blob: 621447dae556134ee8f192c0e27f544bbaf429de [file] [log] [blame]
/*
* 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 <memory>
#include <string>
#include <vector>
#include "api/bitrate_constraints.h"
#include "api/crypto/cryptooptions.h"
#include "api/crypto/frameencryptorinterface.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/network/sent_packet.h"
#include "rtc_base/networkroute.h"
#include "rtc_base/rate_limiter.h"
#include "test/gmock.h"
namespace webrtc {
class MockRtpTransportControllerSend
: public RtpTransportControllerSendInterface {
public:
MOCK_METHOD10(
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*,
std::unique_ptr<FecController>,
const RtpSenderFrameEncryptionConfig&));
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));
MOCK_METHOD1(OnTransportOverheadChanged, void(size_t));
};
} // namespace webrtc
#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_