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/audio/channel_receive_unittest.cc | 50 ++++++++++++++++++++++ 1 file changed, 50 insertions(+) create mode 100644 third_party/libwebrtc/audio/channel_receive_unittest.cc (limited to 'third_party/libwebrtc/audio/channel_receive_unittest.cc') diff --git a/third_party/libwebrtc/audio/channel_receive_unittest.cc b/third_party/libwebrtc/audio/channel_receive_unittest.cc new file mode 100644 index 0000000000..3d9baebe89 --- /dev/null +++ b/third_party/libwebrtc/audio/channel_receive_unittest.cc @@ -0,0 +1,50 @@ +/* + * Copyright 2023 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 "audio/channel_receive.h" + +#include "api/crypto/frame_decryptor_interface.h" +#include "api/task_queue/default_task_queue_factory.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_device/include/mock_audio_device.h" +#include "rtc_base/thread.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/mock_transport.h" +#include "test/time_controller/simulated_time_controller.h" + +namespace webrtc { +namespace voe { + +TEST(ChannelReceiveTest, CreateAndDestroy) { + GlobalSimulatedTimeController time_controller(Timestamp::Seconds(5555)); + uint32_t local_ssrc = 1111; + uint32_t remote_ssrc = 2222; + webrtc::CryptoOptions crypto_options; + rtc::scoped_refptr audio_device_module = + test::MockAudioDeviceModule::CreateNice(); + MockTransport transport; + auto channel = CreateChannelReceive( + time_controller.GetClock(), + /* neteq_factory= */ nullptr, audio_device_module.get(), &transport, + /* rtc_event_log= */ nullptr, local_ssrc, remote_ssrc, + /* jitter_buffer_max_packets= */ 0, + /* jitter_buffer_fast_playout= */ false, + /* jitter_buffer_min_delay_ms= */ 0, + /* enable_non_sender_rtt= */ false, + /* decoder_factory= */ nullptr, + /* codec_pair_id= */ absl::nullopt, + /* frame_decryptor_interface= */ nullptr, crypto_options, + /* frame_transformer= */ nullptr); + EXPECT_TRUE(!!channel); +} + +} // namespace voe +} // namespace webrtc -- cgit v1.2.3