Add m-esr52 at 52.6.0

1 files changed, 429 insertions, 0 deletions

diff --git a/dom/media/AudioSegment.h b/dom/media/AudioSegment.h
new file mode 100644
index 000000000..13f6c2e48
--- /dev/null
+++ b/dom/media/AudioSegment.h
@@ -0,0 +1,429 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* 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/. */
+
+#ifndef MOZILLA_AUDIOSEGMENT_H_
+#define MOZILLA_AUDIOSEGMENT_H_
+
+#include "MediaSegment.h"
+#include "AudioSampleFormat.h"
+#include "AudioChannelFormat.h"
+#include "SharedBuffer.h"
+#include "WebAudioUtils.h"
+#ifdef MOZILLA_INTERNAL_API
+#include "mozilla/TimeStamp.h"
+#endif
+#include <float.h>
+
+namespace mozilla {
+
+template<typename T>
+class SharedChannelArrayBuffer : public ThreadSharedObject {
+public:
+  explicit SharedChannelArrayBuffer(nsTArray<nsTArray<T> >* aBuffers)
+  {
+    mBuffers.SwapElements(*aBuffers);
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = 0;
+    amount += mBuffers.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < mBuffers.Length(); i++) {
+      amount += mBuffers[i].ShallowSizeOfExcludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  nsTArray<nsTArray<T> > mBuffers;
+};
+
+class AudioMixer;
+
+/**
+ * For auto-arrays etc, guess this as the common number of channels.
+ */
+const int GUESS_AUDIO_CHANNELS = 2;
+
+// We ensure that the graph advances in steps that are multiples of the Web
+// Audio block size
+const uint32_t WEBAUDIO_BLOCK_SIZE_BITS = 7;
+const uint32_t WEBAUDIO_BLOCK_SIZE = 1 << WEBAUDIO_BLOCK_SIZE_BITS;
+
+template <typename SrcT, typename DestT>
+static void
+InterleaveAndConvertBuffer(const SrcT* const* aSourceChannels,
+                           uint32_t aLength, float aVolume,
+                           uint32_t aChannels,
+                           DestT* aOutput)
+{
+  DestT* output = aOutput;
+  for (size_t i = 0; i < aLength; ++i) {
+    for (size_t channel = 0; channel < aChannels; ++channel) {
+      float v = AudioSampleToFloat(aSourceChannels[channel][i])*aVolume;
+      *output = FloatToAudioSample<DestT>(v);
+      ++output;
+    }
+  }
+}
+
+template <typename SrcT, typename DestT>
+static void
+DeinterleaveAndConvertBuffer(const SrcT* aSourceBuffer,
+                             uint32_t aFrames, uint32_t aChannels,
+                             DestT** aOutput)
+{
+  for (size_t i = 0; i < aChannels; i++) {
+    size_t interleavedIndex = i;
+    for (size_t j = 0; j < aFrames; j++) {
+      ConvertAudioSample(aSourceBuffer[interleavedIndex],
+                         aOutput[i][j]);
+      interleavedIndex += aChannels;
+    }
+  }
+}
+
+class SilentChannel
+{
+public:
+  static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
+  static const uint8_t gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*AUDIO_PROCESSING_FRAMES];
+  // We take advantage of the fact that zero in float and zero in int have the
+  // same all-zeros bit layout.
+  template<typename T>
+  static const T* ZeroChannel();
+};
+
+
+/**
+ * Given an array of input channels (aChannelData), downmix to aOutputChannels,
+ * interleave the channel data. A total of aOutputChannels*aDuration
+ * interleaved samples will be copied to a channel buffer in aOutput.
