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/*
* 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/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,
((std::map<uint32_t, RtpState>),
(const std::map<uint32_t, RtpPayloadState>&),
const RtpConfig&,
int rtcp_report_interval_ms,
Transport*,
const RtpSenderObservers&,
RtcEventLog*,
std::unique_ptr<FecController>,
const RtpSenderFrameEncryptionConfig&,
rtc::scoped_refptr<FrameTransformerInterface>),
(override));
MOCK_METHOD(void,
DestroyRtpVideoSender,
(RtpVideoSenderInterface*),
(override));
MOCK_METHOD(rtc::TaskQueue*, GetWorkerQueue, (), (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, SetPacingFactor, (float), (override));
MOCK_METHOD(void, SetQueueTimeLimit, (int), (override));
MOCK_METHOD(StreamFeedbackProvider*,
GetStreamFeedbackProvider,
(),
(override));
MOCK_METHOD(void,
RegisterTargetTransferRateObserver,
(TargetTransferRateObserver*),
(override));
MOCK_METHOD(void,
OnNetworkRouteChanged,
(const std::string&, const rtc::NetworkRoute&),
(override));
MOCK_METHOD(void, OnNetworkAvailability, (bool), (override));
MOCK_METHOD(RtcpBandwidthObserver*, GetBandwidthObserver, (), (override));
MOCK_METHOD(int64_t, GetPacerQueuingDelayMs, (), (const, override));
MOCK_METHOD(absl::optional<Timestamp>,
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));
};
} // namespace webrtc
#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
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