Bug 815643 - Part 5: Implement ConvolverNode's processing based on the Blink implementation; r=roc

--HG--
extra : rebase_source : 4c1a8dbaa4ab2d4fcc67c0a5226a6937e0656404

content/media/webaudio/ConvolverNode.cpp

@@ -8,6 +8,7 @@
 #include "mozilla/dom/ConvolverNodeBinding.h"
 #include "AudioNodeEngine.h"
 #include "AudioNodeStream.h"
+#include "blink/Reverb.h"
 
 #include <cmath>
 #include "nsMathUtils.h"
@@ -28,16 +29,32 @@ class ConvolverNodeEngine : public AudioNodeEngine
 public:
   ConvolverNodeEngine(AudioNode* aNode, bool aNormalize)
     : AudioNodeEngine(aNode)
+    , mBufferLength(0)
+    , mSampleRate(0.0f)
+    , mUseBackgroundThreads(!aNode->Context()->IsOffline())
     , mNormalize(aNormalize)
+    , mSeenInput(false)
   {
   }
 
   enum Parameters {
+    BUFFER_LENGTH,
+    SAMPLE_RATE,
     NORMALIZE
   };
   virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam) MOZ_OVERRIDE
   {
     switch (aIndex) {
+    case BUFFER_LENGTH:
+      // BUFFER_LENGTH is the first parameter that we set when setting a new buffer,
+      // so we should be careful to invalidate the rest of our state here.
+      mBuffer = nullptr;
+      mSampleRate = 0.0f;
+      mBufferLength = aParam;
+      break;
+    case SAMPLE_RATE:
+      mSampleRate = aParam;
+      break;
     case NORMALIZE:
       mNormalize = !!aParam;
       break;
@@ -45,9 +62,43 @@ public:
       NS_ERROR("Bad ConvolverNodeEngine Int32Parameter");
     }
   }
+  virtual void SetDoubleParameter(uint32_t aIndex, double aParam) MOZ_OVERRIDE
+  {
+    switch (aIndex) {
+    case SAMPLE_RATE:
+      mSampleRate = aParam;
+      AdjustReverb();
+      break;
+    default:
+      NS_ERROR("Bad ConvolverNodeEngine DoubleParameter");
+    }
+  }
   virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer)
   {
     mBuffer = aBuffer;
+    AdjustReverb();
+  }
+
+  void AdjustReverb()
+  {
+    // Note about empirical tuning (this is copied from Blink)
+    // The maximum FFT size affects reverb performance and accuracy.
+    // If the reverb is single-threaded and processes entirely in the real-time audio thread,
+    // it's important not to make this too high.  In this case 8192 is a good value.
+    // But, the Reverb object is multi-threaded, so we want this as high as possible without losing too much accuracy.
+    // Very large FFTs will have worse phase errors. Given these constraints 32768 is a good compromise.
+    const size_t MaxFFTSize = 32768;
+
+    if (!mBuffer || !mBufferLength || !mSampleRate) {
+      mReverb = nullptr;
+      mSeenInput = false;
+      return;
+    }
+
+    mReverb = new WebCore::Reverb(mBuffer, mBufferLength,
+                                  WEBAUDIO_BLOCK_SIZE,
+                                  MaxFFTSize, 2, mUseBackgroundThreads,
+                                  mNormalize, mSampleRate);
   }
 
   virtual void ProduceAudioBlock(AudioNodeStream* aStream,
@@ -55,12 +106,46 @@ public:
                                  AudioChunk* aOutput,
                                  bool* aFinished)
   {
-    *aOutput = aInput;
+    if (!mSeenInput && aInput.IsNull()) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+    if (!mReverb) {
+      *aOutput = aInput;
+      return;
+    }
+
+    mSeenInput = true;
+    uint32_t numChannels = 2;
+    AudioChunk input = aInput;
+    if (aInput.IsNull()) {
+      AllocateAudioBlock(1, &input);
+      WriteZeroesToAudioBlock(&input, 0, WEBAUDIO_BLOCK_SIZE);
+    } else if (aInput.mVolume != 1.0f) {
+      // Pre-multiply the input's volume
+      numChannels = aInput.mChannelData.Length();
+      AllocateAudioBlock(numChannels, &input);
+      for (uint32_t i = 0; i < numChannels; ++i) {
+        const float* src = static_cast<const float*>(aInput.mChannelData[i]);
+        float* dest = static_cast<float*>(const_cast<void*>(input.mChannelData[i]));
+        AudioBlockAddChannelWithScale(src, aInput.mVolume, dest);
+      }
+    } else {
+      numChannels = aInput.mChannelData.Length();
+    }
+    AllocateAudioBlock(numChannels, aOutput);
+
+    mReverb->process(&input, aOutput, WEBAUDIO_BLOCK_SIZE);
   }
 
 private:
   nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+  nsAutoPtr<WebCore::Reverb> mReverb;
+  int32_t mBufferLength;
+  float mSampleRate;
+  bool mUseBackgroundThreads;
   bool mNormalize;
+  bool mSeenInput;
 };
 
 ConvolverNode::ConvolverNode(AudioContext* aContext)
@@ -100,8 +185,27 @@ ConvolverNode::SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv)
   AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
   MOZ_ASSERT(ns, "Why don't we have a stream here?");
   if (mBuffer) {
+    uint32_t length = mBuffer->Length();
     nsRefPtr<ThreadSharedFloatArrayBufferList> data =
       mBuffer->GetThreadSharedChannelsForRate(aCx);
+    if (length < WEBAUDIO_BLOCK_SIZE) {
+      // For very small impulse response buffers, we need to pad the
+      // buffer with 0 to make sure that the Reverb implementation
+      // has enough data to compute FFTs from.
+      length = WEBAUDIO_BLOCK_SIZE;
+      nsRefPtr<ThreadSharedFloatArrayBufferList> paddedBuffer =
+        new ThreadSharedFloatArrayBufferList(data->GetChannels());
+      float* channelData = (float*) malloc(sizeof(float) * length * data->GetChannels());
+      for (uint32_t i = 0; i < data->GetChannels(); ++i) {
+        PodCopy(channelData + length * i, data->GetData(i), mBuffer->Length());
+        PodZero(channelData + length * i + mBuffer->Length(), WEBAUDIO_BLOCK_SIZE - mBuffer->Length());
+        paddedBuffer->SetData(i, (i == 0) ? channelData : nullptr, channelData);
+      }
+      data = paddedBuffer;
+    }
+    SendInt32ParameterToStream(ConvolverNodeEngine::BUFFER_LENGTH, length);
+    SendDoubleParameterToStream(ConvolverNodeEngine::SAMPLE_RATE,
+                                mBuffer->SampleRate());
     ns->SetBuffer(data.forget());
   } else {
     ns->SetBuffer(nullptr);