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

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 26307b9f95eff204241a5e616f40e5f9fed77dd8..4f3eed88d6df09dc98e85d2d1a0e6ca2c9c6e964 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -78,7 +78,6 @@ class WebRtcAudioCapturer::SinkOwner
                            int audio_delay_milliseconds,
                            double volume) OVERRIDE;
   virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
-  virtual void OnCaptureDeviceStopped() OVERRIDE;
 
   bool IsEqual(const WebRtcAudioCapturerSink* other) const;
   void Reset();
@@ -126,12 +125,6 @@ void WebRtcAudioCapturer::SinkOwner::SetCaptureFormat(
     delegate_->SetCaptureFormat(params);
 }
 
-void WebRtcAudioCapturer::SinkOwner::OnCaptureDeviceStopped() {
-  base::AutoLock lock(lock_);
-  if (delegate_)
-    delegate_->OnCaptureDeviceStopped();
-}
-
 bool WebRtcAudioCapturer::SinkOwner::IsEqual(
     const WebRtcAudioCapturerSink* other) const {
   base::AutoLock lock(lock_);
@@ -212,7 +205,8 @@ bool WebRtcAudioCapturer::Reconfigure(int sample_rate,
 }
 
 bool WebRtcAudioCapturer::Initialize(media::ChannelLayout channel_layout,
-                                     int sample_rate) {
+                                     int sample_rate,
+                                     int session_id) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!sinks_.empty());
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
@@ -221,6 +215,8 @@ bool WebRtcAudioCapturer::Initialize(media::ChannelLayout channel_layout,
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX);
 
+  session_id_ = session_id;
+
   // Verify that the reported input channel configuration is supported.
   if (channel_layout != media::CHANNEL_LAYOUT_MONO &&
       channel_layout != media::CHANNEL_LAYOUT_STEREO) {
@@ -248,7 +244,7 @@ bool WebRtcAudioCapturer::Initialize(media::ChannelLayout channel_layout,
 
   // Create and configure the default audio capturing source. The |source_|
   // will be overwritten if an external client later calls SetCapturerSource()
-  // providing an alternaive media::AudioCapturerSource.
+  // providing an alternative media::AudioCapturerSource.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(),
                     channel_layout,
                     static_cast<float>(sample_rate));
@@ -257,10 +253,10 @@ bool WebRtcAudioCapturer::Initialize(media::ChannelLayout channel_layout,
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer()
-    : main_loop_(base::MessageLoopProxy::current()),
-      source_(NULL),
+    : source_(NULL),
       running_(false),
-      agc_is_enabled_(false) {
+      agc_is_enabled_(false),
+      session_id_(0) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
@@ -307,6 +303,7 @@ void WebRtcAudioCapturer::SetCapturerSource(
     media::ChannelLayout channel_layout,
     float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
+  CHECK_GT(session_id_, 0);
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
            << "sample_rate=" << sample_rate << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
@@ -346,7 +343,7 @@ void WebRtcAudioCapturer::SetCapturerSource(
 
   if (source) {
     // Make sure to grab the new parameters in case they were reconfigured.
-    source->Initialize(current_buffer->params(), this, this);
+    source->Initialize(current_buffer->params(), this, session_id_);
   }
 }
 
@@ -390,14 +387,6 @@ void WebRtcAudioCapturer::SetVolume(double volume) {
     source_->SetVolume(volume);
 }
 
-void WebRtcAudioCapturer::SetDevice(int session_id) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcAudioCapturer::SetDevice(" << session_id << ")";
-  base::AutoLock auto_lock(lock_);
-  if (source_)
-    source_->SetDevice(session_id);
-}
-
 void WebRtcAudioCapturer::SetAutomaticGainControl(bool enable) {
   base::AutoLock auto_lock(lock_);
   // Store the setting since SetAutomaticGainControl() can be called before
@@ -450,34 +439,6 @@ void WebRtcAudioCapturer::OnCaptureError() {
   NOTIMPLEMENTED();
 }
 
-void WebRtcAudioCapturer::OnDeviceStarted(const std::string& device_id) {
-  device_id_ = device_id;
-}
-
-void WebRtcAudioCapturer::OnDeviceStopped() {
-  DCHECK_EQ(MessageLoop::current(), ChildProcess::current()->io_message_loop());
-  main_loop_->PostTask(
-      FROM_HERE, base::Bind(&WebRtcAudioCapturer::DoOnDeviceStopped, this));
-}
-
-// TODO(henrika): this implementation should be removed as soon as we add
-// suppport for proper handling of LocalMediaStream::stop().
-void WebRtcAudioCapturer::DoOnDeviceStopped() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcAudioCapturer::DoOnDeviceStopped()";
-  SinkList sinks;
-  {
-    base::AutoLock auto_lock(lock_);
-    running_ = false;
-    sinks = sinks_;
-  }
-
-  // Inform registered sinks about the stopped device so they can take
-  // appropriate actions.
-  for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it)
-    (*it)->OnCaptureDeviceStopped();
-}
-
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   return buffer_->params();