Move creation of PeerConnection from the RenderView to the RenderThreadImpl.

content/renderer/media/media_stream_impl.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 "content/renderer/media/media_stream_impl.h"
 
 #include <utility>
 
-#include "base/bind.h"
 #include "base/logging.h"
-#include "base/metrics/histogram.h"
 #include "base/string_number_conversions.h"
 #include "base/stringprintf.h"
-#include "base/synchronization/waitable_event.h"
 #include "base/utf_string_conversions.h"
-#include "content/common/child_thread.h"
 #include "content/renderer/media/capture_video_decoder.h"
 #include "content/renderer/media/media_stream_extra_data.h"
+#include "content/renderer/media/media_stream_source_extra_data.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
-#include "content/renderer/media/peer_connection_handler_jsep.h"
+#include "content/renderer/media/rtc_video_decoder.h"
 #include "content/renderer/media/video_capture_impl_manager.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
-#include "content/renderer/p2p/ipc_network_manager.h"
-#include "content/renderer/p2p/ipc_socket_factory.h"
-#include "jingle/glue/thread_wrapper.h"
+#include "content/renderer/media/webrtc_uma_histograms.h"
 #include "media/base/message_loop_factory.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebMediaStreamRegistry.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebSecurityOrigin.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamComponent.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamDescriptor.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebVector.h"
 
 namespace {
 const int kVideoCaptureWidth = 640;
 const int kVideoCaptureHeight = 480;
 const int kVideoCaptureFramePerSecond = 30;
-
-// Helper enum used for histogramming calls to WebRTC APIs from JavaScript.
-enum JavaScriptAPIName {
-  kWebkitGetUserMedia,
-  kWebkitPeerConnection,
-  kInvalidName
-};
 }  // namespace
 
-// Helper method used to collect information about the number of times
-// different WebRTC API:s are called from JavaScript.
-// The histogram can be viewed at chrome://histograms/WebRTC.webkitApiCount.
-static void UpdateWebRTCMethodCount(JavaScriptAPIName api_name) {
-  UMA_HISTOGRAM_ENUMERATION("WebRTC.webkitApiCount", api_name, kInvalidName);
-}
-
 static int g_next_request_id  = 0;
 
-// MediaStreamSourceExtraData contains data stored in the
-// WebKit::WebMediaStreamSource extra data field.
-class MediaStreamSourceExtraData
-    : public WebKit::WebMediaStreamSource::ExtraData {
- public:
-  explicit MediaStreamSourceExtraData(
-      const media_stream::StreamDeviceInfo& device_info)
-      : device_info_(device_info) {
-  }
-  // Return device information about the camera or microphone.
-  const media_stream::StreamDeviceInfo& device_info() const {
-    return device_info_;
-  }
-
- private:
-  media_stream::StreamDeviceInfo device_info_;
-};
-
 // Creates a WebKit representation of a stream sources based on
 // |devices| from the MediaStreamDispatcher.
 static void CreateWebKitSourceVector(
     const std::string& label,
     const media_stream::StreamDeviceInfoArray& devices,
     WebKit::WebMediaStreamSource::Type type,
     WebKit::WebVector<WebKit::WebMediaStreamSource>& webkit_sources) {
-  ASSERT(devices.size() == webkit_sources.size());
+  CHECK_EQ(devices.size(), webkit_sources.size());
   for (size_t i = 0; i < devices.size(); ++i) {
     std::string source_id = StringPrintf("%s%d%u", label.c_str(), type,
                                          static_cast<unsigned int>(i));
     webkit_sources[i].initialize(
           UTF8ToUTF16(source_id),
           type,
           UTF8ToUTF16(devices[i].name));
     webkit_sources[i].setExtraData(
         new MediaStreamSourceExtraData(devices[i]));
   }
 }
 
