Bug 1454998 - Remove AudioNodeStream::SampleRate(). r=karlt

AudioNodeStream is a subclass of MediaStream, which now exposes a
public const mSampleRate member.

Differential Revision: https://phabricator.services.mozilla.com/D47688

--HG--
extra : moz-landing-system : lando

dom/media/MediaStreamGraph.cpp

@@ -1205,7 +1205,7 @@ void MediaStreamGraphImpl::Process() {
           for (uint32_t j = i + 1; j < mStreams.Length(); ++j) {
             AudioNodeStream* nextStream = mStreams[j]->AsAudioNodeStream();
             if (nextStream) {
-              MOZ_ASSERT(n->SampleRate() == nextStream->SampleRate(),
+              MOZ_ASSERT(n->mSampleRate == nextStream->mSampleRate,
                          "All AudioNodeStreams in the graph must have the same "
                          "sampling rate");
             }
@@ -1213,7 +1213,7 @@ void MediaStreamGraphImpl::Process() {
 #endif
           // Since an AudioNodeStream is present, go ahead and
           // produce audio block by block for all the rest of the streams.
-          ProduceDataForStreamsBlockByBlock(i, n->SampleRate());
+          ProduceDataForStreamsBlockByBlock(i, n->mSampleRate);
           doneAllProducing = true;
         } else {
           ps->ProcessInput(mProcessedTime, mStateComputedTime,

dom/media/webaudio/AudioBufferSourceNode.cpp

@@ -97,7 +97,7 @@ class AudioBufferSourceNodeEngine final : public AudioNodeEngine {
     switch (aIndex) {
       case AudioBufferSourceNode::START:
         MOZ_ASSERT(!mStart, "Another START?");
-        mStart = aParam * mDestination->SampleRate();
+        mStart = aParam * mDestination->mSampleRate;
         // Round to nearest
         mBeginProcessing = mStart + 0.5;
         break;
@@ -439,7 +439,7 @@ class AudioBufferSourceNodeEngine final : public AudioNodeEngine {
     float computedPlaybackRate = aPlaybackRate * exp2(aDetune / 1200.f);
     // Make sure the playback rate is something our resampler can work with.
     int32_t rate = WebAudioUtils::TruncateFloatToInt<int32_t>(
-        mSource->SampleRate() / computedPlaybackRate);
+        mSource->mSampleRate / computedPlaybackRate);
     return rate ? rate : mBufferSampleRate;
   }
 

dom/media/webaudio/AudioNodeStream.cpp

@@ -194,7 +194,7 @@ void AudioNodeStream::SendTimelineEvent(uint32_t aIndex,
             const AudioTimelineEvent& aEvent)
         : ControlMessage(aStream),
           mEvent(aEvent),
-          mSampleRate(aStream->SampleRate()),
+          mSampleRate(aStream->mSampleRate),
           mIndex(aIndex) {}
     void Run() override {
       static_cast<AudioNodeStream*>(mStream)->Engine()->RecvTimelineEvent(
@@ -222,11 +222,15 @@ void AudioNodeStream::SetBuffer(AudioChunk&& aBuffer) {
   GraphImpl()->AppendMessage(MakeUnique<Message>(this, std::move(aBuffer)));
 }
 
-void AudioNodeStream::SetReverb(WebCore::Reverb* aReverb, uint32_t aImpulseChannelCount) {
+void AudioNodeStream::SetReverb(WebCore::Reverb* aReverb,
+                                uint32_t aImpulseChannelCount) {
   class Message final : public ControlMessage {
    public:
-    Message(AudioNodeStream* aStream, WebCore::Reverb* aReverb, uint32_t aImpulseChannelCount)
-        : ControlMessage(aStream), mReverb(aReverb), mImpulseChanelCount(aImpulseChannelCount) {}
+    Message(AudioNodeStream* aStream, WebCore::Reverb* aReverb,
+            uint32_t aImpulseChannelCount)
+        : ControlMessage(aStream),
+          mReverb(aReverb),
+          mImpulseChanelCount(aImpulseChannelCount) {}
     void Run() override {
       static_cast<AudioNodeStream*>(mStream)->Engine()->SetReverb(
           mReverb.forget(), mImpulseChanelCount);
@@ -235,7 +239,8 @@ void AudioNodeStream::SetReverb(WebCore::Reverb* aReverb, uint32_t aImpulseChann
     uint32_t mImpulseChanelCount;
   };
 
