| /* |
| * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_ |
| #define CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_ |
| |
| #include <map> |
| #include <string> |
| #include <vector> |
| |
| #include "api/bitrate_constraints.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "modules/congestion_controller/include/network_changed_observer.h" |
| #include "modules/pacing/packet_router.h" |
| #include "rtc_base/networkroute.h" |
| #include "rtc_base/rate_limiter.h" |
| #include "rtc_base/socket.h" |
| #include "test/gmock.h" |
| |
| namespace webrtc { |
| |
| class MockRtpTransportControllerSend |
| : public RtpTransportControllerSendInterface { |
| public: |
| MOCK_METHOD8( |
| CreateRtpVideoSender, |
| RtpVideoSenderInterface*(const std::vector<uint32_t>&, |
| std::map<uint32_t, RtpState>, |
| const std::map<uint32_t, RtpPayloadState>&, |
| const RtpConfig&, |
| const RtcpConfig&, |
| Transport*, |
| const RtpSenderObservers&, |
| RtcEventLog*)); |
| MOCK_METHOD1(DestroyRtpVideoSender, void(RtpVideoSenderInterface*)); |
| MOCK_METHOD0(GetWorkerQueue, rtc::TaskQueue*()); |
| MOCK_METHOD0(packet_router, PacketRouter*()); |
| MOCK_METHOD0(transport_feedback_observer, TransportFeedbackObserver*()); |
| MOCK_METHOD0(packet_sender, RtpPacketSender*()); |
| MOCK_CONST_METHOD0(keepalive_config, RtpKeepAliveConfig&()); |
| MOCK_METHOD3(SetAllocatedSendBitrateLimits, void(int, int, int)); |
| MOCK_METHOD1(SetPacingFactor, void(float)); |
| MOCK_METHOD1(SetQueueTimeLimit, void(int)); |
| MOCK_METHOD0(GetCallStatsObserver, CallStatsObserver*()); |
| MOCK_METHOD1(RegisterPacketFeedbackObserver, void(PacketFeedbackObserver*)); |
| MOCK_METHOD1(DeRegisterPacketFeedbackObserver, void(PacketFeedbackObserver*)); |
| MOCK_METHOD1(RegisterTargetTransferRateObserver, |
| void(TargetTransferRateObserver*)); |
| MOCK_METHOD2(OnNetworkRouteChanged, |
| void(const std::string&, const rtc::NetworkRoute&)); |
| MOCK_METHOD1(OnNetworkAvailability, void(bool)); |
| MOCK_METHOD0(GetBandwidthObserver, RtcpBandwidthObserver*()); |
| MOCK_CONST_METHOD0(GetPacerQueuingDelayMs, int64_t()); |
| MOCK_CONST_METHOD0(GetFirstPacketTimeMs, int64_t()); |
| MOCK_METHOD1(SetPerPacketFeedbackAvailable, void(bool)); |
| MOCK_METHOD1(EnablePeriodicAlrProbing, void(bool)); |
| MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket&)); |
| MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&)); |
| MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&)); |
| MOCK_METHOD1(SetAllocatedBitrateWithoutFeedback, void(uint32_t)); |
| }; |
| } // namespace webrtc |
| #endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_ |