Adding more comments to every header file in api/ subdirectory.

webrtc/api/mediastreaminterface.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 35 matching lines @@
 };
 
 class NotifierInterface {
  public:
   virtual void RegisterObserver(ObserverInterface* observer) = 0;
   virtual void UnregisterObserver(ObserverInterface* observer) = 0;
 
   virtual ~NotifierInterface() {}
 };
 
-// Base class for sources. A MediaStreamTrack have an underlying source that
-// provide media. A source can be shared with multiple tracks.
+// Base class for sources. A MediaStreamTrack has an underlying source that
+// provides media. A source can be shared by multiple tracks.
 class MediaSourceInterface : public rtc::RefCountInterface,
                              public NotifierInterface {
  public:
   enum SourceState {
     kInitializing,
     kLive,
     kEnded,
     kMuted
   };
 
   virtual SourceState state() const = 0;
 
   virtual bool remote() const = 0;
 
  protected:
   virtual ~MediaSourceInterface() {}
 };
 
-// Information about a track.
+// C++ version of MediaStreamTrack.
+// See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack
 class MediaStreamTrackInterface : public rtc::RefCountInterface,
                                   public NotifierInterface {
  public:
   enum TrackState {
     kLive,
     kEnded,
   };
 
   static const char kAudioKind[];
   static const char kVideoKind[];
 
   // The kind() method must return kAudioKind only if the object is a
   // subclass of AudioTrackInterface, and kVideoKind only if the
   // object is a subclass of VideoTrackInterface. It is typically used
   // to protect a static_cast<> to the corresponding subclass.
   virtual std::string kind() const = 0;
+
+  // Track identifier.
   virtual std::string id() const = 0;
+
+  // A disabled track will produce silence (if audio) or black frames (if
+  // video). Can be disabled and re-enabled.
   virtual bool enabled() const = 0;
+  virtual bool set_enabled(bool enable) = 0;
+
+  // Live or ended. A track will never be live again after becoming ended.
   virtual TrackState state() const = 0;
-  virtual bool set_enabled(bool enable) = 0;
 
  protected:
   virtual ~MediaStreamTrackInterface() {}
 };
 
-// VideoTrackSourceInterface is a reference counted source used for VideoTracks.
-// The same source can be used in multiple VideoTracks.
+// VideoTrackSourceInterface is a reference counted source used for
+// VideoTracks. The same source can be used by multiple VideoTracks.
 class VideoTrackSourceInterface
     : public MediaSourceInterface,
       public rtc::VideoSourceInterface<VideoFrame> {
  public:
   struct Stats {
     // Original size of captured frame, before video adaptation.
     int input_width;
     int input_height;
   };
 
   // Indicates that parameters suitable for screencasts should be automatically
   // applied to RtpSenders.
   // TODO(perkj): Remove these once all known applications have moved to
-  // explicitly setting suitable parameters for screencasts and dont' need this
+  // explicitly setting suitable parameters for screencasts and don't need this
   // implicit behavior.
   virtual bool is_screencast() const = 0;
 
   // Indicates that the encoder should denoise video before encoding it.
   // If it is not set, the default configuration is used which is different
   // depending on video codec.
   // TODO(perkj): Remove this once denoising is done by the source, and not by
   // the encoder.
   virtual rtc::Optional<bool> needs_denoising() const = 0;
 
-  // Returns false if no stats are available, e.g, for a remote
-  // source, or a source which has not seen its first frame yet.
-  // Should avoid blocking.
+  // Returns false if no stats are available, e.g, for a remote source, or a
+  // source which has not seen its first frame yet.
+  //
+  // Implementation should avoid blocking.
   virtual bool GetStats(Stats* stats) = 0;
 
  protected:
   virtual ~VideoTrackSourceInterface() {}
 };
 
 class VideoTrackInterface
     : public MediaStreamTrackInterface,
       public rtc::VideoSourceInterface<VideoFrame> {
  public:
   // Video track content hint, used to override the source is_screencast
   // property.
   // See https://crbug.com/653531 and https://github.com/WICG/mst-content-hint.
   enum class ContentHint { kNone, kFluid, kDetailed };
 
-  // Register a video sink for this track.
+  // Register a video sink for this track. Used to connect the track to the
+  // underlying video engine.
   void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
                        const rtc::VideoSinkWants& wants) override {}
   void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
 
   virtual VideoTrackSourceInterface* GetSource() const = 0;
 
   virtual ContentHint content_hint() const { return ContentHint::kNone; }
   virtual void set_content_hint(ContentHint hint) {}
 
  protected:
   virtual ~VideoTrackInterface() {}
 };
 
