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/*
* 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
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* 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
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "platform/audio/AudioDelayDSPKernel.h"
#include "platform/audio/AudioUtilities.h"
#include <algorithm>
using namespace std;
namespace WebCore {
const float SmoothingTimeConstant = 0.020f; // 20ms
AudioDelayDSPKernel::AudioDelayDSPKernel(AudioDSPKernelProcessor* processor, size_t processingSizeInFrames)
: AudioDSPKernel(processor)
, m_writeIndex(0)
, m_firstTime(true)
, m_delayTimes(processingSizeInFrames)
{
}
AudioDelayDSPKernel::AudioDelayDSPKernel(double maxDelayTime, float sampleRate)
: AudioDSPKernel(sampleRate)
, m_maxDelayTime(maxDelayTime)
, m_writeIndex(0)
, m_firstTime(true)
{
ASSERT(maxDelayTime > 0.0);
if (maxDelayTime <= 0.0)
return;
size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate);
ASSERT(bufferLength);
if (!bufferLength)
return;
m_buffer.allocate(bufferLength);
m_buffer.zero();
m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
}
size_t AudioDelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const
{
// Compute the length of the buffer needed to handle a max delay of |maxDelayTime|. One is
// added to handle the case where the actual delay equals the maximum delay.
return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate);
}
bool AudioDelayDSPKernel::hasSampleAccurateValues()
{
return false;
}
void AudioDelayDSPKernel::calculateSampleAccurateValues(float*, size_t)
{
ASSERT_NOT_REACHED();
}
double AudioDelayDSPKernel::delayTime(float sampleRate)
{
return m_desiredDelayFrames / sampleRate;
}
void AudioDelayDSPKernel::process(const float* source, float* destination, size_t framesToProcess)
{
size_t bufferLength = m_buffer.size();
float* buffer = m_buffer.data();
ASSERT(bufferLength);
if (!bufferLength)
return;
ASSERT(source && destination);
if (!source || !destination)
return;
float sampleRate = this->sampleRate();
double delayTime = 0;
float* delayTimes = m_delayTimes.data();
double maxTime = maxDelayTime();
bool sampleAccurate = hasSampleAccurateValues();
if (sampleAccurate) {
calculateSampleAccurateValues(delayTimes, framesToProcess);
} else {
delayTime = this->delayTime(sampleRate);
// Make sure the delay time is in a valid range.
delayTime = min(maxTime, delayTime);
delayTime = max(0.0, delayTime);
if (m_firstTime) {
m_currentDelayTime = delayTime;
m_firstTime = false;
}
}
for (unsigned i = 0; i < framesToProcess; ++i) {
if (sampleAccurate) {
delayTime = delayTimes[i];
delayTime = std::min(maxTime, delayTime);
delayTime = std::max(0.0, delayTime);
m_currentDelayTime = delayTime;
} else {
// Approach desired delay time.
m_currentDelayTime += (delayTime - m_currentDelayTime) * m_smoothingRate;
}
double desiredDelayFrames = m_currentDelayTime * sampleRate;
double readPosition = m_writeIndex + bufferLength - desiredDelayFrames;
if (readPosition >= bufferLength)
readPosition -= bufferLength;
// Linearly interpolate in-between delay times.
int readIndex1 = static_cast<int>(readPosition);
int readIndex2 = (readIndex1 + 1) % bufferLength;
double interpolationFactor = readPosition - readIndex1;
double input = static_cast<float>(*source++);
buffer[m_writeIndex] = static_cast<float>(input);
m_writeIndex = (m_writeIndex + 1) % bufferLength;
double sample1 = buffer[readIndex1];
double sample2 = buffer[readIndex2];
double output = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
*destination++ = static_cast<float>(output);
}
}
void AudioDelayDSPKernel::reset()
{
m_firstTime = true;
m_buffer.zero();
}
double AudioDelayDSPKernel::tailTime() const
{
// Account for worst case delay.
// Don't try to track actual delay time which can change dynamically.
return m_maxDelayTime;
}
double AudioDelayDSPKernel::latencyTime() const
{
return 0;
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)