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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 |
11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
| 17 #include "webrtc/audio/time_interval.h" |
17 #include "webrtc/call/audio_send_stream.h" | 18 #include "webrtc/call/audio_send_stream.h" |
18 #include "webrtc/call/audio_state.h" | 19 #include "webrtc/call/audio_state.h" |
19 #include "webrtc/call/bitrate_allocator.h" | 20 #include "webrtc/call/bitrate_allocator.h" |
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
21 #include "webrtc/rtc_base/constructormagic.h" | 22 #include "webrtc/rtc_base/constructormagic.h" |
22 #include "webrtc/rtc_base/thread_checker.h" | 23 #include "webrtc/rtc_base/thread_checker.h" |
23 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" | 24 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" |
24 | 25 |
25 namespace webrtc { | 26 namespace webrtc { |
26 class VoiceEngine; | 27 class VoiceEngine; |
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69 | 70 |
70 // From PacketFeedbackObserver. | 71 // From PacketFeedbackObserver. |
71 void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; | 72 void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override; |
72 void OnPacketFeedbackVector( | 73 void OnPacketFeedbackVector( |
73 const std::vector<PacketFeedback>& packet_feedback_vector) override; | 74 const std::vector<PacketFeedback>& packet_feedback_vector) override; |
74 | 75 |
75 const webrtc::AudioSendStream::Config& config() const; | 76 const webrtc::AudioSendStream::Config& config() const; |
76 void SetTransportOverhead(int transport_overhead_per_packet); | 77 void SetTransportOverhead(int transport_overhead_per_packet); |
77 | 78 |
78 RtpState GetRtpState() const; | 79 RtpState GetRtpState() const; |
| 80 const TimeInterval& GetActiveLifetime() const; |
79 | 81 |
80 private: | 82 private: |
| 83 class TimedTransport; |
| 84 |
81 VoiceEngine* voice_engine() const; | 85 VoiceEngine* voice_engine() const; |
82 | 86 |
83 // These are all static to make it less likely that (the old) config_ is | 87 // These are all static to make it less likely that (the old) config_ is |
84 // accessed unintentionally. | 88 // accessed unintentionally. |
85 static void ConfigureStream(AudioSendStream* stream, | 89 static void ConfigureStream(AudioSendStream* stream, |
86 const Config& new_config, | 90 const Config& new_config, |
87 bool first_time); | 91 bool first_time); |
88 static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config); | 92 static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config); |
89 static bool ReconfigureSendCodec(AudioSendStream* stream, | 93 static bool ReconfigureSendCodec(AudioSendStream* stream, |
90 const Config& new_config); | 94 const Config& new_config); |
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110 RtpTransportControllerSendInterface* const transport_; | 114 RtpTransportControllerSendInterface* const transport_; |
111 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 115 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
112 | 116 |
113 rtc::CriticalSection packet_loss_tracker_cs_; | 117 rtc::CriticalSection packet_loss_tracker_cs_; |
114 TransportFeedbackPacketLossTracker packet_loss_tracker_ | 118 TransportFeedbackPacketLossTracker packet_loss_tracker_ |
115 GUARDED_BY(&packet_loss_tracker_cs_); | 119 GUARDED_BY(&packet_loss_tracker_cs_); |
116 | 120 |
117 RtpRtcp* rtp_rtcp_module_; | 121 RtpRtcp* rtp_rtcp_module_; |
118 rtc::Optional<RtpState> const suspended_rtp_state_; | 122 rtc::Optional<RtpState> const suspended_rtp_state_; |
119 | 123 |
| 124 std::unique_ptr<TimedTransport> timed_send_transport_adapter_; |
| 125 TimeInterval active_lifetime_; |
| 126 |
120 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 127 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
121 }; | 128 }; |
122 } // namespace internal | 129 } // namespace internal |
123 } // namespace webrtc | 130 } // namespace webrtc |
124 | 131 |
125 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 132 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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