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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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34 AudioReceiveStream(CongestionController* congestion_controller, | 34 AudioReceiveStream(CongestionController* congestion_controller, |
35 const webrtc::AudioReceiveStream::Config& config, | 35 const webrtc::AudioReceiveStream::Config& config, |
36 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); | 36 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); |
37 ~AudioReceiveStream() override; | 37 ~AudioReceiveStream() override; |
38 | 38 |
39 // webrtc::AudioReceiveStream implementation. | 39 // webrtc::AudioReceiveStream implementation. |
40 void Start() override; | 40 void Start() override; |
41 void Stop() override; | 41 void Stop() override; |
42 webrtc::AudioReceiveStream::Stats GetStats() const override; | 42 webrtc::AudioReceiveStream::Stats GetStats() const override; |
43 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; | 43 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
| 44 void SetGain(float gain) override; |
44 | 45 |
45 void SignalNetworkState(NetworkState state); | 46 void SignalNetworkState(NetworkState state); |
46 bool DeliverRtcp(const uint8_t* packet, size_t length); | 47 bool DeliverRtcp(const uint8_t* packet, size_t length); |
47 bool DeliverRtp(const uint8_t* packet, | 48 bool DeliverRtp(const uint8_t* packet, |
48 size_t length, | 49 size_t length, |
49 const PacketTime& packet_time); | 50 const PacketTime& packet_time); |
50 const webrtc::AudioReceiveStream::Config& config() const; | 51 const webrtc::AudioReceiveStream::Config& config() const; |
51 | 52 |
52 private: | 53 private: |
53 VoiceEngine* voice_engine() const; | 54 VoiceEngine* voice_engine() const; |
54 | 55 |
55 rtc::ThreadChecker thread_checker_; | 56 rtc::ThreadChecker thread_checker_; |
56 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; | 57 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; |
57 const webrtc::AudioReceiveStream::Config config_; | 58 const webrtc::AudioReceiveStream::Config config_; |
58 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 59 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
59 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; | 60 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
60 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 61 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
61 | 62 |
62 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 63 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
63 }; | 64 }; |
64 } // namespace internal | 65 } // namespace internal |
65 } // namespace webrtc | 66 } // namespace webrtc |
66 | 67 |
67 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 68 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
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