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

content/renderer/media/rtc_peer_connection_handler_unittest.cc

Index: content/renderer/media/rtc_peer_connection_handler_unittest.cc
diff --git a/content/renderer/media/rtc_peer_connection_handler_unittest.cc b/content/renderer/media/rtc_peer_connection_handler_unittest.cc
index 697f484bdca7930138aed5519bc73a3a05f16904..a8eaf00d8e1fe7722089571d744d31dd4840fc2d 100644
--- a/content/renderer/media/rtc_peer_connection_handler_unittest.cc
+++ b/content/renderer/media/rtc_peer_connection_handler_unittest.cc
@@ -16,7 +16,6 @@
 #include "base/values.h"
 #include "content/child/child_process.h"
 #include "content/renderer/media/media_stream.h"
-#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_source.h"
 #include "content/renderer/media/media_stream_video_track.h"
 #include "content/renderer/media/mock_data_channel_impl.h"
@@ -28,8 +27,8 @@
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.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"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
@@ -250,12 +249,28 @@ class RTCPeerConnectionHandlerTest : public ::testing::Test {
       const std::string& stream_label) {
     std::string video_track_label("video-label");
     std::string audio_track_label("audio-label");
-    blink::WebMediaStreamSource audio_source;
-    audio_source.initialize(blink::WebString::fromUTF8(audio_track_label),
-                            blink::WebMediaStreamSource::TypeAudio,
-                            blink::WebString::fromUTF8("audio_track"),
-                            false /* remote */, true /* readonly */);
-    audio_source.setExtraData(new MediaStreamAudioSource());
+
+    blink::WebMediaStreamSource blink_audio_source;
+    blink_audio_source.initialize(blink::WebString::fromUTF8(audio_track_label),
+                                  blink::WebMediaStreamSource::TypeAudio,
+                                  blink::WebString::fromUTF8("audio_track"),
+                                  false /* remote */, true /* readonly */);
+    ProcessedLocalAudioSource* const audio_source =
+        new ProcessedLocalAudioSource(
+            -1 /* consumer_render_frame_id is N/A for non-browser tests */,
+            StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "Mock device",
+                             "mock_device_id"),
+            mock_dependency_factory_.get());
+    audio_source->SetAllowInvalidRenderFrameIdForTesting(true);
+    audio_source->SetSourceConstraints(
+        MockMediaConstraintFactory().CreateWebMediaConstraints());
+    blink_audio_source.setExtraData(audio_source);  // Takes ownership.
+
+    blink::WebVector<blink::WebMediaStreamTrack> audio_tracks(
+        static_cast<size_t>(1));
+    audio_tracks[0].initialize(blink_audio_source.id(), blink_audio_source);
+    CHECK(audio_source->ConnectToTrack(audio_tracks[0]));
+
     blink::WebMediaStreamSource video_source;
     video_source.initialize(blink::WebString::fromUTF8(video_track_label),
                             blink::WebMediaStreamSource::TypeVideo,
@@ -265,22 +280,6 @@ class RTCPeerConnectionHandlerTest : public ::testing::Test {
         new MockMediaStreamVideoSource(false);
     video_source.setExtraData(native_video_source);
 
-    blink::WebVector<blink::WebMediaStreamTrack> audio_tracks(
-        static_cast<size_t>(1));
-    audio_tracks[0].initialize(audio_source.id(), audio_source);
-    StreamDeviceInfo device_info(MEDIA_DEVICE_AUDIO_CAPTURE, "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,
-                                            nullptr, nullptr));
-    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-        WebRtcLocalAudioTrackAdapter::Create(audio_track_label, nullptr));
-    scoped_ptr<WebRtcLocalAudioTrack> native_track(
-        new WebRtcLocalAudioTrack(adapter.get(), capturer, nullptr));
-    audio_tracks[0].setExtraData(native_track.release());
     blink::WebVector<blink::WebMediaStreamTrack> video_tracks(
         static_cast<size_t>(1));
     blink::WebMediaConstraints video_constraints;
@@ -313,7 +312,8 @@ class RTCPeerConnectionHandlerTest : public ::testing::Test {
     }
     if (!audio_track_label.empty()) {
       scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-          WebRtcLocalAudioTrackAdapter::Create(audio_track_label, NULL));
+          WebRtcLocalAudioTrackAdapter::Create(
+              audio_track_label, nullptr, pc_handler_->signaling_thread()));
       stream->AddTrack(audio_track.get());
     }
     mock_peer_connection_->AddRemoteStream(stream.get());