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 32 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 32 matching lines @@
  public:
   MockWebRtcAudioCapturerSink() {}
   ~MockWebRtcAudioCapturerSink() {}
   int CaptureData(const std::vector<int>& channels,
                   const int16* audio_data,
                   int sample_rate,
                   int number_of_channels,
                   int number_of_frames,
                   int audio_delay_milliseconds,
                   int current_volume,
-                  bool need_audio_processing) OVERRIDE {
-    CaptureData(channels.size(), sample_rate, number_of_channels,
-                number_of_frames, audio_delay_milliseconds, current_volume,
-                need_audio_processing);
+                  bool need_audio_processing,
+                  bool key_pressed) OVERRIDE {
+    CaptureData(channels.size(),
+                sample_rate,
+                number_of_channels,
+                number_of_frames,
+                audio_delay_milliseconds,
+                current_volume,
+                need_audio_processing,
+                key_pressed);
     return 0;
   }
-  MOCK_METHOD7(CaptureData, void(int number_of_network_channels,
-                                 int sample_rate,
-                                 int number_of_channels,
-                                 int number_of_frames,
-                                 int audio_delay_milliseconds,
-                                 int current_volume,
-                                 bool need_audio_processing));
-
+  MOCK_METHOD8(CaptureData,
+               void(int number_of_network_channels,
+                    int sample_rate,
+                    int number_of_channels,
+                    int number_of_frames,
+                    int audio_delay_milliseconds,
+                    int current_volume,
+                    bool need_audio_processing,
+                    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 36 matching lines @@
   static const int kNumberOfNetworkChannels = 4;
   for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track.get())->
         GetRenderer()->AddChannel(i);
   }
   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(
-      kNumberOfNetworkChannels, params.sample_rate(), params.channels(),
-      params.frames_per_buffer(), 0, 0, false))
-      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
+  EXPECT_CALL(*sink,
+              CaptureData(kNumberOfNetworkChannels,
+                          params.sample_rate(),
+                          params.channels(),
+                          params.frames_per_buffer(),
+                          0,
+                          0,
+                          false,
+                          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;
 }
 
@@ skipping 10 matching lines @@
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(track.get())->
       GetRenderer()->AddChannel(0);
   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(
-      1, params.sample_rate(), params.channels(),
-      params.frames_per_buffer(), 0, 0, false))
-      .Times(0);
+  EXPECT_CALL(*sink,
+              CaptureData(1,
+                          params.sample_rate(),
+                          params.channels(),
+                          params.frames_per_buffer(),
+                          0,
+                          0,
+                          false,
+                          false)).Times(0);
   track->AddSink(sink.get());
   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   event.Reset();
-  EXPECT_CALL(*sink, CaptureData(
-      1, params.sample_rate(), params.channels(),
-      params.frames_per_buffer(), 0, 0, false))
-      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
+  EXPECT_CALL(*sink,
+              CaptureData(1,
+                          params.sample_rate(),
+                          params.channels(),
+                          params.frames_per_buffer(),
+                          0,
+                          0,
+                          false,
+                          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();
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   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(
-      1, params.sample_rate(), params.channels(),
-      params.frames_per_buffer(), 0, 0, false))
-      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
+  EXPECT_CALL(*sink_1,
+              CaptureData(1,
+                          params.sample_rate(),
+                          params.channels(),
+                          params.frames_per_buffer(),
+                          0,
+                          0,
+                          false,
+                          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();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
   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(
-      1, params.sample_rate(), params.channels(),
-      params.frames_per_buffer(), 0, 0, false))
-      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
-  EXPECT_CALL(*sink_2, CaptureData(
-      1, params.sample_rate(), params.channels(),
-      params.frames_per_buffer(), 0, 0, false))
-      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_2));
+  EXPECT_CALL(*sink_1,
+              CaptureData(1,
+                          params.sample_rate(),
+                          params.channels(),
+                          params.frames_per_buffer(),
+                          0,
+                          0,
+                          false,
+                          false)).Times(AtLeast(1))
+      .WillRepeatedly(SignalEvent(&event_1));
+  EXPECT_CALL(*sink_2,
+              CaptureData(1,
+                          params.sample_rate(),
+                          params.channels(),
+                          params.frames_per_buffer(),
+                          0,
+                          0,
+                          false,
+                          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());
   track_2->RemoveSink(sink_2.get());
@@ skipping 27 matching lines @@
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Verify the data flow by connecting the sink to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   event.Reset();
-  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
+  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink.get());
 
   // 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();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
 
   // Stop the first audio track will not stop the |capturer_source_|.
   EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
   track_1->RemoveSink(sink.get());
   track_1->Stop();
   track_1 = NULL;
 
-  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
+  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
   track_2->AddSink(sink.get());
 
   // 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->RemoveSink(sink.get());
@@ skipping 36 matching lines @@
 
   // Connect a number of network channels to the |track_1|.
   static const int kNumberOfNetworkChannelsForTrack1 = 2;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_1| to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
-  EXPECT_CALL(*sink_1.get(), CaptureData(kNumberOfNetworkChannelsForTrack1,
-                                         48000, 2, _, 0, 0, false))
+  EXPECT_CALL(
+      *sink_1.get(),
+      CaptureData(
+          kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer());
   scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
   EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
       .WillOnce(Return());
@@ skipping 15 matching lines @@
 
   // Connect a number of network channels to the |track_2|.
   static const int kNumberOfNetworkChannelsForTrack2 = 3;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_2| to |track_2|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
       new MockWebRtcAudioCapturerSink());
-  EXPECT_CALL(*sink_2, CaptureData(kNumberOfNetworkChannelsForTrack2,
-                                     44100, 1, _, 0, 0, false))
+  EXPECT_CALL(
+      *sink_2,
+      CaptureData(
+          kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
   track_2->AddSink(sink_2.get());
 
   // Stop the second audio track will stop the new source.
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*new_source.get(), Stop()).Times(1).WillOnce(SignalEvent(&event));
   track_2->Stop();
   track_2->RemoveSink(sink_2.get());
   track_2 = NULL;
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   audio_thread->Stop();
   audio_thread.reset();
 
   // Stop the first audio track.
   EXPECT_CALL(*capturer_source_.get(), Stop());
   track_1->Stop();
   track_1 = NULL;
 }
 
 }  // namespace content