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Side by Side Diff: content/renderer/media/rtc_peer_connection_handler.cc

Issue 14200016: Added implementation of RemoteMediaStreams. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Fix nits Created 7 years, 8 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "content/renderer/media/rtc_peer_connection_handler.h" 5 #include "content/renderer/media/rtc_peer_connection_handler.h"
6 6
7 #include <string> 7 #include <string>
8 #include <utility> 8 #include <utility>
9 #include <vector> 9 #include <vector>
10 10
11 #include "base/logging.h" 11 #include "base/logging.h"
12 #include "base/memory/scoped_ptr.h" 12 #include "base/memory/scoped_ptr.h"
13 #include "base/stl_util.h"
13 #include "base/utf_string_conversions.h" 14 #include "base/utf_string_conversions.h"
14 #include "content/renderer/media/media_stream_dependency_factory.h" 15 #include "content/renderer/media/media_stream_dependency_factory.h"
15 #include "content/renderer/media/peer_connection_tracker.h" 16 #include "content/renderer/media/peer_connection_tracker.h"
17 #include "content/renderer/media/remote_media_stream_impl.h"
16 #include "content/renderer/media/rtc_data_channel_handler.h" 18 #include "content/renderer/media/rtc_data_channel_handler.h"
17 #include "content/renderer/media/rtc_dtmf_sender_handler.h" 19 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
18 #include "content/renderer/media/rtc_media_constraints.h" 20 #include "content/renderer/media/rtc_media_constraints.h"
19 #include "content/renderer/render_thread_impl.h" 21 #include "content/renderer/render_thread_impl.h"
20 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints .h" 22 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints .h"
21 // TODO(hta): Move the following include to WebRTCStatsRequest.h file. 23 // TODO(hta): Move the following include to WebRTCStatsRequest.h file.
22 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSourc e.h" 24 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSourc e.h"
23 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack .h" 25 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack .h"
24 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCConfiguration .h" 26 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCConfiguration .h"
25 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCICECandidate. h" 27 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCICECandidate. h"
(...skipping 290 matching lines...) Expand 10 before | Expand all | Expand 10 after
316 MediaStreamDependencyFactory* dependency_factory) 318 MediaStreamDependencyFactory* dependency_factory)
317 : PeerConnectionHandlerBase(dependency_factory), 319 : PeerConnectionHandlerBase(dependency_factory),
318 client_(client), 320 client_(client),
319 frame_(NULL), 321 frame_(NULL),
320 peer_connection_tracker_(NULL) { 322 peer_connection_tracker_(NULL) {
321 } 323 }
322 324
323 RTCPeerConnectionHandler::~RTCPeerConnectionHandler() { 325 RTCPeerConnectionHandler::~RTCPeerConnectionHandler() {
324 if (peer_connection_tracker_) 326 if (peer_connection_tracker_)
325 peer_connection_tracker_->UnregisterPeerConnection(this); 327 peer_connection_tracker_->UnregisterPeerConnection(this);
328 STLDeleteValues(&remote_streams_);
326 } 329 }
327 330
328 void RTCPeerConnectionHandler::associateWithFrame(WebKit::WebFrame* frame) { 331 void RTCPeerConnectionHandler::associateWithFrame(WebKit::WebFrame* frame) {
329 DCHECK(frame); 332 DCHECK(frame);
330 frame_ = frame; 333 frame_ = frame;
331 } 334 }
332 335
333 bool RTCPeerConnectionHandler::initialize( 336 bool RTCPeerConnectionHandler::initialize(
334 const WebKit::WebRTCConfiguration& server_configuration, 337 const WebKit::WebRTCConfiguration& server_configuration,
335 const WebKit::WebMediaConstraints& options) { 338 const WebKit::WebMediaConstraints& options) {
(...skipping 317 matching lines...) Expand 10 before | Expand all | Expand 10 after
653 GetWebKitIceGatheringState(new_state); 656 GetWebKitIceGatheringState(new_state);
654 if (peer_connection_tracker_) 657 if (peer_connection_tracker_)
