Interface with webrtc through encoded frames

remoting/protocol/webrtc_transport.cc

Index: remoting/protocol/webrtc_transport.cc
diff --git a/remoting/protocol/webrtc_transport.cc b/remoting/protocol/webrtc_transport.cc
index 9a6d243fe6acd8439b74053871752c51f7149253..a99d7f07ce0cf367dfef6f09b88526cc47008d56 100644
--- a/remoting/protocol/webrtc_transport.cc
+++ b/remoting/protocol/webrtc_transport.cc
@@ -4,6 +4,7 @@
 
 #include "remoting/protocol/webrtc_transport.h"
 
+#include <string>
 #include <utility>
 
 #include "base/base64.h"
@@ -15,11 +16,13 @@
 #include "base/task_runner_util.h"
 #include "base/thread_task_runner_handle.h"
 #include "jingle/glue/thread_wrapper.h"
+#include "remoting/codec/webrtc_video_encoder.h"
 #include "remoting/protocol/authenticator.h"
 #include "remoting/protocol/port_allocator_factory.h"
 #include "remoting/protocol/stream_message_pipe_adapter.h"
 #include "remoting/protocol/transport_context.h"
 #include "third_party/webrtc/api/test/fakeconstraints.h"
+#include "third_party/webrtc/base/stringutils.h"
 #include "third_party/webrtc/libjingle/xmllite/xmlelement.h"
 #include "third_party/webrtc/modules/audio_device/include/fake_audio_device.h"
 
@@ -102,7 +105,7 @@ class SetSessionDescriptionObserver
   }
 
  protected:
-  SetSessionDescriptionObserver(const ResultCallback& result_callback)
+  explicit SetSessionDescriptionObserver(const ResultCallback& result_callback)
       : result_callback_(result_callback) {}
   ~SetSessionDescriptionObserver() override {}
 
@@ -153,10 +156,12 @@ void WebrtcTransport::Start(
   }
 
   fake_audio_device_module_.reset(new webrtc::FakeAudioDeviceModule());
+  video_encoder_factory_ = new remoting::WebRtcVideoEncoderFactory();
 
+  // Takes ownership of video_encoder_factory_
   peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
-      worker_thread_, rtc::Thread::Current(),
-      fake_audio_device_module_.get(), nullptr, nullptr);
+      worker_thread_, rtc::Thread::Current(), fake_audio_device_module_.get(),
+      video_encoder_factory_, nullptr);
 
   webrtc::PeerConnectionInterface::IceServer stun_server;
   stun_server.urls.push_back("stun:stun.l.google.com:19302");
@@ -229,6 +234,17 @@ bool WebrtcTransport::ProcessTransportInfo(XmlElement* transport_info) {
       }
     }
 
+    // TODO(isheriff): The need for this should go away once we have a proper
+    // API to provide max bitrate for the case of handing over encoded
+    // frames to webrtc.
+    // Increase max bitrate from 600 kbps to 5 Mbps.
+    std::string vp8line = "a=rtpmap:100 VP8/90000\n";

[Sergey Ulanov, 2016/03/31 22:06:19]

I think webrtc generates SDP with '\r\n' at the end of each line. Does this code
work properly for you?

This was causing troubles sometimes as \r gets removed sometimes, landed this
workaround today: https://codereview.chromium.org/1843333002/ . We will be
getting \n here after that CL.

[Irfan, 2016/04/05 21:23:28]

It was working for me.

+    std::string vp8line_with_bitrate =
+        vp8line + "a=fmtp:100 x-google-max-bitrate=5000\n";
+    rtc::replace_substrs(vp8line.c_str(), vp8line.length(),

[Sergey Ulanov, 2016/03/31 22:06:19]

Please use base::ReplaceSubstringsAfterOffset:
 base::ReplaceSubstringsAfterOffset(&sdp, 0, vp8line, vp8line_with_bitrate);

[Irfan, 2016/04/05 21:23:28]

Done.

+                         vp8line_with_bitrate.c_str(),
+                         vp8line_with_bitrate.length(), &sdp);
+
     webrtc::SdpParseError error;
     scoped_ptr<webrtc::SessionDescriptionInterface> session_description(
         webrtc::CreateSessionDescription(type, sdp, &error));