m_a1;
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filterCoefficients[4] = m_a2;
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double* inputP = m_inputBuffer.data();
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double* outputP = m_outputBuffer.data();
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double* input2P = inputP + 2;
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double* output2P = outputP + 2;
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// Break up processing into smaller slices (kBufferSize) if necessary.
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int n = framesToProcess;
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while (n > 0) {
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int framesThisTime = n < kBufferSize ? n : kBufferSize;
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// Copy input to input buffer
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for (int i = 0; i < framesThisTime; ++i)
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input2P[i] = *sourceP++;
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processSliceFast(inputP, outputP, filterCoefficients, framesThisTime);
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// Copy output buffer to output (converts float -> double).
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for (int i = 0; i < framesThisTime; ++i)
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*destP++ = static_cast<float>(output2P[i]);
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n -= framesThisTime;
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}
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}
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void Biquad::processSliceFast(double* sourceP, double* destP, double* coefficientsP, size_t framesToProcess)
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{
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// Use double-precision for filter stability
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vDSP_deq22D(sourceP, 1, coefficientsP, destP, 1, framesToProcess);
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// Save history. Note that sourceP and destP reference m_inputBuffer and m_outputBuffer respectively.
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// These buffers are allocated (in the constructor) with space for two extra samples so it's OK to access
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// array values two beyond framesToProcess.
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sourceP[0] = sourceP[framesToProcess - 2 + 2];
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sourceP[1] = sourceP[framesToProcess - 1 + 2];
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destP[0] = destP[framesToProcess - 2 + 2];
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destP[1] = destP[framesToProcess - 1 + 2];
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}
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#endif // OS(DARWIN)
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void Biquad::reset()
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{
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#if OS(DARWIN)
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// Two extra samples for filter history
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double* inputP = m_inputBuffer.data();
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inputP[0] = 0;
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inputP[1] = 0;
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double* outputP = m_outputBuffer.data();
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outputP[0] = 0;
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outputP[1] = 0;
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#elif USE(WEBAUDIO_IPP)
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int bufferSize;
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ippsIIRGetStateSize64f_BiQuad_32f(1, &bufferSize);
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ippsZero_8u(m_ippInternalBuffer, bufferSize);
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#else
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m_x1 = m_x2 = m_y1 = m_y2 = 0;
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#endif
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}
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void Biquad::setLowpassParams(double cutoff, double resonance)
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{
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// Limit cutoff to 0 to 1.
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cutoff = std::max(0.0, std::min(cutoff, 1.0));
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if (cutoff == 1) {
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// When cutoff is 1, the z-transform is 1.
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setNormalizedCoefficients(1, 0, 0,
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@ -219,7 +112,7 @@ void Biquad::setLowpassParams(double cutoff, double resonance)
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double g = pow(10.0, 0.05 * resonance);
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double d = sqrt((4 - sqrt(16 - 16 / (g * g))) / 2);
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double theta = piDouble * cutoff;
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double theta = M_PI * cutoff;
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double sn = 0.5 * d * sin(theta);
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double beta = 0.5 * (1 - sn) / (1 + sn);
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double gamma = (0.5 + beta) * cos(theta);
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@ -255,7 +148,7 @@ void Biquad::setHighpassParams(double cutoff, double resonance)
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double g = pow(10.0, 0.05 * resonance);
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double d = sqrt((4 - sqrt(16 - 16 / (g * g))) / 2);
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double theta = piDouble * cutoff;
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double theta = M_PI * cutoff;
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double sn = 0.5 * d * sin(theta);
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double beta = 0.5 * (1 - sn) / (1 + sn);
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double gamma = (0.5 + beta) * cos(theta);
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@ -287,26 +180,13 @@ void Biquad::setNormalizedCoefficients(double b0, double b1, double b2, double a
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m_b2 = b2 * a0Inverse;
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m_a1 = a1 * a0Inverse;
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m_a2 = a2 * a0Inverse;
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#if USE(WEBAUDIO_IPP)
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Ipp64f taps[6];
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taps[0] = m_b0;
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taps[1] = m_b1;
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taps[2] = m_b2;
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taps[3] = 1;
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taps[4] = m_a1;
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taps[5] = m_a2;
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m_biquadState = 0;
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ippsIIRInit64f_BiQuad_32f(&m_biquadState, taps, 1, 0, m_ippInternalBuffer);
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#endif // USE(WEBAUDIO_IPP)
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}
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void Biquad::setLowShelfParams(double frequency, double dbGain)
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{
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// Clip frequencies to between 0 and 1, inclusive.
