Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 389 insertions, 0 deletions

diff --git a/dom/media/webaudio/AnalyserNode.cpp b/dom/media/webaudio/AnalyserNode.cpp
new file mode 100644
index 0000000000..a3b0508a97
--- /dev/null
+++ b/dom/media/webaudio/AnalyserNode.cpp
@@ -0,0 +1,389 @@
+/* -*- 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 "mozilla/dom/AnalyserNode.h"
+#include "mozilla/dom/AnalyserNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/PodOperations.h"
+#include "nsMathUtils.h"
+#include "Tracing.h"
+
+namespace mozilla {
+
+static const uint32_t MAX_FFT_SIZE = 32768;
+static const size_t CHUNK_COUNT = MAX_FFT_SIZE >> WEBAUDIO_BLOCK_SIZE_BITS;
+static_assert(MAX_FFT_SIZE == CHUNK_COUNT * WEBAUDIO_BLOCK_SIZE,
+              "MAX_FFT_SIZE must be a multiple of WEBAUDIO_BLOCK_SIZE");
+static_assert((CHUNK_COUNT & (CHUNK_COUNT - 1)) == 0,
+              "CHUNK_COUNT must be power of 2 for remainder behavior");
+
+namespace dom {
+
+class AnalyserNodeEngine final : public AudioNodeEngine {
+  class TransferBuffer final : public Runnable {
+   public:
+    TransferBuffer(AudioNodeTrack* aTrack, const AudioChunk& aChunk)
+        : Runnable("dom::AnalyserNodeEngine::TransferBuffer"),
+          mTrack(aTrack),
+          mChunk(aChunk) {}
+
+    NS_IMETHOD Run() override {
+      RefPtr<AnalyserNode> node =
+          static_cast<AnalyserNode*>(mTrack->Engine()->NodeMainThread());
+      if (node) {
+        node->AppendChunk(mChunk);
+      }
+      return NS_OK;
+    }
+
+   private:
+    RefPtr<AudioNodeTrack> mTrack;
+    AudioChunk mChunk;
+  };
+
+ public:
+  explicit AnalyserNodeEngine(AnalyserNode* aNode) : AudioNodeEngine(aNode) {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  virtual void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            const AudioBlock& aInput, AudioBlock* aOutput,
+                            bool* aFinished) override {
+    TRACE("AnalyserNodeEngine::ProcessBlock");
+    *aOutput = aInput;
+
+    if (aInput.IsNull()) {
+      // If AnalyserNode::mChunks has only null chunks, then there is no need
+      // to send further null chunks.
+      if (mChunksToProcess == 0) {
+        return;
+      }
+
+      --mChunksToProcess;
+      if (mChunksToProcess == 0) {
+        aTrack->ScheduleCheckForInactive();
+      }
+
+    } else {
+      // This many null chunks will be required to empty AnalyserNode::mChunks.
+      mChunksToProcess = CHUNK_COUNT;
+    }
+
+    RefPtr<TransferBuffer> transfer =
+        new TransferBuffer(aTrack, aInput.AsAudioChunk());
+    mAbstractMainThread->Dispatch(transfer.forget());
+  }
+
+  virtual bool IsActive() const override { return mChunksToProcess != 0; }
+
+  virtual size_t SizeOfIncludingThis(
+      MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  uint32_t mChunksToProcess = 0;
+};
+
+/* static */
+already_AddRefed<AnalyserNode> AnalyserNode::Create(
+    AudioContext& aAudioContext, const AnalyserOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<AnalyserNode> analyserNode = new AnalyserNode(&aAudioContext);
+
+  analyserNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  analyserNode->SetFftSize(aOptions.mFftSize, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  analyserNode->SetMinAndMaxDecibels(aOptions.mMinDecibels,
+                                     aOptions.mMaxDecibels, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  analyserNode->SetSmoothingTimeConstant(aOptions.mSmoothingTimeConstant, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return analyserNode.forget();
+}
+
+AnalyserNode::AnalyserNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mAnalysisBlock(2048),
+      mMinDecibels(-100.),
+      mMaxDecibels(-30.),
+      mSmoothingTimeConstant(.8) {
+  mTrack =
+      AudioNodeTrack::Create(aContext, new AnalyserNodeEngine(this),
+                             AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+
+  // Enough chunks must be recorded to handle the case of fftSize being
+  // increased to maximum immediately before getFloatTimeDomainData() is
+  // called, for example.
