gecko/content/media/webaudio/AudioContext.cpp

535 lines
14 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/. */
#include "AudioContext.h"
#include "nsContentUtils.h"
#include "nsPIDOMWindow.h"
#include "mozilla/ErrorResult.h"
#include "mozilla/dom/AudioContextBinding.h"
#include "mozilla/dom/OfflineAudioContextBinding.h"
#include "MediaStreamGraph.h"
#include "mozilla/dom/AnalyserNode.h"
#include "AudioDestinationNode.h"
#include "AudioBufferSourceNode.h"
#include "AudioBuffer.h"
#include "GainNode.h"
#include "DelayNode.h"
#include "PannerNode.h"
#include "AudioListener.h"
#include "DynamicsCompressorNode.h"
#include "BiquadFilterNode.h"
#include "ScriptProcessorNode.h"
#include "ChannelMergerNode.h"
#include "ChannelSplitterNode.h"
#include "MediaStreamAudioDestinationNode.h"
#include "WaveShaperNode.h"
#include "PeriodicWave.h"
#include "ConvolverNode.h"
#include "nsNetUtil.h"
namespace mozilla {
namespace dom {
NS_IMPL_CYCLE_COLLECTION_INHERITED_2(AudioContext, nsDOMEventTargetHelper,
mDestination, mListener)
NS_IMPL_ADDREF_INHERITED(AudioContext, nsDOMEventTargetHelper)
NS_IMPL_RELEASE_INHERITED(AudioContext, nsDOMEventTargetHelper)
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioContext)
NS_INTERFACE_MAP_END_INHERITING(nsDOMEventTargetHelper)
static uint8_t gWebAudioOutputKey;
AudioContext::AudioContext(nsPIDOMWindow* aWindow,
bool aIsOffline,
uint32_t aNumberOfChannels,
uint32_t aLength,
float aSampleRate)
: mSampleRate(aIsOffline ? aSampleRate : IdealAudioRate())
, mDestination(new AudioDestinationNode(MOZ_THIS_IN_INITIALIZER_LIST(),
aIsOffline, aNumberOfChannels,
aLength, aSampleRate))
, mNumberOfChannels(aNumberOfChannels)
, mIsOffline(aIsOffline)
{
// Actually play audio
mDestination->Stream()->AddAudioOutput(&gWebAudioOutputKey);
nsDOMEventTargetHelper::BindToOwner(aWindow);
SetIsDOMBinding();
mPannerNodes.Init();
mAudioBufferSourceNodes.Init();
mScriptProcessorNodes.Init();
}
AudioContext::~AudioContext()
{
}
JSObject*
AudioContext::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aScope)
{
if (mIsOffline) {
return OfflineAudioContextBinding::Wrap(aCx, aScope, this);
} else {
return AudioContextBinding::Wrap(aCx, aScope, this);
}
}
/* static */ already_AddRefed<AudioContext>
AudioContext::Constructor(const GlobalObject& aGlobal, ErrorResult& aRv)
{
nsCOMPtr<nsPIDOMWindow> window = do_QueryInterface(aGlobal.Get());
if (!window) {
aRv.Throw(NS_ERROR_FAILURE);
return nullptr;
}
nsRefPtr<AudioContext> object = new AudioContext(window, false);
window->AddAudioContext(object);
return object.forget();
}
/* static */ already_AddRefed<AudioContext>
AudioContext::Constructor(const GlobalObject& aGlobal,
uint32_t aNumberOfChannels,
uint32_t aLength,
float aSampleRate,
ErrorResult& aRv)
{
nsCOMPtr<nsPIDOMWindow> window = do_QueryInterface(aGlobal.Get());
if (!window) {
aRv.Throw(NS_ERROR_FAILURE);
return nullptr;
}
if (aNumberOfChannels == 0 ||
aNumberOfChannels > WebAudioUtils::MaxChannelCount ||
aLength == 0 ||
aSampleRate <= 1.0f ||
aSampleRate >= TRACK_RATE_MAX) {
