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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
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* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
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* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
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*/
#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "platform/audio/AudioBus.h"
#include "platform/audio/DenormalDisabler.h"
#include <assert.h>
#include <math.h>
#include <algorithm>
#include "platform/audio/SincResampler.h"
#include "platform/audio/VectorMath.h"
#include "wtf/OwnPtr.h"
#include "wtf/PassOwnPtr.h"
namespace WebCore {
using namespace VectorMath;
const unsigned MaxBusChannels = 32;
PassRefPtr<AudioBus> AudioBus::create(unsigned numberOfChannels, size_t length, bool allocate)
{
ASSERT(numberOfChannels <= MaxBusChannels);
if (numberOfChannels > MaxBusChannels)
return 0;
return adoptRef(new AudioBus(numberOfChannels, length, allocate));
}
AudioBus::AudioBus(unsigned numberOfChannels, size_t length, bool allocate)
: m_length(length)
, m_busGain(1)
, m_isFirstTime(true)
, m_sampleRate(0)
{
m_channels.reserveInitialCapacity(numberOfChannels);
for (unsigned i = 0; i < numberOfChannels; ++i) {
PassOwnPtr<AudioChannel> channel = allocate ? adoptPtr(new AudioChannel(length)) : adoptPtr(new AudioChannel(0, length));
m_channels.append(channel);
}
m_layout = LayoutCanonical; // for now this is the only layout we define
}
void AudioBus::setChannelMemory(unsigned channelIndex, float* storage, size_t length)
{
if (channelIndex < m_channels.size()) {
channel(channelIndex)->set(storage, length);
m_length = length; // FIXME: verify that this length matches all the other channel lengths
}
}
void AudioBus::resizeSmaller(size_t newLength)
{
ASSERT(newLength <= m_length);
if (newLength <= m_length)
m_length = newLength;
for (unsigned i = 0; i < m_channels.size(); ++i)
m_channels[i]->resizeSmaller(newLength);
}
void AudioBus::zero()
{
for (unsigned i = 0; i < m_channels.size(); ++i)
m_channels[i]->zero();
}
AudioChannel* AudioBus::channelByType(unsigned channelType)
{
// For now we only support canonical channel layouts...
if (m_layout != LayoutCanonical)
return 0;
switch (numberOfChannels()) {
case 1: // mono
if (channelType == ChannelMono || channelType == ChannelLeft)
return channel(0);
return 0;
case 2: // stereo
switch (channelType) {
case ChannelLeft: return channel(0);
case ChannelRight: return channel(1);
default: return 0;
}
case 4: // quad
switch (channelType) {
case ChannelLeft: return channel(0);
case ChannelRight: return channel(1);
case ChannelSurroundLeft: return channel(2);
case ChannelSurroundRight: return channel(3);
default: return 0;
}
case 5: // 5.0
switch (channelType) {
case ChannelLeft: return channel(0);
case ChannelRight: return channel(1);
case ChannelCenter: return channel(2);
case ChannelSurroundLeft: return channel(3);
case ChannelSurroundRight: return channel(4);
default: return 0;
}
case 6: // 5.1
switch (channelType) {
case ChannelLeft: return channel(0);
case ChannelRight: return channel(1);
case ChannelCenter: return channel(2);
case ChannelLFE: return channel(3);
case ChannelSurroundLeft: return channel(4);
case ChannelSurroundRight: return channel(5);
default: return 0;
}
}
ASSERT_NOT_REACHED();
return 0;
}
const AudioChannel* AudioBus::channelByType(unsigned type) const
{
return const_cast<AudioBus*>(this)->channelByType(type);
}
// Returns true if the channel count and frame-size match.
bool AudioBus::topologyMatches(const AudioBus& bus) const
{
if (numberOfChannels() != bus.numberOfChannels())
return false; // channel mismatch
// Make sure source bus has enough frames.