+ */
+template <typename SrcT, typename DestT>
+void
+DownmixAndInterleave(const nsTArray<const SrcT*>& aChannelData,
+                     int32_t aDuration, float aVolume, uint32_t aOutputChannels,
+                     DestT* aOutput)
+{
+
+  if (aChannelData.Length() == aOutputChannels) {
+    InterleaveAndConvertBuffer(aChannelData.Elements(),
+                               aDuration, aVolume, aOutputChannels, aOutput);
+  } else {
+    AutoTArray<SrcT*,GUESS_AUDIO_CHANNELS> outputChannelData;
+    AutoTArray<SrcT, SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS> outputBuffers;
+    outputChannelData.SetLength(aOutputChannels);
+    outputBuffers.SetLength(aDuration * aOutputChannels);
+    for (uint32_t i = 0; i < aOutputChannels; i++) {
+      outputChannelData[i] = outputBuffers.Elements() + aDuration * i;
+    }
+    AudioChannelsDownMix(aChannelData,
+                         outputChannelData.Elements(),
+                         aOutputChannels,
+                         aDuration);
+    InterleaveAndConvertBuffer(outputChannelData.Elements(),
+                               aDuration, aVolume, aOutputChannels, aOutput);
+  }
+}
+
+/**
+ * An AudioChunk represents a multi-channel buffer of audio samples.
+ * It references an underlying ThreadSharedObject which manages the lifetime
+ * of the buffer. An AudioChunk maintains its own duration and channel data
+ * pointers so it can represent a subinterval of a buffer without copying.
+ * An AudioChunk can store its individual channels anywhere; it maintains
+ * separate pointers to each channel's buffer.
+ */
+struct AudioChunk {
+  typedef mozilla::AudioSampleFormat SampleFormat;
+
+  AudioChunk() : mPrincipalHandle(PRINCIPAL_HANDLE_NONE) {}
+
+  // Generic methods
+  void SliceTo(StreamTime aStart, StreamTime aEnd)
+  {
+    MOZ_ASSERT(aStart >= 0 && aStart < aEnd && aEnd <= mDuration,
+               "Slice out of bounds");
+    if (mBuffer) {
+      MOZ_ASSERT(aStart < INT32_MAX, "Can't slice beyond 32-bit sample lengths");
+      for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
+        mChannelData[channel] = AddAudioSampleOffset(mChannelData[channel],
+            mBufferFormat, int32_t(aStart));
+      }
+    }
+    mDuration = aEnd - aStart;
+  }
+  StreamTime GetDuration() const { return mDuration; }
+  bool CanCombineWithFollowing(const AudioChunk& aOther) const
+  {
+    if (aOther.mBuffer != mBuffer) {
+      return false;
+    }
+    if (mBuffer) {
+      NS_ASSERTION(aOther.mBufferFormat == mBufferFormat,
+                   "Wrong metadata about buffer");
+      NS_ASSERTION(aOther.mChannelData.Length() == mChannelData.Length(),
+                   "Mismatched channel count");
+      if (mDuration > INT32_MAX) {
+        return false;
+      }
+      for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
+        if (aOther.mChannelData[channel] != AddAudioSampleOffset(mChannelData[channel],
+            mBufferFormat, int32_t(mDuration))) {
+          return false;
+        }
+      }
+    }
+    return true;
+  }
+  bool IsNull() const { return mBuffer == nullptr; }
+  void SetNull(StreamTime aDuration)
+  {
+    mBuffer = nullptr;
+    mChannelData.Clear();
+    mDuration = aDuration;
+    mVolume = 1.0f;
+    mBufferFormat = AUDIO_FORMAT_SILENCE;
+    mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
+  }
+
+  size_t ChannelCount() const { return mChannelData.Length(); }
+
+  bool IsMuted() const { return mVolume == 0.0f; }
+
+  size_t SizeOfExcludingThisIfUnshared(MallocSizeOf aMallocSizeOf) const
+  {
+    return SizeOfExcludingThis(aMallocSizeOf, true);
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf, bool aUnshared) const
+  {
+    size_t amount = 0;
+
+    // Possibly owned:
+    // - mBuffer - Can hold data that is also in the decoded audio queue. If it
+    //             is not shared, or unshared == false it gets counted.
+    if (mBuffer && (!aUnshared || !mBuffer->IsShared())) {
+      amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    // Memory in the array is owned by mBuffer.