 MediaStreamImpl::MediaStreamImpl(
     content::RenderView* render_view,
     MediaStreamDispatcher* media_stream_dispatcher,
-    content::P2PSocketDispatcher* p2p_socket_dispatcher,
     VideoCaptureImplManager* vc_manager,
     MediaStreamDependencyFactory* dependency_factory)
     : content::RenderViewObserver(render_view),
       dependency_factory_(dependency_factory),
       media_stream_dispatcher_(media_stream_dispatcher),
-      p2p_socket_dispatcher_(p2p_socket_dispatcher),
-      network_manager_(NULL),
-      vc_manager_(vc_manager),
-      signaling_thread_(NULL),
-      worker_thread_(NULL),
-      chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
+      vc_manager_(vc_manager) {
 }
 
 MediaStreamImpl::~MediaStreamImpl() {
-  CleanupPeerConnectionFactory();
-}
-
-WebKit::WebPeerConnection00Handler*
-MediaStreamImpl::CreatePeerConnectionHandlerJsep(
-    WebKit::WebPeerConnection00HandlerClient* client) {
-  // Save histogram data so we can see how much PeerConnetion is used.
-  // The histogram counts the number of calls to the JS API
-  // webKitPeerConnection00.
-  UpdateWebRTCMethodCount(kWebkitPeerConnection);
-  DCHECK(CalledOnValidThread());
-  if (!EnsurePeerConnectionFactory())
-    return NULL;
-
-  PeerConnectionHandlerJsep* pc_handler = new PeerConnectionHandlerJsep(
-      client,
-      dependency_factory_.get());
-  return pc_handler;
 }
 
 void MediaStreamImpl::StopLocalMediaStream(
     const WebKit::WebMediaStreamDescriptor& stream) {
   DVLOG(1) << "MediaStreamImpl::StopLocalMediaStream";
 
   MediaStreamExtraData* extra_data =
       static_cast<MediaStreamExtraData*>(stream.extraData());
   if (extra_data && extra_data->local_stream()) {
     media_stream_dispatcher_->StopStream(extra_data->local_stream()->label());
     local_media_streams_.erase(extra_data->local_stream()->label());
   } else {
     NOTREACHED();
   }
 }
 
-void MediaStreamImpl::CreateMediaStream(
-    WebKit::WebFrame* frame,
-    WebKit::WebMediaStreamDescriptor* stream) {
-  DVLOG(1) << "MediaStreamImpl::CreateMediaStream";
-
-  if (!CreateNativeLocalMediaStream(stream)) {
-    DVLOG(1) << "Failed to create native stream in CreateMediaStream.";
-    return;
-  }
-}
-
 void MediaStreamImpl::requestUserMedia(
     const WebKit::WebUserMediaRequest& user_media_request,
     const WebKit::WebVector<WebKit::WebMediaStreamSource>& audio_sources,
     const WebKit::WebVector<WebKit::WebMediaStreamSource>& video_sources) {
   // Save histogram data so we can see how much GetUserMedia is used.
   // The histogram counts the number of calls to the JS API
   // webGetUserMedia.
-  UpdateWebRTCMethodCount(kWebkitGetUserMedia);
+  UpdateWebRTCMethodCount(WEBKIT_GET_USER_MEDIA);
   DCHECK(CalledOnValidThread());
   int request_id = g_next_request_id++;
   bool audio = false;
   bool video = false;
   WebKit::WebFrame* frame = NULL;
   GURL security_origin;
 
   // |user_media_request| can't be mocked. So in order to test at all we check
   // if it isNull.
   if (user_media_request.isNull()) {
@@ skipping 94 matching lines @@
     DVLOG(1) << "Request ID not found";
     media_stream_dispatcher_->StopStream(label);
     return;
   }
 
   WebKit::WebString webkit_label = UTF8ToUTF16(label);
   WebKit::WebMediaStreamDescriptor description;
   description.initialize(webkit_label, audio_source_vector,
                          video_source_vector);
 
-  if (!CreateNativeLocalMediaStream(&description)) {
+  if (!dependency_factory_->CreateNativeLocalMediaStream(&description)) {
     DVLOG(1) << "Failed to create native stream in OnStreamGenerated.";
     media_stream_dispatcher_->StopStream(label);
     it->second.request_.requestFailed();
     user_media_requests_.erase(it);
     return;
   }
   local_media_streams_[label] = it->second.frame_;
   CompleteGetUserMediaRequest(description, &it->second.request_);
   user_media_requests_.erase(it);
 }
@@ skipping 85 matching lines @@
                << "Stopping stream " << it->first;
       media_stream_dispatcher_->StopStream(it->first);
       local_media_streams_.erase(it);
       it = local_media_streams_.begin();
     } else {
       ++it;
     }
   }
 }
 
