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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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44 int GetRemoteSSRC(int channel, unsigned int& ssrc) override; | 44 int GetRemoteSSRC(int channel, unsigned int& ssrc) override; |
45 | 45 |
46 // RTP Header Extension for Client-to-Mixer Audio Level Indication | 46 // RTP Header Extension for Client-to-Mixer Audio Level Indication |
47 int SetSendAudioLevelIndicationStatus(int channel, | 47 int SetSendAudioLevelIndicationStatus(int channel, |
48 bool enable, | 48 bool enable, |
49 unsigned char id) override; | 49 unsigned char id) override; |
50 int SetReceiveAudioLevelIndicationStatus(int channel, | 50 int SetReceiveAudioLevelIndicationStatus(int channel, |
51 bool enable, | 51 bool enable, |
52 unsigned char id) override; | 52 unsigned char id) override; |
53 | 53 |
54 // RTP Header Extension for Absolute Sender Time | |
55 int SetSendAbsoluteSenderTimeStatus(int channel, | |
56 bool enable, | |
57 unsigned char id) override; | |
58 int SetReceiveAbsoluteSenderTimeStatus(int channel, | |
59 bool enable, | |
60 unsigned char id) override; | |
61 | |
62 // Statistics | 54 // Statistics |
63 int GetRTPStatistics(int channel, | 55 int GetRTPStatistics(int channel, |
64 unsigned int& averageJitterMs, | 56 unsigned int& averageJitterMs, |
65 unsigned int& maxJitterMs, | 57 unsigned int& maxJitterMs, |
66 unsigned int& discardedPackets) override; | 58 unsigned int& discardedPackets) override; |
67 | 59 |
68 int GetRTCPStatistics(int channel, CallStatistics& stats) override; | 60 int GetRTCPStatistics(int channel, CallStatistics& stats) override; |
69 | 61 |
70 int GetRemoteRTCPReportBlocks( | 62 int GetRemoteRTCPReportBlocks( |
71 int channel, | 63 int channel, |
72 std::vector<ReportBlock>* report_blocks) override; | 64 std::vector<ReportBlock>* report_blocks) override; |
73 | 65 |
74 // NACK | 66 // NACK |
75 int SetNACKStatus(int channel, bool enable, int maxNoPackets) override; | 67 int SetNACKStatus(int channel, bool enable, int maxNoPackets) override; |
76 | 68 |
77 protected: | 69 protected: |
78 VoERTP_RTCPImpl(voe::SharedData* shared); | 70 VoERTP_RTCPImpl(voe::SharedData* shared); |
79 ~VoERTP_RTCPImpl() override; | 71 ~VoERTP_RTCPImpl() override; |
80 | 72 |
81 private: | 73 private: |
82 voe::SharedData* _shared; | 74 voe::SharedData* _shared; |
83 }; | 75 }; |
84 | 76 |
85 } // namespace webrtc | 77 } // namespace webrtc |
86 | 78 |
87 #endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H | 79 #endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H |
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