655 peer_connection_tracker_->TrackIceGatheringStateChange(this, state); 658 peer_connection_tracker_->TrackIceGatheringStateChange(this, state);
656 client_->didChangeICEGatheringState(state); 659 client_->didChangeICEGatheringState(state);
657 } 660 }
658 661
659 void RTCPeerConnectionHandler::OnAddStream( 662 void RTCPeerConnectionHandler::OnAddStream(
660 webrtc::MediaStreamInterface* stream_interface) { 663 webrtc::MediaStreamInterface* stream_interface) {
661 DCHECK(stream_interface); 664 DCHECK(stream_interface);
662 DCHECK(remote_streams_.find(stream_interface) == remote_streams_.end()); 665 DCHECK(remote_streams_.find(stream_interface) == remote_streams_.end());
663 WebKit::WebMediaStream stream = 666
664 CreateRemoteWebKitMediaStream(stream_interface); 667 RemoteMediaStreamImpl* remote_stream =
668 new RemoteMediaStreamImpl(stream_interface);
669 remote_streams_.insert(
670 std::pair<webrtc::MediaStreamInterface*, RemoteMediaStreamImpl*> (
671 stream_interface, remote_stream));
665 672
666 if (peer_connection_tracker_) 673 if (peer_connection_tracker_)
667 peer_connection_tracker_->TrackAddStream( 674 peer_connection_tracker_->TrackAddStream(
668 this, stream, PeerConnectionTracker::SOURCE_REMOTE); 675 this, remote_stream->webkit_stream(),
676 PeerConnectionTracker::SOURCE_REMOTE);
669 677
670 remote_streams_.insert( 678 client_->didAddRemoteStream(remote_stream->webkit_stream());
671 std::pair<webrtc::MediaStreamInterface*,
672 WebKit::WebMediaStream>(stream_interface, stream));
673 client_->didAddRemoteStream(stream);
674 } 679 }
675 680
676 void RTCPeerConnectionHandler::OnRemoveStream( 681 void RTCPeerConnectionHandler::OnRemoveStream(
677 webrtc::MediaStreamInterface* stream_interface) { 682 webrtc::MediaStreamInterface* stream_interface) {
678 DCHECK(stream_interface); 683 DCHECK(stream_interface);
679 RemoteStreamMap::iterator it = remote_streams_.find(stream_interface); 684 RemoteStreamMap::iterator it = remote_streams_.find(stream_interface);
680 if (it == remote_streams_.end()) { 685 if (it == remote_streams_.end()) {
681 NOTREACHED() << "Stream not found"; 686 NOTREACHED() << "Stream not found";
682 return; 687 return;
683 } 688 }
684 WebKit::WebMediaStream stream = it->second; 689
685 DCHECK(!stream.isNull()); 690 scoped_ptr<RemoteMediaStreamImpl> remote_stream(it->second);
691 const WebKit::WebMediaStream& webkit_stream = remote_stream->webkit_stream();
692 DCHECK(!webkit_stream.isNull());
686 remote_streams_.erase(it); 693 remote_streams_.erase(it);
687 694
688 if (peer_connection_tracker_) 695 if (peer_connection_tracker_)
689 peer_connection_tracker_->TrackRemoveStream( 696 peer_connection_tracker_->TrackRemoveStream(
690 this, stream, PeerConnectionTracker::SOURCE_REMOTE); 697 this, webkit_stream, PeerConnectionTracker::SOURCE_REMOTE);
691 698
692 client_->didRemoveRemoteStream(stream); 699 client_->didRemoveRemoteStream(webkit_stream);
693 } 700 }
694 701
695 void RTCPeerConnectionHandler::OnIceCandidate( 702 void RTCPeerConnectionHandler::OnIceCandidate(
696 const webrtc::IceCandidateInterface* candidate) { 703 const webrtc::IceCandidateInterface* candidate) {
697 DCHECK(candidate); 704 DCHECK(candidate);
698 std::string sdp; 705 std::string sdp;
699 if (!candidate->ToString(&sdp)) { 706 if (!candidate->ToString(&sdp)) {
700 NOTREACHED() << "OnIceCandidate: Could not get SDP string."; 707 NOTREACHED() << "OnIceCandidate: Could not get SDP string.";
701 return; 708 return;
702 } 709 }
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
741 webrtc::SessionDescriptionInterface* native_desc = 748 webrtc::SessionDescriptionInterface* native_desc =
742 dependency_factory_->CreateSessionDescription(type, sdp, error); 749 dependency_factory_->CreateSessionDescription(type, sdp, error);
743 750
744 LOG_IF(ERROR, !native_desc) << "Failed to create native session description." 751 LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
745 << " Type: " << type << " SDP: " << sdp; 752 << " Type: " << type << " SDP: " << sdp;
746 753
747 return native_desc; 754 return native_desc;
748 } 755 }
749 756
750 } // namespace content 757 } // namespace content
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