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