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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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108 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 108 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
109 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 109 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
110 RtcEventLog* const event_log_; | 110 RtcEventLog* const event_log_; |
111 | 111 |
112 BitrateAllocator* const bitrate_allocator_; | 112 BitrateAllocator* const bitrate_allocator_; |
113 RtpTransportControllerSendInterface* const transport_; | 113 RtpTransportControllerSendInterface* const transport_; |
114 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 114 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
115 | 115 |
116 rtc::CriticalSection packet_loss_tracker_cs_; | 116 rtc::CriticalSection packet_loss_tracker_cs_; |
117 TransportFeedbackPacketLossTracker packet_loss_tracker_ | 117 TransportFeedbackPacketLossTracker packet_loss_tracker_ |
118 GUARDED_BY(&packet_loss_tracker_cs_); | 118 RTC_GUARDED_BY(&packet_loss_tracker_cs_); |
119 | 119 |
120 RtpRtcp* rtp_rtcp_module_; | 120 RtpRtcp* rtp_rtcp_module_; |
121 rtc::Optional<RtpState> const suspended_rtp_state_; | 121 rtc::Optional<RtpState> const suspended_rtp_state_; |
122 | 122 |
123 std::unique_ptr<TimedTransport> timed_send_transport_adapter_; | 123 std::unique_ptr<TimedTransport> timed_send_transport_adapter_; |
124 TimeInterval active_lifetime_; | 124 TimeInterval active_lifetime_; |
125 | 125 |
126 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 126 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
127 }; | 127 }; |
128 } // namespace internal | 128 } // namespace internal |
129 } // namespace webrtc | 129 } // namespace webrtc |
130 | 130 |
131 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 131 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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