This CL adds an unittest that a ChannelReceive can be constructed and destroyed without crashing. It is a basis for further testing. Lack of unit test was discovered while pursuing bug mentioned below. Bug: webrtc:13931 Change-Id: Iddb110f2df25e3806c74a5d00bbfab6d6d8e267f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291338 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39200}
51 lines
1.8 KiB
C++
51 lines
1.8 KiB
C++
/*
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* Copyright 2023 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/channel_receive.h"
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_device/include/mock_audio_device.h"
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#include "rtc_base/thread.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/mock_transport.h"
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#include "test/time_controller/simulated_time_controller.h"
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namespace webrtc {
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namespace voe {
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TEST(ChannelReceiveTest, CreateAndDestroy) {
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GlobalSimulatedTimeController time_controller(Timestamp::Seconds(5555));
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uint32_t local_ssrc = 1111;
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uint32_t remote_ssrc = 2222;
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webrtc::CryptoOptions crypto_options;
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rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_module =
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test::MockAudioDeviceModule::CreateNice();
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MockTransport transport;
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auto channel = CreateChannelReceive(
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time_controller.GetClock(),
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/* neteq_factory= */ nullptr, audio_device_module.get(), &transport,
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/* rtc_event_log= */ nullptr, local_ssrc, remote_ssrc,
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/* jitter_buffer_max_packets= */ 0,
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/* jitter_buffer_fast_playout= */ false,
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/* jitter_buffer_min_delay_ms= */ 0,
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/* enable_non_sender_rtt= */ false,
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/* decoder_factory= */ nullptr,
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/* codec_pair_id= */ absl::nullopt,
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/* frame_decryptor_interface= */ nullptr, crypto_options,
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/* frame_transformer= */ nullptr);
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EXPECT_TRUE(!!channel);
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}
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} // namespace voe
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} // namespace webrtc
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