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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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