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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 <string> | 17 #include <string> |
| 18 #include <vector> | |
| 19 | 18 |
| 20 #include "webrtc/api/mediatypes.h" | 19 #include "webrtc/api/mediatypes.h" |
| 21 #include "webrtc/api/mediastreaminterface.h" | 20 #include "webrtc/api/mediastreaminterface.h" |
| 22 #include "webrtc/api/proxy.h" | 21 #include "webrtc/api/proxy.h" |
| 23 #include "webrtc/api/rtpparameters.h" | 22 #include "webrtc/api/rtpparameters.h" |
| 24 #include "webrtc/base/refcount.h" | 23 #include "webrtc/base/refcount.h" |
| 25 #include "webrtc/base/scoped_ref_ptr.h" | 24 #include "webrtc/base/scoped_ref_ptr.h" |
| 26 | 25 |
| 27 namespace webrtc { | 26 namespace webrtc { |
| 28 | 27 |
| 29 enum class RtpSourceType { | |
| 30 SSRC, | |
| 31 CSRC, | |
| 32 }; | |
| 33 | |
| 34 class RtpSource { | |
| 35 public: | |
| 36 RtpSource() = delete; | |
| 37 RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type) | |
| 38 : timestamp_ms_(timestamp_ms), | |
| 39 source_id_(source_id), | |
| 40 source_type_(source_type) {} | |
| 41 | |
| 42 int64_t timestamp_ms() const { return timestamp_ms_; } | |
| 43 void update_timestamp_ms(int64_t timestamp_ms) { | |
| 44 RTC_DCHECK_LE(timestamp_ms_, timestamp_ms); | |
| 45 timestamp_ms_ = timestamp_ms; | |
| 46 } | |
| 47 | |
| 48 // The identifier of the source can be the CSRC or the SSRC. | |
| 49 uint32_t source_id() const { return source_id_; } | |
| 50 | |
| 51 // The source can be either a contributing source or a synchronization source. | |
| 52 RtpSourceType source_type() const { return source_type_; } | |
| 53 | |
| 54 // This isn't implemented yet and will always return an empty Optional. | |
| 55 // TODO(zhihuang): Implement this to return real audio level. | |
| 56 rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); } | |
| 57 | |
| 58 private: | |
| 59 int64_t timestamp_ms_; | |
| 60 uint32_t source_id_; | |
| 61 RtpSourceType source_type_; | |
| 62 }; | |
| 63 | |
| 64 class RtpReceiverObserverInterface { | 28 class RtpReceiverObserverInterface { |
| 65 public: | 29 public: |
| 66 // Note: Currently if there are multiple RtpReceivers of the same media type, | 30 // Note: Currently if there are multiple RtpReceivers of the same media type, |
| 67 // they will all call OnFirstPacketReceived at once. | 31 // they will all call OnFirstPacketReceived at once. |
| 68 // | 32 // |
| 69 // In the future, it's likely that an RtpReceiver will only call | 33 // In the future, it's likely that an RtpReceiver will only call |
| 70 // OnFirstPacketReceived when a packet is received specifically for its | 34 // OnFirstPacketReceived when a packet is received specifically for its |
| 71 // SSRC/mid. | 35 // SSRC/mid. |
| 72 virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; | 36 virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; |
| 73 | 37 |
| (...skipping 16 matching lines...) Expand all Loading... |
| 90 // but this API also applies them to receivers, similar to ORTC: | 54 // but this API also applies them to receivers, similar to ORTC: |
| 91 // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. | 55 // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. |
| 92 virtual RtpParameters GetParameters() const = 0; | 56 virtual RtpParameters GetParameters() const = 0; |
| 93 // Currently, doesn't support changing any parameters, but may in the future. | 57 // Currently, doesn't support changing any parameters, but may in the future. |
| 94 virtual bool SetParameters(const RtpParameters& parameters) = 0; | 58 virtual bool SetParameters(const RtpParameters& parameters) = 0; |
| 95 | 59 |
| 96 // Does not take ownership of observer. | 60 // Does not take ownership of observer. |
| 97 // Must call SetObserver(nullptr) before the observer is destroyed. | 61 // Must call SetObserver(nullptr) before the observer is destroyed. |
| 98 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; | 62 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; |
| 99 | 63 |
| 100 // TODO(zhihuang): Remove the default implementation once the subclasses | |
| 101 // implement this. Currently, the only relevant subclass is the | |
| 102 // content::FakeRtpReceiver in Chromium. | |
| 103 virtual std::vector<RtpSource> GetSources() const { | |
| 104 return std::vector<RtpSource>(); | |
| 105 } | |
| 106 | |
| 107 protected: | 64 protected: |
| 108 virtual ~RtpReceiverInterface() {} | 65 virtual ~RtpReceiverInterface() {} |
| 109 }; | 66 }; |
| 110 | 67 |
| 111 // Define proxy for RtpReceiverInterface. | 68 // Define proxy for RtpReceiverInterface. |
| 112 // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods | 69 // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods |
| 113 // are called on is an implementation detail. | 70 // are called on is an implementation detail. |
| 114 BEGIN_SIGNALING_PROXY_MAP(RtpReceiver) | 71 BEGIN_SIGNALING_PROXY_MAP(RtpReceiver) |
| 115 PROXY_SIGNALING_THREAD_DESTRUCTOR() | 72 PROXY_SIGNALING_THREAD_DESTRUCTOR() |
| 116 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) | 73 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) |
| 117 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) | 74 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) |
| 118 PROXY_CONSTMETHOD0(std::string, id) | 75 PROXY_CONSTMETHOD0(std::string, id) |
| 119 PROXY_CONSTMETHOD0(RtpParameters, GetParameters); | 76 PROXY_CONSTMETHOD0(RtpParameters, GetParameters); |
| 120 PROXY_METHOD1(bool, SetParameters, const RtpParameters&) | 77 PROXY_METHOD1(bool, SetParameters, const RtpParameters&) |
| 121 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*); | 78 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*); |
| 122 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources); | 79 END_PROXY_MAP() |
| 123 END_PROXY_MAP() | |
| 124 | 80 |
| 125 } // namespace webrtc | 81 } // namespace webrtc |
| 126 | 82 |
| 127 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ | 83 #endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ |
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