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| 1 /* | 1 /* |
| 2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_MEDIA_BASE_RTPDUMP_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_RTPDUMP_H_ |
| 12 #define WEBRTC_MEDIA_BASE_RTPDUMP_H_ | 12 #define WEBRTC_MEDIA_BASE_RTPDUMP_H_ |
| 13 | 13 |
| 14 #include <string.h> | 14 #include <string.h> |
| 15 | 15 |
| 16 #include <string> | 16 #include <string> |
| 17 #include <vector> | 17 #include <vector> |
| 18 | 18 |
| 19 #include "webrtc/base/basictypes.h" | 19 #include "webrtc/base/basictypes.h" |
| 20 #include "webrtc/base/bytebuffer.h" | 20 #include "webrtc/base/bytebuffer.h" |
| 21 #include "webrtc/base/constructormagic.h" |
| 21 #include "webrtc/base/stream.h" | 22 #include "webrtc/base/stream.h" |
| 22 | 23 |
| 23 namespace cricket { | 24 namespace cricket { |
| 24 | 25 |
| 25 // We use the RTP dump file format compatible to the format used by rtptools | 26 // We use the RTP dump file format compatible to the format used by rtptools |
| 26 // (http://www.cs.columbia.edu/irt/software/rtptools/) and Wireshark | 27 // (http://www.cs.columbia.edu/irt/software/rtptools/) and Wireshark |
| 27 // (http://wiki.wireshark.org/rtpdump). In particular, the file starts with the | 28 // (http://wiki.wireshark.org/rtpdump). In particular, the file starts with the |
| 28 // first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header. | 29 // first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header. |
| 29 // For each packet, the file contains a 8 byte dump packet header, followed by | 30 // For each packet, the file contains a 8 byte dump packet header, followed by |
| 30 // the actual RTP or RTCP packet. | 31 // the actual RTP or RTCP packet. |
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| 208 bool file_header_written_; | 209 bool file_header_written_; |
| 209 uint32_t start_time_ms_; // Time when the record starts. | 210 uint32_t start_time_ms_; // Time when the record starts. |
| 210 // If writing to the stream takes longer than this many ms, log a warning. | 211 // If writing to the stream takes longer than this many ms, log a warning. |
| 211 uint32_t warn_slow_writes_delay_; | 212 uint32_t warn_slow_writes_delay_; |
| 212 RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter); | 213 RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter); |
| 213 }; | 214 }; |
| 214 | 215 |
| 215 } // namespace cricket | 216 } // namespace cricket |
| 216 | 217 |
| 217 #endif // WEBRTC_MEDIA_BASE_RTPDUMP_H_ | 218 #endif // WEBRTC_MEDIA_BASE_RTPDUMP_H_ |
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