| OLD | NEW |
| 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 |
| (...skipping 18 matching lines...) Expand all Loading... |
| 29 | 29 |
| 30 namespace internal { | 30 namespace internal { |
| 31 | 31 |
| 32 class AudioReceiveStream final : public webrtc::AudioReceiveStream { | 32 class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
| 33 public: | 33 public: |
| 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::ReceiveStream implementation. | 39 // webrtc::AudioReceiveStream implementation. |
| 40 void Start() override; | 40 void Start() override; |
| 41 void Stop() override; | 41 void Stop() override; |
| 42 void SignalNetworkState(NetworkState state) override; | 42 webrtc::AudioReceiveStream::Stats GetStats() const override; |
| 43 bool DeliverRtcp(const uint8_t* packet, size_t length) override; | 43 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
| 44 |
| 45 void SignalNetworkState(NetworkState state); |
| 46 bool DeliverRtcp(const uint8_t* packet, size_t length); |
| 44 bool DeliverRtp(const uint8_t* packet, | 47 bool DeliverRtp(const uint8_t* packet, |
| 45 size_t length, | 48 size_t length, |
| 46 const PacketTime& packet_time) override; | 49 const PacketTime& packet_time); |
| 47 | |
| 48 // webrtc::AudioReceiveStream implementation. | |
| 49 webrtc::AudioReceiveStream::Stats GetStats() const override; | |
| 50 | |
| 51 void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; | |
| 52 | |
| 53 const webrtc::AudioReceiveStream::Config& config() const; | 50 const webrtc::AudioReceiveStream::Config& config() const; |
| 54 | 51 |
| 55 private: | 52 private: |
| 56 VoiceEngine* voice_engine() const; | 53 VoiceEngine* voice_engine() const; |
| 57 | 54 |
| 58 rtc::ThreadChecker thread_checker_; | 55 rtc::ThreadChecker thread_checker_; |
| 59 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; | 56 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; |
| 60 const webrtc::AudioReceiveStream::Config config_; | 57 const webrtc::AudioReceiveStream::Config config_; |
| 61 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 58 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| 62 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; | 59 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
| 63 std::unique_ptr<voe::ChannelProxy> channel_proxy_; | 60 std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
| 64 | 61 |
| 65 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); | 62 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
| 66 }; | 63 }; |
| 67 } // namespace internal | 64 } // namespace internal |
| 68 } // namespace webrtc | 65 } // namespace webrtc |
| 69 | 66 |
| 70 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ | 67 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
| OLD | NEW |