+    amount += mChannelData.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  template<typename T>
+  const nsTArray<const T*>& ChannelData()
+  {
+    MOZ_ASSERT(AudioSampleTypeToFormat<T>::Format == mBufferFormat);
+    return *reinterpret_cast<nsTArray<const T*>*>(&mChannelData);
+  }
+
+  PrincipalHandle GetPrincipalHandle() const { return mPrincipalHandle; }
+
+  StreamTime mDuration; // in frames within the buffer
+  RefPtr<ThreadSharedObject> mBuffer; // the buffer object whose lifetime is managed; null means data is all zeroes
+  nsTArray<const void*> mChannelData; // one pointer per channel; empty if and only if mBuffer is null
+  float mVolume; // volume multiplier to apply (1.0f if mBuffer is nonnull)
+  SampleFormat mBufferFormat; // format of frames in mBuffer (only meaningful if mBuffer is nonnull)
+#ifdef MOZILLA_INTERNAL_API
+  mozilla::TimeStamp mTimeStamp;           // time at which this has been fetched from the MediaEngine
+#endif
+  // principalHandle for the data in this chunk.
+  // This can be compared to an nsIPrincipal* when back on main thread.
+  PrincipalHandle mPrincipalHandle;
+};
+
+/**
+ * A list of audio samples consisting of a sequence of slices of SharedBuffers.
+ * The audio rate is determined by the track, not stored in this class.
+ */
+class AudioSegment : public MediaSegmentBase<AudioSegment, AudioChunk> {
+public:
+  typedef mozilla::AudioSampleFormat SampleFormat;
+
+  AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO) {}
+
+  // Resample the whole segment in place.
+  template<typename T>
+  void Resample(SpeexResamplerState* aResampler, uint32_t aInRate, uint32_t aOutRate)
+  {
+    mDuration = 0;
+#ifdef DEBUG
+    uint32_t segmentChannelCount = ChannelCount();
+#endif
+
+    for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
+      AutoTArray<nsTArray<T>, GUESS_AUDIO_CHANNELS> output;
+      AutoTArray<const T*, GUESS_AUDIO_CHANNELS> bufferPtrs;
+      AudioChunk& c = *ci;
+      // If this chunk is null, don't bother resampling, just alter its duration
+      if (c.IsNull()) {
+        c.mDuration = (c.mDuration * aOutRate) / aInRate;
+        mDuration += c.mDuration;
+        continue;
+      }
+      uint32_t channels = c.mChannelData.Length();
+      MOZ_ASSERT(channels == segmentChannelCount);
+      output.SetLength(channels);
+      bufferPtrs.SetLength(channels);
+      uint32_t inFrames = c.mDuration;
+      // Round up to allocate; the last frame may not be used.
+      NS_ASSERTION((UINT32_MAX - aInRate + 1) / c.mDuration >= aOutRate,
+                   "Dropping samples");
+      uint32_t outSize = (c.mDuration * aOutRate + aInRate - 1) / aInRate;
+      for (uint32_t i = 0; i < channels; i++) {
+        T* out = output[i].AppendElements(outSize);
+        uint32_t outFrames = outSize;
+
+        const T* in = static_cast<const T*>(c.mChannelData[i]);
+        dom::WebAudioUtils::SpeexResamplerProcess(aResampler, i,
+                                                  in, &inFrames,
+                                                  out, &outFrames);
+        MOZ_ASSERT(inFrames == c.mDuration);
+
+        bufferPtrs[i] = out;
+        output[i].SetLength(outFrames);
+      }
+      MOZ_ASSERT(channels > 0);
+      c.mDuration = output[0].Length();
+      c.mBuffer = new mozilla::SharedChannelArrayBuffer<T>(&output);
+      for (uint32_t i = 0; i < channels; i++) {
+        c.mChannelData[i] = bufferPtrs[i];
+      }
+      mDuration += c.mDuration;
+    }
+  }
+
+  void ResampleChunks(SpeexResamplerState* aResampler,
+                      uint32_t aInRate,
+                      uint32_t aOutRate);
+
+  void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
+                    const nsTArray<const float*>& aChannelData,
+                    int32_t aDuration, const PrincipalHandle& aPrincipalHandle)
+  {
+    AudioChunk* chunk = AppendChunk(aDuration);
+    chunk->mBuffer = aBuffer;
+    for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
+      chunk->mChannelData.AppendElement(aChannelData[channel]);
+    }
+    chunk->mVolume = 1.0f;
+    chunk->mBufferFormat = AUDIO_FORMAT_FLOAT32;
+#ifdef MOZILLA_INTERNAL_API
+    chunk->mTimeStamp = TimeStamp::Now();
+#endif
+    chunk->mPrincipalHandle = aPrincipalHandle;
+  }
+  void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
+                    const nsTArray<const int16_t*>& aChannelData,
+                    int32_t aDuration, const PrincipalHandle& aPrincipalHandle)
+  {
+    AudioChunk* chunk = AppendChunk(aDuration);
+    chunk->mBuffer = aBuffer;
+    for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
+      chunk->mChannelData.AppendElement(aChannelData[channel]);
+    }
+    chunk->mVolume = 1.0f;
+    chunk->mBufferFormat = AUDIO_FORMAT_S16;
+#ifdef MOZILLA_INTERNAL_API
+    chunk->mTimeStamp = TimeStamp::Now();
+#endif
+    chunk->mPrincipalHandle = aPrincipalHandle;
+  }
+  // Consumes aChunk, and returns a pointer to the persistent copy of aChunk
+  // in the segment.
