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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 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 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 "AudioParam.h"
#include "AudioNode.h"
#include "AudioNodeOutput.h"
#include "AudioUtilities.h"
#include "FloatConversion.h"
#include <wtf/MathExtras.h>
namespace WebCore {
const double AudioParam::DefaultSmoothingConstant = 0.05;
const double AudioParam::SnapThreshold = 0.001;
float AudioParam::value()
{
// Update value for timeline.
if (context() && context()->isAudioThread()) {
bool hasValue;
float timelineValue = m_timeline.valueForContextTime(context(), narrowPrecisionToFloat(m_value), hasValue);
if (hasValue)
m_value = timelineValue;
}
return narrowPrecisionToFloat(m_value);
}
void AudioParam::setValue(float value)
{
// Check against JavaScript giving us bogus floating-point values.
// Don't ASSERT, since this can happen if somebody writes bad JS.
if (!std::isnan(value) && !std::isinf(value))
m_value = value;
}
float AudioParam::smoothedValue()
{
return narrowPrecisionToFloat(m_smoothedValue);
}
bool AudioParam::smooth()
{
// If values have been explicitly scheduled on the timeline, then use the exact value.
// Smoothing effectively is performed by the timeline.
bool useTimelineValue = false;
if (context())
m_value = m_timeline.valueForContextTime(context(), narrowPrecisionToFloat(m_value), useTimelineValue);
if (m_smoothedValue == m_value) {
// Smoothed value has already approached and snapped to value.
return true;
}
if (useTimelineValue)
m_smoothedValue = m_value;
else {
// Dezipper - exponential approach.
m_smoothedValue += (m_value - m_smoothedValue) * m_smoothingConstant;
// If we get close enough then snap to actual value.
if (fabs(m_smoothedValue - m_value) < SnapThreshold) // FIXME: the threshold needs to be adjustable depending on range - but this is OK general purpose value.
m_smoothedValue = m_value;
}
return false;
}
float AudioParam::finalValue()
{
float value;
calculateFinalValues(&value, 1, false);
return value;
}
void AudioParam::calculateSampleAccurateValues(float* values, unsigned numberOfValues)
{
bool isSafe = context() && context()->isAudioThread() && values && numberOfValues;
ASSERT(isSafe);
if (!isSafe)
return;
calculateFinalValues(values, numberOfValues, true);
}
void AudioParam::calculateFinalValues(float* values, unsigned numberOfValues, bool sampleAccurate)
{
bool isGood = context() && context()->isAudioThread() && values && numberOfValues;
ASSERT(isGood);
if (!isGood)
return;
// The calculated result will be the "intrinsic" value summed with all audio-rate connections.
if (sampleAccurate) {
// Calculate sample-accurate (a-rate) intrinsic values.
calculateTimelineValues(values, numberOfValues);
} else {
// Calculate control-rate (k-rate) intrinsic value.
bool hasValue;
float timelineValue = m_timeline.valueForContextTime(context(), narrowPrecisionToFloat(m_value), hasValue);
if (hasValue)
m_value = timelineValue;
values[0] = narrowPrecisionToFloat(m_value);
}
// Now sum all of the audio-rate connections together (unity-gain summing junction).
// Note that connections would normally be mono, but we mix down to mono if necessary.
RefPtr<AudioBus> summingBus = AudioBus::create(1, numberOfValues, false);
summingBus->setChannelMemory(0, values, numberOfValues);
for (unsigned i = 0; i < numberOfRenderingConnections(); ++i) {
AudioNodeOutput* output = renderingOutput(i);
ASSERT(output);
// Render audio from this output.
AudioBus* connectionBus = output->pull(0, AudioNode::ProcessingSizeInFrames);
// Sum, with unity-gain.
summingBus->sumFrom(*connectionBus);
}
}
void AudioParam::calculateTimelineValues(float* values, unsigned numberOfValues)
{
// Calculate values for this render quantum.
// Normally numberOfValues will equal AudioNode::ProcessingSizeInFrames (the render quantum size).
double sampleRate = context()->sampleRate();
double startTime = context()->currentTime();
double endTime = startTime + numberOfValues / sampleRate;
// Note we're running control rate at the sample-rate.
// Pass in the current value as default value.
m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues, sampleRate, sampleRate);
}
void AudioParam::connect(AudioNodeOutput* output)
{
ASSERT(context()->isGraphOwner());
ASSERT(output);
if (!output)
return;
if (m_outputs.contains(output))
return;
output->addParam(this);
m_outputs.add(output);
changedOutputs();
}
void AudioParam::disconnect(AudioNodeOutput* output)
{
ASSERT(context()->isGraphOwner());
ASSERT(output);
if (!output)
return;
if (m_outputs.contains(output)) {
m_outputs.remove(output);
changedOutputs();
output->removeParam(this);
}
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
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