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

Issue 2770233003: Implemented the GetSources() in native code. (Closed)
Patch Set: Renaming. Add a unit test. Resolve the comments. 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
11 // This file contains interfaces for RtpReceivers 11 // This file contains interfaces for RtpReceivers
12 // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface 12 // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
13 13
14 #ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_ 14 #ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_
15 #define WEBRTC_API_RTPRECEIVERINTERFACE_H_ 15 #define WEBRTC_API_RTPRECEIVERINTERFACE_H_
16 16
17 #include <memory>
the sun 2017/04/05 14:58:29 Why the need to add this?
Zhi Huang 2017/04/06 03:09:49 Oh, I missed this when removing the unique_ptr.
17 #include <string> 18 #include <string>
19 #include <vector>
18 20
19 #include "webrtc/api/mediatypes.h" 21 #include "webrtc/api/mediatypes.h"
20 #include "webrtc/api/mediastreaminterface.h" 22 #include "webrtc/api/mediastreaminterface.h"
21 #include "webrtc/api/proxy.h" 23 #include "webrtc/api/proxy.h"
22 #include "webrtc/api/rtpparameters.h" 24 #include "webrtc/api/rtpparameters.h"
23 #include "webrtc/base/refcount.h" 25 #include "webrtc/base/refcount.h"
24 #include "webrtc/base/scoped_ref_ptr.h" 26 #include "webrtc/base/scoped_ref_ptr.h"
25 27
26 namespace webrtc { 28 namespace webrtc {
27 29
30 enum class RtpSourceType {
hbos 2017/04/05 15:20:16 nit: With enum class you have to qualify the name
Zhi Huang 2017/04/06 03:09:49 Done.
31 RTP_SSRC_SOURCE,
32 RTP_CSRC_SOURCE,
33 };
34
35 struct RtpSource {
hbos 2017/04/05 11:15:41 Considering the getters/setters and private member
the sun 2017/04/05 14:58:29 +1, make it a class! Btw, thanks for not adding a
Zhi Huang 2017/04/06 03:09:49 Done.
36 public:
37 RtpSource() = delete;
38 RtpSource(int64_t timestamp, uint32_t source_id, RtpSourceType source_type)
39 : timestamp_(timestamp),
40 source_id_(source_id),
41 source_type_(source_type) {}
42
43 int64_t timestamp() const { return timestamp_; }
danilchap 2017/04/05 16:24:21 can you add units, is it timestamp_ms?
Zhi Huang 2017/04/06 03:09:49 Done.
44 void update_timestamp(int64_t timestamp) {
45 RTC_DCHECK_LE(timestamp_, timestamp);
46 timestamp_ = timestamp;
47 }
48
49 // The identifier of the source can be the CSRC or the SSRC.
50 uint32_t source_id() const { return source_id_; }
51
52 // The source can be either a contributing source or a synchronization source.
53 RtpSourceType source_type() const { return source_type_; }
54
55 // This isn't implemented yet and will always return an empty Optional.
56 // TODO(zhihuang): Implement this to return real audio level.
57 rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); }
58
59 private:
60 int64_t timestamp_;
61 uint32_t source_id_;
62 RtpSourceType source_type_;
63 };
64
28 class RtpReceiverObserverInterface { 65 class RtpReceiverObserverInterface {
29 public: 66 public:
30 // Note: Currently if there are multiple RtpReceivers of the same media type, 67 // Note: Currently if there are multiple RtpReceivers of the same media type,
31 // they will all call OnFirstPacketReceived at once. 68 // they will all call OnFirstPacketReceived at once.
32 // 69 //
33 // In the future, it's likely that an RtpReceiver will only call 70 // In the future, it's likely that an RtpReceiver will only call
34 // OnFirstPacketReceived when a packet is received specifically for its 71 // OnFirstPacketReceived when a packet is received specifically for its
35 // SSRC/mid. 72 // SSRC/mid.
36 virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; 73 virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
37 74
(...skipping 16 matching lines...) Expand all
54 // but this API also applies them to receivers, similar to ORTC: 91 // but this API also applies them to receivers, similar to ORTC:
55 // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. 92 // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
56 virtual RtpParameters GetParameters() const = 0; 93 virtual RtpParameters GetParameters() const = 0;
57 // Currently, doesn't support changing any parameters, but may in the future. 94 // Currently, doesn't support changing any parameters, but may in the future.
58 virtual bool SetParameters(const RtpParameters& parameters) = 0; 95 virtual bool SetParameters(const RtpParameters& parameters) = 0;
59 96
60 // Does not take ownership of observer. 97 // Does not take ownership of observer.
61 // Must call SetObserver(nullptr) before the observer is destroyed. 98 // Must call SetObserver(nullptr) before the observer is destroyed.
62 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; 99 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
63 100
101 // TODO(zhihuang): Remove the default implementation once the subclasses
102 // implement this. Currently, the only relevant subclass is the
103 // content::FakeRtpReceiver in Chromium.
104 virtual std::vector<RtpSource> GetSources() {
the sun 2017/04/05 14:58:29 Make this method const: virtual std::vector<RtpSou
Zhi Huang 2017/04/06 03:09:49 Done.
105 return std::vector<RtpSource>();
106 }
107
64 protected: 108 protected:
65 virtual ~RtpReceiverInterface() {} 109 virtual ~RtpReceiverInterface() {}
66 }; 110 };
67 111
68 // Define proxy for RtpReceiverInterface. 112 // Define proxy for RtpReceiverInterface.
69 // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods 113 // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
70 // are called on is an implementation detail. 114 // are called on is an implementation detail.
71 BEGIN_SIGNALING_PROXY_MAP(RtpReceiver) 115 BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
72 PROXY_SIGNALING_THREAD_DESTRUCTOR() 116 PROXY_SIGNALING_THREAD_DESTRUCTOR()
73 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) 117 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
74 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) 118 PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
75 PROXY_CONSTMETHOD0(std::string, id) 119 PROXY_CONSTMETHOD0(std::string, id)
76 PROXY_CONSTMETHOD0(RtpParameters, GetParameters); 120 PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
77 PROXY_METHOD1(bool, SetParameters, const RtpParameters&) 121 PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
78 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*); 122 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
79 END_PROXY_MAP() 123 PROXY_METHOD0(std::vector<RtpSource>, GetSources);
124 END_PROXY_MAP()
80 125
81 } // namespace webrtc 126 } // namespace webrtc
82 127
83 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ 128 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_
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