Bug 890716. Fix race in Reverb::Reverb. r=padenot

--HG--
extra : rebase_source : 65af577f5ffcf7f7a0b650f1693f4159ddac4ac6

content/media/AudioNodeEngine.cpp

@@ -39,6 +39,20 @@ WriteZeroesToAudioBlock(AudioChunk* aChunk, uint32_t aStart, uint32_t aLength)
   }
 }
 
+void AudioBufferCopyWithScale(const float* aInput,
+                              float aScale,
+                              float* aOutput,
+                              uint32_t aSize)
+{
+  if (aScale == 1.0f) {
+    PodCopy(aOutput, aInput, aSize);
+  } else {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] = aInput[i]*aScale;
+    }
+  }
+}
+
 void AudioBufferAddWithScale(const float* aInput,
                              float aScale,
                              float* aOutput,

content/media/AudioNodeEngine.h

@@ -87,6 +87,14 @@ void AllocateAudioBlock(uint32_t aChannelCount, AudioChunk* aChunk);
  */
 void WriteZeroesToAudioBlock(AudioChunk* aChunk, uint32_t aStart, uint32_t aLength);
 
+/**
+ * Copy with scale. aScale == 1.0f should be optimized.
+ */
+void AudioBufferCopyWithScale(const float* aInput,
+                              float aScale,
+                              float* aOutput,
+                              uint32_t aSize);
+
 /**
  * Pointwise multiply-add operation. aScale == 1.0f should be optimized.
  */

content/media/webaudio/blink/Reverb.cpp

@@ -80,41 +80,43 @@ Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulse
 {
     float scale = 1;
 
+    nsAutoTArray<const float*,4> irChannels;
+    for (size_t i = 0; i < impulseResponse->GetChannels(); ++i) {
+        irChannels.AppendElement(impulseResponse->GetData(i));
+    }
+    nsAutoTArray<float,1024> tempBuf;
+
     if (normalize) {
         scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
 
         if (scale) {
-            for (uint32_t i = 0; i < impulseResponse->GetChannels(); ++i) {
+            tempBuf.SetLength(irChannels.Length()*impulseResponseBufferLength);
-                AudioBufferInPlaceScale(const_cast<float*>(impulseResponse->GetData(i)),
+            for (uint32_t i = 0; i < irChannels.Length(); ++i) {
-                                        1, scale, impulseResponseBufferLength);
+                float* buf = &tempBuf[i*impulseResponseBufferLength];
+                AudioBufferCopyWithScale(irChannels[i], scale, buf,
+                                         impulseResponseBufferLength);
+                irChannels[i] = buf;
             }
         }
     }
 
-    initialize(impulseResponse, impulseResponseBufferLength, renderSliceSize, maxFFTSize, numberOfChannels, useBackgroundThreads);
+    initialize(irChannels, impulseResponseBufferLength, renderSliceSize,
-
+               maxFFTSize, numberOfChannels, useBackgroundThreads);
-    // Undo scaling since this shouldn't be a destructive operation on impulseResponse.
-    // FIXME: What about roundoff? Perhaps consider making a temporary scaled copy
-    // instead of scaling and unscaling in place.
-    if (normalize && scale) {
-        for (uint32_t i = 0; i < impulseResponse->GetChannels(); ++i) {
-            AudioBufferInPlaceScale(const_cast<float*>(impulseResponse->GetData(i)),
-                                    1, 1 / scale, impulseResponseBufferLength);
-        }
-    }
 }
 
-void Reverb::initialize(ThreadSharedFloatArrayBufferList* impulseResponseBuffer, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                        size_t impulseResponseBufferLength, size_t renderSliceSize,
+                        size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
 {
     m_impulseResponseLength = impulseResponseBufferLength;
 
     // The reverb can handle a mono impulse response and still do stereo processing
-    size_t numResponseChannels = impulseResponseBuffer->GetChannels();
+    size_t numResponseChannels = impulseResponseBuffer.Length();
     m_convolvers.SetCapacity(numberOfChannels);
 
     int convolverRenderPhase = 0;
     for (size_t i = 0; i < numResponseChannels; ++i) {
-        const float* channel = impulseResponseBuffer->GetData(i);
+        const float* channel = impulseResponseBuffer[i];
         size_t length = impulseResponseBufferLength;
 
         nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));

content/media/webaudio/blink/Reverb.h

@@ -55,7 +55,7 @@ public:
     size_t latencyFrames() const;
 
 private:
-    void initialize(mozilla::ThreadSharedFloatArrayBufferList* impulseResponseBuffer, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads);
+    void initialize(const nsTArray<const float*>& impulseResponseBuffer, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads);
 
     size_t m_impulseResponseLength;