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

Issue 3000713002: Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket. (Closed)
Patch Set: Move set_audio_level before RemoveOutdatedSources. Created 3 years, 4 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
(...skipping 19 matching lines...) Expand all
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.
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
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.
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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