// The DOM binding protects us against infinity and NaN
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return nullptr;
}
nsRefPtr<AudioContext> object = new AudioContext(window,
true,
aNumberOfChannels,
aLength,
aSampleRate);
window->AddAudioContext(object);
return object.forget();
}
already_AddRefed<AudioBufferSourceNode>
AudioContext::CreateBufferSource()
{
nsRefPtr<AudioBufferSourceNode> bufferNode =
new AudioBufferSourceNode(this);
mAudioBufferSourceNodes.PutEntry(bufferNode);
return bufferNode.forget();
}
already_AddRefed<AudioBuffer>
AudioContext::CreateBuffer(JSContext* aJSContext, uint32_t aNumberOfChannels,
uint32_t aLength, float aSampleRate,
ErrorResult& aRv)
{
if (aSampleRate < 8000 || aSampleRate > 96000 || !aLength) {
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return nullptr;
}
if (aLength > INT32_MAX) {
aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
return nullptr;
}
nsRefPtr<AudioBuffer> buffer =
new AudioBuffer(this, int32_t(aLength), aSampleRate);
if (!buffer->InitializeBuffers(aNumberOfChannels, aJSContext)) {
aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
return nullptr;
}
return buffer.forget();
}
already_AddRefed<AudioBuffer>
AudioContext::CreateBuffer(JSContext* aJSContext, ArrayBuffer& aBuffer,
bool aMixToMono, ErrorResult& aRv)
{
// Sniff the content of the media.
// Failed type sniffing will be handled by SyncDecodeMedia.
nsAutoCString contentType;
NS_SniffContent(NS_DATA_SNIFFER_CATEGORY, nullptr,
aBuffer.Data(), aBuffer.Length(),
contentType);
WebAudioDecodeJob job(contentType, this);
if (mDecoder.SyncDecodeMedia(contentType.get(),
aBuffer.Data(), aBuffer.Length(), job) &&
job.mOutput) {
nsRefPtr<AudioBuffer> buffer = job.mOutput.forget();
if (aMixToMono) {
buffer->MixToMono(aJSContext);
}
return buffer.forget();
}
return nullptr;
}
namespace {
bool IsValidBufferSize(uint32_t aBufferSize) {
switch (aBufferSize) {
case 0: // let the implementation choose the buffer size
case 256:
case 512:
case 1024:
case 2048:
case 4096:
case 8192:
case 16384:
return true;
default:
return false;
}
}
}
already_AddRefed<MediaStreamAudioDestinationNode>
AudioContext::CreateMediaStreamDestination()
{
nsRefPtr<MediaStreamAudioDestinationNode> node =
new MediaStreamAudioDestinationNode(this);
return node.forget();
}
already_AddRefed<ScriptProcessorNode>
AudioContext::CreateScriptProcessor(uint32_t aBufferSize,
uint32_t aNumberOfInputChannels,
uint32_t aNumberOfOutputChannels,
ErrorResult& aRv)
{
if ((aNumberOfInputChannels == 0 && aNumberOfOutputChannels == 0) ||
aNumberOfInputChannels > WebAudioUtils::MaxChannelCount ||
aNumberOfOutputChannels > WebAudioUtils::MaxChannelCount ||
!IsValidBufferSize(aBufferSize)) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return nullptr;
}
nsRefPtr<ScriptProcessorNode> scriptProcessor =
new ScriptProcessorNode(this, aBufferSize, aNumberOfInputChannels,
aNumberOfOutputChannels);
mScriptProcessorNodes.PutEntry(scriptProcessor);
return scriptProcessor.forget();
}
already_AddRefed<AnalyserNode>
AudioContext::CreateAnalyser()
{