if (length() > bus.length())
return false; // frame-size mismatch
return true;
}
PassRefPtr<AudioBus> AudioBus::createBufferFromRange(const AudioBus* sourceBuffer, unsigned startFrame, unsigned endFrame)
{
size_t numberOfSourceFrames = sourceBuffer->length();
unsigned numberOfChannels = sourceBuffer->numberOfChannels();
// Sanity checking
bool isRangeSafe = startFrame < endFrame && endFrame <= numberOfSourceFrames;
ASSERT(isRangeSafe);
if (!isRangeSafe)
return 0;
size_t rangeLength = endFrame - startFrame;
RefPtr<AudioBus> audioBus = create(numberOfChannels, rangeLength);
audioBus->setSampleRate(sourceBuffer->sampleRate());
for (unsigned i = 0; i < numberOfChannels; ++i)
audioBus->channel(i)->copyFromRange(sourceBuffer->channel(i), startFrame, endFrame);
return audioBus;
}
float AudioBus::maxAbsValue() const
{
float max = 0.0f;
for (unsigned i = 0; i < numberOfChannels(); ++i) {
const AudioChannel* channel = this->channel(i);
max = std::max(max, channel->maxAbsValue());
}
return max;
}
void AudioBus::normalize()
{
float max = maxAbsValue();
if (max)
scale(1.0f / max);
}
void AudioBus::scale(float scale)
{
for (unsigned i = 0; i < numberOfChannels(); ++i)
channel(i)->scale(scale);
}
void AudioBus::copyFrom(const AudioBus& sourceBus, ChannelInterpretation channelInterpretation)
{
if (&sourceBus == this)
return;
unsigned numberOfSourceChannels = sourceBus.numberOfChannels();
unsigned numberOfDestinationChannels = numberOfChannels();
if (numberOfDestinationChannels == numberOfSourceChannels) {
for (unsigned i = 0; i < numberOfSourceChannels; ++i)
channel(i)->copyFrom(sourceBus.channel(i));
} else {
switch (channelInterpretation) {
case Speakers:
speakersCopyFrom(sourceBus);
break;
case Discrete:
discreteCopyFrom(sourceBus);
break;
default:
ASSERT_NOT_REACHED();
}
}
}
void AudioBus::sumFrom(const AudioBus& sourceBus, ChannelInterpretation channelInterpretation)
{
if (&sourceBus == this)
return;
unsigned numberOfSourceChannels = sourceBus.numberOfChannels();
unsigned numberOfDestinationChannels = numberOfChannels();
if (numberOfDestinationChannels == numberOfSourceChannels) {
for (unsigned i = 0; i < numberOfSourceChannels; ++i)
channel(i)->sumFrom(sourceBus.channel(i));
} else {
switch (channelInterpretation) {
case Speakers:
speakersSumFrom(sourceBus);
break;
case Discrete:
discreteSumFrom(sourceBus);
break;
default:
ASSERT_NOT_REACHED();
}
}
}
void AudioBus::speakersCopyFrom(const AudioBus& sourceBus)
{
// FIXME: Implement down mixing 5.1 to stereo.
// https://bugs.webkit.org/show_bug.cgi?id=79192
unsigned numberOfSourceChannels = sourceBus.numberOfChannels();
unsigned numberOfDestinationChannels = numberOfChannels();
if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 1) {
// Handle mono -> stereo case (for now simply copy mono channel into both left and right)
// FIXME: Really we should apply an equal-power scaling factor here, since we're effectively panning center...
const AudioChannel* sourceChannel = sourceBus.channel(0);
channel(0)->copyFrom(sourceChannel);
channel(1)->copyFrom(sourceChannel);
} else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 2) {
// Handle stereo -> mono case. output = 0.5 * (input.L + input.R).
AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus);
const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data();
const float* sourceR = sourceBusSafe.channelByType(ChannelRight)->data();
float* destination = channelByType(ChannelLeft)->mutableData();
vadd(sourceL, 1, sourceR, 1, destination, 1, length());
float scale = 0.5;
vsmul(destination, 1, &scale, destination, 1, length());
} else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 1) {
// Handle mono -> 5.1 case, copy mono channel to center.
channel(2)->copyFrom(sourceBus.channel(0));
channel(0)->zero();
channel(1)->zero();
channel(3)->zero();
channel(4)->zero();
channel(5)->zero();
} else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 6) {
// Handle 5.1 -> mono case.
zero();
speakersSumFrom5_1_ToMono(sourceBus);
} else {
// Fallback for unknown combinations.
discreteCopyFrom(sourceBus);
}
}
void AudioBus::speakersSumFrom(const AudioBus& sourceBus)
{
// FIXME: Implement down mixing 5.1 to stereo.
