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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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| 92 int SetSendAbsoluteSendTimeStatus(bool enable, int id); | 92 int SetSendAbsoluteSendTimeStatus(bool enable, int id); |
| 93 int SetSendVideoRotationStatus(bool enable, int id); | 93 int SetSendVideoRotationStatus(bool enable, int id); |
| 94 int SetSendTransportSequenceNumber(bool enable, int id); | 94 int SetSendTransportSequenceNumber(bool enable, int id); |
| 95 | 95 |
| 96 RtpState GetRtpStateForSsrc(uint32_t ssrc) const; | 96 RtpState GetRtpStateForSsrc(uint32_t ssrc) const; |
| 97 | 97 |
| 98 // Gets send statistics for the rtp and rtx stream. | 98 // Gets send statistics for the rtp and rtx stream. |
| 99 void GetSendStreamDataCounters(StreamDataCounters* rtp_counters, | 99 void GetSendStreamDataCounters(StreamDataCounters* rtp_counters, |
| 100 StreamDataCounters* rtx_counters) const; | 100 StreamDataCounters* rtx_counters) const; |
| 101 | 101 |
| 102 // Gets received stream data counters. | |
| 103 void GetReceiveStreamDataCounters(StreamDataCounters* rtp_counters, | |
| 104 StreamDataCounters* rtx_counters) const; | |
| 105 | |
| 106 void GetSendRtcpPacketTypeCounter( | 102 void GetSendRtcpPacketTypeCounter( |
| 107 RtcpPacketTypeCounter* packet_counter) const; | 103 RtcpPacketTypeCounter* packet_counter) const; |
| 108 | 104 |
| 109 void GetReceiveRtcpPacketTypeCounter( | 105 void GetReceiveRtcpPacketTypeCounter( |
| 110 RtcpPacketTypeCounter* packet_counter) const; | 106 RtcpPacketTypeCounter* packet_counter) const; |
| 111 | 107 |
| 112 void RegisterSendSideDelayObserver(SendSideDelayObserver* observer); | 108 void RegisterSendSideDelayObserver(SendSideDelayObserver* observer); |
| 113 | 109 |
| 114 // Called on any new send bitrate estimate. | 110 // Called on any new send bitrate estimate. |
| 115 void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer); | 111 void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer); |
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| 336 int64_t last_rtt_ms_ GUARDED_BY(crit_); | 332 int64_t last_rtt_ms_ GUARDED_BY(crit_); |
| 337 | 333 |
| 338 // RtpRtcp modules, declared last as they use other members on construction. | 334 // RtpRtcp modules, declared last as they use other members on construction. |
| 339 const std::vector<RtpRtcp*> rtp_rtcp_modules_; | 335 const std::vector<RtpRtcp*> rtp_rtcp_modules_; |
| 340 size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_); | 336 size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_); |
| 341 }; | 337 }; |
| 342 | 338 |
| 343 } // namespace webrtc | 339 } // namespace webrtc |
| 344 | 340 |
| 345 #endif // WEBRTC_VIDEO_VIE_CHANNEL_H_ | 341 #endif // WEBRTC_VIDEO_VIE_CHANNEL_H_ |
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