nsRefPtr<AnalyserNode> analyserNode = new AnalyserNode(this);
return analyserNode.forget();
}
already_AddRefed<GainNode>
AudioContext::CreateGain()
{
nsRefPtr<GainNode> gainNode = new GainNode(this);
return gainNode.forget();
}
already_AddRefed<WaveShaperNode>
AudioContext::CreateWaveShaper()
{
nsRefPtr<WaveShaperNode> waveShaperNode = new WaveShaperNode(this);
return waveShaperNode.forget();
}
already_AddRefed<DelayNode>
AudioContext::CreateDelay(double aMaxDelayTime, ErrorResult& aRv)
{
if (aMaxDelayTime > 0. && aMaxDelayTime < 180.) {
nsRefPtr<DelayNode> delayNode = new DelayNode(this, aMaxDelayTime);
return delayNode.forget();
}
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return nullptr;
}
already_AddRefed<PannerNode>
AudioContext::CreatePanner()
{
nsRefPtr<PannerNode> pannerNode = new PannerNode(this);
mPannerNodes.PutEntry(pannerNode);
return pannerNode.forget();
}
already_AddRefed<ConvolverNode>
AudioContext::CreateConvolver()
{
nsRefPtr<ConvolverNode> convolverNode = new ConvolverNode(this);
return convolverNode.forget();
}
already_AddRefed<ChannelSplitterNode>
AudioContext::CreateChannelSplitter(uint32_t aNumberOfOutputs, ErrorResult& aRv)
{
if (aNumberOfOutputs == 0 ||
aNumberOfOutputs > WebAudioUtils::MaxChannelCount) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return nullptr;
}
nsRefPtr<ChannelSplitterNode> splitterNode =
new ChannelSplitterNode(this, aNumberOfOutputs);
return splitterNode.forget();
}
already_AddRefed<ChannelMergerNode>
AudioContext::CreateChannelMerger(uint32_t aNumberOfInputs, ErrorResult& aRv)
{
if (aNumberOfInputs == 0 ||
aNumberOfInputs > WebAudioUtils::MaxChannelCount) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return nullptr;
}
nsRefPtr<ChannelMergerNode> mergerNode =
new ChannelMergerNode(this, aNumberOfInputs);
return mergerNode.forget();
}
already_AddRefed<DynamicsCompressorNode>
AudioContext::CreateDynamicsCompressor()
{
nsRefPtr<DynamicsCompressorNode> compressorNode =
new DynamicsCompressorNode(this);
return compressorNode.forget();
}
already_AddRefed<BiquadFilterNode>
AudioContext::CreateBiquadFilter()
{
nsRefPtr<BiquadFilterNode> filterNode =
new BiquadFilterNode(this);
return filterNode.forget();
}
already_AddRefed<PeriodicWave>
AudioContext::CreatePeriodicWave(const Float32Array& aRealData,
const Float32Array& aImagData,
ErrorResult& aRv)
{
if (aRealData.Length() != aImagData.Length() ||
aRealData.Length() == 0 ||
aRealData.Length() > 4096) {
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return nullptr;
}
nsRefPtr<PeriodicWave> periodicWave =
new PeriodicWave(this, aRealData.Data(), aRealData.Length(),
aImagData.Data(), aImagData.Length());
return periodicWave.forget();
}
AudioListener*
AudioContext::Listener()
{
if (!mListener) {
mListener = new AudioListener(this);
}
return mListener;
}
void
AudioContext::DecodeAudioData(const ArrayBuffer& aBuffer,
DecodeSuccessCallback& aSuccessCallback,
const Optional<OwningNonNull<DecodeErrorCallback> >& aFailureCallback)
{
// Sniff the content of the media.
// Failed type sniffing will be handled by AsyncDecodeMedia.
