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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 classes that implement RtpReceiverInterface. | 11 // This file contains classes that implement RtpReceiverInterface. |
12 // An RtpReceiver associates a MediaStreamTrackInterface with an underlying | 12 // An RtpReceiver associates a MediaStreamTrackInterface with an underlying |
13 // transport (provided by AudioProviderInterface/VideoProviderInterface) | 13 // transport (provided by AudioProviderInterface/VideoProviderInterface) |
14 | 14 |
15 #ifndef WEBRTC_API_RTPRECEIVER_H_ | 15 #ifndef WEBRTC_API_RTPRECEIVER_H_ |
16 #define WEBRTC_API_RTPRECEIVER_H_ | 16 #define WEBRTC_API_RTPRECEIVER_H_ |
17 | 17 |
18 #include <string> | 18 #include <string> |
19 | 19 |
20 #include "webrtc/api/mediastreamprovider.h" | 20 #include "webrtc/api/mediastreamprovider.h" |
21 #include "webrtc/api/rtpreceiverinterface.h" | 21 #include "webrtc/api/rtpreceiverinterface.h" |
22 #include "webrtc/api/remoteaudiosource.h" | 22 #include "webrtc/api/remoteaudiosource.h" |
23 #include "webrtc/api/videotracksource.h" | 23 #include "webrtc/api/videotracksource.h" |
24 #include "webrtc/base/basictypes.h" | 24 #include "webrtc/base/basictypes.h" |
| 25 #include "webrtc/base/sigslot.h" |
25 #include "webrtc/media/base/videobroadcaster.h" | 26 #include "webrtc/media/base/videobroadcaster.h" |
26 | 27 |
27 namespace webrtc { | 28 namespace webrtc { |
28 | 29 |
29 // Internal class used by PeerConnection. | 30 // Internal class used by PeerConnection. |
30 class RtpReceiverInternal : public RtpReceiverInterface { | 31 class RtpReceiverInternal : public RtpReceiverInterface { |
31 public: | 32 public: |
32 virtual void Stop() = 0; | 33 virtual void Stop() = 0; |
33 }; | 34 }; |
34 | 35 |
35 class AudioRtpReceiver : public ObserverInterface, | 36 class AudioRtpReceiver : public ObserverInterface, |
36 public AudioSourceInterface::AudioObserver, | 37 public AudioSourceInterface::AudioObserver, |
37 public rtc::RefCountedObject<RtpReceiverInternal> { | 38 public rtc::RefCountedObject<RtpReceiverInternal>, |
| 39 public sigslot::has_slots<> { |
38 public: | 40 public: |
39 AudioRtpReceiver(MediaStreamInterface* stream, | 41 AudioRtpReceiver(MediaStreamInterface* stream, |
40 const std::string& track_id, | 42 const std::string& track_id, |
41 uint32_t ssrc, | 43 uint32_t ssrc, |
42 AudioProviderInterface* provider); | 44 AudioProviderInterface* provider); |
43 | 45 |
44 virtual ~AudioRtpReceiver(); | 46 virtual ~AudioRtpReceiver(); |
45 | 47 |
46 // ObserverInterface implementation | 48 // ObserverInterface implementation |
47 void OnChanged() override; | 49 void OnChanged() override; |
(...skipping 11 matching lines...) Expand all Loading... |
59 } | 61 } |
60 | 62 |
61 std::string id() const override { return id_; } | 63 std::string id() const override { return id_; } |
62 | 64 |
63 RtpParameters GetParameters() const override; | 65 RtpParameters GetParameters() const override; |
64 bool SetParameters(const RtpParameters& parameters) override; | 66 bool SetParameters(const RtpParameters& parameters) override; |
65 | 67 |
66 // RtpReceiverInternal implementation. | 68 // RtpReceiverInternal implementation. |
67 void Stop() override; | 69 void Stop() override; |
68 | 70 |
| 71 void SetObserver(RtpReceiverObserverInterface* observer) override; |
| 72 |
| 73 cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; } |
| 74 |
69 private: | 75 private: |
70 void Reconfigure(); | 76 void Reconfigure(); |
| 77 void OnFirstAudioPacketReceived(); |
71 | 78 |
72 const std::string id_; | 79 const std::string id_; |
73 const uint32_t ssrc_; | 80 const uint32_t ssrc_; |