-void MediaStreamImpl::InitializeWorkerThread(talk_base::Thread** thread,
-                                             base::WaitableEvent* event) {
-  jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
-  jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
-  *thread = jingle_glue::JingleThreadWrapper::current();
-  event->Signal();
-}
-
-void MediaStreamImpl::CreateIpcNetworkManagerOnWorkerThread(
-    base::WaitableEvent* event) {
-  DCHECK_EQ(MessageLoop::current(), chrome_worker_thread_.message_loop());
-  network_manager_ = new content::IpcNetworkManager(p2p_socket_dispatcher_);
-  event->Signal();
-}
-
-void MediaStreamImpl::DeleteIpcNetworkManager() {
-  DCHECK_EQ(MessageLoop::current(), chrome_worker_thread_.message_loop());
-  delete network_manager_;
-  network_manager_ = NULL;
-}
-
-bool MediaStreamImpl::EnsurePeerConnectionFactory() {
-  DCHECK(CalledOnValidThread());
-  if (!signaling_thread_) {
-    jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
-    jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
-    signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
-  }
-
-  if (!worker_thread_) {
-    if (!chrome_worker_thread_.IsRunning()) {
-      if (!chrome_worker_thread_.Start()) {
-        LOG(ERROR) << "Could not start worker thread";
-        signaling_thread_ = NULL;
-        return false;
-      }
-    }
-    base::WaitableEvent event(true, false);
-    chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
-        &MediaStreamImpl::InitializeWorkerThread,
-        base::Unretained(this),
-        &worker_thread_,
-        &event));
-    event.Wait();
-    DCHECK(worker_thread_);
-  }
-
-  if (!network_manager_) {
-    base::WaitableEvent event(true, false);
-    chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
-          &MediaStreamImpl::CreateIpcNetworkManagerOnWorkerThread,
-          base::Unretained(this),
-          &event));
-    event.Wait();
-  }
-
-  if (!socket_factory_.get()) {
-    socket_factory_.reset(
-        new content::IpcPacketSocketFactory(p2p_socket_dispatcher_));
-  }
-
-  if (!dependency_factory_->PeerConnectionFactoryCreated()) {
-    if (!dependency_factory_->CreatePeerConnectionFactory(
-            worker_thread_,
-            signaling_thread_,
-            p2p_socket_dispatcher_,
-            network_manager_,
-            socket_factory_.get())) {
-      LOG(ERROR) << "Could not create PeerConnection factory";
-      return false;
-    }
-  }
-
-  return true;
-}
-
-void MediaStreamImpl::CleanupPeerConnectionFactory() {
-  if (dependency_factory_.get())
-    dependency_factory_->ReleasePeerConnectionFactory();
-  if (network_manager_) {
-    // The network manager needs to free its resources on the thread they were
-    // created, which is the worked thread.
-    if (chrome_worker_thread_.IsRunning()) {
-      chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
-          &MediaStreamImpl::DeleteIpcNetworkManager,
-          base::Unretained(this)));
-      // Stopping the thread will wait until all tasks have been
-      // processed before returning. We wait for the above task to finish before
-      // letting the the function continue to avoid any potential race issues.
-      chrome_worker_thread_.Stop();
-    } else {
-      NOTREACHED() << "Worker thread not running.";
-    }
-  }
-}
-
 scoped_refptr<media::VideoDecoder> MediaStreamImpl::CreateLocalVideoDecoder(
     webrtc::MediaStreamInterface* stream,
     media::MessageLoopFactory* message_loop_factory) {
   if (!stream->video_tracks() || stream->video_tracks()->count() == 0)
     return NULL;
 
   int video_session_id =
       media_stream_dispatcher_->video_session_id(stream->label(), 0);
   media::VideoCaptureCapability capability;
   capability.width = kVideoCaptureWidth;
@@ skipping 22 matching lines @@
 
