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| 1 /* | |
| 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 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 | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ | |
| 12 #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ | |
| 13 | |
| 14 #include "webrtc/base/platform_thread.h" | |
| 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | |
| 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | |
| 17 #include "webrtc/test/channel_transport/udp_transport.h" | |
| 18 #include "webrtc/typedefs.h" | |
| 19 | |
| 20 class TestLoadGenerator; | |
| 21 namespace webrtc { | |
| 22 class CriticalSectionWrapper; | |
| 23 class EventWrapper; | |
| 24 | |
| 25 #define MAX_BITRATE_KBPS 50000 | |
| 26 | |
| 27 | |
| 28 class SendRecCB | |
| 29 { | |
| 30 public: | |
| 31 virtual void OnOnNetworkChanged(const uint32_t bitrateTarget, | |
| 32 const uint8_t fractionLost, | |
| 33 const uint16_t roundTripTimeMs, | |
| 34 const uint16_t bwEstimateKbitMin, | |
| 35 const uint16_t bwEstimateKbitMax) = 0; | |
| 36 | |
| 37 virtual ~SendRecCB() {}; | |
| 38 }; | |
| 39 | |
| 40 | |
| 41 class TestSenderReceiver : public RtpFeedback, public RtpData, public UdpTranspo
rtData, public RtpVideoFeedback | |
| 42 { | |
| 43 | |
| 44 public: | |
| 45 TestSenderReceiver (void); | |
| 46 | |
| 47 ~TestSenderReceiver (void); | |
| 48 | |
| 49 void SetCallback (SendRecCB *cb) { _sendRecCB = cb; }; | |
| 50 | |
| 51 int32_t Start(); | |
| 52 | |
| 53 int32_t Stop(); | |
| 54 | |
| 55 bool ProcLoop(); | |
| 56 | |
| 57 ///////////////////////////////////////////// | |
| 58 // Receiver methods | |
| 59 | |
| 60 int32_t InitReceiver (const uint16_t rtpPort, | |
| 61 const uint16_t rtcpPort = 0, | |
| 62 const int8_t payloadType = 127); | |
| 63 | |
| 64 int32_t ReceiveBitrateKbps (); | |
| 65 | |
| 66 int32_t SetPacketTimeout(const uint32_t timeoutMS); | |
| 67 | |
| 68 // Inherited from RtpFeedback | |
| 69 int32_t OnInitializeDecoder(const int32_t id, | |
| 70 const int8_t payloadType, | |
| 71 const int8_t payloadName[RTP_PAYLOAD_NAME_SIZE], | |
| 72 const uint32_t frequency, | |
| 73 const uint8_t channels, | |
| 74 const uint32_t rate) override { | |
| 75 return 0; | |
| 76 } | |
| 77 | |
| 78 void OnIncomingSSRCChanged(const int32_t id, const uint32_t SSRC) override { | |
| 79 } | |
| 80 | |
| 81 void OnIncomingCSRCChanged(const int32_t id, | |
| 82 const uint32_t CSRC, | |
| 83 const bool added) override {} | |
| 84 | |
| 85 // Inherited from RtpData | |
| 86 int32_t OnReceivedPayloadData( | |
| 87 const uint8_t* payloadData, | |
| 88 const size_t payloadSize, | |
| 89 const webrtc::WebRtcRTPHeader* rtpHeader) override; | |
| 90 | |
| 91 // Inherited from UdpTransportData | |
| 92 void IncomingRTPPacket(const int8_t* incomingRtpPacket, | |
| 93 const size_t rtpPacketLength, | |
| 94 const int8_t* fromIP, | |
| 95 const uint16_t fromPort) override; | |
| 96 | |
| 97 void IncomingRTCPPacket(const int8_t* incomingRtcpPacket, | |
| 98 const size_t rtcpPacketLength, | |
| 99 const int8_t* fromIP, | |
| 100 const uint16_t fromPort) override; | |
| 101 | |
| 102 ///////////////////////////////// | |
| 103 // Sender methods | |
| 104 | |
| 105 int32_t InitSender (const uint32_t startBitrateKbps, | |
| 106 const int8_t* ipAddr, | |
| 107 const uint16_t rtpPort, | |
| 108 const uint16_t rtcpPort = 0, | |
| 109 const int8_t payloadType = 127); | |
| 110 | |
| 111 int32_t SendOutgoingData(const uint32_t timeStamp, | |
| 112 const uint8_t* payloadData, | |
| 113 const size_t payloadSize, | |
| 114 const webrtc::FrameType frameType = webrtc::kVideoFrameDelta); | |
| 115 | |
| 116 int32_t SetLoadGenerator(TestLoadGenerator *generator); | |
| 117 | |
| 118 uint32_t BitrateSent() { return (_rtp->BitrateSent()); }; | |
| 119 | |
| 120 | |
| 121 // Inherited from RtpVideoFeedback | |
| 122 virtual void OnReceivedIntraFrameRequest(const int32_t id, | |
| 123 const uint8_t message = 0) {}; | |
| 124 | |
| 125 virtual void OnNetworkChanged(const int32_t id, | |
| 126 const uint32_t minBitrateBps, | |
| 127 const uint32_t maxBitrateBps, | |
| 128 const uint8_t fractionLost, | |
| 129 const uint16_t roundTripTimeMs, | |
| 130 const uint16_t bwEstimateKbitMin, | |
| 131 const uint16_t bwEstimateKbitMax); | |
| 132 | |
| 133 private: | |
| 134 RtpRtcp* _rtp; | |
| 135 UdpTransport* _transport; | |
| 136 webrtc::CriticalSectionWrapper* _critSect; | |
| 137 webrtc::EventWrapper *_eventPtr; | |
| 138 rtc::PlatformThread _procThread; | |
| 139 bool _running; | |
| 140 int8_t _payloadType; | |
| 141 TestLoadGenerator* _loadGenerator; | |
| 142 bool _isSender; | |
| 143 bool _isReceiver; | |
| 144 SendRecCB * _sendRecCB; | |
| 145 size_t _lastBytesReceived; | |
| 146 int64_t _lastTime; | |
| 147 | |
| 148 }; | |
| 149 } // namespace webrtc | |
| 150 #endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ | |
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Chromium Code Reviews
Issue 1525573002:
rtp_rtcp/test/BWEStandAlone deleted as obsolete (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master