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Side by Side Diff: webrtc/video/audio_receive_stream.h

Issue 1333483002: Wire up PacketTime to ReceiveStreams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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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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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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