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

content/renderer/media/webrtc_local_audio_track_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/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_capturer_source.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 using ::testing::_;
+using ::testing::AnyNumber;
 using ::testing::AtLeast;
 using ::testing::Return;
 
 namespace content {
 
 namespace {
 
 ACTION_P(SignalEvent, event) {
   event->Signal();
 }
@@ skipping 82 matching lines @@
  protected:
   virtual void SetUp() OVERRIDE {
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
         .WillOnce(Return());
     capturer_->SetCapturerSource(capturer_source_,
                                  media::CHANNEL_LAYOUT_STEREO,
                                  48000);
 
-    EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
         .WillOnce(Return());
-    capturer_->Start();
     audio_thread_.reset(new FakeAudioThread(capturer_));
     audio_thread_->Start();
   }
 
   virtual void TearDown() {
     audio_thread_->Stop();
     audio_thread_.reset();
-    EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
-    capturer_->Stop();
   }
 
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
   scoped_ptr<FakeAudioThread> audio_thread_;
 };
 
 // Creates a capturer and audio track, fakes its audio thread, and
 // connect/disconnect the sink to the audio track on the fly, the sink should
 // get data callback when the track is connected to the capturer but not when
 // the track is disconnected from the capturer.
 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
+  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track->Start();
   EXPECT_TRUE(track->enabled());
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
 
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
+
+  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
+  track->Stop();
   track = NULL;
 }
 
 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the
 // audio track on the fly. When the audio track is disabled, there is no data
 // callback to the sink; when the audio track is enabled, there comes data
 // callback.
 // TODO(xians): Enable this test after resolving the racing issue that TSAN
 // reports on MediaStreamTrack::enabled();
 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
+  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   scoped_refptr<WebRtcLocalAudioTrack> track =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track->Start();
   EXPECT_TRUE(track->enabled());
   EXPECT_TRUE(track->set_enabled(false));
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(0);
   track->AddSink(sink.get());
   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   event.Reset();
   EXPECT_CALL(*sink, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   EXPECT_TRUE(track->set_enabled(true));
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
+
+  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
+  track->Stop();
   track = NULL;
 }
 
 // Create multiple audio tracks and enable/disable them, verify that the audio
 // callbacks appear/disappear.
 // TODO(xians): Enable the test after the racing problem is resolved.
 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
+  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track_1->Start();
   EXPECT_TRUE(track_1->enabled());
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event_1(false, false);
   EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
   track_1->AddSink(sink_1.get());
 
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track_2->Start();
   EXPECT_TRUE(track_2->set_enabled(false));
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
       new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_2, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(0);
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
 
   // Enable |track_2|, and verify the data callback comes to |sink_2|;
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
   EXPECT_CALL(*sink_1, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
   EXPECT_CALL(*sink_2, CaptureData(
       _, params.channels(), params.frames_per_buffer(), 0, 0))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_2));
   EXPECT_TRUE(track_2->set_enabled(true));
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 
   track_1->RemoveSink(sink_1.get());
+  track_1->Stop();
+  track_1 = NULL;
+
+  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
   track_2->RemoveSink(sink_2.get());
-  track_1 = NULL;
+  track_2->Stop();
   track_2 = NULL;
 }
 
+
+// Start one track and verify the capturer is correctly starting its source.
+// And it should be fine to not to call Stop() explicitly.
+TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
+  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
+  scoped_refptr<WebRtcLocalAudioTrack> track =
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track->Start();
+}
+
+// Start/Stop tracks and verify the capturer is correctly starting/stopping
+// its source.
+TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
+  // SetDefaultSink() should not trigger the Start() on |capturer_source_|.
+  base::WaitableEvent event(false, false);
+  scoped_ptr<MockWebRtcAudioCapturerSink> default_sink(
+      new MockWebRtcAudioCapturerSink());
+  EXPECT_CALL(*default_sink, SetCaptureFormat(_)).WillOnce(Return());
+  EXPECT_CALL(*default_sink, CaptureData(
+      _, _, _, 0, 0))
+      .Times(AnyNumber()).WillRepeatedly(Return());
+  capturer_->SetDefaultSink(default_sink.get());
+  EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
+  EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
+
+  // Starting the first audio track will start the |capturer_source_|.
+  event.Reset();
+  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
+  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track_1->Start();
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
+
+  // Start the second audio track will not start the |capturer_source_|
+  // since it has been started.
+  EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
+  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
+  track_2->Start();
+
+  // Stop the first audio track will not stop the |capturer_source_|.
+  EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
+  track_1->Stop();
+  track_1 = NULL;
+
+  // Stop the last audio track will stop the |capturer_source_|.
+  event.Reset();
+  EXPECT_CALL(*capturer_source_.get(), Stop())
+      .Times(1).WillOnce(SignalEvent(&event));
+  track_2->Stop();
+  track_2 = NULL;
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
+
+  capturer_->SetDefaultSink(NULL);
+}
+
 }  // namespace content