| /* |
| * 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 "absl/strings/string_view.h" |
| #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, |
| ((const 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, |
| (absl::string_view, 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)); |
| MOCK_METHOD(void, EnsureStarted, (), (override)); |
| }; |
| } // namespace webrtc |
| #endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_ |