BASELINE: Update Chromium to 67.0.3396.47

Change-Id: Idcb1341782e417561a2473eeecc82642dafda5b7
Reviewed-by: Michal Klocek <michal.klocek@qt.io>

1 files changed, 187 insertions, 0 deletions

diff --git a/chromium/third_party/blink/renderer/platform/audio/audio_bus.h b/chromium/third_party/blink/renderer/platform/audio/audio_bus.h
new file mode 100644
index 00000000000..0d29198b79d
--- /dev/null
+++ b/chromium/third_party/blink/renderer/platform/audio/audio_bus.h
@@ -0,0 +1,187 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_BUS_H_
+#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_BUS_H_
+
+#include <memory>
+#include "third_party/blink/renderer/platform/audio/audio_channel.h"
+#include "third_party/blink/renderer/platform/wtf/noncopyable.h"
+#include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
+#include "third_party/blink/renderer/platform/wtf/vector.h"
+
+namespace blink {
+
+// An AudioBus represents a collection of one or more AudioChannels.
+// The data layout is "planar" as opposed to "interleaved".  An AudioBus with
+// one channel is mono, an AudioBus with two channels is stereo, etc.
+class PLATFORM_EXPORT AudioBus : public ThreadSafeRefCounted<AudioBus> {
+  WTF_MAKE_NONCOPYABLE(AudioBus);
+
+ public:
+  enum {
+    kChannelLeft = 0,
+    kChannelRight = 1,
+    kChannelCenter = 2,  // center and mono are the same
+    kChannelMono = 2,
+    kChannelLFE = 3,
+    kChannelSurroundLeft = 4,
+    kChannelSurroundRight = 5,
+  };
+
+  enum {
+    kLayoutCanonical = 0
+    // Can define non-standard layouts here
+  };
+
+  enum ChannelInterpretation {
+    kSpeakers,
+    kDiscrete,
+  };
+
+  // allocate indicates whether or not to initially have the AudioChannels
+  // created with managed storage.  Normal usage is to pass true here, in which
+  // case the AudioChannels will memory-manage their own storage.  If allocate
+  // is false then setChannelMemory() has to be called later on for each
+  // channel before the AudioBus is useable...
+  static scoped_refptr<AudioBus> Create(unsigned number_of_channels,
+                                        size_t length,
+                                        bool allocate = true);
+
+  // Tells the given channel to use an externally allocated buffer.
+  void SetChannelMemory(unsigned channel_index, float* storage, size_t length);
+
+  // Channels
+  unsigned NumberOfChannels() const { return channels_.size(); }
+
+  AudioChannel* Channel(unsigned channel) { return channels_[channel].get(); }
+  const AudioChannel* Channel(unsigned channel) const {
+    return channels_[channel].get();
+  }
+  AudioChannel* ChannelByType(unsigned type);
+  const AudioChannel* ChannelByType(unsigned type) const;
+
+  // Number of sample-frames
+  size_t length() const { return length_; }
+
+  // resizeSmaller() can only be called with a new length <= the current length.
+  // The data stored in the bus will remain undisturbed.
+  void ResizeSmaller(size_t new_length);
+
+  // Sample-rate : 0.0 if unknown or "don't care"
+  float SampleRate() const { return sample_rate_; }
+  void SetSampleRate(float sample_rate) { sample_rate_ = sample_rate; }
+
+  // Zeroes all channels.
+  void Zero();
+
+  // Clears the silent flag on all channels.
+  void ClearSilentFlag();
+
+  // Returns true if the silent bit is set on all channels.
+  bool IsSilent() const;
+
+  // Returns true if the channel count and frame-size match.
+  bool TopologyMatches(const AudioBus& source_bus) const;
+
+  // Creates a new buffer from a range in the source buffer.
+  // 0 may be returned if the range does not fit in the sourceBuffer
+  static scoped_refptr<AudioBus> CreateBufferFromRange(
+      const AudioBus* source_buffer,
+      unsigned start_frame,
+      unsigned end_frame);
+
+  // Creates a new AudioBus by sample-rate converting sourceBus to the
+  // newSampleRate.
+  // setSampleRate() must have been previously called on sourceBus.
+  // Note: sample-rate conversion is already handled in the file-reading code
+  // for the mac port, so we don't need this.
+  static scoped_refptr<AudioBus> CreateBySampleRateConverting(
+      const AudioBus* source_bus,
+      bool mix_to_mono,
+      double new_sample_rate);
+
+  // Creates a new AudioBus by mixing all the channels down to mono.
+  // If sourceBus is already mono, then the returned AudioBus will simply be a
+  // copy.
+  static scoped_refptr<AudioBus> CreateByMixingToMono(
+      const AudioBus* source_bus);
+
+  // Scales all samples by the same amount.
+  void Scale(float scale);
+
+  // Copies the samples from the source bus to this one.
+  // This is just a simple per-channel copy if the number of channels match,
+  // otherwise an up-mix or down-mix is done.
+  void CopyFrom(const AudioBus& source_bus, ChannelInterpretation = kSpeakers);
+
+  // Sums the samples from the source bus to this one.
+  // This is just a simple per-channel summing if the number of channels match,
+  // otherwise an up-mix or down-mix is done.
+  void SumFrom(const AudioBus& source_bus, ChannelInterpretation = kSpeakers);
+
+  // Copy each channel from |source_bus| into our corresponding channel.  We
+  // scale |source_bus| by |gain| before copying into the bus.
+  void CopyWithGainFrom(const AudioBus& source_bus, float gain);
+
+  // Copies the sourceBus by scaling with sample-accurate gain values.
+  void CopyWithSampleAccurateGainValuesFrom(const AudioBus& source_bus,
+                                            float* gain_values,
+                                            unsigned number_of_gain_values);
+
+  // Returns maximum absolute value across all channels (useful for
+  // normalization).
+  float MaxAbsValue() const;
+
+  // Makes maximum absolute value == 1.0 (if possible).
+  void Normalize();
+
+  static scoped_refptr<AudioBus> GetDataResource(const char* name,
+                                                 float sample_rate);
+
+ protected:
+  AudioBus() = default;
+
+  AudioBus(unsigned number_of_channels, size_t length, bool allocate);
+
+  void DiscreteSumFrom(const AudioBus&);
+
+  // Up/down-mix by in-place summing upon the existing channel content.
+  // http://webaudio.github.io/web-audio-api/#channel-up-mixing-and-down-mixing
+  void SumFromByUpMixing(const AudioBus&);
+  void SumFromByDownMixing(const AudioBus&);
+
+  size_t length_;
+  Vector<std::unique_ptr<AudioChannel>> channels_;
+  int layout_;
+  float sample_rate_;  // 0.0 if unknown or N/A
+};
+
+}  // namespace blink
+
+#endif  // THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_BUS_H_