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

Issue 1855193002: Move the call to enable the WebRTC event log from PeerConnectionFactory to PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Merge with existing WebRtcEventLogHandler. Created 4 years, 7 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 #ifndef CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
6 #define CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_ 6 #define CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
7 7
8 #include <stddef.h> 8 #include <stddef.h>
9 9
10 #include <map> 10 #include <map>
11 #include <memory> 11 #include <memory>
12 #include <string> 12 #include <string>
13 13
14 #include "base/compiler_specific.h" 14 #include "base/compiler_specific.h"
15 #include "base/macros.h" 15 #include "base/macros.h"
16 #include "base/memory/ref_counted.h" 16 #include "base/memory/ref_counted.h"
17 #include "base/memory/weak_ptr.h" 17 #include "base/memory/weak_ptr.h"
18 #include "base/single_thread_task_runner.h" 18 #include "base/single_thread_task_runner.h"
19 #include "base/threading/thread.h" 19 #include "base/threading/thread.h"
20 #include "base/threading/thread_checker.h" 20 #include "base/threading/thread_checker.h"
21 #include "content/common/content_export.h" 21 #include "content/common/content_export.h"
22 #include "content/renderer/media/webrtc/media_stream_track_metrics.h" 22 #include "content/renderer/media/webrtc/media_stream_track_metrics.h"
23 #include "ipc/ipc_platform_file.h"
23 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" 24 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
24 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandler.h" 25 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandler.h"
25 #include "third_party/WebKit/public/platform/WebRTCStatsRequest.h" 26 #include "third_party/WebKit/public/platform/WebRTCStatsRequest.h"
26 #include "third_party/WebKit/public/platform/WebRTCStatsResponse.h" 27 #include "third_party/WebKit/public/platform/WebRTCStatsResponse.h"
27 28
28 namespace blink { 29 namespace blink {
29 class WebFrame; 30 class WebFrame;
30 class WebRTCAnswerOptions; 31 class WebRTCAnswerOptions;
31 class WebRTCDataChannelHandler; 32 class WebRTCDataChannelHandler;
32 class WebRTCOfferOptions; 33 class WebRTCOfferOptions;
(...skipping 126 matching lines...) Expand 10 before | Expand all | Expand 10 after
159 // Asynchronously calls native_peer_connection_->getStats on the signaling 160 // Asynchronously calls native_peer_connection_->getStats on the signaling
160 // thread. If the |track_id| is empty, the |track_type| parameter is ignored. 161 // thread. If the |track_id| is empty, the |track_type| parameter is ignored.
161 void GetStats(webrtc::StatsObserver* observer, 162 void GetStats(webrtc::StatsObserver* observer,
162 webrtc::PeerConnectionInterface::StatsOutputLevel level, 163 webrtc::PeerConnectionInterface::StatsOutputLevel level,
163 const std::string& track_id, 164 const std::string& track_id,
164 blink::WebMediaStreamSource::Type track_type); 165 blink::WebMediaStreamSource::Type track_type);
165 166
166 // Tells the |client_| to close RTCPeerConnection. 167 // Tells the |client_| to close RTCPeerConnection.
167 void CloseClientPeerConnection(); 168 void CloseClientPeerConnection();
168 169
170 // Start recording an event log.
171 void StartEventLog(IPC::PlatformFileForTransit file,
172 int64_t max_file_size_bytes);
173 // Stop recording an event log.
174 void StopEventLog();
175
169 protected: 176 protected:
170 webrtc::PeerConnectionInterface* native_peer_connection() { 177 webrtc::PeerConnectionInterface* native_peer_connection() {
171 return native_peer_connection_.get(); 178 return native_peer_connection_.get();
172 } 179 }
173 180
174 class Observer; 181 class Observer;
175 friend class Observer; 182 friend class Observer;
176 183
177 void OnSignalingChange( 184 void OnSignalingChange(
178 webrtc::PeerConnectionInterface::SignalingState new_state); 185 webrtc::PeerConnectionInterface::SignalingState new_state);
(...skipping 99 matching lines...) Expand 10 before | Expand all | Expand 10 after
278 bool ice_state_seen_[webrtc::PeerConnectionInterface::kIceConnectionMax] = {}; 285 bool ice_state_seen_[webrtc::PeerConnectionInterface::kIceConnectionMax] = {};
279 286
280 base::WeakPtrFactory<RTCPeerConnectionHandler> weak_factory_; 287 base::WeakPtrFactory<RTCPeerConnectionHandler> weak_factory_;
281 288
282 DISALLOW_COPY_AND_ASSIGN(RTCPeerConnectionHandler); 289 DISALLOW_COPY_AND_ASSIGN(RTCPeerConnectionHandler);
283 }; 290 };
284 291
285 } // namespace content 292 } // namespace content
286 293
287 #endif // CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_ 294 #endif // CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
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