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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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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65 | 65 |
66 virtual RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const = 0; | 66 virtual RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const = 0; |
67 virtual bool SetAudioRtpSendParameters(uint32_t ssrc, | 67 virtual bool SetAudioRtpSendParameters(uint32_t ssrc, |
68 const RtpParameters& parameters) = 0; | 68 const RtpParameters& parameters) = 0; |
69 | 69 |
70 virtual RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const = 0; | 70 virtual RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const = 0; |
71 virtual bool SetAudioRtpReceiveParameters( | 71 virtual bool SetAudioRtpReceiveParameters( |
72 uint32_t ssrc, | 72 uint32_t ssrc, |
73 const RtpParameters& parameters) = 0; | 73 const RtpParameters& parameters) = 0; |
74 | 74 |
| 75 // Called when the first audio packet is received. |
| 76 sigslot::signal0<> SignalFirstAudioPacketReceived; |
| 77 |
75 protected: | 78 protected: |
76 virtual ~AudioProviderInterface() {} | 79 virtual ~AudioProviderInterface() {} |
77 }; | 80 }; |
78 | 81 |
79 // This interface is called by VideoRtpSender/Receivers to change the settings | 82 // This interface is called by VideoRtpSender/Receivers to change the settings |
80 // of a video track connected to a certain PeerConnection. | 83 // of a video track connected to a certain PeerConnection. |
81 class VideoProviderInterface { | 84 class VideoProviderInterface { |
82 public: | 85 public: |
83 // Enable/disable the video playout of a remote video track with |ssrc|. | 86 // Enable/disable the video playout of a remote video track with |ssrc|. |
84 virtual void SetVideoPlayout( | 87 virtual void SetVideoPlayout( |
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98 | 101 |
99 virtual RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const = 0; | 102 virtual RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const = 0; |
100 virtual bool SetVideoRtpSendParameters(uint32_t ssrc, | 103 virtual bool SetVideoRtpSendParameters(uint32_t ssrc, |
101 const RtpParameters& parameters) = 0; | 104 const RtpParameters& parameters) = 0; |
102 | 105 |
103 virtual RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const = 0; | 106 virtual RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const = 0; |
104 virtual bool SetVideoRtpReceiveParameters( | 107 virtual bool SetVideoRtpReceiveParameters( |
105 uint32_t ssrc, | 108 uint32_t ssrc, |
106 const RtpParameters& parameters) = 0; | 109 const RtpParameters& parameters) = 0; |
107 | 110 |
| 111 // Called when the first video packet is received. |
| 112 sigslot::signal0<> SignalFirstVideoPacketReceived; |
| 113 |
108 protected: | 114 protected: |
109 virtual ~VideoProviderInterface() {} | 115 virtual ~VideoProviderInterface() {} |
110 }; | 116 }; |
111 | 117 |
112 } // namespace webrtc | 118 } // namespace webrtc |
113 | 119 |
114 #endif // WEBRTC_API_MEDIASTREAMPROVIDER_H_ | 120 #endif // WEBRTC_API_MEDIASTREAMPROVIDER_H_ |
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