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

content/renderer/media/webrtc/processed_local_audio_source.h

-// Copyright (c) 2012 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_PROCESSED_LOCAL_AUDIO_SOURCE_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
-
-#include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/synchronization/lock.h"
-#include "content/common/media/media_stream_options.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/media_stream_audio_source.h"
-#include "media/base/audio_capturer_source.h"
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-
-namespace media {
-class AudioBus;
-}
-
-namespace webrtc {
-class AudioSourceInterface;
-}
-
-namespace content {
-
-class PeerConnectionDependencyFactory;
-
-// Represents a local source of audio data that is routed through the WebRTC
-// audio pipeline for post-processing (e.g., for echo cancellation during a
-// video conferencing call). Owns a media::AudioCapturerSource and the
-// MediaStreamProcessor that modifies its audio. Modified audio is delivered to
-// one or more MediaStreamAudioTracks.
-class CONTENT_EXPORT ProcessedLocalAudioSource final
-    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
-      NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
- public:
-  // |consumer_render_frame_id| references the RenderFrame that will consume the
-  // audio data. Audio parameters and (optionally) a pre-existing audio session
-  // ID are derived from |device_info|. |factory| must outlive this instance.
-  ProcessedLocalAudioSource(int consumer_render_frame_id,
-                            const StreamDeviceInfo& device_info,
-                            PeerConnectionDependencyFactory* factory);
-
-  ~ProcessedLocalAudioSource() final;
-
-  // If |source| is an instance of ProcessedLocalAudioSource, return a
-  // type-casted pointer to it. Otherwise, return null.
-  static ProcessedLocalAudioSource* From(MediaStreamAudioSource* source);
-
-  // Non-browser unit tests cannot provide RenderFrame implementations at
-  // run-time. This is used to skip the otherwise mandatory check for a valid
-  // render frame ID when the source is started.
-  void SetAllowInvalidRenderFrameIdForTesting(bool allowed) {
-    allow_invalid_render_frame_id_for_testing_ = allowed;
-  }
-
-  // Gets/Sets source constraints. Using this is optional, but must be done
-  // before the first call to ConnectToTrack().
-  const blink::WebMediaConstraints& source_constraints() const {
-    return constraints_;
-  }
-  void SetSourceConstraints(const blink::WebMediaConstraints& constraints);
-
-  // The following accessors are not valid until after the source is started
-  // (when the first track is connected).
-  webrtc::AudioSourceInterface* rtc_source() const { return rtc_source_.get(); }
-  const scoped_refptr<MediaStreamAudioProcessor>& audio_processor() const {
-    return audio_processor_;
-  }
-  const scoped_refptr<MediaStreamAudioLevelCalculator::Level>& audio_level()
-      const {
-    return level_calculator_.level();
-  }
-
-  // Thread-safe volume accessors used by WebRtcAudioDeviceImpl.
-  void SetVolume(int volume);
-  int Volume() const;
-  int MaxVolume() const;
-
-  // Audio parameters utilized by the source of the audio capturer.
-  // TODO(phoglund): Think over the implications of this accessor and if we can
-  // remove it.
-  media::AudioParameters GetInputFormat() const;
-
- protected:
-  // MediaStreamAudioSource implementation.
-  void* GetClassIdentifier() const final;
-  bool EnsureSourceIsStarted() final;
-  void EnsureSourceIsStopped() final;
-
-  // AudioCapturerSource::CaptureCallback implementation.
-  // Called on the AudioCapturerSource audio thread.
-  void Capture(const media::AudioBus* audio_source,
-               int audio_delay_milliseconds,
-               double volume,
-               bool key_pressed) override;
-  void OnCaptureError(const std::string& message) override;
-
- private:
-  // Helper function to get the source buffer size based on whether audio
-  // processing will take place.
-  int GetBufferSize(int sample_rate) const;
-
-  // The RenderFrame that will consume the audio data. Used when creating
-  // AudioCapturerSources.
-  const int consumer_render_frame_id_;
-
-  PeerConnectionDependencyFactory* const pc_factory_;
-
-  // In debug builds, check that all methods that could cause object graph
-  // or data flow changes are being called on the main thread.
-  base::ThreadChecker thread_checker_;
-
-  // Cached audio constraints for the capturer.
-  blink::WebMediaConstraints constraints_;
-
-  // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
-  // data is in a unit of 10 ms data chunk.
-  scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
-
-  // The device created by the AudioDeviceFactory in EnsureSourceIsStarted().
-  scoped_refptr<media::AudioCapturerSource> source_;
-
-  // Holder for WebRTC audio pipeline objects. Created in
-  // EnsureSourceIsStarted().
-  scoped_refptr<webrtc::AudioSourceInterface> rtc_source_;
-
-  // Protects data elements from concurrent access when using the volume
-  // methods.
-  mutable base::Lock volume_lock_;
-
-  // Stores latest microphone volume received in a CaptureData() callback.
-  // Range is [0, 255].
-  int volume_;
-
-  // Used to calculate the signal level that shows in the UI.
-  MediaStreamAudioLevelCalculator level_calculator_;
-
-  bool allow_invalid_render_frame_id_for_testing_;
-
-  DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
-};
-
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_