+  AudioChunk* AppendAndConsumeChunk(AudioChunk* aChunk)
+  {
+    AudioChunk* chunk = AppendChunk(aChunk->mDuration);
+    chunk->mBuffer = aChunk->mBuffer.forget();
+    chunk->mChannelData.SwapElements(aChunk->mChannelData);
+    chunk->mVolume = aChunk->mVolume;
+    chunk->mBufferFormat = aChunk->mBufferFormat;
+#ifdef MOZILLA_INTERNAL_API
+    chunk->mTimeStamp = TimeStamp::Now();
+#endif
+    chunk->mPrincipalHandle = aChunk->mPrincipalHandle;
+    return chunk;
+  }
+  void ApplyVolume(float aVolume);
+  // Mix the segment into a mixer, interleaved. This is useful to output a
+  // segment to a system audio callback. It up or down mixes to aChannelCount
+  // channels.
+  void WriteTo(uint64_t aID, AudioMixer& aMixer, uint32_t aChannelCount,
+               uint32_t aSampleRate);
+  // Mix the segment into a mixer, keeping it planar, up or down mixing to
+  // aChannelCount channels.
+  void Mix(AudioMixer& aMixer, uint32_t aChannelCount, uint32_t aSampleRate);
+
+  int ChannelCount() {
+    NS_WARNING_ASSERTION(
+      !mChunks.IsEmpty(),
+      "Cannot query channel count on a AudioSegment with no chunks.");
+    // Find the first chunk that has non-zero channels. A chunk that hs zero
+    // channels is just silence and we can simply discard it.
+    for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
+      if (ci->ChannelCount()) {
+        return ci->ChannelCount();
+      }
+    }
+    return 0;
+  }
+
+  bool IsNull() const {
+    for (ChunkIterator ci(*const_cast<AudioSegment*>(this)); !ci.IsEnded();
+         ci.Next()) {
+      if (!ci->IsNull()) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  static Type StaticType() { return AUDIO; }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+};
+
+template<typename SrcT>
+void WriteChunk(AudioChunk& aChunk,
+                uint32_t aOutputChannels,
+                AudioDataValue* aOutputBuffer)
+{
+  AutoTArray<const SrcT*,GUESS_AUDIO_CHANNELS> channelData;
+
+  channelData = aChunk.ChannelData<SrcT>();
+
+  if (channelData.Length() < aOutputChannels) {
+    // Up-mix. Note that this might actually make channelData have more
+    // than aOutputChannels temporarily.
+    AudioChannelsUpMix(&channelData, aOutputChannels, SilentChannel::ZeroChannel<SrcT>());
+  }
+  if (channelData.Length() > aOutputChannels) {
+    // Down-mix.
+    DownmixAndInterleave(channelData, aChunk.mDuration,
+        aChunk.mVolume, aOutputChannels, aOutputBuffer);
+  } else {
+    InterleaveAndConvertBuffer(channelData.Elements(),
+        aChunk.mDuration, aChunk.mVolume,
+        aOutputChannels,
+        aOutputBuffer);
+  }
+}
+
+
+
+} // namespace mozilla
+
+#endif /* MOZILLA_AUDIOSEGMENT_H_ */