 // Interface for receiving audio data from a AudioTrack.
 class AudioTrackSinkInterface {
  public:
   virtual void OnData(const void* audio_data,
                       int bits_per_sample,
                       int sample_rate,
                       size_t number_of_channels,
                       size_t number_of_frames) = 0;
 
  protected:
   virtual ~AudioTrackSinkInterface() {}
 };
 
 // AudioSourceInterface is a reference counted source used for AudioTracks.
-// The same source can be used in multiple AudioTracks.
+// The same source can be used by multiple AudioTracks.
 class AudioSourceInterface : public MediaSourceInterface {
  public:
   class AudioObserver {
    public:
     virtual void OnSetVolume(double volume) = 0;
 
    protected:
     virtual ~AudioObserver() {}
   };
 
-  // TODO(xians): Makes all the interface pure virtual after Chrome has their
-  // implementations.
-  // Sets the volume to the source. |volume| is in  the range of [0, 10].
+  // TODO(deadbeef): Makes all the interfaces pure virtual after they're
+  // implemented in chromium.
+
+  // Sets the volume of the source. |volume| is in  the range of [0, 10].
   // TODO(tommi): This method should be on the track and ideally volume should
   // be applied in the track in a way that does not affect clones of the track.
   virtual void SetVolume(double volume) {}
 
-  // Registers/unregisters observer to the audio source.
+  // Registers/unregisters observers to the audio source.
   virtual void RegisterAudioObserver(AudioObserver* observer) {}
   virtual void UnregisterAudioObserver(AudioObserver* observer) {}
 
   // TODO(tommi): Make pure virtual.
   virtual void AddSink(AudioTrackSinkInterface* sink) {}
   virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
 };
 
 // Interface of the audio processor used by the audio track to collect
 // statistics.
@@ skipping 25 matching lines @@
 
   // Get audio processor statistics.
   virtual void GetStats(AudioProcessorStats* stats) = 0;
 
  protected:
   virtual ~AudioProcessorInterface() {}
 };
 
 class AudioTrackInterface : public MediaStreamTrackInterface {
  public:
-  // TODO(xians): Figure out if the following interface should be const or not.
+  // TODO(deadbeef): Figure out if the following interface should be const or
+  // not.
   virtual AudioSourceInterface* GetSource() const =  0;
 
   // Add/Remove a sink that will receive the audio data from the track.
   virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
   virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
 
   // Get the signal level from the audio track.
   // Return true on success, otherwise false.
-  // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
-  // after Chrome has the correct implementation of the interface.
+  // TODO(deadbeef): Change the interface to int GetSignalLevel() and pure
+  // virtual after it's implemented in chromium.
   virtual bool GetSignalLevel(int* level) { return false; }
 
   // Get the audio processor used by the audio track. Return NULL if the track
   // does not have any processor.
-  // TODO(xians): Make the interface pure virtual.
-  virtual rtc::scoped_refptr<AudioProcessorInterface>
-      GetAudioProcessor() { return NULL; }
+  // TODO(deadbeef): Make the interface pure virtual.
+  virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() {
+    return nullptr;
+  }
 
  protected:
   virtual ~AudioTrackInterface() {}
 };
 
 typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
     AudioTrackVector;
 typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
     VideoTrackVector;
 
+// C++ version of https://www.w3.org/TR/mediacapture-streams/#mediastream.
+//
+// A major difference is that remote audio/video tracks (received by a
+// PeerConnection/RtpReceiver) are not synchronized simply by adding them to
+// the same stream; a session description with the correct "a=msid" attributes
+// must be pushed down.
+//
+// Thus, this interface acts as simply a container for tracks.
 class MediaStreamInterface : public rtc::RefCountInterface,
                              public NotifierInterface {
  public:
   virtual std::string label() const = 0;
 
   virtual AudioTrackVector GetAudioTracks() = 0;
   virtual VideoTrackVector GetVideoTracks() = 0;
   virtual rtc::scoped_refptr<AudioTrackInterface>
       FindAudioTrack(const std::string& track_id) = 0;
   virtual rtc::scoped_refptr<VideoTrackInterface>
       FindVideoTrack(const std::string& track_id) = 0;
 
   virtual bool AddTrack(AudioTrackInterface* track) = 0;
   virtual bool AddTrack(VideoTrackInterface* track) = 0;
   virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
   virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
 
  protected:
   virtual ~MediaStreamInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_MEDIASTREAMINTERFACE_H_