-  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aReverb, aImpulseChannelCount));
+  GraphImpl()->AppendMessage(
+      MakeUnique<Message>(this, aReverb, aImpulseChannelCount));
 }
 
 void AudioNodeStream::SetRawArrayData(nsTArray<float>& aData) {

dom/media/webaudio/AudioNodeStream.h

@@ -153,7 +153,6 @@ class AudioNodeStream : public ProcessedMediaStream {
 
   // Any thread
   AudioNodeEngine* Engine() { return mEngine; }
-  TrackRate SampleRate() const { return mSampleRate; }
 
   size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
   size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;

dom/media/webaudio/BiquadFilterNode.cpp

@@ -188,8 +188,8 @@ class BiquadFilterNodeEngine final : public AudioNodeEngine {
           input = alignedInputBuffer;
         }
       }
-      SetParamsOnBiquad(mBiquads[i], aStream->SampleRate(), mType, freq, q,
-                        gain, detune);
+      SetParamsOnBiquad(mBiquads[i], aStream->mSampleRate, mType, freq, q, gain,
+                        detune);
 
       mBiquads[i].process(input, aOutput->ChannelFloatsForWrite(i),
                           aInput.GetDuration());

dom/media/webaudio/DelayNode.cpp

@@ -65,7 +65,7 @@ class DelayNodeEngine final : public AudioNodeEngine {
   void ProcessBlock(AudioNodeStream* aStream, GraphTime aFrom,
                     const AudioBlock& aInput, AudioBlock* aOutput,
                     bool* aFinished) override {
-    MOZ_ASSERT(aStream->SampleRate() == mDestination->SampleRate());
+    MOZ_ASSERT(aStream->mSampleRate == mDestination->mSampleRate);
 
     if (!aInput.IsSilentOrSubnormal()) {
       if (mLeftOverData <= 0) {
@@ -106,7 +106,7 @@ class DelayNodeEngine final : public AudioNodeEngine {
   void UpdateOutputBlock(AudioNodeStream* aStream, GraphTime aFrom,
                          AudioBlock* aOutput, float minDelay) {
     float maxDelay = mMaxDelay;
-    float sampleRate = aStream->SampleRate();
+    float sampleRate = aStream->mSampleRate;
     ChannelInterpretation channelInterpretation =
         aStream->GetChannelInterpretation();
     if (mDelay.HasSimpleValue()) {

dom/media/webaudio/DynamicsCompressorNode.cpp

@@ -41,7 +41,7 @@ class DynamicsCompressorNodeEngine final : public AudioNodeEngine {
         mRatio(12.f),
         mAttack(0.003f),
         mRelease(0.25f),
-        mCompressor(new DynamicsCompressor(mDestination->SampleRate(), 2)) {}
+        mCompressor(new DynamicsCompressor(mDestination->mSampleRate, 2)) {}
 
   enum Parameters { THRESHOLD, KNEE, RATIO, ATTACK, RELEASE };
   void RecvTimelineEvent(uint32_t aIndex, AudioTimelineEvent& aEvent) override {
@@ -82,7 +82,7 @@ class DynamicsCompressorNodeEngine final : public AudioNodeEngine {
     const uint32_t channelCount = aInput.ChannelCount();
     if (mCompressor->numberOfChannels() != channelCount) {
       // Create a new compressor object with a new channel count
-      mCompressor = new WebCore::DynamicsCompressor(aStream->SampleRate(),
+      mCompressor = new WebCore::DynamicsCompressor(aStream->mSampleRate,
                                                     aInput.ChannelCount());
     }
 

