1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 13 14 #include "webrtc/audio_send_stream.h" 15 #include "webrtc/audio_state.h" 16 #include "webrtc/base/thread_checker.h" 17 #include "webrtc/base/scoped_ptr.h" 18 19 namespace webrtc { 20 class CongestionController; 21 class VoiceEngine; 22 23 namespace voe { 24 class ChannelProxy; 25 } // namespace voe 26 27 namespace internal { 28 class AudioSendStream final : public webrtc::AudioSendStream { 29 public: 30 AudioSendStream(const webrtc::AudioSendStream::Config& config, 31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 32 CongestionController* congestion_controller); 33 ~AudioSendStream() override; 34 35 // webrtc::SendStream implementation. 36 void Start() override; 37 void Stop() override; 38 void SignalNetworkState(NetworkState state) override; 39 bool DeliverRtcp(const uint8_t* packet, size_t length) override; 40 41 // webrtc::AudioSendStream implementation. 42 bool SendTelephoneEvent(int payload_type, uint8_t event, 43 uint32_t duration_ms) override; 44 webrtc::AudioSendStream::Stats GetStats() const override; 45 46 const webrtc::AudioSendStream::Config& config() const; 47 48 private: 49 VoiceEngine* voice_engine() const; 50 51 rtc::ThreadChecker thread_checker_; 52 const webrtc::AudioSendStream::Config config_; 53 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 54 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_; 55 56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 57 }; 58 } // namespace internal 59 } // namespace webrtc 60 61 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 62