mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
39ac25d6ec
Using the Api, BWE components are recreated and new settings can be applied. Initially, the only configuration available is allowing BWE probes without media". Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration. Integration test in https://webrtc-review.googlesource.com/c/src/+/337322 Bug: webrtc:14928 Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41687}
114 lines
4.3 KiB
C++
114 lines
4.3 KiB
C++
/*
|
|
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
|
|
#define CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
|
|
|
|
#include <map>
|
|
#include <memory>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "absl/strings/string_view.h"
|
|
#include "api/crypto/crypto_options.h"
|
|
#include "api/crypto/frame_encryptor_interface.h"
|
|
#include "api/frame_transformer_interface.h"
|
|
#include "api/transport/bitrate_settings.h"
|
|
#include "call/rtp_transport_controller_send_interface.h"
|
|
#include "modules/pacing/packet_router.h"
|
|
#include "rtc_base/network/sent_packet.h"
|
|
#include "rtc_base/network_route.h"
|
|
#include "rtc_base/rate_limiter.h"
|
|
#include "test/gmock.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class MockRtpTransportControllerSend
|
|
: public RtpTransportControllerSendInterface {
|
|
public:
|
|
MOCK_METHOD(RtpVideoSenderInterface*,
|
|
CreateRtpVideoSender,
|
|
((const std::map<uint32_t, RtpState>&),
|
|
(const std::map<uint32_t, RtpPayloadState>&),
|
|
const RtpConfig&,
|
|
int rtcp_report_interval_ms,
|
|
Transport*,
|
|
const RtpSenderObservers&,
|
|
RtcEventLog*,
|
|
std::unique_ptr<FecController>,
|
|
const RtpSenderFrameEncryptionConfig&,
|
|
rtc::scoped_refptr<FrameTransformerInterface>),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
DestroyRtpVideoSender,
|
|
(RtpVideoSenderInterface*),
|
|
(override));
|
|
MOCK_METHOD(void, RegisterSendingRtpStream, (RtpRtcpInterface&), (override));
|
|
MOCK_METHOD(void,
|
|
DeRegisterSendingRtpStream,
|
|
(RtpRtcpInterface&),
|
|
(override));
|
|
MOCK_METHOD(PacketRouter*, packet_router, (), (override));
|
|
MOCK_METHOD(NetworkStateEstimateObserver*,
|
|
network_state_estimate_observer,
|
|
(),
|
|
(override));
|
|
MOCK_METHOD(TransportFeedbackObserver*,
|
|
transport_feedback_observer,
|
|
(),
|
|
(override));
|
|
MOCK_METHOD(RtpPacketSender*, packet_sender, (), (override));
|
|
MOCK_METHOD(void,
|
|
SetAllocatedSendBitrateLimits,
|
|
(BitrateAllocationLimits),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
ReconfigureBandwidthEstimation,
|
|
(const BandwidthEstimationSettings&),
|
|
(override));
|
|
MOCK_METHOD(void, SetPacingFactor, (float), (override));
|
|
MOCK_METHOD(void, SetQueueTimeLimit, (int), (override));
|
|
MOCK_METHOD(StreamFeedbackProvider*,
|
|
GetStreamFeedbackProvider,
|
|
(),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
RegisterTargetTransferRateObserver,
|
|
(TargetTransferRateObserver*),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
OnNetworkRouteChanged,
|
|
(absl::string_view, const rtc::NetworkRoute&),
|
|
(override));
|
|
MOCK_METHOD(void, OnNetworkAvailability, (bool), (override));
|
|
MOCK_METHOD(NetworkLinkRtcpObserver*, GetRtcpObserver, (), (override));
|
|
MOCK_METHOD(int64_t, GetPacerQueuingDelayMs, (), (const, override));
|
|
MOCK_METHOD(absl::optional<Timestamp>,
|
|
GetFirstPacketTime,
|
|
(),
|
|
(const, override));
|
|
MOCK_METHOD(void, EnablePeriodicAlrProbing, (bool), (override));
|
|
MOCK_METHOD(void, OnSentPacket, (const rtc::SentPacket&), (override));
|
|
MOCK_METHOD(void,
|
|
SetSdpBitrateParameters,
|
|
(const BitrateConstraints&),
|
|
(override));
|
|
MOCK_METHOD(void,
|
|
SetClientBitratePreferences,
|
|
(const BitrateSettings&),
|
|
(override));
|
|
MOCK_METHOD(void, OnTransportOverheadChanged, (size_t), (override));
|
|
MOCK_METHOD(void, AccountForAudioPacketsInPacedSender, (bool), (override));
|
|
MOCK_METHOD(void, IncludeOverheadInPacedSender, (), (override));
|
|
MOCK_METHOD(void, OnReceivedPacket, (const ReceivedPacket&), (override));
|
|
MOCK_METHOD(void, EnsureStarted, (), (override));
|
|
};
|
|
} // namespace webrtc
|
|
#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
|