mirror of
https://github.com/mollyim/webrtc.git
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f298855981
Removes all unused features, reducing the exposed interface surface. This makes refactoring and maintenance simpler as we can change TransportFeedbackAdapter without making corresponding changes to RtpVideoSender. Bug: webrtc:9883 Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29649}
73 lines
3.2 KiB
C++
73 lines
3.2 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 "api/crypto/crypto_options.h"
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#include "api/crypto/frame_encryptor_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 "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_METHOD9(
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CreateRtpVideoSender,
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RtpVideoSenderInterface*(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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RtcEventLog*,
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std::unique_ptr<FecController>,
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const RtpSenderFrameEncryptionConfig&));
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MOCK_METHOD1(DestroyRtpVideoSender, void(RtpVideoSenderInterface*));
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MOCK_METHOD0(GetWorkerQueue, rtc::TaskQueue*());
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MOCK_METHOD0(packet_router, PacketRouter*());
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MOCK_METHOD0(network_state_estimate_observer,
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NetworkStateEstimateObserver*());
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MOCK_METHOD0(transport_feedback_observer, TransportFeedbackObserver*());
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MOCK_METHOD0(packet_sender, RtpPacketSender*());
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MOCK_METHOD1(SetAllocatedSendBitrateLimits, void(BitrateAllocationLimits));
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MOCK_METHOD1(SetPacingFactor, void(float));
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MOCK_METHOD1(SetQueueTimeLimit, void(int));
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MOCK_METHOD0(GetStreamFeedbackProvider, StreamFeedbackProvider*());
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MOCK_METHOD1(RegisterTargetTransferRateObserver,
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void(TargetTransferRateObserver*));
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MOCK_METHOD2(OnNetworkRouteChanged,
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void(const std::string&, const rtc::NetworkRoute&));
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MOCK_METHOD1(OnNetworkAvailability, void(bool));
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MOCK_METHOD0(GetBandwidthObserver, RtcpBandwidthObserver*());
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MOCK_CONST_METHOD0(GetPacerQueuingDelayMs, int64_t());
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MOCK_CONST_METHOD0(GetFirstPacketTime, absl::optional<Timestamp>());
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MOCK_METHOD1(EnablePeriodicAlrProbing, void(bool));
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MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket&));
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MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&));
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MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&));
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MOCK_METHOD1(OnTransportOverheadChanged, void(size_t));
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MOCK_METHOD1(AccountForAudioPacketsInPacedSender, void(bool));
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MOCK_METHOD1(OnReceivedPacket, void(const ReceivedPacket&));
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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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