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Side by Side Diff: webrtc/video/vie_channel.h

Issue 1674663002: Initialize VideoSendStream members in constructor. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: feedback + test update Created 4 years, 10 months ago
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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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105 105
106 // Gets SSRC for outgoing stream number |idx|. 106 // Gets SSRC for outgoing stream number |idx|.
107 int32_t GetLocalSSRC(uint8_t idx, unsigned int* ssrc); 107 int32_t GetLocalSSRC(uint8_t idx, unsigned int* ssrc);
108 108
109 // Gets SSRC for the incoming stream. 109 // Gets SSRC for the incoming stream.
110 uint32_t GetRemoteSSRC(); 110 uint32_t GetRemoteSSRC();
111 111
112 int SetRtxSendPayloadType(int payload_type, int associated_payload_type); 112 int SetRtxSendPayloadType(int payload_type, int associated_payload_type);
113 113
114 void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state); 114 void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
115 RtpState GetRtpStateForSsrc(uint32_t ssrc); 115 RtpState GetRtpStateForSsrc(uint32_t ssrc) const;
116 116
117 // Sets the CName for the outgoing stream on the channel. 117 // Sets the CName for the outgoing stream on the channel.
118 int32_t SetRTCPCName(const char* rtcp_cname); 118 int32_t SetRTCPCName(const char* rtcp_cname);
119 119
120 // Gets the CName of the incoming stream. 120 // Gets the CName of the incoming stream.
121 int32_t GetRemoteRTCPCName(char rtcp_cname[]); 121 int32_t GetRemoteRTCPCName(char rtcp_cname[]);
122 122
123 // Returns statistics reported by the remote client in an RTCP packet. 123 // Returns statistics reported by the remote client in an RTCP packet.
124 // TODO(pbos): Remove this along with VideoSendStream::GetRtt(). 124 // TODO(pbos): Remove this along with VideoSendStream::GetRtt().
125 int32_t GetSendRtcpStatistics(uint16_t* fraction_lost, 125 int32_t GetSendRtcpStatistics(uint16_t* fraction_lost,
126 uint32_t* cumulative_lost, 126 uint32_t* cumulative_lost,
127 uint32_t* extended_max, 127 uint32_t* extended_max,
128 uint32_t* jitter_samples, 128 uint32_t* jitter_samples,
129 int64_t* rtt_ms); 129 int64_t* rtt_ms) const;
130 130
131 // Called on receipt of RTCP report block from remote side. 131 // Called on receipt of RTCP report block from remote side.
132 void RegisterSendChannelRtcpStatisticsCallback( 132 void RegisterSendChannelRtcpStatisticsCallback(
133 RtcpStatisticsCallback* callback); 133 RtcpStatisticsCallback* callback);
134 134
135 // Gets send statistics for the rtp and rtx stream. 135 // Gets send statistics for the rtp and rtx stream.
136 void GetSendStreamDataCounters(StreamDataCounters* rtp_counters, 136 void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
137 StreamDataCounters* rtx_counters) const; 137 StreamDataCounters* rtx_counters) const;
138 138
139 // Gets received stream data counters. 139 // Gets received stream data counters.
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406 size_t num_rtts_ GUARDED_BY(crit_); 406 size_t num_rtts_ GUARDED_BY(crit_);
407 407
408 // RtpRtcp modules, declared last as they use other members on construction. 408 // RtpRtcp modules, declared last as they use other members on construction.
409 const std::vector<RtpRtcp*> rtp_rtcp_modules_; 409 const std::vector<RtpRtcp*> rtp_rtcp_modules_;
410 size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_); 410 size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_);
411 }; 411 };
412 412
413 } // namespace webrtc 413 } // namespace webrtc
414 414
415 #endif // WEBRTC_VIDEO_VIE_CHANNEL_H_ 415 #endif // WEBRTC_VIDEO_VIE_CHANNEL_H_
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