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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #include <algorithm> | 10 #include <algorithm> |
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91 } | 91 } |
92 if (need_rtcp_) { | 92 if (need_rtcp_) { |
93 ADD_FAILURE() << "Expected RTCP packet not sent."; | 93 ADD_FAILURE() << "Expected RTCP packet not sent."; |
94 } | 94 } |
95 } | 95 } |
96 | 96 |
97 private: | 97 private: |
98 bool SendRtp(const uint8_t* packet, | 98 bool SendRtp(const uint8_t* packet, |
99 size_t length, | 99 size_t length, |
100 const PacketOptions& options) override { | 100 const PacketOptions& options) override { |
| 101 rtc::CritScope lock(&crit_); |
101 need_rtp_ = false; | 102 need_rtp_ = false; |
102 return true; | 103 return true; |
103 } | 104 } |
104 | 105 |
105 bool SendRtcp(const uint8_t* packet, size_t length) override { | 106 bool SendRtcp(const uint8_t* packet, size_t length) override { |
| 107 rtc::CritScope lock(&crit_); |
106 need_rtcp_ = false; | 108 need_rtcp_ = false; |
107 return true; | 109 return true; |
108 } | 110 } |
109 bool need_rtp_; | 111 bool need_rtp_; |
110 bool need_rtcp_; | 112 bool need_rtcp_; |
| 113 rtc::CriticalSection crit_; |
111 }; | 114 }; |
112 | 115 |
113 void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red); | 116 void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red); |
114 void ReceivesPliAndRecovers(int rtp_history_ms); | 117 void ReceivesPliAndRecovers(int rtp_history_ms); |
115 void RespectsRtcpMode(RtcpMode rtcp_mode); | 118 void RespectsRtcpMode(RtcpMode rtcp_mode); |
116 void TestXrReceiverReferenceTimeReport(bool enable_rrtr); | 119 void TestXrReceiverReferenceTimeReport(bool enable_rrtr); |
117 void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first); | 120 void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first); |
118 void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp); | 121 void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp); |
119 void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare); | 122 void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare); |
120 void VerifyNewVideoSendStreamsRespectNetworkState( | 123 void VerifyNewVideoSendStreamsRespectNetworkState( |
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3731 private: | 3734 private: |
3732 bool video_observed_; | 3735 bool video_observed_; |
3733 bool audio_observed_; | 3736 bool audio_observed_; |
3734 SequenceNumberUnwrapper unwrapper_; | 3737 SequenceNumberUnwrapper unwrapper_; |
3735 std::set<int64_t> received_packet_ids_; | 3738 std::set<int64_t> received_packet_ids_; |
3736 } test; | 3739 } test; |
3737 | 3740 |
3738 RunBaseTest(&test); | 3741 RunBaseTest(&test); |
3739 } | 3742 } |
3740 } // namespace webrtc | 3743 } // namespace webrtc |
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