// https://bugs.webkit.org/show_bug.cgi?id=79192
unsigned numberOfSourceChannels = sourceBus.numberOfChannels();
unsigned numberOfDestinationChannels = numberOfChannels();
if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 1) {
// Handle mono -> stereo case (summing mono channel into both left and right).
const AudioChannel* sourceChannel = sourceBus.channel(0);
channel(0)->sumFrom(sourceChannel);
channel(1)->sumFrom(sourceChannel);
} else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 2) {
// Handle stereo -> mono case. output += 0.5 * (input.L + input.R).
AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus);
const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data();
const float* sourceR = sourceBusSafe.channelByType(ChannelRight)->data();
float* destination = channelByType(ChannelLeft)->mutableData();
float scale = 0.5;
vsma(sourceL, 1, &scale, destination, 1, length());
vsma(sourceR, 1, &scale, destination, 1, length());
} else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 1) {
// Handle mono -> 5.1 case, sum mono channel into center.
channel(2)->sumFrom(sourceBus.channel(0));
} else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 6) {
// Handle 5.1 -> mono case.
speakersSumFrom5_1_ToMono(sourceBus);
} else {
// Fallback for unknown combinations.
discreteSumFrom(sourceBus);
}
}
void AudioBus::speakersSumFrom5_1_ToMono(const AudioBus& sourceBus)
{
AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus);
const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data();
const float* sourceR = sourceBusSafe.channelByType(ChannelRight)->data();
const float* sourceC = sourceBusSafe.channelByType(ChannelCenter)->data();
const float* sourceSL = sourceBusSafe.channelByType(ChannelSurroundLeft)->data();
const float* sourceSR = sourceBusSafe.channelByType(ChannelSurroundRight)->data();
float* destination = channelByType(ChannelLeft)->mutableData();
AudioFloatArray temp(length());
float* tempData = temp.data();
// Sum in L and R.
vadd(sourceL, 1, sourceR, 1, tempData, 1, length());
float scale = 0.7071;
vsmul(tempData, 1, &scale, tempData, 1, length());
vadd(tempData, 1, destination, 1, destination, 1, length());
// Sum in SL and SR.
vadd(sourceSL, 1, sourceSR, 1, tempData, 1, length());
scale = 0.5;
vsmul(tempData, 1, &scale, tempData, 1, length());
vadd(tempData, 1, destination, 1, destination, 1, length());
// Sum in center.
vadd(sourceC, 1, destination, 1, destination, 1, length());
}
void AudioBus::discreteCopyFrom(const AudioBus& sourceBus)
{
unsigned numberOfSourceChannels = sourceBus.numberOfChannels();
unsigned numberOfDestinationChannels = numberOfChannels();
if (numberOfDestinationChannels < numberOfSourceChannels) {
// Down-mix by copying channels and dropping the remaining.
for (unsigned i = 0; i < numberOfDestinationChannels; ++i)
channel(i)->copyFrom(sourceBus.channel(i));
} else if (numberOfDestinationChannels > numberOfSourceChannels) {
// Up-mix by copying as many channels as we have, then zeroing remaining channels.
for (unsigned i = 0; i < numberOfSourceChannels; ++i)
channel(i)->copyFrom(sourceBus.channel(i));
for (unsigned i = numberOfSourceChannels; i < numberOfDestinationChannels; ++i)
channel(i)->zero();
}
}
void AudioBus::discreteSumFrom(const AudioBus& sourceBus)
{
unsigned numberOfSourceChannels = sourceBus.numberOfChannels();
unsigned numberOfDestinationChannels = numberOfChannels();
if (numberOfDestinationChannels < numberOfSourceChannels) {
// Down-mix by summing channels and dropping the remaining.
for (unsigned i = 0; i < numberOfDestinationChannels; ++i)
channel(i)->sumFrom(sourceBus.channel(i));
} else if (numberOfDestinationChannels > numberOfSourceChannels) {
// Up-mix by summing as many channels as we have.
