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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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30 SSRC, | 30 SSRC, |
31 CSRC, | 31 CSRC, |
32 }; | 32 }; |
33 | 33 |
34 class RtpSource { | 34 class RtpSource { |
35 public: | 35 public: |
36 RtpSource() = delete; | 36 RtpSource() = delete; |
37 RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type) | 37 RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type) |
38 : timestamp_ms_(timestamp_ms), | 38 : timestamp_ms_(timestamp_ms), |
39 source_id_(source_id), | 39 source_id_(source_id), |
40 source_type_(source_type) {} | 40 source_type_(source_type), |
41 audio_level_() {} | |
pthatcher
2017/08/21 22:55:39
Do you even need to do audio_level_() here? I tho
Zach Stein
2017/08/22 21:29:59
I think the right thing will happen without this.
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42 | |
43 RtpSource(int64_t timestamp_ms, | |
44 uint32_t source_id, | |
45 RtpSourceType source_type, | |
46 uint8_t audio_level) | |
47 : timestamp_ms_(timestamp_ms), | |
48 source_id_(source_id), | |
49 source_type_(source_type), | |
50 audio_level_(audio_level) {} | |
pthatcher
2017/08/21 22:55:40
Is the other constructor (without an audio_level)
Zhi Huang
2017/08/22 00:01:19
Some internal tests will call the other constructo
Zach Stein
2017/08/22 21:29:59
Yup, some external code relies on the existing con
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41 | 51 |
42 int64_t timestamp_ms() const { return timestamp_ms_; } | 52 int64_t timestamp_ms() const { return timestamp_ms_; } |
43 void update_timestamp_ms(int64_t timestamp_ms) { | 53 void update_timestamp_ms(int64_t timestamp_ms) { |
44 RTC_DCHECK_LE(timestamp_ms_, timestamp_ms); | 54 RTC_DCHECK_LE(timestamp_ms_, timestamp_ms); |
45 timestamp_ms_ = timestamp_ms; | 55 timestamp_ms_ = timestamp_ms; |
46 } | 56 } |
47 | 57 |
48 // The identifier of the source can be the CSRC or the SSRC. | 58 // The identifier of the source can be the CSRC or the SSRC. |
49 uint32_t source_id() const { return source_id_; } | 59 uint32_t source_id() const { return source_id_; } |
50 | 60 |
51 // The source can be either a contributing source or a synchronization source. | 61 // The source can be either a contributing source or a synchronization source. |
52 RtpSourceType source_type() const { return source_type_; } | 62 RtpSourceType source_type() const { return source_type_; } |
53 | 63 |
54 // This isn't implemented yet and will always return an empty Optional. | 64 rtc::Optional<uint8_t> audio_level() const { return audio_level_; } |
55 // TODO(zhihuang): Implement this to return real audio level. | 65 void set_audio_level(rtc::Optional<uint8_t> level) { audio_level_ = level; } |
pthatcher
2017/08/21 22:55:39
I think "const rtc::Optional<T>&" is usually how w
Zach Stein
2017/08/22 21:29:59
Done.
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56 rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); } | |
57 | 66 |
58 bool operator==(const RtpSource& o) const { | 67 bool operator==(const RtpSource& o) const { |
59 return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() && | 68 return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() && |
60 source_type_ == o.source_type(); | 69 source_type_ == o.source_type() && audio_level_ == o.audio_level_; |
61 } | 70 } |
62 | 71 |
63 private: | 72 private: |
64 int64_t timestamp_ms_; | 73 int64_t timestamp_ms_; |
65 uint32_t source_id_; | 74 uint32_t source_id_; |
66 RtpSourceType source_type_; | 75 RtpSourceType source_type_; |
76 rtc::Optional<uint8_t> audio_level_; | |
67 }; | 77 }; |
68 | 78 |
69 class RtpReceiverObserverInterface { | 79 class RtpReceiverObserverInterface { |
70 public: | 80 public: |
71 // Note: Currently if there are multiple RtpReceivers of the same media type, | 81 // Note: Currently if there are multiple RtpReceivers of the same media type, |
72 // they will all call OnFirstPacketReceived at once. | 82 // they will all call OnFirstPacketReceived at once. |
73 // | 83 // |
74 // In the future, it's likely that an RtpReceiver will only call | 84 // In the future, it's likely that an RtpReceiver will only call |
75 // OnFirstPacketReceived when a packet is received specifically for its | 85 // OnFirstPacketReceived when a packet is received specifically for its |
76 // SSRC/mid. | 86 // SSRC/mid. |
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123 PROXY_CONSTMETHOD0(std::string, id) | 133 PROXY_CONSTMETHOD0(std::string, id) |
124 PROXY_CONSTMETHOD0(RtpParameters, GetParameters); | 134 PROXY_CONSTMETHOD0(RtpParameters, GetParameters); |
125 PROXY_METHOD1(bool, SetParameters, const RtpParameters&) | 135 PROXY_METHOD1(bool, SetParameters, const RtpParameters&) |
126 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*); | 136 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*); |
127 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources); | 137 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources); |
128 END_PROXY_MAP() | 138 END_PROXY_MAP() |
129 | 139 |
130 } // namespace webrtc | 140 } // namespace webrtc |
131 | 141 |
132 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ | 142 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ |
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