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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 "webrtc/audio_send_stream.h" | 14 #include "webrtc/audio_send_stream.h" |
15 #include "webrtc/audio_state.h" | 15 #include "webrtc/audio_state.h" |
16 #include "webrtc/base/thread_checker.h" | 16 #include "webrtc/base/thread_checker.h" |
17 #include "webrtc/base/scoped_ptr.h" | 17 #include "webrtc/base/scoped_ptr.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
| 20 class CongestionController; |
20 class VoiceEngine; | 21 class VoiceEngine; |
21 | 22 |
22 namespace voe { | 23 namespace voe { |
23 class ChannelProxy; | 24 class ChannelProxy; |
24 } // namespace voe | 25 } // namespace voe |
25 | 26 |
26 namespace internal { | 27 namespace internal { |
27 class AudioSendStream final : public webrtc::AudioSendStream { | 28 class AudioSendStream final : public webrtc::AudioSendStream { |
28 public: | 29 public: |
29 AudioSendStream(const webrtc::AudioSendStream::Config& config, | 30 AudioSendStream(const webrtc::AudioSendStream::Config& config, |
30 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); | 31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| 32 CongestionController* congestion_controller); |
31 ~AudioSendStream() override; | 33 ~AudioSendStream() override; |
32 | 34 |
33 // webrtc::SendStream implementation. | 35 // webrtc::SendStream implementation. |
34 void Start() override; | 36 void Start() override; |
35 void Stop() override; | 37 void Stop() override; |
36 void SignalNetworkState(NetworkState state) override; | 38 void SignalNetworkState(NetworkState state) override; |
37 bool DeliverRtcp(const uint8_t* packet, size_t length) override; | 39 bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
38 | 40 |
39 // webrtc::AudioSendStream implementation. | 41 // webrtc::AudioSendStream implementation. |
40 bool SendTelephoneEvent(int payload_type, uint8_t event, | 42 bool SendTelephoneEvent(int payload_type, uint8_t event, |
41 uint32_t duration_ms) override; | 43 uint32_t duration_ms) override; |
42 webrtc::AudioSendStream::Stats GetStats() const override; | 44 webrtc::AudioSendStream::Stats GetStats() const override; |
43 | 45 |
44 const webrtc::AudioSendStream::Config& config() const; | 46 const webrtc::AudioSendStream::Config& config() const; |
45 | 47 |
46 private: | 48 private: |
47 VoiceEngine* voice_engine() const; | 49 VoiceEngine* voice_engine() const; |
48 | 50 |
49 rtc::ThreadChecker thread_checker_; | 51 rtc::ThreadChecker thread_checker_; |
50 const webrtc::AudioSendStream::Config config_; | 52 const webrtc::AudioSendStream::Config config_; |
51 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 53 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
52 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_; | 54 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_; |
53 | 55 |
54 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); | 56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
55 }; | 57 }; |
56 } // namespace internal | 58 } // namespace internal |
57 } // namespace webrtc | 59 } // namespace webrtc |
58 | 60 |
59 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 61 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |
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