gecko-dev/content/media/webaudio/GainProcessor.h
Ehsan Akhgari f781f8cc60 Bug 871201 - Part 1: Refactor the gain processing logic of GainNodeEngine into a reusable base class; r=roc
X-Git-Commit-ID: e3c9ccaf984c74bb041e2a35fb092dd91e643555
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--HG--
extra : rebase_source : c26ad3b5953f15b599baa9fea0cf66580f2a3262
2013-05-13 00:17:36 -04:00

86 lines
2.7 KiB
C++

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef GainProcessor_h_
#define GainProcessor_h_
#include "AudioNodeStream.h"
#include "AudioDestinationNode.h"
#include "WebAudioUtils.h"
namespace mozilla {
namespace dom {
// This class implements the gain processing logic used by GainNodeEngine
// and AudioBufferSourceNodeEngine.
class GainProcessor
{
public:
explicit GainProcessor(AudioDestinationNode* aDestination)
: mSource(nullptr)
, mDestination(static_cast<AudioNodeStream*> (aDestination->Stream()))
, mGain(1.f)
{
}
void SetSourceStream(AudioNodeStream* aSource)
{
mSource = aSource;
}
void SetGainParameter(const AudioParamTimeline& aValue)
{
MOZ_ASSERT(mSource && mDestination);
mGain = aValue;
WebAudioUtils::ConvertAudioParamToTicks(mGain, mSource, mDestination);
}
void ProcessGain(AudioNodeStream* aStream,
float aInputVolume,
const nsTArray<const void*>& aInputChannelData,
AudioChunk* aOutput)
{
MOZ_ASSERT(mSource == aStream, "Invalid source stream");
if (mGain.HasSimpleValue()) {
// Optimize the case where we only have a single value set as the volume
aOutput->mVolume *= mGain.GetValue();
} else {
// First, compute a vector of gains for each track tick based on the
// timeline at hand, and then for each channel, multiply the values
// in the buffer with the gain vector.
// Compute the gain values for the duration of the input AudioChunk
// XXX we need to add a method to AudioEventTimeline to compute this buffer directly.
float computedGain[WEBAUDIO_BLOCK_SIZE];
for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
TrackTicks tick = aStream->GetCurrentPosition();
computedGain[counter] = mGain.GetValueAtTime(tick, counter) * aInputVolume;
}
// Apply the gain to the output buffer
MOZ_ASSERT(aInputChannelData.Length() == aOutput->mChannelData.Length());
for (size_t channel = 0; channel < aOutput->mChannelData.Length(); ++channel) {
const float* inputBuffer = static_cast<const float*> (aInputChannelData[channel]);
float* buffer = static_cast<float*> (const_cast<void*>
(aOutput->mChannelData[channel]));
AudioBlockCopyChannelWithScale(inputBuffer, computedGain, buffer);
}
}
}
protected:
AudioNodeStream* mSource;
AudioNodeStream* mDestination;
AudioParamTimeline mGain;
};
}
}
#endif