Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

4 files changed, 213 insertions, 0 deletions

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 <memory>
+
+#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<webrtc::AudioFrame> 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 <string>
+
+#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..b24e5a59ec
--- /dev/null
+++ b/third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h
@@ -0,0 +1,105 @@
+/*
+ *  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(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(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_