Update To 11.40.268.0

[platform/framework/web/crosswalk.git] / src / content / renderer / media / webrtc / webrtc_local_audio_track_adapter_unittest.cc

diff --git a/src/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/src/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
index d798b31..184240c 100644 (file)
--- a/src/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
+++ b/src/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
@@ -4,6 +4,7 @@
 
 #include "base/command_line.h"
 #include "content/public/common/content_switches.h"
+#include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
@@ -35,14 +36,16 @@ class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
   WebRtcLocalAudioTrackAdapterTest()
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
-        adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)),
-        capturer_(WebRtcAudioCapturer::CreateCapturer(
-            -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
-            blink::WebMediaConstraints(), NULL, NULL)),
-        track_(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL)) {}
+        adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) {
+    MockMediaConstraintFactory constraint_factory;
+    capturer_ = WebRtcAudioCapturer::CreateCapturer(
+        -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
+        constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
+    track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
+  }
 
  protected:
-  virtual void SetUp() OVERRIDE {
+  void SetUp() override {
     track_->OnSetFormat(params_);
     EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
   }
@@ -71,26 +74,26 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   EXPECT_CALL(*sink,
               OnData(_, 16, params_.sample_rate(), params_.channels(),
                      params_.frames_per_buffer()));
-  track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
+  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
   sink.reset();
 
   // Verify that no more callback gets into the sink.
-  track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
+  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
 }
 
 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
   webrtc::AudioTrackInterface* webrtc_track =
       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
   int signal_level = 0;
-  EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
+  EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
 
-  // Enable the audio processing in the audio track.
+  // Disable the audio processing in the audio track.
   CommandLine::ForCurrentProcess()->AppendSwitch(
-      switches::kEnableAudioTrackProcessing);
-  EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
+      switches::kDisableAudioTrackProcessing);
+  EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
 }
 
 }  // namespace content