Notify the track before source provider goes away.

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

 // Copyright 2013 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.
 
 #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 {
 
 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                          media::CHANNEL_LAYOUT_MONO, 1, 0, 48000, 16, 480);
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
         media::CHANNEL_LAYOUT_STEREO, 2, 0, 44100, 16,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     const int length =
         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_);
   }
 
   media::AudioParameters source_params_;
   scoped_ptr<int16[]> source_data_;
   media::AudioParameters sink_params_;
   scoped_ptr<media::AudioBus> sink_bus_;
+  blink::WebMediaStreamTrack blink_track_;
   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 };
 
 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
   // Point the WebVector into memory owned by |sink_bus_|.
   blink::WebVector<float*> audio_data(
       static_cast<size_t>(sink_bus_->channels()));
   for (size_t i = 0; i < audio_data.size(); ++i)
     audio_data[i] = sink_bus_->channel(i);
 
@@ skipping 39 matching lines @@
        i += sink_params_.frames_per_buffer()) {
     sink_bus_->Zero();
     source_provider_->provideInput(audio_data,
                                    sink_params_.frames_per_buffer());
     EXPECT_GT(sink_bus_->channel(0)[0], 0);
     EXPECT_GT(sink_bus_->channel(1)[0], 0);
     EXPECT_DOUBLE_EQ(sink_bus_->channel(0)[0], sink_bus_->channel(1)[0]);
   }
 }
 
+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