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f459fb31b9
Bug: None Change-Id: If5976873027981c08ea0a99bfbe29f9bfe4ba34b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347861 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42118}
110 lines
4.2 KiB
C++
110 lines
4.2 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
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#define CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/strings/string_view.h"
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#include "api/crypto/crypto_options.h"
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#include "api/crypto/frame_encryptor_interface.h"
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#include "api/frame_transformer_interface.h"
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#include "api/transport/bitrate_settings.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/rate_limiter.h"
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#include "test/gmock.h"
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namespace webrtc {
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class MockRtpTransportControllerSend
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: public RtpTransportControllerSendInterface {
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public:
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MOCK_METHOD(RtpVideoSenderInterface*,
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CreateRtpVideoSender,
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((const std::map<uint32_t, RtpState>&),
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(const std::map<uint32_t, RtpPayloadState>&),
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const RtpConfig&,
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int rtcp_report_interval_ms,
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Transport*,
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const RtpSenderObservers&,
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std::unique_ptr<FecController>,
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const RtpSenderFrameEncryptionConfig&,
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rtc::scoped_refptr<FrameTransformerInterface>),
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(override));
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MOCK_METHOD(void,
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DestroyRtpVideoSender,
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(RtpVideoSenderInterface*),
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(override));
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MOCK_METHOD(void, RegisterSendingRtpStream, (RtpRtcpInterface&), (override));
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MOCK_METHOD(void,
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DeRegisterSendingRtpStream,
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(RtpRtcpInterface&),
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(override));
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MOCK_METHOD(PacketRouter*, packet_router, (), (override));
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MOCK_METHOD(NetworkStateEstimateObserver*,
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network_state_estimate_observer,
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(),
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(override));
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MOCK_METHOD(RtpPacketSender*, packet_sender, (), (override));
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MOCK_METHOD(void,
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SetAllocatedSendBitrateLimits,
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(BitrateAllocationLimits),
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(override));
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MOCK_METHOD(void,
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ReconfigureBandwidthEstimation,
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(const BandwidthEstimationSettings&),
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(override));
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MOCK_METHOD(void, SetPacingFactor, (float), (override));
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MOCK_METHOD(void, SetQueueTimeLimit, (int), (override));
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MOCK_METHOD(StreamFeedbackProvider*,
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GetStreamFeedbackProvider,
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(),
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(override));
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MOCK_METHOD(void,
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RegisterTargetTransferRateObserver,
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(TargetTransferRateObserver*),
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(override));
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MOCK_METHOD(void,
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OnNetworkRouteChanged,
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(absl::string_view, const rtc::NetworkRoute&),
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(override));
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MOCK_METHOD(void, OnNetworkAvailability, (bool), (override));
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MOCK_METHOD(NetworkLinkRtcpObserver*, GetRtcpObserver, (), (override));
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MOCK_METHOD(int64_t, GetPacerQueuingDelayMs, (), (const, override));
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MOCK_METHOD(absl::optional<Timestamp>,
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GetFirstPacketTime,
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(),
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(const, override));
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MOCK_METHOD(void, EnablePeriodicAlrProbing, (bool), (override));
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MOCK_METHOD(void, OnSentPacket, (const rtc::SentPacket&), (override));
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MOCK_METHOD(void,
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SetSdpBitrateParameters,
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(const BitrateConstraints&),
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(override));
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MOCK_METHOD(void,
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SetClientBitratePreferences,
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(const BitrateSettings&),
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(override));
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MOCK_METHOD(void, OnTransportOverheadChanged, (size_t), (override));
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MOCK_METHOD(void, AccountForAudioPacketsInPacedSender, (bool), (override));
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MOCK_METHOD(void, IncludeOverheadInPacedSender, (), (override));
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MOCK_METHOD(void, OnReceivedPacket, (const ReceivedPacket&), (override));
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MOCK_METHOD(void, EnsureStarted, (), (override));
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};
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} // namespace webrtc
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#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
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