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1 /* | 1 /* |
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2011 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 |
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44 // from 0 after wrap-around. | 44 // from 0 after wrap-around. |
45 uint16_t base_seq_no_; // The sequence number of the first received packet. | 45 uint16_t base_seq_no_; // The sequence number of the first received packet. |
46 uint32_t received_packets_; // The number of packets that have been received. | 46 uint32_t received_packets_; // The number of packets that have been received. |
47 uint32_t received_packets_prior_; // Number of packets received when last | 47 uint32_t received_packets_prior_; // Number of packets received when last |
48 // report was generated. | 48 // report was generated. |
49 uint32_t expected_prior_; // Expected number of packets, at the time of the | 49 uint32_t expected_prior_; // Expected number of packets, at the time of the |
50 // last report. | 50 // last report. |
51 uint32_t jitter_; // Current jitter value. | 51 uint32_t jitter_; // Current jitter value. |
52 int32_t transit_; // Clock difference for previous packet. | 52 int32_t transit_; // Clock difference for previous packet. |
53 | 53 |
54 DISALLOW_COPY_AND_ASSIGN(Rtcp); | 54 RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp); |
55 }; | 55 }; |
56 | 56 |
57 } // namespace webrtc | 57 } // namespace webrtc |
58 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ | 58 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_ |
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