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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 |
| (...skipping 13 matching lines...) Expand all Loading... |
| 24 public: | 24 public: |
| 25 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, | 25 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, |
| 26 const webrtc::AudioReceiveStream::Config& config); | 26 const webrtc::AudioReceiveStream::Config& config); |
| 27 ~AudioReceiveStream() override {} | 27 ~AudioReceiveStream() override {} |
| 28 | 28 |
| 29 // webrtc::ReceiveStream implementation. | 29 // webrtc::ReceiveStream implementation. |
| 30 void Start() override; | 30 void Start() override; |
| 31 void Stop() override; | 31 void Stop() override; |
| 32 void SignalNetworkState(NetworkState state) override; | 32 void SignalNetworkState(NetworkState state) override; |
| 33 bool DeliverRtcp(const uint8_t* packet, size_t length) override; | 33 bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
| 34 bool DeliverRtp(const uint8_t* packet, size_t length) override; | 34 bool DeliverRtp(const uint8_t* packet, |
| 35 size_t length, |
| 36 const PacketTime& packet_time) override; |
| 35 | 37 |
| 36 // webrtc::AudioReceiveStream implementation. | 38 // webrtc::AudioReceiveStream implementation. |
| 37 webrtc::AudioReceiveStream::Stats GetStats() const override; | 39 webrtc::AudioReceiveStream::Stats GetStats() const override; |
| 38 | 40 |
| 39 const webrtc::AudioReceiveStream::Config& config() const { | 41 const webrtc::AudioReceiveStream::Config& config() const { |
| 40 return config_; | 42 return config_; |
| 41 } | 43 } |
| 42 | 44 |
| 43 private: | 45 private: |
| 44 RemoteBitrateEstimator* const remote_bitrate_estimator_; | 46 RemoteBitrateEstimator* const remote_bitrate_estimator_; |
| 45 const webrtc::AudioReceiveStream::Config config_; | 47 const webrtc::AudioReceiveStream::Config config_; |
| 46 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; | 48 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
| 47 }; | 49 }; |
| 48 } // namespace internal | 50 } // namespace internal |
| 49 } // namespace webrtc | 51 } // namespace webrtc |
| 50 | 52 |
| 51 #endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ | 53 #endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ |
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