From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- .../libwebrtc/call/test/mock_audio_send_stream.h | 51 ++++++++++ .../libwebrtc/call/test/mock_bitrate_allocator.h | 32 +++++++ .../call/test/mock_rtp_packet_sink_interface.h | 25 +++++ .../call/test/mock_rtp_transport_controller_send.h | 106 +++++++++++++++++++++ 4 files changed, 214 insertions(+) create mode 100644 third_party/libwebrtc/call/test/mock_audio_send_stream.h create mode 100644 third_party/libwebrtc/call/test/mock_bitrate_allocator.h create mode 100644 third_party/libwebrtc/call/test/mock_rtp_packet_sink_interface.h create mode 100644 third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h (limited to 'third_party/libwebrtc/call/test') diff --git a/third_party/libwebrtc/call/test/mock_audio_send_stream.h b/third_party/libwebrtc/call/test/mock_audio_send_stream.h new file mode 100644 index 0000000000..1993de8de0 --- /dev/null +++ b/third_party/libwebrtc/call/test/mock_audio_send_stream.h @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2015 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_AUDIO_SEND_STREAM_H_ +#define CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_ + +#include + +#include "call/audio_send_stream.h" +#include "test/gmock.h" + +namespace webrtc { +namespace test { + +class MockAudioSendStream : public AudioSendStream { + public: + MOCK_METHOD(const webrtc::AudioSendStream::Config&, + GetConfig, + (), + (const, override)); + MOCK_METHOD(void, + Reconfigure, + (const Config& config, SetParametersCallback callback), + (override)); + MOCK_METHOD(void, Start, (), (override)); + MOCK_METHOD(void, Stop, (), (override)); + // GMock doesn't like move-only types, such as std::unique_ptr. + void SendAudioData(std::unique_ptr audio_frame) override { + SendAudioDataForMock(audio_frame.get()); + } + MOCK_METHOD(void, SendAudioDataForMock, (webrtc::AudioFrame*)); + MOCK_METHOD( + bool, + SendTelephoneEvent, + (int payload_type, int payload_frequency, int event, int duration_ms), + (override)); + MOCK_METHOD(void, SetMuted, (bool muted), (override)); + MOCK_METHOD(Stats, GetStats, (), (const, override)); + MOCK_METHOD(Stats, GetStats, (bool has_remote_tracks), (const, override)); +}; +} // namespace test +} // namespace webrtc + +#endif // CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_ diff --git a/third_party/libwebrtc/call/test/mock_bitrate_allocator.h b/third_party/libwebrtc/call/test/mock_bitrate_allocator.h new file mode 100644 index 0000000000..b08916fe4f --- /dev/null +++ b/third_party/libwebrtc/call/test/mock_bitrate_allocator.h @@ -0,0 +1,32 @@ +/* + * Copyright 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_BITRATE_ALLOCATOR_H_ +#define CALL_TEST_MOCK_BITRATE_ALLOCATOR_H_ + +#include + +#include "call/bitrate_allocator.h" +#include "test/gmock.h" + +namespace webrtc { +class MockBitrateAllocator : public BitrateAllocatorInterface { + public: + MOCK_METHOD(void, + AddObserver, + (BitrateAllocatorObserver*, MediaStreamAllocationConfig), + (override)); + MOCK_METHOD(void, RemoveObserver, (BitrateAllocatorObserver*), (override)); + MOCK_METHOD(int, + GetStartBitrate, + (BitrateAllocatorObserver*), + (const, override)); +}; +} // namespace webrtc +#endif // CALL_TEST_MOCK_BITRATE_ALLOCATOR_H_ diff --git a/third_party/libwebrtc/call/test/mock_rtp_packet_sink_interface.h b/third_party/libwebrtc/call/test/mock_rtp_packet_sink_interface.h new file mode 100644 index 0000000000..e6d14f05c5 --- /dev/null +++ b/third_party/libwebrtc/call/test/mock_rtp_packet_sink_interface.h @@ -0,0 +1,25 @@ +/* + * Copyright (c) 2017 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_PACKET_SINK_INTERFACE_H_ +#define CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_ + +#include "call/rtp_packet_sink_interface.h" +#include "test/gmock.h" + +namespace webrtc { + +class MockRtpPacketSink : public RtpPacketSinkInterface { + public: + MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived&), (override)); +}; + +} // namespace webrtc + +#endif // CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_ 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 +#include +#include +#include + +#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&), + (const std::map&), + const RtpConfig&, + int rtcp_report_interval_ms, + Transport*, + const RtpSenderObservers&, + RtcEventLog*, + std::unique_ptr, + const RtpSenderFrameEncryptionConfig&, + rtc::scoped_refptr), + (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, + 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_ -- cgit v1.2.3