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

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index 960b4e02c4db8104028ad8161fff7312781f97a8..38990d7f3c7e1ee342eaead491d7efe576fa9f25 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -33,23 +33,22 @@ class WebRtcLocalAudioRenderer;
 // 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
@@ -84,10 +83,6 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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);
@@ -109,11 +104,6 @@ class CONTENT_EXPORT WebRtcAudioCapturer
                        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();
@@ -127,21 +117,11 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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.
@@ -154,12 +134,14 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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);
 };