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

Issue 1999853002: Forward the SignalFirstPacketReceived to RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Modified the unit test. Created 4 years, 6 months ago
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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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