dom/media/webaudio/OscillatorNode.cpp

@@ -131,7 +131,7 @@ class OscillatorNodeEngine final : public AudioNodeEngine {
                "PeriodicWave should have sent two channels");
     MOZ_ASSERT(aBuffer.mVolume == 1.0f);
     mPeriodicWave = WebCore::PeriodicWave::create(
-        mSource->SampleRate(), aBuffer.ChannelData<float>()[0],
+        mSource->mSampleRate, aBuffer.ChannelData<float>()[0],
         aBuffer.ChannelData<float>()[1], aBuffer.mDuration,
         mCustomDisableNormalization);
   }
@@ -172,7 +172,7 @@ class OscillatorNodeEngine final : public AudioNodeEngine {
     }
 
     float finalFrequency = frequency * exp2(detune / 1200.);
-    float signalPeriod = mSource->SampleRate() / finalFrequency;
+    float signalPeriod = mSource->mSampleRate / finalFrequency;
     mRecomputeParameters = false;
 
     mPhaseIncrement = 2 * M_PI / signalPeriod;

dom/media/webaudio/ScriptProcessorNode.cpp

@@ -244,7 +244,7 @@ class ScriptProcessorNodeEngine final : public AudioNodeEngine {
                             uint32_t aNumberOfInputChannels)
       : AudioNodeEngine(aNode),
         mDestination(aDestination->Stream()),
-        mSharedBuffers(new SharedBuffers(mDestination->SampleRate())),
+        mSharedBuffers(new SharedBuffers(mDestination->mSampleRate)),
         mBufferSize(aBufferSize),
         mInputChannelCount(aNumberOfInputChannels),
         mInputWriteIndex(0) {}

dom/media/webaudio/WaveShaperNode.cpp

@@ -242,19 +242,19 @@ class WaveShaperNodeEngine final : public AudioNodeEngine {
 
       switch (mType) {
         case OverSampleType::None:
-          mResampler.Reset(channelCount, aStream->SampleRate(),
+          mResampler.Reset(channelCount, aStream->mSampleRate,
                            OverSampleType::None);
           ProcessCurve<1>(inputSamples, outputBuffer);
           break;
         case OverSampleType::_2x:
-          mResampler.Reset(channelCount, aStream->SampleRate(),
+          mResampler.Reset(channelCount, aStream->mSampleRate,
                            OverSampleType::_2x);
           sampleBuffer = mResampler.UpSample(i, inputSamples, 2);
           ProcessCurve<2>(sampleBuffer, sampleBuffer);
           mResampler.DownSample(i, outputBuffer, 2);
           break;
         case OverSampleType::_4x:
-          mResampler.Reset(channelCount, aStream->SampleRate(),
+          mResampler.Reset(channelCount, aStream->mSampleRate,
                            OverSampleType::_4x);
           sampleBuffer = mResampler.UpSample(i, inputSamples, 4);
           ProcessCurve<4>(sampleBuffer, sampleBuffer);

dom/media/webaudio/WebAudioUtils.cpp

@@ -23,8 +23,8 @@ void WebAudioUtils::ConvertAudioTimelineEventToTicks(AudioTimelineEvent& aEvent,
                                                      AudioNodeStream* aDest) {
   aEvent.SetTimeInTicks(
       aDest->SecondsToNearestStreamTime(aEvent.Time<double>()));
-  aEvent.mTimeConstant *= aDest->SampleRate();
-  aEvent.mDuration *= aDest->SampleRate();
+  aEvent.mTimeConstant *= aDest->mSampleRate;
+  aEvent.mDuration *= aDest->mSampleRate;
 }
 
 void WebAudioUtils::Shutdown() { WebCore::HRTFDatabaseLoader::shutdown(); }