   DVLOG(1) << "MediaStreamImpl::CreateRemoteVideoDecoder label:"
            << stream->label();
 
   return new RTCVideoDecoder(
       message_loop_factory->GetMessageLoop(media::MessageLoopFactory::kDecoder),
       base::MessageLoopProxy::current(),
       stream->video_tracks()->at(0));
 }
 
-bool MediaStreamImpl::CreateNativeLocalMediaStream(
-    WebKit::WebMediaStreamDescriptor* description) {
-  // Creating the peer connection factory can fail if for example the audio
-  // (input or output) or video device cannot be opened. Handling such cases
-  // better is a higher level design discussion which involves the media
-  // manager, webrtc and libjingle. We cannot create any native
-  // track objects however, so we'll just have to skip that. Furthermore,
-  // creating a peer connection later on will fail if we don't have a factory.
-  if (!EnsurePeerConnectionFactory())
-    return false;
-
-  std::string label = UTF16ToUTF8(description->label());
-  LocalNativeStreamPtr native_stream =
-      dependency_factory_->CreateLocalMediaStream(label);
-
-  // Add audio tracks.
-  WebKit::WebVector<WebKit::WebMediaStreamComponent> audio_components;
-  description->audioSources(audio_components);
-  for (size_t i = 0; i < audio_components.size(); ++i) {
-    const WebKit::WebMediaStreamSource& source = audio_components[i].source();
-    MediaStreamSourceExtraData* source_data =
-        static_cast<MediaStreamSourceExtraData*>(source.extraData());
-    if (!source_data) {
-      // TODO(perkj): Implement support for sources from remote MediaStreams.
-      NOTIMPLEMENTED();
-      continue;
-    }
-    // TODO(perkj): Refactor the creation of audio tracks to use a proper
-    // interface for receiving audio input data. Currently NULL is passed since
-    // the |audio_device| is the wrong class and is unused.
-    talk_base::scoped_refptr<webrtc::LocalAudioTrackInterface> audio_track(
-        dependency_factory_->CreateLocalAudioTrack(
-            UTF16ToUTF8(source.id()), NULL));
-    native_stream->AddTrack(audio_track);
-    audio_track->set_enabled(audio_components[i].isEnabled());
-    // TODO(xians): This set the source of all audio tracks to the same
-    // microphone. Implement support for setting the source per audio track
-    // instead.
-    dependency_factory_->SetAudioDeviceSessionId(
-        source_data->device_info().session_id);
-  }
-
-  // Add video tracks.
-  WebKit::WebVector<WebKit::WebMediaStreamComponent> video_components;
-  description->videoSources(video_components);
-  for (size_t i = 0; i < video_components.size(); ++i) {
-    const WebKit::WebMediaStreamSource& source = video_components[i].source();
-    MediaStreamSourceExtraData* source_data =
-        static_cast<MediaStreamSourceExtraData*>(source.extraData());
-    if (!source_data) {
-      // TODO(perkj): Implement support for sources from remote MediaStreams.
-      NOTIMPLEMENTED();
-      continue;
-    }
-    talk_base::scoped_refptr<webrtc::LocalVideoTrackInterface> video_track(
-        dependency_factory_->CreateLocalVideoTrack(
-            UTF16ToUTF8(source.id()), source_data->device_info().session_id));
-    native_stream->AddTrack(video_track);
-    video_track->set_enabled(video_components[i].isEnabled());
-  }
-
-  description->setExtraData(new MediaStreamExtraData(native_stream));
-
-  return true;
-}
-
 MediaStreamExtraData::MediaStreamExtraData(
     webrtc::MediaStreamInterface* remote_stream)
     : remote_stream_(remote_stream) {
 }
 MediaStreamExtraData::MediaStreamExtraData(
     webrtc::LocalMediaStreamInterface* local_stream)
     : local_stream_(local_stream) {
 }
 MediaStreamExtraData::~MediaStreamExtraData() {
 }