mirror of
https://github.com/mollyim/webrtc.git
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b4d4bbcebd
This reverts commit 2f4bc64166.
Reason for revert: Breaks downstream test
Original change's description:
> Clean up last_packet_received_time_ as it's no longer used.
>
> Bug: webrtc:15377
> Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40792}
Bug: webrtc:15377
Change-Id: Ifa57671cc479cdd86f543c4edc236221beb76f90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321340
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40797}
105 lines
4 KiB
C++
105 lines
4 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 "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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RtcEventLog*,
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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(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(TransportFeedbackObserver*,
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transport_feedback_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, 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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