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

content/renderer/media/rtc_peer_connection_handler_unittest.cc

 // Copyright (c) 2012 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 <stddef.h>
 
 #include <string>
 #include <vector>
 
 #include "base/location.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/run_loop.h"
 #include "base/single_thread_task_runner.h"
 #include "base/strings/utf_string_conversions.h"
 #include "base/thread_task_runner_handle.h"
 #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"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/mock_media_stream_video_source.h"
 #include "content/renderer/media/mock_peer_connection_impl.h"
 #include "content/renderer/media/mock_web_rtc_peer_connection_handler_client.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #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"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
@@ skipping 200 matching lines @@
     mock_dependency_factory_.reset();
     mock_client_.reset();
     blink::WebHeap::collectAllGarbageForTesting();
   }
 
   // Creates a WebKit local MediaStream.
   blink::WebMediaStream CreateLocalMediaStream(
       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,
                             blink::WebString::fromUTF8("video_track"),
                             false /* remote */, true /* readonly */);
     MockMediaStreamVideoSource* native_video_source =
         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;
     video_constraints.initialize();
     video_tracks[0] = MediaStreamVideoTrack::CreateVideoTrack(
         native_video_source, video_constraints,
         MediaStreamVideoSource::ConstraintsCallback(), true);
 
     blink::WebMediaStream local_stream;
     local_stream.initialize(base::UTF8ToUTF16(stream_label), audio_tracks,
@@ skipping 12 matching lines @@
         mock_dependency_factory_->CreateLocalMediaStream(stream_label));
     if (!video_track_label.empty()) {
       webrtc::VideoSourceInterface* source = NULL;
       scoped_refptr<webrtc::VideoTrackInterface> video_track(
           mock_dependency_factory_->CreateLocalVideoTrack(
               video_track_label, source));
       stream->AddTrack(video_track.get());
     }
     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());
     return stream;
   }
 
   base::MessageLoop message_loop_;
   scoped_ptr<ChildProcess> child_process_;
   scoped_ptr<MockWebRTCPeerConnectionHandlerClient> mock_client_;
   scoped_ptr<MockPeerConnectionDependencyFactory> mock_dependency_factory_;
@@ skipping 711 matching lines @@
   EXPECT_CALL(*mock_tracker_.get(),
               TrackCreateDTMFSender(pc_handler_.get(),
                                     testing::Ref(tracks[0])));
 
   scoped_ptr<blink::WebRTCDTMFSenderHandler> sender(
       pc_handler_->createDTMFSender(tracks[0]));
   EXPECT_TRUE(sender.get());
 }
 
 }  // namespace content