Adding upstream version 115.8.0esr.upstream/115.8.0esr

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

3 files changed, 189 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/voip/test/compile_all_headers.cc b/third_party/libwebrtc/api/voip/test/compile_all_headers.cc
new file mode 100644
index 0000000000..73a0f0d1c4
--- /dev/null
+++ b/third_party/libwebrtc/api/voip/test/compile_all_headers.cc
@@ -0,0 +1,14 @@
+/*
+ *  Copyright (c) 2021 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.
+ */
+
+// This file verifies that all include files in this directory can be
+// compiled without errors or other required includes.
+
+#include "api/voip/test/mock_voip_engine.h"
diff --git a/third_party/libwebrtc/api/voip/test/mock_voip_engine.h b/third_party/libwebrtc/api/voip/test/mock_voip_engine.h
new file mode 100644
index 0000000000..74b880d652
--- /dev/null
+++ b/third_party/libwebrtc/api/voip/test/mock_voip_engine.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2021 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 API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
+#define API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
+
+#include <map>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_codec.h"
+#include "api/voip/voip_dtmf.h"
+#include "api/voip/voip_engine.h"
+#include "api/voip/voip_network.h"
+#include "api/voip/voip_statistics.h"
+#include "api/voip/voip_volume_control.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockVoipBase : public VoipBase {
+ public:
+  MOCK_METHOD(ChannelId,
+              CreateChannel,
+              (Transport*, absl::optional<uint32_t>),
+              (override));
+  MOCK_METHOD(VoipResult, ReleaseChannel, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StartSend, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StopSend, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StartPlayout, (ChannelId), (override));
+  MOCK_METHOD(VoipResult, StopPlayout, (ChannelId), (override));
+};
+
+class MockVoipCodec : public VoipCodec {
+ public:
+  MOCK_METHOD(VoipResult,
+              SetSendCodec,
+              (ChannelId, int, const SdpAudioFormat&),
+              (override));
+  MOCK_METHOD(VoipResult,
+              SetReceiveCodecs,
+              (ChannelId, (const std::map<int, SdpAudioFormat>&)),
+              (override));
+};
+
+class MockVoipDtmf : public VoipDtmf {
+ public:
+  MOCK_METHOD(VoipResult,
+              RegisterTelephoneEventType,
+              (ChannelId, int, int),
+              (override));
+  MOCK_METHOD(VoipResult,
+              SendDtmfEvent,
+              (ChannelId, DtmfEvent, int),
+              (override));
+};
+
+class MockVoipNetwork : public VoipNetwork {
+ public:
+  MOCK_METHOD(VoipResult,
+              ReceivedRTPPacket,
+              (ChannelId channel_id, rtc::ArrayView<const uint8_t> rtp_packet),
+              (override));
+  MOCK_METHOD(VoipResult,
+              ReceivedRTCPPacket,
+              (ChannelId channel_id, rtc::ArrayView<const uint8_t> rtcp_packet),
+              (override));
+};
+
+class MockVoipStatistics : public VoipStatistics {
+ public:
+  MOCK_METHOD(VoipResult,
+              GetIngressStatistics,
+              (ChannelId, IngressStatistics&),
+              (override));
+  MOCK_METHOD(VoipResult,
+              GetChannelStatistics,
+              (ChannelId channel_id, ChannelStatistics&),
+              (override));
+};
+
+class MockVoipVolumeControl : public VoipVolumeControl {
+ public:
+  MOCK_METHOD(VoipResult, SetInputMuted, (ChannelId, bool), (override));
+
+  MOCK_METHOD(VoipResult,
+              GetInputVolumeInfo,
+              (ChannelId, VolumeInfo&),
+              (override));
+  MOCK_METHOD(VoipResult,
+              GetOutputVolumeInfo,
+              (ChannelId, VolumeInfo&),
+              (override));
+};
+
+class MockVoipEngine : public VoipEngine {
+ public:
+  VoipBase& Base() override { return base_; }
+  VoipNetwork& Network() override { return network_; }
+  VoipCodec& Codec() override { return codec_; }
+  VoipDtmf& Dtmf() override { return dtmf_; }
+  VoipStatistics& Statistics() override { return statistics_; }
+  VoipVolumeControl& VolumeControl() override { return volume_; }
+
+  // Direct access to underlying members are required for testing.
+  MockVoipBase base_;
+  MockVoipNetwork network_;
+  MockVoipCodec codec_;
+  MockVoipDtmf dtmf_;
+  MockVoipStatistics statistics_;
+  MockVoipVolumeControl volume_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
diff --git a/third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc b/third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc
new file mode 100644
index 0000000000..f967a0ba8f
--- /dev/null
+++ b/third_party/libwebrtc/api/voip/test/voip_engine_factory_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#include <utility>
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/voip/voip_engine_factory.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_audio_encoder_factory.h"
+
+namespace webrtc {
+namespace {
+
+// Create voip engine with mock modules as normal use case.
+TEST(VoipEngineFactoryTest, CreateEngineWithMockModules) {
+  VoipEngineConfig config;
+  config.encoder_factory = rtc::make_ref_counted<MockAudioEncoderFactory>();
+  config.decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  config.task_queue_factory = CreateDefaultTaskQueueFactory();
+  config.audio_processing =
+      rtc::make_ref_counted<testing::NiceMock<test::MockAudioProcessing>>();
+  config.audio_device_module = test::MockAudioDeviceModule::CreateNice();
+
+  auto voip_engine = CreateVoipEngine(std::move(config));
+  EXPECT_NE(voip_engine, nullptr);
+}
+
+// Create voip engine without setting audio processing as optional component.
+TEST(VoipEngineFactoryTest, UseNoAudioProcessing) {
+  VoipEngineConfig config;
+  config.encoder_factory = rtc::make_ref_counted<MockAudioEncoderFactory>();
+  config.decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  config.task_queue_factory = CreateDefaultTaskQueueFactory();
+  config.audio_device_module = test::MockAudioDeviceModule::CreateNice();
+
+  auto voip_engine = CreateVoipEngine(std::move(config));
+  EXPECT_NE(voip_engine, nullptr);
+}
+
+}  // namespace
+}  // namespace webrtc