root/RealtimeAnalyser.cpp
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DEFINITIONS
This source file includes following definitions.
- doFFTAnalysis
- getFloatFrequencyData
- getByteFrequencyData
- getByteTimeDomainData
/*
* Copyright (C) 2010, Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
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* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
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#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "modules/webaudio/RealtimeAnalyser.h"
#include "core/platform/audio/AudioBus.h"
#include "core/platform/audio/AudioUtilities.h"
#include "core/platform/audio/FFTFrame.h"
#include "core/platform/audio/VectorMath.h"
#include <algorithm>
#include <limits.h>
#include "wtf/Complex.h"
#include "wtf/Float32Array.h"
#include "wtf/MainThread.h"
#include "wtf/MathExtras.h"
#include "wtf/Uint8Array.h"
using namespace std;
namespace WebCore {
const double RealtimeAnalyser::DefaultSmoothingTimeConstant = 0.8;
const double RealtimeAnalyser::DefaultMinDecibels = -100;
const double RealtimeAnalyser::DefaultMaxDecibels = -30;
const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
// All FFT implementations are expected to handle power-of-two sizes MinFFTSize <= size <= MaxFFTSize.
const unsigned RealtimeAnalyser::MinFFTSize = 32;
const unsigned RealtimeAnalyser::MaxFFTSize = 2048;
const unsigned RealtimeAnalyser::InputBufferSize = RealtimeAnalyser::MaxFFTSize * 2;
RealtimeAnalyser::RealtimeAnalyser()
: m_inputBuffer(InputBufferSize)
, m_writeIndex(0)
, m_fftSize(DefaultFFTSize)
, m_magnitudeBuffer(DefaultFFTSize / 2)
, m_smoothingTimeConstant(DefaultSmoothingTimeConstant)
, m_minDecibels(DefaultMinDecibels)
, m_maxDecibels(DefaultMaxDecibels)
{
m_analysisFrame = adoptPtr(new FFTFrame(DefaultFFTSize));
}
RealtimeAnalyser::~RealtimeAnalyser()
{
}
void RealtimeAnalyser::reset()
{
m_writeIndex = 0;
m_inputBuffer.zero();
m_magnitudeBuffer.zero();
}
bool RealtimeAnalyser::setFftSize(size_t size)
{
ASSERT(isMainThread());
// Only allow powers of two.
unsigned log2size = static_cast<unsigned>(log2(size));
bool isPOT(1UL << log2size == size);
if (!isPOT || size > MaxFFTSize || size < MinFFTSize)
return false;
if (m_fftSize != size) {
m_analysisFrame = adoptPtr(new FFTFrame(size));
// m_magnitudeBuffer has size = fftSize / 2 because it contains floats reduced from complex values in m_analysisFrame.
m_magnitudeBuffer.allocate(size / 2);
m_fftSize = size;
}
return true;
}
void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess)
{
bool isBusGood = bus && bus->numberOfChannels() > 0 && bus->channel(0)->length() >= framesToProcess;
ASSERT(isBusGood);
if (!isBusGood)
return;
// FIXME : allow to work with non-FFTSize divisible chunking
bool isDestinationGood = m_writeIndex < m_inputBuffer.size() && m_writeIndex + framesToProcess <= m_inputBuffer.size();
ASSERT(isDestinationGood);
if (!isDestinationGood)
return;
// Perform real-time analysis
const float* source = bus->channel(0)->data();
float* dest = m_inputBuffer.data() + m_writeIndex;
// The source has already been sanity checked with isBusGood above.
memcpy(dest, source, sizeof(float) * framesToProcess);
// Sum all channels in one if numberOfChannels > 1.
unsigned numberOfChannels = bus->numberOfChannels();
if (numberOfChannels > 1) {
for (unsigned i = 1; i < numberOfChannels; i++) {
source = bus->channel(i)->data();
VectorMath::vadd(dest, 1, source, 1, dest, 1, framesToProcess);
}
const float scale = 1.0 / numberOfChannels;
VectorMath::vsmul(dest, 1, &scale, dest, 1, framesToProcess);
}
m_writeIndex += framesToProcess;
if (m_writeIndex >= InputBufferSize)
m_writeIndex = 0;
}
namespace {
void applyWindow(float* p, size_t n)
{
ASSERT(isMainThread());
// Blackman window
double alpha = 0.16;
double a0 = 0.5 * (1 - alpha);
double a1 = 0.5;
double a2 = 0.5 * alpha;
for (unsigned i = 0; i < n; ++i) {
double x = static_cast<double>(i) / static_cast<double>(n);
double window = a0 - a1 * cos(2 * piDouble * x) + a2 * cos(4 * piDouble * x);
p[i] *= float(window);
}
}
} // namespace
void RealtimeAnalyser::doFFTAnalysis()
{
ASSERT(isMainThread());
// Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
size_t fftSize = this->fftSize();
AudioFloatArray temporaryBuffer(fftSize);
float* inputBuffer = m_inputBuffer.data();
float* tempP = temporaryBuffer.data();
// Take the previous fftSize values from the input buffer and copy into the temporary buffer.
