/* * 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 #include #include #include #include "absl/strings/string_view.h" #include "api/crypto/crypto_options.h" #include "api/crypto/frame_encryptor_interface.h" #include "api/frame_transformer_interface.h" #include "api/transport/bitrate_settings.h" #include "call/rtp_transport_controller_send_interface.h" #include "modules/pacing/packet_router.h" #include "rtc_base/network/sent_packet.h" #include "rtc_base/network_route.h" #include "rtc_base/rate_limiter.h" #include "test/gmock.h" namespace webrtc { class MockRtpTransportControllerSend : public RtpTransportControllerSendInterface { public: MOCK_METHOD(RtpVideoSenderInterface*, CreateRtpVideoSender, ((const std::map&), (const std::map&), const RtpConfig&, int rtcp_report_interval_ms, Transport*, const RtpSenderObservers&, RtcEventLog*, std::unique_ptr, const RtpSenderFrameEncryptionConfig&, rtc::scoped_refptr), (override)); MOCK_METHOD(void, DestroyRtpVideoSender, (RtpVideoSenderInterface*), (override)); MOCK_METHOD(void, RegisterSendingRtpStream, (RtpRtcpInterface&), (override)); MOCK_METHOD(void, DeRegisterSendingRtpStream, (RtpRtcpInterface&), (override)); MOCK_METHOD(PacketRouter*, packet_router, (), (override)); MOCK_METHOD(NetworkStateEstimateObserver*, network_state_estimate_observer, (), (override)); MOCK_METHOD(TransportFeedbackObserver*, transport_feedback_observer, (), (override)); MOCK_METHOD(RtpPacketSender*, packet_sender, (), (override)); MOCK_METHOD(void, SetAllocatedSendBitrateLimits, (BitrateAllocationLimits), (override)); MOCK_METHOD(void, ReconfigureBandwidthEstimation, (const BandwidthEstimationSettings&), (override)); MOCK_METHOD(void, SetPacingFactor, (float), (override)); MOCK_METHOD(void, SetQueueTimeLimit, (int), (override)); MOCK_METHOD(StreamFeedbackProvider*, GetStreamFeedbackProvider, (), (override)); MOCK_METHOD(void, RegisterTargetTransferRateObserver, (TargetTransferRateObserver*), (override)); MOCK_METHOD(void, OnNetworkRouteChanged, (absl::string_view, const rtc::NetworkRoute&), (override)); MOCK_METHOD(void, OnNetworkAvailability, (bool), (override)); MOCK_METHOD(NetworkLinkRtcpObserver*, GetRtcpObserver, (), (override)); MOCK_METHOD(int64_t, GetPacerQueuingDelayMs, (), (const, override)); MOCK_METHOD(absl::optional, GetFirstPacketTime, (), (const, override)); MOCK_METHOD(void, EnablePeriodicAlrProbing, (bool), (override)); MOCK_METHOD(void, OnSentPacket, (const rtc::SentPacket&), (override)); MOCK_METHOD(void, SetSdpBitrateParameters, (const BitrateConstraints&), (override)); MOCK_METHOD(void, SetClientBitratePreferences, (const BitrateSettings&), (override)); MOCK_METHOD(void, OnTransportOverheadChanged, (size_t), (override)); MOCK_METHOD(void, AccountForAudioPacketsInPacedSender, (bool), (override)); MOCK_METHOD(void, IncludeOverheadInPacedSender, (), (override)); MOCK_METHOD(void, OnReceivedPacket, (const ReceivedPacket&), (override)); MOCK_METHOD(void, EnsureStarted, (), (override)); }; } // namespace webrtc #endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_