Interface with webrtc through encoded frames

remoting/protocol/webrtc_transport.cc

 // Copyright 2015 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 "remoting/protocol/webrtc_transport.h"
 
+#include <string>
 #include <utility>
 
 #include "base/base64.h"
 #include "base/callback_helpers.h"
 #include "base/command_line.h"
 #include "base/macros.h"
 #include "base/single_thread_task_runner.h"
 #include "base/strings/string_number_conversions.h"
 #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"
 
 using buzz::QName;
 using buzz::XmlElement;
 
 namespace remoting {
 namespace protocol {
 
 namespace {
@@ skipping 62 matching lines @@
 
   void OnSuccess() override {
     base::ResetAndReturn(&result_callback_).Run(true, std::string());
   }
 
   void OnFailure(const std::string& error) override {
     base::ResetAndReturn(&result_callback_).Run(false, error);
   }
 
  protected:
-  SetSessionDescriptionObserver(const ResultCallback& result_callback)
+  explicit SetSessionDescriptionObserver(const ResultCallback& result_callback)
       : result_callback_(result_callback) {}
   ~SetSessionDescriptionObserver() override {}
 
  private:
   ResultCallback result_callback_;
 
   DISALLOW_COPY_AND_ASSIGN(SetSessionDescriptionObserver);
 };
 
 
@@ skipping 30 matching lines @@
   // TODO(sergeyu): Investigate if it's possible to avoid Send().
   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 
   send_transport_info_callback_ = std::move(send_transport_info_callback);
 
   if (!handshake_hmac_.Init(authenticator->GetAuthKey())) {
     LOG(FATAL) << "HMAC::Init() failed.";
   }
 
   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");
   webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.servers.push_back(stun_server);
 
   webrtc::FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            webrtc::MediaConstraintsInterface::kValueTrue);
 
@@ skipping 52 matching lines @@
 #if !defined(NDEBUG)
       ignore_error = base::CommandLine::ForCurrentProcess()->HasSwitch(
           kDisableAuthenticationSwitchName);
 #endif
       if (!ignore_error) {
         Close(AUTHENTICATION_FAILED);
         return true;
       }
     }
 
+    // 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));
     if (!session_description) {
       LOG(ERROR) << "Failed to parse the session description: "
                  << error.description << " line: " << error.line;
       return false;
     }
 
     peer_connection_->SetRemoteDescription(
@@ skipping 287 matching lines @@
   peer_connection_->Close();
   peer_connection_ = nullptr;
   peer_connection_factory_ = nullptr;
 
   if (error != OK)
     event_handler_->OnWebrtcTransportError(error);
 }
 
 }  // namespace protocol
 }  // namespace remoting