418dd0b96a
There are two RTT values reported to GoogCC. They come from the same source initially but one is calculated and smoothed in the video call stats. However, there's not really any technical reasons why this value should be received via the stats, this has just been maintained for legacy reasons. Experiments shows no real difference between the modes, therefore the stats-reported RTT is removed in this CL as a cleanup. Bug: None Change-Id: If1462d6c91570ffb883ecef2ba034f04a571c9b5 Reviewed-on: https://webrtc-review.googlesource.com/c/123883 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26833}
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/bitrate_constraints.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 "call/rtp_transport_controller_send_interface.h"
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#include "modules/congestion_controller/include/network_changed_observer.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(transport_feedback_observer, TransportFeedbackObserver*());
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MOCK_METHOD0(packet_sender, RtpPacketSender*());
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MOCK_CONST_METHOD0(keepalive_config, RtpKeepAliveConfig&());
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MOCK_METHOD3(SetAllocatedSendBitrateLimits, void(int, int, int));
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MOCK_METHOD1(SetPacingFactor, void(float));
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MOCK_METHOD1(SetQueueTimeLimit, void(int));
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MOCK_METHOD1(RegisterPacketFeedbackObserver, void(PacketFeedbackObserver*));
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MOCK_METHOD1(DeRegisterPacketFeedbackObserver, void(PacketFeedbackObserver*));
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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(GetFirstPacketTimeMs, int64_t());
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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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};
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} // namespace webrtc
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#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
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