74 AudioProviderInterface* provider_; // Set to null in Stop(). | 81 AudioProviderInterface* provider_; // Set to null in Stop(). |
75 const rtc::scoped_refptr<AudioTrackInterface> track_; | 82 const rtc::scoped_refptr<AudioTrackInterface> track_; |
76 bool cached_track_enabled_; | 83 bool cached_track_enabled_; |
| 84 RtpReceiverObserverInterface* observer_ = nullptr; |
| 85 bool received_first_packet_ = false; |
77 }; | 86 }; |
78 | 87 |
79 class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> { | 88 class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>, |
| 89 public sigslot::has_slots<> { |
80 public: | 90 public: |
81 VideoRtpReceiver(MediaStreamInterface* stream, | 91 VideoRtpReceiver(MediaStreamInterface* stream, |
82 const std::string& track_id, | 92 const std::string& track_id, |
83 rtc::Thread* worker_thread, | 93 rtc::Thread* worker_thread, |
84 uint32_t ssrc, | 94 uint32_t ssrc, |
85 VideoProviderInterface* provider); | 95 VideoProviderInterface* provider); |
86 | 96 |
87 virtual ~VideoRtpReceiver(); | 97 virtual ~VideoRtpReceiver(); |
88 | 98 |
89 rtc::scoped_refptr<VideoTrackInterface> video_track() const { | 99 rtc::scoped_refptr<VideoTrackInterface> video_track() const { |
90 return track_.get(); | 100 return track_.get(); |
91 } | 101 } |
92 | 102 |
93 // RtpReceiverInterface implementation | 103 // RtpReceiverInterface implementation |
94 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 104 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
95 return track_.get(); | 105 return track_.get(); |
96 } | 106 } |
97 | 107 |
98 std::string id() const override { return id_; } | 108 std::string id() const override { return id_; } |
99 | 109 |
100 RtpParameters GetParameters() const override; | 110 RtpParameters GetParameters() const override; |
101 bool SetParameters(const RtpParameters& parameters) override; | 111 bool SetParameters(const RtpParameters& parameters) override; |
102 | 112 |
103 // RtpReceiverInternal implementation. | 113 // RtpReceiverInternal implementation. |
104 void Stop() override; | 114 void Stop() override; |
105 | 115 |
| 116 void SetObserver(RtpReceiverObserverInterface* observer) override; |
| 117 |
| 118 cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; } |
| 119 |
106 private: | 120 private: |
| 121 void OnFirstVideoPacketReceived(); |
| 122 |
107 std::string id_; | 123 std::string id_; |
108 uint32_t ssrc_; | 124 uint32_t ssrc_; |
109 VideoProviderInterface* provider_; | 125 VideoProviderInterface* provider_; |
110 // |broadcaster_| is needed since the decoder can only handle one sink. | 126 // |broadcaster_| is needed since the decoder can only handle one sink. |
111 // It might be better if the decoder can handle multiple sinks and consider | 127 // It might be better if the decoder can handle multiple sinks and consider |
112 // the VideoSinkWants. | 128 // the VideoSinkWants. |
113 rtc::VideoBroadcaster broadcaster_; | 129 rtc::VideoBroadcaster broadcaster_; |
114 // |source_| is held here to be able to change the state of the source when | 130 // |source_| is held here to be able to change the state of the source when |
115 // the VideoRtpReceiver is stopped. | 131 // the VideoRtpReceiver is stopped. |
116 rtc::scoped_refptr<VideoTrackSource> source_; | 132 rtc::scoped_refptr<VideoTrackSource> source_; |
117 rtc::scoped_refptr<VideoTrackInterface> track_; | 133 rtc::scoped_refptr<VideoTrackInterface> track_; |
| 134 RtpReceiverObserverInterface* observer_ = nullptr; |
| 135 bool received_first_packet_ = false; |
118 }; | 136 }; |
119 | 137 |
120 } // namespace webrtc | 138 } // namespace webrtc |
121 | 139 |
122 #endif // WEBRTC_API_RTPRECEIVER_H_ | 140 #endif // WEBRTC_API_RTPRECEIVER_H_ |
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