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Side by Side Diff: chrome/browser/media/webrtc_logging_handler_host.h

Issue 305363004: Fix a bug in passing the RTP header from P2PSocketHost to WebRtcLogggingHandlerHost. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 6 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 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 CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_ 5 #ifndef CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_
6 #define CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_ 6 #define CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_
7 7
8 #include "base/basictypes.h" 8 #include "base/basictypes.h"
9 #include "base/memory/shared_memory.h" 9 #include "base/memory/shared_memory.h"
10 #include "chrome/browser/media/rtp_dump_type.h" 10 #include "chrome/browser/media/rtp_dump_type.h"
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85 stop_callback); 85 stop_callback);
86 86
87 // Stops dumping the RTP headers for the specified direction. Must be called 87 // Stops dumping the RTP headers for the specified direction. Must be called
88 // on the IO thread. |type| specifies which direction(s) of RTP packet dumping 88 // on the IO thread. |type| specifies which direction(s) of RTP packet dumping
89 // should be stopped. |callback| will be called when stopping the dump is 89 // should be stopped. |callback| will be called when stopping the dump is
90 // done. 90 // done.
91 void StopRtpDump(RtpDumpType type, const GenericDoneCallback& callback); 91 void StopRtpDump(RtpDumpType type, const GenericDoneCallback& callback);
92 92
93 // Called when an RTP packet is sent or received. Must be called on the UI 93 // Called when an RTP packet is sent or received. Must be called on the UI
94 // thread. 94 // thread.
95 void OnRtpPacket(const uint8* packet_header, 95 void OnRtpPacket(scoped_ptr<uint8[]> packet_header,
96 size_t header_length, 96 size_t header_length,
97 size_t packet_length, 97 size_t packet_length,
98 bool incoming); 98 bool incoming);
99 99
100 private: 100 private:
101 // States used for protecting from function calls made at non-allowed points 101 // States used for protecting from function calls made at non-allowed points
102 // in time. For example, StartLogging() is only allowed in CLOSED state. 102 // in time. For example, StartLogging() is only allowed in CLOSED state.
103 // Transitions: SetMetaData(): CLOSED -> CLOSED. 103 // Transitions: SetMetaData(): CLOSED -> CLOSED.
104 // StartLogging(): CLOSED -> STARTING. 104 // StartLogging(): CLOSED -> STARTING.
105 // Start done: STARTING -> STARTED. 105 // Start done: STARTING -> STARTED.
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209 // The RTP dump handler responsible for creating the RTP header dump files. 209 // The RTP dump handler responsible for creating the RTP header dump files.
210 scoped_ptr<WebRtcRtpDumpHandler> rtp_dump_handler_; 210 scoped_ptr<WebRtcRtpDumpHandler> rtp_dump_handler_;
211 211
212 // The callback to call when StopRtpDump is called. 212 // The callback to call when StopRtpDump is called.
213 content::RenderProcessHost::WebRtcStopRtpDumpCallback stop_rtp_dump_callback_; 213 content::RenderProcessHost::WebRtcStopRtpDumpCallback stop_rtp_dump_callback_;
214 214
215 DISALLOW_COPY_AND_ASSIGN(WebRtcLoggingHandlerHost); 215 DISALLOW_COPY_AND_ASSIGN(WebRtcLoggingHandlerHost);
216 }; 216 };
217 217
218 #endif // CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_ 218 #endif // CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_
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