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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

Issue 1513303003: [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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64 64
65 TelephoneEventHandler* GetTelephoneEventHandler() override; 65 TelephoneEventHandler* GetTelephoneEventHandler() override;
66 66
67 private: 67 private:
68 bool HaveReceivedFrame() const; 68 bool HaveReceivedFrame() const;
69 69
70 void CheckSSRCChanged(const RTPHeader& rtp_header); 70 void CheckSSRCChanged(const RTPHeader& rtp_header);
71 void CheckCSRC(const WebRtcRTPHeader& rtp_header); 71 void CheckCSRC(const WebRtcRTPHeader& rtp_header);
72 int32_t CheckPayloadChanged(const RTPHeader& rtp_header, 72 int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
73 const int8_t first_payload_byte, 73 const int8_t first_payload_byte,
74 bool& is_red, 74 bool* is_red,
75 PayloadUnion* payload); 75 PayloadUnion* payload);
76 76
77 Clock* clock_; 77 Clock* clock_;
78 RTPPayloadRegistry* rtp_payload_registry_; 78 RTPPayloadRegistry* rtp_payload_registry_;
79 rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_; 79 rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
80 80
81 RtpFeedback* cb_rtp_feedback_; 81 RtpFeedback* cb_rtp_feedback_;
82 82
83 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_; 83 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
84 int64_t last_receive_time_; 84 int64_t last_receive_time_;
85 size_t last_received_payload_length_; 85 size_t last_received_payload_length_;
86 86
87 // SSRCs. 87 // SSRCs.
88 uint32_t ssrc_; 88 uint32_t ssrc_;
89 uint8_t num_csrcs_; 89 uint8_t num_csrcs_;
90 uint32_t current_remote_csrc_[kRtpCsrcSize]; 90 uint32_t current_remote_csrc_[kRtpCsrcSize];
91 91
92 uint32_t last_received_timestamp_; 92 uint32_t last_received_timestamp_;
93 int64_t last_received_frame_time_ms_; 93 int64_t last_received_frame_time_ms_;
94 uint16_t last_received_sequence_number_; 94 uint16_t last_received_sequence_number_;
95 95
96 NACKMethod nack_method_; 96 NACKMethod nack_method_;
97 }; 97 };
98 } // namespace webrtc 98 } // namespace webrtc
99 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ 99 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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