unsigned writeIndex = m_writeIndex;
if (writeIndex < fftSize) {
memcpy(tempP, inputBuffer + writeIndex - fftSize + InputBufferSize, sizeof(*tempP) * (fftSize - writeIndex));
memcpy(tempP + fftSize - writeIndex, inputBuffer, sizeof(*tempP) * writeIndex);
} else
memcpy(tempP, inputBuffer + writeIndex - fftSize, sizeof(*tempP) * fftSize);
// Window the input samples.
applyWindow(tempP, fftSize);
// Do the analysis.
m_analysisFrame->doFFT(tempP);
float* realP = m_analysisFrame->realData();
float* imagP = m_analysisFrame->imagData();
// Blow away the packed nyquist component.
imagP[0] = 0;
// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
const double magnitudeScale = 1.0 / DefaultFFTSize;
// A value of 0 does no averaging with the previous result. Larger values produce slower, but smoother changes.
double k = m_smoothingTimeConstant;
k = max(0.0, k);
k = min(1.0, k);
// Convert the analysis data from complex to magnitude and average with the previous result.
float* destination = magnitudeBuffer().data();
size_t n = magnitudeBuffer().size();
for (size_t i = 0; i < n; ++i) {
Complex c(realP[i], imagP[i]);
double scalarMagnitude = abs(c) * magnitudeScale;
destination[i] = float(k * destination[i] + (1 - k) * scalarMagnitude);
}
}
void RealtimeAnalyser::getFloatFrequencyData(Float32Array* destinationArray)
{
ASSERT(isMainThread());
if (!destinationArray)
return;
doFFTAnalysis();
// Convert from linear magnitude to floating-point decibels.
const double minDecibels = m_minDecibels;
unsigned sourceLength = magnitudeBuffer().size();
size_t len = min(sourceLength, destinationArray->length());
if (len > 0) {
const float* source = magnitudeBuffer().data();
float* destination = destinationArray->data();
for (unsigned i = 0; i < len; ++i) {
float linearValue = source[i];
double dbMag = !linearValue ? minDecibels : AudioUtilities::linearToDecibels(linearValue);
destination[i] = float(dbMag);
}
}
}
void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
{
ASSERT(isMainThread());
if (!destinationArray)
return;
doFFTAnalysis();
// Convert from linear magnitude to unsigned-byte decibels.
unsigned sourceLength = magnitudeBuffer().size();
size_t len = min(sourceLength, destinationArray->length());
if (len > 0) {
const double rangeScaleFactor = m_maxDecibels == m_minDecibels ? 1 : 1 / (m_maxDecibels - m_minDecibels);
const double minDecibels = m_minDecibels;
const float* source = magnitudeBuffer().data();
unsigned char* destination = destinationArray->data();
for (unsigned i = 0; i < len; ++i) {
float linearValue = source[i];
double dbMag = !linearValue ? minDecibels : AudioUtilities::linearToDecibels(linearValue);
// The range m_minDecibels to m_maxDecibels will be scaled to byte values from 0 to UCHAR_MAX.
double scaledValue = UCHAR_MAX * (dbMag - minDecibels) * rangeScaleFactor;
// Clip to valid range.
if (scaledValue < 0)
scaledValue = 0;
if (scaledValue > UCHAR_MAX)
scaledValue = UCHAR_MAX;
destination[i] = static_cast<unsigned char>(scaledValue);
}
}
}
void RealtimeAnalyser::getByteTimeDomainData(Uint8Array* destinationArray)
{
ASSERT(isMainThread());
if (!destinationArray)
return;
unsigned fftSize = this->fftSize();
size_t len = min(fftSize, destinationArray->length());
if (len > 0) {
bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
ASSERT(isInputBufferGood);
if (!isInputBufferGood)
return;
float* inputBuffer = m_inputBuffer.data();
unsigned char* destination = destinationArray->data();
unsigned writeIndex = m_writeIndex;
for (unsigned i = 0; i < len; ++i) {
// Buffer access is protected due to modulo operation.
float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
// Scale from nominal -1 -> +1 to unsigned byte.
double scaledValue = 128 * (value + 1);
// Clip to valid range.
if (scaledValue < 0)
scaledValue = 0;
if (scaledValue > UCHAR_MAX)
scaledValue = UCHAR_MAX;
destination[i] = static_cast<unsigned char>(scaledValue);
}
}
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)