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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 #ifndef WEBRTC_CALL_CALL_H_ | 10 #ifndef WEBRTC_CALL_CALL_H_ |
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105 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. | 105 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. |
106 rtc::scoped_refptr<AudioState> audio_state; | 106 rtc::scoped_refptr<AudioState> audio_state; |
107 | 107 |
108 // Audio Processing Module to be used in this call. | 108 // Audio Processing Module to be used in this call. |
109 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. | 109 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. |
110 AudioProcessing* audio_processing = nullptr; | 110 AudioProcessing* audio_processing = nullptr; |
111 | 111 |
112 // RtcEventLog to use for this call. Required. | 112 // RtcEventLog to use for this call. Required. |
113 // Use webrtc::RtcEventLog::CreateNull() for a null implementation. | 113 // Use webrtc::RtcEventLog::CreateNull() for a null implementation. |
114 RtcEventLog* event_log = nullptr; | 114 RtcEventLog* event_log = nullptr; |
115 | |
116 // Enables periodic sending if empty keep-alive messages that helps prevent | |
117 // network time-out events. The packets adhere to RFC6263 section 4.6, and | |
118 // by default use payload type 20, as described in 3GPP TS 24.229, | |
119 // Appendix K.5.2.1. | |
120 RtpKeepAliveConfig keepalive_config; | |
121 }; | 115 }; |
122 | 116 |
123 struct Stats { | 117 struct Stats { |
124 std::string ToString(int64_t time_ms) const; | 118 std::string ToString(int64_t time_ms) const; |
125 | 119 |
126 int send_bandwidth_bps = 0; // Estimated available send bandwidth. | 120 int send_bandwidth_bps = 0; // Estimated available send bandwidth. |
127 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. | 121 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. |
128 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. | 122 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. |
129 int64_t pacer_delay_ms = 0; | 123 int64_t pacer_delay_ms = 0; |
130 int64_t rtt_ms = -1; | 124 int64_t rtt_ms = -1; |
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204 const rtc::NetworkRoute& network_route) = 0; | 198 const rtc::NetworkRoute& network_route) = 0; |
205 | 199 |
206 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 200 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
207 | 201 |
208 virtual ~Call() {} | 202 virtual ~Call() {} |
209 }; | 203 }; |
210 | 204 |
211 } // namespace webrtc | 205 } // namespace webrtc |
212 | 206 |
213 #endif // WEBRTC_CALL_CALL_H_ | 207 #endif // WEBRTC_CALL_CALL_H_ |
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