Adding key press detection in the browser process.

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"
@@ skipping 31 matching lines @@
   // base::PlatformThread::Delegate:
   virtual void ThreadMain() OVERRIDE {
     while (true) {
       if (closure_.IsSignaled())
         return;
 
       media::AudioCapturerSource::CaptureCallback* callback =
           static_cast<media::AudioCapturerSource::CaptureCallback*>(
               capturer_.get());
       audio_bus_->Zero();
-      callback->Capture(audio_bus_.get(), 0, 0);
+      callback->Capture(audio_bus_.get(), 0, 0, false);
 
       // Sleep 1ms to yield the resource for the main thread.
       base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
     }
   }
 
   void Start() {
     base::PlatformThread::CreateWithPriority(
         0, this, &thread_, base::kThreadPriority_RealtimeAudio);
     CHECK(!thread_.is_null());
@@ skipping 25 matching lines @@
   MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
 
  protected:
   virtual ~MockCapturerSource() {}
 };
 
 class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
  public:
   MockWebRtcAudioCapturerSink() {}
   ~MockWebRtcAudioCapturerSink() {}
-  MOCK_METHOD5(CaptureData, void(const int16* audio_data,
-                                 int number_of_channels,
-                                 int number_of_frames,
-                                 int audio_delay_milliseconds,
-                                 double volume));
+  MOCK_METHOD6(CaptureData,
+               void(const int16* audio_data,
+                    int number_of_channels,
+                    int number_of_frames,
+                    int audio_delay_milliseconds,
+                    double volume,
+                    bool key_pressed));
   MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
     capturer_source_ = new MockCapturerSource();
@@ skipping 27 matching lines @@
   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))
+  EXPECT_CALL(
+      *sink,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .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))
+  EXPECT_CALL(
+      *sink,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .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))
+  EXPECT_CALL(
+      *sink,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .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.
 TEST_F(WebRtcLocalAudioTrackTest, 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))
+  EXPECT_CALL(
+      *sink_1,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track_2->Start();
   EXPECT_TRUE(track_2->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
         new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*sink_1, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink_1,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
-  EXPECT_CALL(*sink_2, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink_2,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_2));
   track_2->AddSink(sink_2.get());
   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());
@@ skipping 18 matching lines @@
 }
 
 // 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))
+  EXPECT_CALL(*default_sink, CaptureData(_, _, _, 0, 0, false))
       .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);
@@ skipping 42 matching lines @@
                                media::CHANNEL_LAYOUT_STEREO,
                                48000);
 
   // Stop the track.
   EXPECT_CALL(*new_source.get(), Stop());
   track->Stop();
   track = NULL;
 }
 
 }  // namespace content