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/*
* 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<test::MockAudioDeviceModule> 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
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