for (unsigned i = 0; i < numberOfSourceChannels; ++i)
channel(i)->sumFrom(sourceBus.channel(i));
}
}
void AudioBus::copyWithGainFrom(const AudioBus &sourceBus, float* lastMixGain, float targetGain)
{
if (!topologyMatches(sourceBus)) {
ASSERT_NOT_REACHED();
zero();
return;
}
if (sourceBus.isSilent()) {
zero();
return;
}
unsigned numberOfChannels = this->numberOfChannels();
ASSERT(numberOfChannels <= MaxBusChannels);
if (numberOfChannels > MaxBusChannels)
return;
// If it is copying from the same bus and no need to change gain, just return.
if (this == &sourceBus && *lastMixGain == targetGain && targetGain == 1)
return;
AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus);
const float* sources[MaxBusChannels];
float* destinations[MaxBusChannels];
for (unsigned i = 0; i < numberOfChannels; ++i) {
sources[i] = sourceBusSafe.channel(i)->data();
destinations[i] = channel(i)->mutableData();
}
// We don't want to suddenly change the gain from mixing one time slice to the next,
// so we "de-zipper" by slowly changing the gain each sample-frame until we've achieved the target gain.
// Take master bus gain into account as well as the targetGain.
float totalDesiredGain = static_cast<float>(m_busGain * targetGain);
// First time, snap directly to totalDesiredGain.
float gain = static_cast<float>(m_isFirstTime ? totalDesiredGain : *lastMixGain);
m_isFirstTime = false;
const float DezipperRate = 0.005f;
unsigned framesToProcess = length();
// If the gain is within epsilon of totalDesiredGain, we can skip dezippering.
// FIXME: this value may need tweaking.
const float epsilon = 0.001f;
float gainDiff = fabs(totalDesiredGain - gain);
// Number of frames to de-zipper before we are close enough to the target gain.
// FIXME: framesToDezipper could be smaller when target gain is close enough within this process loop.
unsigned framesToDezipper = (gainDiff < epsilon) ? 0 : framesToProcess;
if (framesToDezipper) {
if (!m_dezipperGainValues.get() || m_dezipperGainValues->size() < framesToDezipper)
m_dezipperGainValues = adoptPtr(new AudioFloatArray(framesToDezipper));
float* gainValues = m_dezipperGainValues->data();
for (unsigned i = 0; i < framesToDezipper; ++i) {
gain += (totalDesiredGain - gain) * DezipperRate;
// FIXME: If we are clever enough in calculating the framesToDezipper value, we can probably get
// rid of this DenormalDisabler::flushDenormalFloatToZero() call.
gain = DenormalDisabler::flushDenormalFloatToZero(gain);
*gainValues++ = gain;
}
for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex) {
vmul(sources[channelIndex], 1, m_dezipperGainValues->data(), 1, destinations[channelIndex], 1, framesToDezipper);
sources[channelIndex] += framesToDezipper;
destinations[channelIndex] += framesToDezipper;
}
} else
gain = totalDesiredGain;
// Apply constant gain after de-zippering has converged on target gain.
if (framesToDezipper < framesToProcess) {
for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex)
vsmul(sources[channelIndex], 1, &gain, destinations[channelIndex], 1, framesToProcess - framesToDezipper);
}
// Save the target gain as the starting point for next time around.
*lastMixGain = gain;
}
void AudioBus::copyWithSampleAccurateGainValuesFrom(const AudioBus &sourceBus, float* gainValues, unsigned numberOfGainValues)
{
// Make sure we're processing from the same type of bus.
// We *are* able to process from mono -> stereo
if (sourceBus.numberOfChannels() != 1 && !topologyMatches(sourceBus)) {
ASSERT_NOT_REACHED();
return;
}
if (!gainValues || numberOfGainValues > sourceBus.length()) {
ASSERT_NOT_REACHED();
return;
}
if (sourceBus.length() == numberOfGainValues && sourceBus.length() == length() && sourceBus.isSilent()) {
zero();
return;
}
// We handle both the 1 -> N and N -> N case here.
const float* source = sourceBus.channel(0)->data();
for (unsigned channelIndex = 0; channelIndex < numberOfChannels(); ++channelIndex) {
if (sourceBus.numberOfChannels() == numberOfChannels())
source = sourceBus.channel(channelIndex)->data();
float* destination = channel(channelIndex)->mutableData();
vmul(source, 1, gainValues, 1, destination, 1, numberOfGainValues);
}
}
PassRefPtr<AudioBus> AudioBus::createBySampleRateConverting(const AudioBus* sourceBus, bool mixToMono, double newSampleRate)
{
// sourceBus's sample-rate must be known.
