Do not pass the string device_id via IPC message to create an audio input stream

content/renderer/media/webrtc_audio_capturer.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_
 
 #include <list>
 #include <string>
 
@@ skipping 15 matching lines @@
 class WebRtcLocalAudioRenderer;
 
 // This class manages the capture data flow by getting data from its
 // |source_|, and passing it to its |sink_|.
 // It allows clients to inject their own capture data source by calling
 // SetCapturerSource().
 // 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. In addition, the
-// AudioCapturerSource::CaptureEventHandler methods are called on the IO thread
-// and requests for data to render is done on the AudioOutputDevice thread.
+// if an alternative AudioCapturerSource has been set.
 class CONTENT_EXPORT WebRtcAudioCapturer
     : public base::RefCountedThreadSafe<WebRtcAudioCapturer>,
-      NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback),
-      NON_EXPORTED_BASE(
-          public media::AudioCapturerSource::CaptureEventHandler) {
+      NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
  public:
   // Use to construct the audio capturer.
   // Called on the main render thread.
   static scoped_refptr<WebRtcAudioCapturer> CreateCapturer();
 
   // Creates and configures the default audio capturing source using the
-  // provided audio parameters.
+  // provided audio parameters,  |session_id| is used to be passed to the

[palmer, 2013/03/12 17:45:11]

NIT: Extra space again. :)

[no longer working on chromium, 2013/03/14 10:47:32]

Done.

+  // browser to decide which device to use.
   // Called on the main render thread.
-  bool Initialize(media::ChannelLayout channel_layout, int sample_rate);
+  bool Initialize(media::ChannelLayout channel_layout,
+                  int sample_rate,
+                  int session_id);
 
   // Called by the client on the sink side to add a sink.
   // 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.
   // Called on the main render thread.
   void AddCapturerSink(WebRtcAudioCapturerSink* sink);
 
   // Called by the client on the sink side to remove a sink.
   // Called on the main render thread.
@@ skipping 14 matching lines @@
   void Start();
 
   // Stops recording audio.
   // Called on the main render thread or a Libjingle working thread.
   void Stop();
 
   // Sets the microphone volume.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(double volume);
 
-  // Specifies the |session_id| to query which device to use.
-  // Called on the main render thread.
-  void SetDevice(int session_id);
-
   // Enables or disables the WebRtc AGC control.
   // Called from a Libjingle working thread.
   void SetAutomaticGainControl(bool enable);
 
   bool is_recording() const { return running_; }
 
   // Audio parameters utilized by the audio capturer. Can be utilized by
   // a local renderer to set up a renderer using identical parameters as the
   // capturer.
   // TODO(phoglund): This accessor is inherently unsafe since the returned
   // parameters can become outdated at any time. Think over the implications
   // of this accessor and if we can remove it.
   media::AudioParameters audio_parameters() const;
 
   // AudioCapturerSource::CaptureCallback implementation.
   // Called on the AudioInputDevice audio thread.
   virtual void Capture(media::AudioBus* audio_source,
                        int audio_delay_milliseconds,
                        double volume) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
-  // AudioCapturerSource::CaptureEventHandler implementation.
-  // Called on the IO thread.
-  virtual void OnDeviceStarted(const std::string& device_id) OVERRIDE;
-  virtual void OnDeviceStopped() OVERRIDE;
-
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   virtual ~WebRtcAudioCapturer();
 
  private:
   typedef std::list<WebRtcAudioCapturerSink*> SinkList;
 
   WebRtcAudioCapturer();
 
   // Reconfigures the capturer with a new buffer size and capture parameters.
   // Must be called without holding the lock. Returns true on success.
   bool Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
 
-  // Distributes information about a stopped capture device to all registered
-  // capture sinks.
-  // Runs on the main render thread.
-  void DoOnDeviceStopped();
-
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
-  // Message loop for the main render thread. Utilized in OnDeviceStopped() to
-  // ensure that OnSourceCaptureDeviceStopped() is called on the main thread
-  // instead of the originating IO thread.
-  scoped_refptr<base::MessageLoopProxy> main_loop_;
-
-  // Protects |source_|, |sinks_|, |running_|, |on_device_stopped_cb_|,
-  // |loopback_fifo_|, |params_|, |buffering_| and |agc_is_enabled_|.
+  // Protects |source_|, |sinks_|, |running_|, |loopback_fifo_|, |params_|,
+  // |buffering_| and |agc_is_enabled_|.
   mutable base::Lock lock_;
 
   // A list of sinks that the audio data is fed to.
   SinkList sinks_;
 
   // The audio data source from the browser process.
   scoped_refptr<media::AudioCapturerSource> source_;
 
   // Buffers used for temporary storage during capture callbacks.
   // Allocated during initialization.
   class ConfiguredBuffer;
   scoped_refptr<ConfiguredBuffer> buffer_;
-  std::string device_id_;
   bool running_;
 
   // True when automatic gain control is enabled, false otherwise.
   bool agc_is_enabled_;
 
+  // The media session ID used to identify which input device to be started.
+  int session_id_;

[palmer, 2013/03/12 17:45:11]

As discussed in other comments, it *might* make sense for this to be unsigned
and possibly also less wide.

[no longer working on chromium, 2013/03/14 10:47:32]

Answered in that comment, anyhow this should be done on a separate CL and all
the session_id should be changed to one type.

+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_