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| 1 /* | 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ | 11 #ifndef WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ |
| 12 #define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ | 12 #define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ |
| 13 | 13 |
| 14 #include <atomic> | |
| 14 #include <memory> | 15 #include <memory> |
| 15 | 16 |
| 16 #include "webrtc/call/flexfec_receive_stream.h" | 17 #include "webrtc/call/flexfec_receive_stream.h" |
| 17 #include "webrtc/call/rtp_packet_sink_interface.h" | 18 #include "webrtc/call/rtp_packet_sink_interface.h" |
| 18 #include "webrtc/rtc_base/criticalsection.h" | 19 #include "webrtc/rtc_base/criticalsection.h" |
| 19 | 20 |
| 20 namespace webrtc { | 21 namespace webrtc { |
| 21 | 22 |
| 22 class FlexfecReceiver; | 23 class FlexfecReceiver; |
| 23 class ProcessThread; | 24 class ProcessThread; |
| (...skipping 22 matching lines...) Expand all Loading... | |
| 46 void OnRtpPacket(const RtpPacketReceived& packet) override; | 47 void OnRtpPacket(const RtpPacketReceived& packet) override; |
| 47 | 48 |
| 48 // Implements FlexfecReceiveStream. | 49 // Implements FlexfecReceiveStream. |
| 49 void Start() override; | 50 void Start() override; |
| 50 void Stop() override; | 51 void Stop() override; |
| 51 Stats GetStats() const override; | 52 Stats GetStats() const override; |
| 52 | 53 |
| 53 private: | 54 private: |
| 54 // Config. | 55 // Config. |
| 55 const Config config_; | 56 const Config config_; |
| 56 bool started_ GUARDED_BY(crit_); | 57 std::atomic<bool> started_; |
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brandtr
2017/08/01 07:08:02
The start/stop functionality doesn't really serve
eladalon
2017/08/01 08:27:46
Okay, thanks.
In the meantime, do you think should
brandtr
2017/08/01 08:39:03
Might as well land it, since it's a nice improveme
| |
| 57 rtc::CriticalSection crit_; | |
| 58 | 58 |
| 59 // Erasure code interfacing. | 59 // Erasure code interfacing. |
| 60 const std::unique_ptr<FlexfecReceiver> receiver_; | 60 const std::unique_ptr<FlexfecReceiver> receiver_; |
| 61 | 61 |
| 62 // RTCP reporting. | 62 // RTCP reporting. |
| 63 const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; | 63 const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; |
| 64 const std::unique_ptr<RtpRtcp> rtp_rtcp_; | 64 const std::unique_ptr<RtpRtcp> rtp_rtcp_; |
| 65 ProcessThread* process_thread_; | 65 ProcessThread* process_thread_; |
| 66 | 66 |
| 67 std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_; | 67 std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_; |
| 68 }; | 68 }; |
| 69 | 69 |
| 70 } // namespace webrtc | 70 } // namespace webrtc |
| 71 | 71 |
| 72 #endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ | 72 #endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ |
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