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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h')
-rw-r--r-- | third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h | 106 |
1 files changed, 106 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h b/third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h new file mode 100644 index 0000000000..6e78534de2 --- /dev/null +++ b/third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h @@ -0,0 +1,106 @@ +/* + * 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(MaybeWorkerThread*, GetWorkerQueue, (), (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, 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(RtcpBandwidthObserver*, GetBandwidthObserver, (), (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_ |