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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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27 } // namespace voe | 27 } // namespace voe |
28 | 28 |
29 namespace internal { | 29 namespace internal { |
30 class AudioSendStream final : public webrtc::AudioSendStream { | 30 class AudioSendStream final : public webrtc::AudioSendStream { |
31 public: | 31 public: |
32 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 32 AudioSendStream(const webrtc::AudioSendStream::Config& config, |
33 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 33 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
34 CongestionController* congestion_controller); | 34 CongestionController* congestion_controller); |
35 ~AudioSendStream() override; | 35 ~AudioSendStream() override; |
36 | 36 |
37 // webrtc::SendStream implementation. | 37 // webrtc::AudioSendStream implementation. |
38 void Start() override; | 38 void Start() override; |
39 void Stop() override; | 39 void Stop() override; |
40 void SignalNetworkState(NetworkState state) override; | |
41 bool DeliverRtcp(const uint8_t* packet, size_t length) override; | |
42 | |
43 // webrtc::AudioSendStream implementation. | |
44 bool SendTelephoneEvent(int payload_type, int event, | 40 bool SendTelephoneEvent(int payload_type, int event, |
45 int duration_ms) override; | 41 int duration_ms) override; |
46 webrtc::AudioSendStream::Stats GetStats() const override; | 42 webrtc::AudioSendStream::Stats GetStats() const override; |
47 | 43 |
| 44 void SignalNetworkState(NetworkState state); |
| 45 bool DeliverRtcp(const uint8_t* packet, size_t length); |
48 const webrtc::AudioSendStream::Config& config() const; | 46 const webrtc::AudioSendStream::Config& config() const; |
49 | 47 |
50 private: | 48 private: |
51 VoiceEngine* voice_engine() const; | 49 VoiceEngine* voice_engine() const; |
52 | 50 |
53 rtc::ThreadChecker thread_checker_; | 51 rtc::ThreadChecker thread_checker_; |
54 const webrtc::AudioSendStream::Config config_; | 52 const webrtc::AudioSendStream::Config config_; |
55 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 53 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
56 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 54 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
57 | 55 |
58 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
59 }; | 57 }; |
60 } // namespace internal | 58 } // namespace internal |
61 } // namespace webrtc | 59 } // namespace webrtc |
62 | 60 |
63 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 61 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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