15583c19d7
The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545}
135 lines
3.9 KiB
C++
135 lines
3.9 KiB
C++
/*
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* Copyright 2012 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_
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#define WEBRTC_EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_
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#pragma once
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#include <deque>
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#include <map>
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#include <set>
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#include <string>
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/peerconnectioninterface.h"
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#include "webrtc/examples/peerconnection/client/main_wnd.h"
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#include "webrtc/examples/peerconnection/client/peer_connection_client.h"
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#include "webrtc/base/scoped_ptr.h"
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namespace webrtc {
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class VideoCaptureModule;
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} // namespace webrtc
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namespace cricket {
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class VideoRenderer;
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} // namespace cricket
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class Conductor
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: public webrtc::PeerConnectionObserver,
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public webrtc::CreateSessionDescriptionObserver,
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public PeerConnectionClientObserver,
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public MainWndCallback {
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public:
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enum CallbackID {
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MEDIA_CHANNELS_INITIALIZED = 1,
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PEER_CONNECTION_CLOSED,
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SEND_MESSAGE_TO_PEER,
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NEW_STREAM_ADDED,
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STREAM_REMOVED,
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};
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Conductor(PeerConnectionClient* client, MainWindow* main_wnd);
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bool connection_active() const;
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virtual void Close();
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protected:
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~Conductor();
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bool InitializePeerConnection();
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bool ReinitializePeerConnectionForLoopback();
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bool CreatePeerConnection(bool dtls);
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void DeletePeerConnection();
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void EnsureStreamingUI();
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void AddStreams();
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cricket::VideoCapturer* OpenVideoCaptureDevice();
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//
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// PeerConnectionObserver implementation.
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//
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void OnSignalingChange(
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webrtc::PeerConnectionInterface::SignalingState new_state) override{};
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void OnAddStream(webrtc::MediaStreamInterface* stream) override;
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void OnRemoveStream(webrtc::MediaStreamInterface* stream) override;
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void OnDataChannel(webrtc::DataChannelInterface* channel) override {}
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void OnRenegotiationNeeded() override {}
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void OnIceConnectionChange(
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webrtc::PeerConnectionInterface::IceConnectionState new_state) override{};
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void OnIceGatheringChange(
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webrtc::PeerConnectionInterface::IceGatheringState new_state) override{};
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void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
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void OnIceConnectionReceivingChange(bool receiving) override {}
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//
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// PeerConnectionClientObserver implementation.
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//
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virtual void OnSignedIn();
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virtual void OnDisconnected();
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virtual void OnPeerConnected(int id, const std::string& name);
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virtual void OnPeerDisconnected(int id);
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virtual void OnMessageFromPeer(int peer_id, const std::string& message);
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virtual void OnMessageSent(int err);
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virtual void OnServerConnectionFailure();
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//
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// MainWndCallback implementation.
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//
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virtual void StartLogin(const std::string& server, int port);
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virtual void DisconnectFromServer();
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virtual void ConnectToPeer(int peer_id);
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virtual void DisconnectFromCurrentPeer();
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virtual void UIThreadCallback(int msg_id, void* data);
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// CreateSessionDescriptionObserver implementation.
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virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc);
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virtual void OnFailure(const std::string& error);
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protected:
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// Send a message to the remote peer.
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void SendMessage(const std::string& json_object);
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int peer_id_;
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bool loopback_;
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rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
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rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
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peer_connection_factory_;
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PeerConnectionClient* client_;
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MainWindow* main_wnd_;
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std::deque<std::string*> pending_messages_;
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std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> >
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active_streams_;
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std::string server_;
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};
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#endif // WEBRTC_EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_
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