webrtc_m130/call/test/mock_rtp_transport_controller_send.h
Erik Språng 59b8654045 Switch from RtpPacketSender to RtpPacketPacer interface usage.
RtpPacketSender interface will be removed when downstream projects have
been updated.

Bug: webrtc:10633
Change-Id: Ie127b9814f39bd213d00ded0f7b98380f2f01084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143175
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28350}
2019-06-24 10:46:06 +00:00

72 lines
3.1 KiB
C++

/*
* 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/crypto_options.h"
#include "api/crypto/frame_encryptor_interface.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_METHOD9(
CreateRtpVideoSender,
RtpVideoSenderInterface*(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&));
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, RtpPacketPacer*());
MOCK_METHOD3(SetAllocatedSendBitrateLimits, void(int, int, int));
MOCK_METHOD1(SetPacingFactor, void(float));
MOCK_METHOD1(SetQueueTimeLimit, void(int));
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(EnablePeriodicAlrProbing, void(bool));
MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket&));
MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&));
MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&));
MOCK_METHOD1(OnTransportOverheadChanged, void(size_t));
MOCK_METHOD1(OnReceivedPacket, void(const ReceivedPacket&));
};
} // namespace webrtc
#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_