e8964903a9
This reverts commit ab65d8aab5fe63619033371fca1ce2711c2c2137. Reason for revert: Fails video_engine_tests ExtendedReportsEndToEndTest.TestExtendedReportsCanSignalZeroTargetBitrate https://ci.chromium.org/p/webrtc/builders/ci/Linux%20MSan/18366 Original change's description: > Fix target bitrate RTCP messages behavior for SVC streams > > Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders > were created. The RTCP target bitrate messages were treated as simulcast > and were split and send for each separate spatial layer in a separate SSRC. > > To fix that an svc flag is now wired to VideoSendStream config > and filled based on the encoder config in WebrtcVideoEngine. This flag is > used to differentiate between simulcast and SVC mode in RtpVideoSender. > > Bug: webrtc:10485 > Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929 > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27345} TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org Change-Id: I184f87289d9dccc67de165038d76a5690158a3b5 No-Tree-Checks: True No-Try: True Bug: webrtc:10485 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130467 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27355}
72 lines
3.1 KiB
C++
72 lines
3.1 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_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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