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Side by Side Diff: webrtc/api/rtpreceiverinterface.h

Issue 2813753002: Modified the rtp_receiver_unittests. (Closed)
Patch Set: Modified the rtp_receiver_unittests. Created 3 years, 8 months ago
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1 /* 1 /*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright 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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48 // The identifier of the source can be the CSRC or the SSRC. 48 // The identifier of the source can be the CSRC or the SSRC.
49 uint32_t source_id() const { return source_id_; } 49 uint32_t source_id() const { return source_id_; }
50 50
51 // The source can be either a contributing source or a synchronization source. 51 // The source can be either a contributing source or a synchronization source.
52 RtpSourceType source_type() const { return source_type_; } 52 RtpSourceType source_type() const { return source_type_; }
53 53
54 // This isn't implemented yet and will always return an empty Optional. 54 // This isn't implemented yet and will always return an empty Optional.
55 // TODO(zhihuang): Implement this to return real audio level. 55 // TODO(zhihuang): Implement this to return real audio level.
56 rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); } 56 rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); }
57 57
58 bool operator==(const RtpSource& o) const {
59 return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
60 source_type_ == o.source_type();
61 }
62
58 private: 63 private:
59 int64_t timestamp_ms_; 64 int64_t timestamp_ms_;
60 uint32_t source_id_; 65 uint32_t source_id_;
61 RtpSourceType source_type_; 66 RtpSourceType source_type_;
62 }; 67 };
63 68
64 class RtpReceiverObserverInterface { 69 class RtpReceiverObserverInterface {
65 public: 70 public:
66 // Note: Currently if there are multiple RtpReceivers of the same media type, 71 // Note: Currently if there are multiple RtpReceivers of the same media type,
67 // they will all call OnFirstPacketReceived at once. 72 // they will all call OnFirstPacketReceived at once.
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118 PROXY_CONSTMETHOD0(std::string, id) 123 PROXY_CONSTMETHOD0(std::string, id)
119 PROXY_CONSTMETHOD0(RtpParameters, GetParameters); 124 PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
120 PROXY_METHOD1(bool, SetParameters, const RtpParameters&) 125 PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
121 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*); 126 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
122 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources); 127 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
123 END_PROXY_MAP() 128 END_PROXY_MAP()
124 129
125 } // namespace webrtc 130 } // namespace webrtc
126 131
127 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ 132 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_
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