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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_RECEIVE_STREAM_H_ | 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
17 #include "webrtc/api/audio/audio_mixer.h" | 17 #include "webrtc/api/audio/audio_mixer.h" |
18 #include "webrtc/audio/audio_state.h" | 18 #include "webrtc/audio/audio_state.h" |
19 #include "webrtc/call/audio_receive_stream.h" | 19 #include "webrtc/call/audio_receive_stream.h" |
20 #include "webrtc/call/rtp_packet_sink_interface.h" | 20 #include "webrtc/call/rtp_packet_sink_interface.h" |
21 #include "webrtc/call/syncable.h" | 21 #include "webrtc/call/syncable.h" |
| 22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h" |
22 #include "webrtc/rtc_base/constructormagic.h" | 23 #include "webrtc/rtc_base/constructormagic.h" |
23 #include "webrtc/rtc_base/thread_checker.h" | 24 #include "webrtc/rtc_base/thread_checker.h" |
24 | 25 |
25 namespace webrtc { | 26 namespace webrtc { |
26 class PacketRouter; | 27 class PacketRouter; |
27 class RtcEventLog; | 28 class RtcEventLog; |
28 class RtpPacketReceived; | 29 class RtpPacketReceived; |
29 class RtpStreamReceiverControllerInterface; | 30 class RtpStreamReceiverControllerInterface; |
30 class RtpStreamReceiverInterface; | 31 class RtpStreamReceiverInterface; |
31 | 32 |
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80 const webrtc::AudioReceiveStream::Config& config() const; | 81 const webrtc::AudioReceiveStream::Config& config() const; |
81 | 82 |
82 private: | 83 private: |
83 VoiceEngine* voice_engine() const; | 84 VoiceEngine* voice_engine() const; |
84 AudioState* audio_state() const; | 85 AudioState* audio_state() const; |
85 int SetVoiceEnginePlayout(bool playout); | 86 int SetVoiceEnginePlayout(bool playout); |
86 | 87 |
87 rtc::ThreadChecker worker_thread_checker_; | 88 rtc::ThreadChecker worker_thread_checker_; |
88 rtc::ThreadChecker module_process_thread_checker_; | 89 rtc::ThreadChecker module_process_thread_checker_; |
89 const webrtc::AudioReceiveStream::Config config_; | 90 const webrtc::AudioReceiveStream::Config config_; |
| 91 RtpHeaderExtensionMap rtp_header_extensions_; |
| 92 |
90 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 93 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
91 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 94 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
92 | 95 |
93 bool playing_ ACCESS_ON(worker_thread_checker_) = false; | 96 bool playing_ ACCESS_ON(worker_thread_checker_) = false; |
94 | 97 |
95 std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_; | 98 std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_; |
96 | 99 |
97 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 100 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
98 }; | 101 }; |
99 } // namespace internal | 102 } // namespace internal |
100 } // namespace webrtc | 103 } // namespace webrtc |
101 | 104 |
102 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 105 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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