Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/webrtc_audio_sink.h

-// Copyright 2016 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_H_
-
-#include <stdint.h>
-
-#include <memory>
-#include <vector>
-
-#include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/single_thread_task_runner.h"
-#include "base/synchronization/lock.h"
-#include "content/common/content_export.h"
-#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "media/base/audio_parameters.h"
-#include "media/base/audio_push_fifo.h"
-#include "third_party/webrtc/api/mediastreamtrack.h"
-#include "third_party/webrtc/media/base/audiorenderer.h"
-
-namespace content {
-
-// Provides an implementation of the MediaStreamAudioSink which re-chunks audio
-// data into the 10ms chunks required by WebRTC and then delivers the audio to
-// one or more objects implementing the webrtc::AudioTrackSinkInterface.
-//
-// The inner class, Adapter, implements the webrtc::AudioTrackInterface and
-// manages one or more "WebRTC sinks" (i.e., instances of
-// webrtc::AudioTrackSinkInterface) which are added/removed on the WebRTC
-// signaling thread.
-class CONTENT_EXPORT WebRtcAudioSink : public MediaStreamAudioSink {
- public:
-  WebRtcAudioSink(
-      const std::string& label,
-      scoped_refptr<webrtc::AudioSourceInterface> track_source,
-      scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
-
-  ~WebRtcAudioSink() override;
-
-  webrtc::AudioTrackInterface* webrtc_audio_track() const {
-    return adapter_.get();
-  }
-
-  // Set the object that provides shared access to the current audio signal
-  // level. This is passed via the Adapter to libjingle. This method may only
-  // be called once, before the audio data flow starts, and before any calls to
-  // Adapter::GetSignalLevel() might be made.
-  void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
-
-  // Set the processor that applies signal processing on the data from the
-  // source. This is passed via the Adapter to libjingle. This method may only
-  // be called once, before the audio data flow starts, and before any calls to
-  // GetAudioProcessor() might be made.
-  void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
-
-  // MediaStreamSink override.
-  void OnEnabledChanged(bool enabled) override;
-
- private:
-  // Private implementation of the webrtc::AudioTrackInterface whose control
-  // methods are all called on the WebRTC signaling thread. This class is
-  // ref-counted, per the requirements of webrtc::AudioTrackInterface.
-  class Adapter
-      : NON_EXPORTED_BASE(
-            public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
-   public:
-    Adapter(const std::string& label,
-            scoped_refptr<webrtc::AudioSourceInterface> source,
-            scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
-
-    base::SingleThreadTaskRunner* signaling_task_runner() const {
-      return signaling_task_runner_.get();
-    }
-
-    // These setters are called before the audio data flow starts, and before
-    // any methods called on the signaling thread reference these objects.
-    void set_processor(scoped_refptr<MediaStreamAudioProcessor> processor) {
-      audio_processor_ = std::move(processor);
-    }
-    void set_level(
-        scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
-      level_ = std::move(level);
-    }
-
-    // Delivers a 10ms chunk of audio to all WebRTC sinks managed by this
-    // Adapter. This is called on the audio thread.
-    void DeliverPCMToWebRtcSinks(const int16_t* audio_data,
-                                 int sample_rate,
-                                 size_t number_of_channels,
-                                 size_t number_of_frames);
-
-    // webrtc::MediaStreamTrack implementation.
-    std::string kind() const override;
-    bool set_enabled(bool enable) override;
-
-    // webrtc::AudioTrackInterface implementation.
-    void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
-    void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
-    bool GetSignalLevel(int* level) override;
-    rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
-        override;
-    webrtc::AudioSourceInterface* GetSource() const override;
-
-   protected:
-    ~Adapter() override;
-
-   private:
-    const scoped_refptr<webrtc::AudioSourceInterface> source_;
-
-    // Task runner for operations that must be done on libjingle's signaling
-    // thread.
-    const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
-
-    // Task runner used for the final de-referencing of |audio_processor_| at
-    // destruction time.
-    //
-    // TODO(miu): Remove this once MediaStreamAudioProcessor is fixed.
-    const scoped_refptr<base::SingleThreadTaskRunner> main_task_runner_;
-
-    // The audio processsor that applies audio post-processing on the source
-    // audio. This is null if there is no audio processing taking place
-    // upstream. This must be set before calls to GetAudioProcessor() are made.
-    scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
-
-    // Thread-safe accessor to current audio signal level. This may be null, if
-    // not applicable to the current use case. This must be set before calls to
-    // GetSignalLevel() are made.
-    scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
-
-    // Lock that protects concurrent access to the |sinks_| list.
-    base::Lock lock_;
-
-    // A vector of pointers to unowned WebRTC-internal objects which each
-    // receive the audio data.
-    std::vector<webrtc::AudioTrackSinkInterface*> sinks_;
-
-    DISALLOW_COPY_AND_ASSIGN(Adapter);
-  };
-
-  // MediaStreamAudioSink implementation.
-  void OnData(const media::AudioBus& audio_bus,
-              base::TimeTicks estimated_capture_time) override;
-  void OnSetFormat(const media::AudioParameters& params) override;
-
-  // Called by AudioPushFifo zero or more times during the call to OnData().
-  // Delivers audio data with the required 10ms buffer size to |adapter_|.
-  void DeliverRebufferedAudio(const media::AudioBus& audio_bus,
-                              int frame_delay);
-
-  // Owner of the WebRTC sinks. May outlive this WebRtcAudioSink (if references
-  // are held by libjingle).
-  const scoped_refptr<Adapter> adapter_;
-
-  // The current format of the audio passing through this sink.
-  media::AudioParameters params_;
-
-  // Light-weight fifo used for re-chunking audio into the 10ms chunks required
-  // by the WebRTC sinks.
-  media::AudioPushFifo fifo_;
-
-  // Buffer used for converting into the required signed 16-bit integer
-  // interleaved samples.
-  std::unique_ptr<int16_t[]> interleaved_data_;
-
-  // In debug builds, check that WebRtcAudioSink's public methods are all being
-  // called on the main render thread.
-  base::ThreadChecker thread_checker_;
-
-  // Used to DCHECK that OnSetFormat() and OnData() are called on the same
-  // thread.
-  base::ThreadChecker audio_thread_checker_;
-
-  DISALLOW_COPY_AND_ASSIGN(WebRtcAudioSink);
-};
-
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_H_