e77912ba8c
Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
77 lines
3.4 KiB
C++
77 lines
3.4 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 "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_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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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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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(network_state_estimate_observer,
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NetworkStateEstimateObserver*());
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MOCK_METHOD0(transport_feedback_observer, TransportFeedbackObserver*());
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MOCK_METHOD0(packet_sender, RtpPacketSender*());
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MOCK_METHOD1(SetAllocatedSendBitrateLimits, void(BitrateAllocationLimits));
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MOCK_METHOD1(SetPacingFactor, void(float));
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MOCK_METHOD1(SetQueueTimeLimit, void(int));
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MOCK_METHOD0(GetStreamFeedbackProvider, StreamFeedbackProvider*());
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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(GetFirstPacketTime, absl::optional<Timestamp>());
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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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MOCK_METHOD1(AccountForAudioPacketsInPacedSender, void(bool));
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MOCK_METHOD0(IncludeOverheadInPacedSender, void());
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MOCK_METHOD1(OnReceivedPacket, void(const ReceivedPacket&));
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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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