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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/call/test
parentInitial commit. (diff)
downloadfirefox-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')
-rw-r--r--third_party/libwebrtc/call/test/mock_audio_send_stream.h51
-rw-r--r--third_party/libwebrtc/call/test/mock_bitrate_allocator.h32
-rw-r--r--third_party/libwebrtc/call/test/mock_rtp_packet_sink_interface.h25
-rw-r--r--third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h106
4 files changed, 214 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..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_