ab65d8aab5
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}
73 lines
3.1 KiB
C++
73 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_METHOD10(
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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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bool,
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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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