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frequency = std::max(0.0, std::min(frequency, 1.0));
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double A = pow(10.0, dbGain / 40);
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if (frequency == 1) {
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@ -314,7 +194,7 @@ void Biquad::setLowShelfParams(double frequency, double dbGain)
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setNormalizedCoefficients(A * A, 0, 0,
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1, 0, 0);
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} else if (frequency > 0) {
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double w0 = piDouble * frequency;
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double w0 = M_PI * frequency;
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double S = 1; // filter slope (1 is max value)
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double alpha = 0.5 * sin(w0) * sqrt((A + 1 / A) * (1 / S - 1) + 2);
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double k = cos(w0);
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@ -349,7 +229,7 @@ void Biquad::setHighShelfParams(double frequency, double dbGain)
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setNormalizedCoefficients(1, 0, 0,
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1, 0, 0);
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} else if (frequency > 0) {
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double w0 = piDouble * frequency;
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double w0 = M_PI * frequency;
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double S = 1; // filter slope (1 is max value)
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double alpha = 0.5 * sin(w0) * sqrt((A + 1 / A) * (1 / S - 1) + 2);
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double k = cos(w0);
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@ -384,7 +264,7 @@ void Biquad::setPeakingParams(double frequency, double Q, double dbGain)
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if (frequency > 0 && frequency < 1) {
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if (Q > 0) {
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double w0 = piDouble * frequency;
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double w0 = M_PI * frequency;
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double alpha = sin(w0) / (2 * Q);
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double k = cos(w0);
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@ -420,7 +300,7 @@ void Biquad::setAllpassParams(double frequency, double Q)
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if (frequency > 0 && frequency < 1) {
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if (Q > 0) {
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double w0 = piDouble * frequency;
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double w0 = M_PI * frequency;
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double alpha = sin(w0) / (2 * Q);
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double k = cos(w0);
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@ -456,7 +336,7 @@ void Biquad::setNotchParams(double frequency, double Q)
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if (frequency > 0 && frequency < 1) {
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if (Q > 0) {
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double w0 = piDouble * frequency;
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double w0 = M_PI * frequency;
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double alpha = sin(w0) / (2 * Q);
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double k = cos(w0);
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@ -491,7 +371,7 @@ void Biquad::setBandpassParams(double frequency, double Q)
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Q = std::max(0.0, Q);
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if (frequency > 0 && frequency < 1) {
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double w0 = piDouble * frequency;
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double w0 = M_PI * frequency;
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if (Q > 0) {
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double alpha = sin(w0) / (2 * Q);
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double k = cos(w0);
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@ -572,7 +452,7 @@ void Biquad::getFrequencyResponse(int nFrequencies,
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double a2 = m_a2;
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for (int k = 0; k < nFrequencies; ++k) {
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double omega = -piDouble * frequency[k];
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double omega = -M_PI * frequency[k];
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Complex z = Complex(cos(omega), sin(omega));
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Complex numerator = b0 + (b1 + b2 * z) * z;
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Complex denominator = Complex(1, 0) + (a1 + a2 * z) * z;
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@ -584,4 +464,3 @@ void Biquad::getFrequencyResponse(int nFrequencies,
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} // namespace WebCore
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#endif // ENABLE(WEB_AUDIO)
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@ -29,25 +29,21 @@
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#ifndef Biquad_h
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#define Biquad_h
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#include "AudioArray.h"
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#include <sys/types.h>
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#include <wtf/Complex.h>
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#if USE(WEBAUDIO_IPP)
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#include <ipps.h>
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#endif // USE(WEBAUDIO_IPP)
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#include <complex>
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namespace WebCore {
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typedef std::complex<double> Complex;
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// A basic biquad (two-zero / two-pole digital filter)
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//
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// It can be configured to a number of common and very useful filters:
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// lowpass, highpass, shelving, parameteric, notch, allpass, ...
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class Biquad {
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public:
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public:
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Biquad();
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virtual ~Biquad();
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~Biquad();
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void process(const float* sourceP, float* destP, size_t framesToProcess);
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@ -82,7 +78,7 @@ public:
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float* phaseResponse);
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private:
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void setNormalizedCoefficients(double b0, double b1, double b2, double a0, double a1, double a2);
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// Filter coefficients. The filter is defined as
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//
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// y[n] + m_a1*y[n-1] + m_a2*y[n-2] = m_b0*x[n] + m_b1*x[n-1] + m_b2*x[n-2].
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@ -92,24 +88,11 @@ private:
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double m_a1;
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double m_a2;
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#if OS(DARWIN)
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void processFast(const float* sourceP, float* destP, size_t framesToProcess);
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void processSliceFast(double* sourceP, double* destP, double* coefficientsP, size_t framesToProcess);
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AudioDoubleArray m_inputBuffer;
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AudioDoubleArray m_outputBuffer;
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#elif USE(WEBAUDIO_IPP)
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IppsIIRState64f_32f* m_biquadState;
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Ipp8u* m_ippInternalBuffer;
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#else
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// Filter memory
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double m_x1; // input delayed by 1 sample
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double m_x2; // input delayed by 2 samples
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double m_y1; // output delayed by 1 sample
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double m_y2; // output delayed by 2 samples
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#endif
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};
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} // namespace WebCore
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@ -13,6 +13,7 @@ LIBRARY_NAME := gkconwebaudio_blink_s
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LIBXUL_LIBRARY := 1
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CPPSRCS := \
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Biquad.cpp \
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DynamicsCompressor.cpp \
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DynamicsCompressorKernel.cpp \
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ZeroPole.cpp \
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