nsAutoCString contentType;
NS_SniffContent(NS_DATA_SNIFFER_CATEGORY, nullptr,
aBuffer.Data(), aBuffer.Length(),
contentType);
nsCOMPtr<DecodeErrorCallback> failureCallback;
if (aFailureCallback.WasPassed()) {
failureCallback = &aFailureCallback.Value();
}
nsAutoPtr<WebAudioDecodeJob> job(
new WebAudioDecodeJob(contentType, this,
&aSuccessCallback, failureCallback));
mDecoder.AsyncDecodeMedia(contentType.get(),
aBuffer.Data(), aBuffer.Length(), *job);
// Transfer the ownership to mDecodeJobs
mDecodeJobs.AppendElement(job.forget());
}
void
AudioContext::RemoveFromDecodeQueue(WebAudioDecodeJob* aDecodeJob)
{
mDecodeJobs.RemoveElement(aDecodeJob);
}
void
AudioContext::UnregisterAudioBufferSourceNode(AudioBufferSourceNode* aNode)
{
mAudioBufferSourceNodes.RemoveEntry(aNode);
UpdatePannerSource();
}
void
AudioContext::UnregisterPannerNode(PannerNode* aNode)
{
mPannerNodes.RemoveEntry(aNode);
}
void
AudioContext::UnregisterScriptProcessorNode(ScriptProcessorNode* aNode)
{
mScriptProcessorNodes.RemoveEntry(aNode);
}
static PLDHashOperator
FindConnectedSourcesOn(nsPtrHashKey<PannerNode>* aEntry, void* aData)
{
aEntry->GetKey()->FindConnectedSources();
return PL_DHASH_NEXT;
}
void
AudioContext::UpdatePannerSource()
{
mPannerNodes.EnumerateEntries(FindConnectedSourcesOn, nullptr);
}
uint32_t
AudioContext::MaxChannelCount() const
{
return mIsOffline ? mNumberOfChannels : AudioStream::MaxNumberOfChannels();
}
MediaStreamGraph*
AudioContext::Graph() const
{
return Destination()->Stream()->Graph();
}
MediaStream*
AudioContext::DestinationStream() const
{
return Destination()->Stream();
}
double
AudioContext::CurrentTime() const
{
return MediaTimeToSeconds(Destination()->Stream()->GetCurrentTime());
}
template <class T>
static PLDHashOperator
GetHashtableEntry(nsPtrHashKey<T>* aEntry, void* aData)
{
nsTArray<T*>* array = static_cast<nsTArray<T*>*>(aData);
array->AppendElement(aEntry->GetKey());
return PL_DHASH_NEXT;
}
template <class T>
static void
GetHashtableElements(nsTHashtable<nsPtrHashKey<T> >& aHashtable, nsTArray<T*>& aArray)
{
aHashtable.EnumerateEntries(&GetHashtableEntry<T>, &aArray);
}
void
AudioContext::Shutdown()
{
Suspend();
mDecoder.Shutdown();
// Stop all audio buffer source nodes, to make sure that they release
// their self-references.
// We first gather an array of the nodes and then call Stop on each one,
// since Stop may delete the object and therefore trigger a re-entrant
// hashtable call to remove the pointer from the hashtable, which is
// not safe.
nsTArray<AudioBufferSourceNode*> sourceNodes;
GetHashtableElements(mAudioBufferSourceNodes, sourceNodes);
for (uint32_t i = 0; i < sourceNodes.Length(); ++i) {
ErrorResult rv;
sourceNodes[i]->Stop(0.0, rv, true);
}
// Stop all script processor nodes, to make sure that they release
// their self-references.
nsTArray<ScriptProcessorNode*> spNodes;
GetHashtableElements(mScriptProcessorNodes, spNodes);
for (uint32_t i = 0; i < spNodes.Length(); ++i) {
spNodes[i]->Stop();
}
// For offline contexts, we can destroy the MediaStreamGraph at this point.
if (mIsOffline) {
mDestination->DestroyGraph();
}
}
void
AudioContext::Suspend()
{
MediaStream* ds = DestinationStream();
if (ds) {
ds->ChangeExplicitBlockerCount(1);
}
}
void
AudioContext::Resume()
{
MediaStream* ds = DestinationStream();
if (ds) {
ds->ChangeExplicitBlockerCount(-1);
}
}
JSContext*
AudioContext::GetJSContext() const
{
MOZ_ASSERT(NS_IsMainThread());
nsCOMPtr<nsIScriptGlobalObject> scriptGlobal =
do_QueryInterface(GetParentObject());
if (!scriptGlobal) {
return nullptr;
}
nsIScriptContext* scriptContext = scriptGlobal->GetContext();
if (!scriptContext) {
return nullptr;
}
return scriptContext->GetNativeContext();
}
void
AudioContext::StartRendering()
{
MOZ_ASSERT(mIsOffline, "This should only be called on OfflineAudioContext");
mDestination->StartRendering();
}
}
}