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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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93 | 93 |
94 int send_bandwidth_bps = 0; // Estimated available send bandwidth. | 94 int send_bandwidth_bps = 0; // Estimated available send bandwidth. |
95 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. | 95 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. |
96 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. | 96 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. |
97 int64_t pacer_delay_ms = 0; | 97 int64_t pacer_delay_ms = 0; |
98 int64_t rtt_ms = -1; | 98 int64_t rtt_ms = -1; |
99 }; | 99 }; |
100 | 100 |
101 static Call* Create(const Call::Config& config); | 101 static Call* Create(const Call::Config& config); |
102 | 102 |
103 // TODO(nisse): Should move to RtpTransportController. | |
104 // Rtp header extensions can be renegotiated mid-call. | |
105 virtual void SetVideoReceiveRtpHeaderExtensions( | |
106 const std::vector<RtpExtension>& extensions) = 0; | |
pthatcher1
2017/04/29 00:25:18
It makes sense for this to be per-RtpTransport, bu
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107 | |
103 virtual AudioSendStream* CreateAudioSendStream( | 108 virtual AudioSendStream* CreateAudioSendStream( |
104 const AudioSendStream::Config& config) = 0; | 109 const AudioSendStream::Config& config) = 0; |
105 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; | 110 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; |
106 | 111 |
107 virtual AudioReceiveStream* CreateAudioReceiveStream( | 112 virtual AudioReceiveStream* CreateAudioReceiveStream( |
108 const AudioReceiveStream::Config& config) = 0; | 113 const AudioReceiveStream::Config& config) = 0; |
109 virtual void DestroyAudioReceiveStream( | 114 virtual void DestroyAudioReceiveStream( |
110 AudioReceiveStream* receive_stream) = 0; | 115 AudioReceiveStream* receive_stream) = 0; |
111 | 116 |
112 virtual VideoSendStream* CreateVideoSendStream( | 117 virtual VideoSendStream* CreateVideoSendStream( |
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159 const rtc::NetworkRoute& network_route) = 0; | 164 const rtc::NetworkRoute& network_route) = 0; |
160 | 165 |
161 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 166 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
162 | 167 |
163 virtual ~Call() {} | 168 virtual ~Call() {} |
164 }; | 169 }; |
165 | 170 |
166 } // namespace webrtc | 171 } // namespace webrtc |
167 | 172 |
168 #endif // WEBRTC_CALL_CALL_H_ | 173 #endif // WEBRTC_CALL_CALL_H_ |
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