Add m-esr52 at 52.6.0

1 files changed, 234 insertions, 0 deletions

diff --git a/dom/media/webaudio/DelayNode.cpp b/dom/media/webaudio/DelayNode.cpp
new file mode 100644
index 000000000..17dc72514
--- /dev/null
+++ b/dom/media/webaudio/DelayNode.cpp
@@ -0,0 +1,234 @@
+/* -*- 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 "DelayNode.h"
+#include "mozilla/dom/DelayNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+#include "DelayBuffer.h"
+#include "PlayingRefChangeHandler.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(DelayNode, AudioNode,
+                                   mDelay)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(DelayNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(DelayNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(DelayNode, AudioNode)
+
+class DelayNodeEngine final : public AudioNodeEngine
+{
+  typedef PlayingRefChangeHandler PlayingRefChanged;
+public:
+  DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
+                  double aMaxDelayTicks)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Keep the default value in sync with the default value in DelayNode::DelayNode.
+    , mDelay(0.f)
+    // Use a smoothing range of 20ms
+    , mBuffer(std::max(aMaxDelayTicks,
+                       static_cast<double>(WEBAUDIO_BLOCK_SIZE)),
+              WebAudioUtils::ComputeSmoothingRate(0.02,
+                                                  mDestination->SampleRate()))
+    , mMaxDelay(aMaxDelayTicks)
+    , mHaveProducedBeforeInput(false)
+    , mLeftOverData(INT32_MIN)
+  {
+  }
+
+  DelayNodeEngine* AsDelayNodeEngine() override
+  {
+    return this;
+  }
+
+  enum Parameters {
+    DELAY,
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case DELAY:
+      mDelay.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad DelayNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    MOZ_ASSERT(aStream->SampleRate() == mDestination->SampleRate());
+
+    if (!aInput.IsSilentOrSubnormal()) {
+      if (mLeftOverData <= 0) {
+        RefPtr<PlayingRefChanged> refchanged =
+          new PlayingRefChanged(aStream, PlayingRefChanged::ADDREF);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      }
+      mLeftOverData = mBuffer.MaxDelayTicks();
+    } else if (mLeftOverData > 0) {
+      mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
+    } else {
+      if (mLeftOverData != INT32_MIN) {
+        mLeftOverData = INT32_MIN;
+        aStream->ScheduleCheckForInactive();
+
+        // Delete our buffered data now we no longer need it
+        mBuffer.Reset();
+
+        RefPtr<PlayingRefChanged> refchanged =
+          new PlayingRefChanged(aStream, PlayingRefChanged::RELEASE);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      }
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    mBuffer.Write(aInput);
+
+    // Skip output update if mLastChunks has already been set by
+    // ProduceBlockBeforeInput() when in a cycle.
+    if (!mHaveProducedBeforeInput) {
+      UpdateOutputBlock(aStream, aFrom, aOutput, 0.0);
+    }
+    mHaveProducedBeforeInput = false;
+    mBuffer.NextBlock();
+  }
+
+  void UpdateOutputBlock(AudioNodeStream* aStream, GraphTime aFrom,
+                         AudioBlock* aOutput, double minDelay)
+  {
+    double maxDelay = mMaxDelay;
+    double sampleRate = aStream->SampleRate();
+    ChannelInterpretation channelInterpretation =
+      aStream->GetChannelInterpretation();
+    if (mDelay.HasSimpleValue()) {
+      // If this DelayNode is in a cycle, make sure the delay value is at least
+      // one block, even if that is greater than maxDelay.
+      double delayFrames = mDelay.GetValue() * sampleRate;
+      double delayFramesClamped =
+        std::max(minDelay, std::min(delayFrames, maxDelay));
+      mBuffer.Read(delayFramesClamped, aOutput, channelInterpretation);
+    } else {
+      // Compute the delay values for the duration of the input AudioChunk
+      // If this DelayNode is in a cycle, make sure the delay value is at least
+      // one block.
+      StreamTime tick = mDestination->GraphTimeToStreamTime(aFrom);
+      float values[WEBAUDIO_BLOCK_SIZE];
+      mDelay.GetValuesAtTime(tick, values,WEBAUDIO_BLOCK_SIZE);
+
+      double computedDelay[WEBAUDIO_BLOCK_SIZE];
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        double delayAtTick = values[counter] * sampleRate;
+        double delayAtTickClamped =
+          std::max(minDelay, std::min(delayAtTick, maxDelay));
+        computedDelay[counter] = delayAtTickClamped;
+      }
+      mBuffer.Read(computedDelay, aOutput, channelInterpretation);
+    }
+  }
+
+  void ProduceBlockBeforeInput(AudioNodeStream* aStream,
+                               GraphTime aFrom,
+                               AudioBlock* aOutput) override
+  {
+    if (mLeftOverData <= 0) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else {
+      UpdateOutputBlock(aStream, aFrom, aOutput, WEBAUDIO_BLOCK_SIZE);
+    }
+    mHaveProducedBeforeInput = true;
+  }
+
+  bool IsActive() const override
+  {
+    return mLeftOverData != INT32_MIN;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    // Not owned:
+    // - mDestination - probably not owned
+    // - mDelay - shares ref with AudioNode, don't count
+    amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mDestination;
+  AudioParamTimeline mDelay;
+  DelayBuffer mBuffer;
+  double mMaxDelay;
+  bool mHaveProducedBeforeInput;
+  // How much data we have in our buffer which needs to be flushed out when our inputs
+  // finish.
+  int32_t mLeftOverData;
+};
+
+DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mDelay(new AudioParam(this, DelayNodeEngine::DELAY, 0.0f, "delayTime"))
+{
+  DelayNodeEngine* engine =
+    new DelayNodeEngine(this, aContext->Destination(),
+                        aContext->SampleRate() * aMaxDelay);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+DelayNode::~DelayNode()
+{
+}
+
+size_t
+DelayNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mDelay->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+DelayNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+DelayNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return DelayNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla