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

content/renderer/media/speech_recognition_audio_sink_unittest.cc

Index: content/renderer/media/speech_recognition_audio_sink_unittest.cc
diff --git a/content/renderer/media/speech_recognition_audio_sink_unittest.cc b/content/renderer/media/speech_recognition_audio_sink_unittest.cc
index 64ee4348f10d53570d6f09f0271141479d168596..c88ef61cb8708d2eb4ca996f46c389bc30a5c08f 100644
--- a/content/renderer/media/speech_recognition_audio_sink_unittest.cc
+++ b/content/renderer/media/speech_recognition_audio_sink_unittest.cc
@@ -12,10 +12,10 @@
 #include "base/bind.h"
 #include "base/macros.h"
 #include "base/strings/utf_string_conversions.h"
-#include "content/renderer/media/media_stream_audio_source.h"
+#include "content/renderer/media/media_stream_audio_track.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 "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gmock/include/gmock/gmock.h"
@@ -243,9 +243,8 @@ class SpeechRecognitionAudioSinkTest : public testing::Test {
     PrepareBlinkTrackOfType(MEDIA_DEVICE_AUDIO_CAPTURE, &blink_track);
 
     // Get the native track from the blink track and initialize.
-    native_track_ =
-        static_cast<WebRtcLocalAudioTrack*>(blink_track.extraData());
-    native_track_->OnSetFormat(source_params_);
+    native_track_ = MediaStreamAudioTrack::Get(blink_track);
+    native_track_->SetFormat(source_params_);
 
     // Create and initialize the consumer.
     recognizer_.reset(new FakeSpeechRecognizer());
@@ -272,32 +271,26 @@ class SpeechRecognitionAudioSinkTest : public testing::Test {
  protected:
   // Prepares a blink track of a given MediaStreamType and attaches the native
   // track which can be used to capture audio data and pass it to the producer.
-  static void PrepareBlinkTrackOfType(
-      const MediaStreamType device_type,
-      blink::WebMediaStreamTrack* blink_track) {
-    StreamDeviceInfo device_info(device_type, "Mock device",
-                                 "mock_device_id");
-    MockMediaConstraintFactory constraint_factory;
-    const blink::WebMediaConstraints constraints =
-        constraint_factory.CreateWebMediaConstraints();
-    scoped_refptr<WebRtcAudioCapturer> capturer(
-        WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL,
-                                            NULL));
-    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-    scoped_ptr<WebRtcLocalAudioTrack> native_track(
-        new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
+  void PrepareBlinkTrackOfType(const MediaStreamType device_type,
+                               blink::WebMediaStreamTrack* blink_track) {
     blink::WebMediaStreamSource blink_audio_source;
     blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
                                   blink::WebMediaStreamSource::TypeAudio,
                                   base::UTF8ToUTF16("dummy_source_name"),
                                   false /* remote */, true /* readonly */);
-    MediaStreamSource::SourceStoppedCallback cb;
-    blink_audio_source.setExtraData(
-        new MediaStreamAudioSource(-1, device_info, cb, NULL));
+    ProcessedLocalAudioSource* const audio_source =
+        new ProcessedLocalAudioSource(
+            -1 /* consumer_render_frame_id is N/A for non-browser tests */,
+            StreamDeviceInfo(device_type, "Mock device", "mock_device_id"),
+            &mock_dependency_factory_);
+    audio_source->SetAllowInvalidRenderFrameIdForTesting(true);
+    audio_source->SetSourceConstraints(
+        MockMediaConstraintFactory().CreateWebMediaConstraints());
+    blink_audio_source.setExtraData(audio_source);  // Takes ownership.
+
     blink_track->initialize(blink::WebString::fromUTF8("dummy_track"),
                             blink_audio_source);
-    blink_track->setExtraData(native_track.release());
+    ASSERT_TRUE(audio_source->ConnectToTrack(*blink_track));
   }
 
   // Emulates an audio capture device capturing data from the source.
@@ -306,7 +299,7 @@ class SpeechRecognitionAudioSinkTest : public testing::Test {
       const base::TimeTicks estimated_capture_time = first_frame_capture_time_ +
           (sample_frames_captured_ * base::TimeDelta::FromSeconds(1) /
                source_params_.sample_rate());
-      native_track()->Capture(*source_bus_, estimated_capture_time, false);
+      native_track()->DeliverDataToSinks(*source_bus_, estimated_capture_time);
       sample_frames_captured_ += source_bus_->frames();
     }
   }
@@ -360,9 +353,11 @@ class SpeechRecognitionAudioSinkTest : public testing::Test {
 
   const media::AudioParameters& sink_params() const { return sink_params_; }
 
-  WebRtcLocalAudioTrack* native_track() const { return native_track_; }
+  MediaStreamAudioTrack* native_track() const { return native_track_; }
 
  private:
+  MockPeerConnectionDependencyFactory mock_dependency_factory_;
+
   // Producer.
   scoped_ptr<SpeechRecognitionAudioSink> speech_audio_sink_;
 
@@ -373,7 +368,7 @@ class SpeechRecognitionAudioSinkTest : public testing::Test {
   scoped_ptr<media::AudioBus> source_bus_;
   media::AudioParameters source_params_;
   media::AudioParameters sink_params_;
-  WebRtcLocalAudioTrack* native_track_;
+  MediaStreamAudioTrack* native_track_;
 
   base::TimeTicks first_frame_capture_time_;
   int64_t sample_frames_captured_;