+  Unused << mChunks.SetLength(CHUNK_COUNT, fallible);
+
+  AllocateBuffer();
+}
+
+size_t AnalyserNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mAnalysisBlock.SizeOfExcludingThis(aMallocSizeOf);
+  amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += mOutputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t AnalyserNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* AnalyserNode::WrapObject(JSContext* aCx,
+                                   JS::Handle<JSObject*> aGivenProto) {
+  return AnalyserNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void AnalyserNode::SetFftSize(uint32_t aValue, ErrorResult& aRv) {
+  // Disallow values that are not a power of 2 and outside the [32,32768] range
+  if (aValue < 32 || aValue > MAX_FFT_SIZE || (aValue & (aValue - 1)) != 0) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "FFT size %u is not a power of two in the range 32 to 32768", aValue));
+    return;
+  }
+  if (FftSize() != aValue) {
+    mAnalysisBlock.SetFFTSize(aValue);
+    AllocateBuffer();
+  }
+}
+
+void AnalyserNode::SetMinDecibels(double aValue, ErrorResult& aRv) {
+  if (aValue >= mMaxDecibels) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "%g is not strictly smaller than current maxDecibels (%g)", aValue,
+        mMaxDecibels));
+    return;
+  }
+  mMinDecibels = aValue;
+}
+
+void AnalyserNode::SetMaxDecibels(double aValue, ErrorResult& aRv) {
+  if (aValue <= mMinDecibels) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "%g is not strictly larger than current minDecibels (%g)", aValue,
+        mMinDecibels));
+    return;
+  }
+  mMaxDecibels = aValue;
+}
+
+void AnalyserNode::SetMinAndMaxDecibels(double aMinValue, double aMaxValue,
+                                        ErrorResult& aRv) {
+  if (aMinValue >= aMaxValue) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "minDecibels value (%g) must be smaller than maxDecibels value (%g)",
+        aMinValue, aMaxValue));
+    return;
+  }
+  mMinDecibels = aMinValue;
+  mMaxDecibels = aMaxValue;
+}
+
+void AnalyserNode::SetSmoothingTimeConstant(double aValue, ErrorResult& aRv) {
+  if (aValue < 0 || aValue > 1) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("%g is not in the range [0, 1]", aValue));
+    return;
+  }
+  mSmoothingTimeConstant = aValue;
+}
+
+void AnalyserNode::GetFloatFrequencyData(const Float32Array& aArray) {
+  if (!FFTAnalysis()) {
+    // Might fail to allocate memory
+    return;
+  }
+
+  aArray.ComputeState();
+
+  float* buffer = aArray.Data();
+  size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length());
+
+  for (size_t i = 0; i < length; ++i) {
+    buffer[i] = WebAudioUtils::ConvertLinearToDecibels(
+        mOutputBuffer[i], -std::numeric_limits<float>::infinity());
+  }
+}
+
+void AnalyserNode::GetByteFrequencyData(const Uint8Array& aArray) {
+  if (!FFTAnalysis()) {
+    // Might fail to allocate memory
+    return;
+  }
+
+  const double rangeScaleFactor = 1.0 / (mMaxDecibels - mMinDecibels);
+
+  aArray.ComputeState();
+
+  unsigned char* buffer = aArray.Data();
+  size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length());
+
+  for (size_t i = 0; i < length; ++i) {
+    const double decibels =
+        WebAudioUtils::ConvertLinearToDecibels(mOutputBuffer[i], mMinDecibels);
+    // scale down the value to the range of [0, UCHAR_MAX]
+    const double scaled = std::max(
+        0.0, std::min(double(UCHAR_MAX), UCHAR_MAX * (decibels - mMinDecibels) *
+                                             rangeScaleFactor));
+    buffer[i] = static_cast<unsigned char>(scaled);
+  }
+}
+
+void AnalyserNode::GetFloatTimeDomainData(const Float32Array& aArray) {
+  aArray.ComputeState();
+
+  float* buffer = aArray.Data();
+  size_t length = std::min(aArray.Length(), FftSize());
+
+  GetTimeDomainData(buffer, length);
+}
+
+void AnalyserNode::GetByteTimeDomainData(const Uint8Array& aArray) {
+  aArray.ComputeState();
+
+  size_t length = std::min(aArray.Length(), FftSize());
+
+  AlignedTArray<float> tmpBuffer;
+  if (!tmpBuffer.SetLength(length, fallible)) {
+    return;
+  }
+
+  GetTimeDomainData(tmpBuffer.Elements(), length);
+
+  unsigned char* buffer = aArray.Data();
+  for (size_t i = 0; i < length; ++i) {
+    const float value = tmpBuffer[i];
+    // scale the value to the range of [0, UCHAR_MAX]
+    const float scaled =
+        std::max(0.0f, std::min(float(UCHAR_MAX), 128.0f * (value + 1.0f)));
+    buffer[i] = static_cast<unsigned char>(scaled);
+  }
+}
+
+bool AnalyserNode::FFTAnalysis() {
+  AlignedTArray<float> tmpBuffer;
+  size_t fftSize = FftSize();
+  if (!tmpBuffer.SetLength(fftSize, fallible)) {
+    return false;
+  }
+
+  float* inputBuffer = tmpBuffer.Elements();
+  GetTimeDomainData(inputBuffer, fftSize);
+  ApplyBlackmanWindow(inputBuffer, fftSize);
+  mAnalysisBlock.PerformFFT(inputBuffer);
+
+  // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
+  // scaling factor).
