Notify the track before source provider goes away.

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

Index: content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
index 5b7e8526898f93cd6287148093c658bda7e3ec80..8b0fd6b27c157e9ce14da82c82b8d83f0135de39 100644
--- a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
@@ -3,10 +3,16 @@
 // found in the LICENSE file.
 
 #include "base/logging.h"
+#include "base/strings/utf_string_conversions.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
+#include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace content {
 
@@ -23,7 +29,22 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
         source_params_.frames_per_buffer() * source_params_.channels();
     source_data_.reset(new int16[length]);
     sink_bus_ = media::AudioBus::Create(sink_params_);
-    source_provider_.reset(new WebRtcLocalAudioSourceProvider());
+    blink::WebMediaConstraints constraints;
+    scoped_refptr<WebRtcAudioCapturer> capturer(
+        WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo(),
+                                            constraints, NULL));
+    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
+        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+    scoped_ptr<WebRtcLocalAudioTrack> native_track(
+        new WebRtcLocalAudioTrack(adapter, capturer, NULL));
+    blink::WebMediaStreamSource audio_source;
+    audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
+                            blink::WebMediaStreamSource::TypeAudio,
+                            base::UTF8ToUTF16("dummy_source_name"));
+    blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
+                            audio_source);
+    blink_track_.setExtraData(native_track.release());
+    source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
   }
@@ -32,6 +53,7 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
   scoped_ptr<int16[]> source_data_;
   media::AudioParameters sink_params_;
   scoped_ptr<media::AudioBus> sink_bus_;
+  blink::WebMediaStreamTrack blink_track_;
   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 };
 
@@ -91,4 +113,25 @@ TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
   }
 }
 
+TEST_F(WebRtcLocalAudioSourceProviderTest,
+       DeleteSourceProviderBeforeStoppingTrack) {
+  source_provider_.reset();
+
+  // Stop the audio track.
+  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(

[perkj_chrome, 2014/04/03 07:45:43]

Can you make sure this is always the case?

[no longer working on chromium, 2014/04/03 09:18:57]

Yes, we set this up in SetUp() method, otherwise it will crash.

[perkj_chrome, 2014/04/03 09:42:06]

 I think what I meant was, The sourceProvider should be destroyed before the
track. But as discussed, don't care at the moment.

[no longer working on chromium, 2014/04/03 09:48:20]

Yes, it has been the case that track will always outlive the source provider.
This two test cases verify that we can stop the track before and after the
deletion of the source provider.

+      MediaStreamTrack::GetTrack(blink_track_));
+  native_track->Stop();
+}
+
+TEST_F(WebRtcLocalAudioSourceProviderTest,
+       StopTrackBeforeDeletingSourceProvider) {
+  // Stop the audio track.
+  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
+      MediaStreamTrack::GetTrack(blink_track_));
+  native_track->Stop();
+
+  // Delete the source provider.
+  source_provider_.reset();
+}
+
 }  // namespace content