start/stop the source of the capturer when 1st audiotrack/last audiotrack is added/removed

content/renderer/media/webrtc_audio_device_unittest.cc

Index: content/renderer/media/webrtc_audio_device_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index 37d5401c558b9e1aadfed78d7aadc2ddf2e87135..0384b6c6d3c4adf7d909a5b0ef8c681e64f06808 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -11,6 +11,7 @@
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
+#include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/render_thread_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager_base.h"
@@ -582,7 +583,13 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_StartRecording) {
   ASSERT_EQ(0, err);
 
   EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
-  webrtc_audio_device->capturer()->Start();
+
+  // Adds a local audio track to trigger starting the capturer.

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

Could you make this comment a bit more clear. "..to trigger starting..." is a
bit vague. Do you mean, "create local audio track and connect it to the source
and then start the source by...." 

[no longer working on chromium, 2013/06/05 16:29:45]

Done.

+  scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
+      WebRtcLocalAudioTrack::Create(std::string(),
+                                    webrtc_audio_device->capturer(),
+                                    NULL));
+  local_audio_track->Start();

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

No EXPECT?

[no longer working on chromium, 2013/06/05 16:29:45]

It is void

 
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);
@@ -619,7 +626,7 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_StartRecording) {
       ch, webrtc::kRecordingPerChannel));
   EXPECT_EQ(0, base->StopSend(ch));
 
-  webrtc_audio_device->capturer()->Stop();
+  local_audio_track->Stop();
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
@@ -745,7 +752,12 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
   ASSERT_EQ(0, err);
 
   EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
-  webrtc_audio_device->capturer()->Start();
+  // Adds a local audio track to trigger starting the capturer.

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

dito

[no longer working on chromium, 2013/06/05 16:29:45]

Done.

+  scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
+      WebRtcLocalAudioTrack::Create(std::string(),
+                                    webrtc_audio_device->capturer(),
+                                    NULL));
+  local_audio_track->Start();
 
   ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
   ASSERT_TRUE(audio_processing.valid());
@@ -781,7 +793,7 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
                                 base::TimeDelta::FromSeconds(2));
   message_loop_.Run();
 
-  webrtc_audio_device->capturer()->Stop();
+  local_audio_track->Stop();
   renderer->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
@@ -815,7 +827,12 @@ TEST_F(WebRTCAudioDeviceTest, WebRtcRecordingSetupTime) {
   ASSERT_EQ(0, err);
 
   EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get()));
-  webrtc_audio_device->capturer()->Start();
+  // Adds a local audio track to trigger starting the capturer.

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

same here

[no longer working on chromium, 2013/06/05 16:29:45]

Done.

+  scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
+      WebRtcLocalAudioTrack::Create(std::string(),
+                                    webrtc_audio_device->capturer(),
+                                    NULL));
+  local_audio_track->Start();
 
   base::WaitableEvent event(false, false);
   scoped_ptr<MockWebRtcAudioCapturerSink> capturer_sink(
@@ -834,7 +851,7 @@ TEST_F(WebRTCAudioDeviceTest, WebRtcRecordingSetupTime) {
   PrintPerfResultMs("webrtc_recording_setup_c", "t", delay);
 
   capturer->RemoveSink(capturer_sink.get());
-  webrtc_audio_device->capturer()->Stop();
+  local_audio_track->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());