MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
index 0230e0642a6a34ad78aa3808ea76a047d3358768..6336e9d0b2a575532d700616921b24f0e671fc04 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
@@ -5,9 +5,9 @@
 #include <stddef.h>
 
 #include "content/renderer/media/mock_media_constraint_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_track.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_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
@@ -37,24 +37,21 @@ 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)) {
+        adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), nullptr,
+                                                      nullptr)) {
     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));
+    track_.reset(new ProcessedLocalAudioTrack(adapter_.get()));
   }
 
  protected:
   void SetUp() override {
-    track_->OnSetFormat(params_);
-    EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
+    track_->SetFormat(params_);
+    EXPECT_TRUE(track_->adapter()->enabled());
   }
 
   media::AudioParameters params_;
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
-  scoped_refptr<WebRtcAudioCapturer> capturer_;
-  scoped_ptr<WebRtcLocalAudioTrack> track_;
+  scoped_ptr<ProcessedLocalAudioTrack> track_;
 };
 
 // Adds and Removes a WebRtcAudioSink to a local audio track.
@@ -70,16 +67,16 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   const scoped_ptr<media::AudioBus> audio_bus =
       media::AudioBus::Create(params_);
   // While this test is not checking the signal data being passed around, the
-  // implementation in WebRtcLocalAudioTrack reads the data for its signal level
-  // computation.  Initialize all samples to zero to make the memory sanitizer
-  // happy.
+  // implementation in ProcessedLocalAudioSource reads the data for its signal
+  // level computation.  Initialize all samples to zero to make the memory
+  // sanitizer happy.
   audio_bus->Zero();
 
   base::TimeTicks estimated_capture_time = base::TimeTicks::Now();
   EXPECT_CALL(*sink,
               OnData(_, 16, params_.sample_rate(), params_.channels(),
                      params_.frames_per_buffer()));
-  track_->Capture(*audio_bus, estimated_capture_time, false);
+  track_->DeliverDataToSinks(*audio_bus, estimated_capture_time);
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
@@ -89,7 +86,7 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   estimated_capture_time +=
       params_.frames_per_buffer() * base::TimeDelta::FromSeconds(1) /
           params_.sample_rate();
-  track_->Capture(*audio_bus, estimated_capture_time, false);
+  track_->DeliverDataToSinks(*audio_bus, estimated_capture_time);
 }
 
 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {