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_AUDIO_CAPTURER_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 
-#include <list>
-#include <string>
-
-#include "base/callback.h"
-#include "base/files/file.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
-#include "base/time/time.h"
 #include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/tagged_list.h"
-#include "media/audio/audio_input_device.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 MediaStreamAudioProcessor;
-class MediaStreamAudioSource;
-class WebRtcAudioDeviceImpl;
-class WebRtcLocalAudioRenderer;
-class WebRtcLocalAudioTrack;
+class PeerConnectionDependencyFactory;
 
-// This class manages the capture data flow by getting data from its
-// |source_|, and passing it to its |tracks_|.
-// The threading model for this class is rather complex since it will be
-// created on the main render thread, captured data is provided on a dedicated
-// AudioInputDevice thread, and methods can be called either on the Libjingle
-// thread or on the main render thread but also other client threads
-// if an alternative AudioCapturerSource has been set.
-class CONTENT_EXPORT WebRtcAudioCapturer
-    : NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
+// 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
+// WebRtcLocalAudioTracks.
+class CONTENT_EXPORT ProcessedLocalAudioSource final
+    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+      NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
  public:
-  // Used to construct the audio capturer. |render_frame_id| specifies the
-  // RenderFrame consuming audio for capture; -1 is used for tests.
-  // |device_info| contains all the device information that the capturer is
-  // created for. |constraints| contains the settings for audio processing.
-  // TODO(xians): Implement the interface for the audio source and move the
-  // |constraints| to ApplyConstraints(). Called on the main render thread.
-  static scoped_ptr<WebRtcAudioCapturer> CreateCapturer(
-      int render_frame_id,
-      const StreamDeviceInfo& device_info,
-      const blink::WebMediaConstraints& constraints,
-      WebRtcAudioDeviceImpl* audio_device,
-      MediaStreamAudioSource* audio_source);
+  // |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);
 
-  ~WebRtcAudioCapturer() override;
+  ~ProcessedLocalAudioSource() final;
 
-  // Add a audio track to the sinks of the capturer.
-  // WebRtcAudioDeviceImpl calls this method on the main render thread but
-  // other clients may call it from other threads. The current implementation
-  // does not support multi-thread calling.
-  // The first AddTrack will implicitly trigger the Start() of this object.
-  void AddTrack(WebRtcLocalAudioTrack* track);
+  // If |source| is an instance of ProcessedLocalAudioSource, return a
+  // type-casted pointer to it.  Otherwise, return null.
+  static ProcessedLocalAudioSource* From(MediaStreamAudioSource* source);
 
-  // Remove a audio track from the sinks of the capturer.
-  // If the track has been added to the capturer, it  must call RemoveTrack()
-  // before it goes away.
-  // Called on the main render thread or libjingle working thread.
-  void RemoveTrack(WebRtcLocalAudioTrack* track);
+  // 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;
+  }
 
-  // Called when a stream is connecting to a peer connection. This will set
-  // up the native buffer size for the stream in order to optimize the
-  // performance for peer connection.
-  void EnablePeerConnectionMode();
+  // Gets/Sets source constraints.  Using this is optional, but must be done
+  // before the first call to ConnectToTrack().
+  blink::WebMediaConstraints source_constraints() const { return constraints_; }

[o1ka, 2016/04/01 15:11:41]

I just don't know if it's ok from WebRTC point of view to be able to do it only
ones. Is there a possibility that they are changed dynamically during the call?

[perkj_chrome, 2016/04/08 14:05:42]

Constraints are supposed to be able to change from js
MediaStreamTrack.appy_constraints or something like that. But implementing that
will probably require tons of other changes and not in the scope of this cl. 