+  const double magnitudeScale = 1.0 / fftSize;
+
+  for (uint32_t i = 0; i < mOutputBuffer.Length(); ++i) {
+    double scalarMagnitude =
+        NS_hypot(mAnalysisBlock.RealData(i), mAnalysisBlock.ImagData(i)) *
+        magnitudeScale;
+    mOutputBuffer[i] = mSmoothingTimeConstant * mOutputBuffer[i] +
+                       (1.0 - mSmoothingTimeConstant) * scalarMagnitude;
+  }
+
+  return true;
+}
+
+void AnalyserNode::ApplyBlackmanWindow(float* aBuffer, uint32_t aSize) {
+  double alpha = 0.16;
+  double a0 = 0.5 * (1.0 - alpha);
+  double a1 = 0.5;
+  double a2 = 0.5 * alpha;
+
+  for (uint32_t i = 0; i < aSize; ++i) {
+    double x = double(i) / aSize;
+    double window = a0 - a1 * cos(2 * M_PI * x) + a2 * cos(4 * M_PI * x);
+    aBuffer[i] *= window;
+  }
+}
+
+bool AnalyserNode::AllocateBuffer() {
+  bool result = true;
+  if (mOutputBuffer.Length() != FrequencyBinCount()) {
+    if (!mOutputBuffer.SetLength(FrequencyBinCount(), fallible)) {
+      return false;
+    }
+    memset(mOutputBuffer.Elements(), 0, sizeof(float) * FrequencyBinCount());
+  }
+  return result;
+}
+
+void AnalyserNode::AppendChunk(const AudioChunk& aChunk) {
+  if (mChunks.Length() == 0) {
+    return;
+  }
+
+  ++mCurrentChunk;
+  mChunks[mCurrentChunk & (CHUNK_COUNT - 1)] = aChunk;
+}
+
+// Reads into aData the oldest aLength samples of the fftSize most recent
+// samples.
+void AnalyserNode::GetTimeDomainData(float* aData, size_t aLength) {
+  size_t fftSize = FftSize();
+  MOZ_ASSERT(aLength <= fftSize);
+
+  if (mChunks.Length() == 0) {
+    PodZero(aData, aLength);
+    return;
+  }
+
+  size_t readChunk =
+      mCurrentChunk - ((fftSize - 1) >> WEBAUDIO_BLOCK_SIZE_BITS);
+  size_t readIndex = (0 - fftSize) & (WEBAUDIO_BLOCK_SIZE - 1);
+  MOZ_ASSERT(readIndex == 0 || readIndex + fftSize == WEBAUDIO_BLOCK_SIZE);
+
+  for (size_t writeIndex = 0; writeIndex < aLength;) {
+    const AudioChunk& chunk = mChunks[readChunk & (CHUNK_COUNT - 1)];
+    const size_t channelCount = chunk.ChannelCount();
+    size_t copyLength =
+        std::min<size_t>(aLength - writeIndex, WEBAUDIO_BLOCK_SIZE);
+    float* dataOut = &aData[writeIndex];
+
+    if (channelCount == 0) {
+      PodZero(dataOut, copyLength);
+    } else {
+      float scale = chunk.mVolume / channelCount;
+      {  // channel 0
+        auto channelData =
+            static_cast<const float*>(chunk.mChannelData[0]) + readIndex;
+        AudioBufferCopyWithScale(channelData, scale, dataOut, copyLength);
+      }
+      for (uint32_t i = 1; i < channelCount; ++i) {
+        auto channelData =
+            static_cast<const float*>(chunk.mChannelData[i]) + readIndex;
+        AudioBufferAddWithScale(channelData, scale, dataOut, copyLength);
+      }
+    }
+
+    readChunk++;
+    writeIndex += copyLength;
+  }
+}
+
+}  // namespace dom
+}  // namespace mozilla