ASSERT(sourceBus && sourceBus->sampleRate());
if (!sourceBus || !sourceBus->sampleRate())
return 0;
double sourceSampleRate = sourceBus->sampleRate();
double destinationSampleRate = newSampleRate;
double sampleRateRatio = sourceSampleRate / destinationSampleRate;
unsigned numberOfSourceChannels = sourceBus->numberOfChannels();
if (numberOfSourceChannels == 1)
mixToMono = false; // already mono
if (sourceSampleRate == destinationSampleRate) {
// No sample-rate conversion is necessary.
if (mixToMono)
return AudioBus::createByMixingToMono(sourceBus);
// Return exact copy.
return AudioBus::createBufferFromRange(sourceBus, 0, sourceBus->length());
}
if (sourceBus->isSilent()) {
RefPtr<AudioBus> silentBus = create(numberOfSourceChannels, sourceBus->length() / sampleRateRatio);
silentBus->setSampleRate(newSampleRate);
return silentBus;
}
// First, mix to mono (if necessary) then sample-rate convert.
const AudioBus* resamplerSourceBus;
RefPtr<AudioBus> mixedMonoBus;
if (mixToMono) {
mixedMonoBus = AudioBus::createByMixingToMono(sourceBus);
resamplerSourceBus = mixedMonoBus.get();
} else {
// Directly resample without down-mixing.
resamplerSourceBus = sourceBus;
}
// Calculate destination length based on the sample-rates.
int sourceLength = resamplerSourceBus->length();
int destinationLength = sourceLength / sampleRateRatio;
// Create destination bus with same number of channels.
unsigned numberOfDestinationChannels = resamplerSourceBus->numberOfChannels();
RefPtr<AudioBus> destinationBus = create(numberOfDestinationChannels, destinationLength);
// Sample-rate convert each channel.
for (unsigned i = 0; i < numberOfDestinationChannels; ++i) {
const float* source = resamplerSourceBus->channel(i)->data();
float* destination = destinationBus->channel(i)->mutableData();
SincResampler resampler(sampleRateRatio);
resampler.process(source, destination, sourceLength);
}
destinationBus->clearSilentFlag();
destinationBus->setSampleRate(newSampleRate);
return destinationBus;
}
PassRefPtr<AudioBus> AudioBus::createByMixingToMono(const AudioBus* sourceBus)
{
if (sourceBus->isSilent())
return create(1, sourceBus->length());
switch (sourceBus->numberOfChannels()) {
case 1:
// Simply create an exact copy.
return AudioBus::createBufferFromRange(sourceBus, 0, sourceBus->length());
case 2:
{
unsigned n = sourceBus->length();
RefPtr<AudioBus> destinationBus = create(1, n);
const float* sourceL = sourceBus->channel(0)->data();
const float* sourceR = sourceBus->channel(1)->data();
float* destination = destinationBus->channel(0)->mutableData();
// Do the mono mixdown.
for (unsigned i = 0; i < n; ++i)
destination[i] = (sourceL[i] + sourceR[i]) / 2;
destinationBus->clearSilentFlag();
destinationBus->setSampleRate(sourceBus->sampleRate());
return destinationBus;
}
}
ASSERT_NOT_REACHED();
return 0;
}
bool AudioBus::isSilent() const
{
for (size_t i = 0; i < m_channels.size(); ++i) {
if (!m_channels[i]->isSilent())
return false;
}
return true;
}
void AudioBus::clearSilentFlag()
{
for (size_t i = 0; i < m_channels.size(); ++i)
m_channels[i]->clearSilentFlag();
}
} // WebCore
#endif // ENABLE(WEB_AUDIO)