+  void SetSourceConstraints(const blink::WebMediaConstraints& constraints);
 
-  // Volume APIs used by WebRtcAudioDeviceImpl.
-  // Called on the AudioInputDevice audio thread.
+  // Not valid until after the source is started (when the first track is
+  // connected).
+  webrtc::AudioSourceInterface* rtc_source() const { return rtc_source_.get(); }
+
+  // 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;
 
-  const StreamDeviceInfo& device_info() const { return device_info_; }
-
-  // Stops recording audio. This method will empty its track lists since
-  // stopping the capturer will implicitly invalidate all its tracks.
-  // This method is exposed to the public because the MediaStreamAudioSource can
-  // call Stop()
-  void Stop();
-
-  // Returns the output format.
-  // Called on the main render thread.
-  media::AudioParameters GetOutputFormat() const;
-
-  // Used by clients to inject their own source to the capturer.
-  void SetCapturerSource(
-      const scoped_refptr<media::AudioCapturerSource>& source,
-      media::AudioParameters params);
-
- private:
-  class TrackOwner;
-  typedef TaggedList<TrackOwner> TrackList;
-
-  WebRtcAudioCapturer(int render_frame_id,
-                      const StreamDeviceInfo& device_info,
-                      const blink::WebMediaConstraints& constraints,
-                      WebRtcAudioDeviceImpl* audio_device,
-                      MediaStreamAudioSource* audio_source);
+ protected:
+  // MediaStreamAudioSource implementation.
+  void* GetClassIdentifier() const final;
+  scoped_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
+      const std::string& id) final;
+  bool EnsureSourceIsStarted() final;
+  void EnsureSourceIsStopped() final;
 
   // AudioCapturerSource::CaptureCallback implementation.
-  // Called on the AudioInputDevice audio thread.
+  // 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;
 
-  // Initializes the default audio capturing source using the provided render
-  // frame id and device information. Return true if success, otherwise false.
-  bool Initialize();
-
-  // SetCapturerSourceInternal() is called if the client on the source side
-  // desires to provide their own captured audio data. Client is responsible
-  // for calling Start() on its own source to get the ball rolling.
-  // Called on the main render thread.
-  // buffer_size is optional. Set to 0 to let it be chosen automatically.
-  void SetCapturerSourceInternal(
-      const scoped_refptr<media::AudioCapturerSource>& source,
-      media::ChannelLayout channel_layout,
-      int sample_rate);
-
-  // Starts recording audio.
-  // Triggered by AddSink() on the main render thread or a Libjingle working
-  // thread. It should NOT be called under |lock_|.
-  void Start();
-
-  // Helper function to get the buffer size based on |peer_connection_mode_|
-  // and sample rate;
+ private:
+  // Helper function to get the source buffer size based on whether audio
+  // processing will take place.
   int GetBufferSize(int sample_rate) const;
 
-  // Used to DCHECK that we are called on the correct thread.
+  // 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_;
 
-  // Protects |source_|, |audio_tracks_|, |running_|, |loopback_fifo_|,
-  // |params_| and |buffering_|.
-  mutable base::Lock lock_;
-
-  // A tagged list of audio tracks that the audio data is fed
-  // to. Tagged items need to be notified that the audio format has
-  // changed.
-  TrackList tracks_;
-
-  // The audio data source from the browser process.
-  scoped_refptr<media::AudioCapturerSource> source_;
-
   // 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.
-  const scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
+  scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
-  bool running_;
+  // The device created by the AudioDeviceFactory in EnsureSourceIsStarted().
+  scoped_refptr<media::AudioCapturerSource> source_;
 
-  int render_frame_id_;
+  // Holder for WebRTC audio pipeline objects.  Created in
+  // EnsureSourceIsStarted().
+  scoped_refptr<webrtc::AudioSourceInterface> rtc_source_;
 
-  // Cached information of the device used by the capturer.
-  const StreamDeviceInfo device_info_;
+  // 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_;
 
-  // Flag which affects the buffer size used by the capturer.
-  bool peer_connection_mode_;
-
-  // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
-  // of RenderThread.
-  WebRtcAudioDeviceImpl* audio_device_;
-
-  // Raw pointer to the MediaStreamAudioSource object that holds a reference
-  // to this WebRtcAudioCapturer.
-  // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and
-  // blink guarantees that the blink::WebMediaStreamSource outlives any
-  // blink::WebMediaStreamTrack connected to the source, |audio_source_| is
-  // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
-  // WebRtcAudioCapturer.
-  MediaStreamAudioSource* const audio_source_;
-
   // Used to calculate the signal level that shows in the UI.
   MediaStreamAudioLevelCalculator level_calculator_;
 
-  DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
+  bool allow_invalid_render_frame_id_for_testing_;
+
+  DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
 };
 
 }  // namespace content
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_