| Index: webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..034523215e3d9bc26c59eb6afd34b63e1590f5de
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
|
| @@ -0,0 +1,76 @@
|
| +/*
|
| + * 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.
|
| + */
|
| +
|
| +#include <array>
|
| +#include <utility>
|
| +
|
| +#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
|
| +
|
| +#include "webrtc/base/task_queue.h"
|
| +#include "webrtc/modules/include/module_common_types.h"
|
| +#include "webrtc/test/gtest.h"
|
| +#include "webrtc/test/testsupport/fileutils.h"
|
| +
|
| +TEST(AecDumper, APICallsDoNotCrash) {
|
| + // Note order of initialization: AecDump uses the task queue in its
|
| + // d-tor. The task queue has to be initialized first.
|
| + rtc::TaskQueue file_writer_queue("file_writer_queue");
|
| +
|
| + std::vector<std::string> file_names;
|
| + for (const std::string prefix : {"aecdump0", "aecdump1", "aecdump2"}) {
|
| + file_names.push_back(
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| + webrtc::test::TempFilename(webrtc::test::OutputPath(), prefix));
|
| + }
|
| +
|
| + auto aec_dump =
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| + webrtc::AecDumpFactory::Create(file_names[0], -1, &file_writer_queue);
|
| + EXPECT_TRUE(aec_dump);
|
| +
|
| + aec_dump =
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| + webrtc::AecDumpFactory::Create(file_names[1], -1, &file_writer_queue);
|
| +
|
| + aec_dump =
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| + webrtc::AecDumpFactory::Create(file_names[2], 10, &file_writer_queue);
|
| +
|
| + FILE* fid = nullptr;
|
| + aec_dump = webrtc::AecDumpFactory::Create(fid, 10, &file_writer_queue);
|
| +
|
| + const webrtc::AudioFrame frame;
|
| + aec_dump->WriteRenderStreamMessage(frame);
|
| +
|
| + auto capture_stream_info = aec_dump->GetCaptureStreamInfo();
|
| +
|
| + capture_stream_info->AddInput(frame);
|
| + capture_stream_info->AddOutput(frame);
|
| +
|
| + aec_dump->WriteCaptureStreamMessage(std::move(capture_stream_info));
|
| +
|
| + capture_stream_info = aec_dump->GetCaptureStreamInfo();
|
| + std::vector<rtc::ArrayView<const float>> audio_channels;
|
| + std::array<float, 160> audio_data;
|
| + audio_channels.emplace_back(audio_data);
|
| +
|
| + capture_stream_info->AddInput(audio_channels);
|
| + capture_stream_info->AddOutput(audio_channels);
|
| +
|
| + aec_dump->WriteCaptureStreamMessage(std::move(capture_stream_info));
|
| +
|
| + webrtc::InternalAPMConfig apm_config;
|
| + aec_dump->WriteConfig(apm_config, false);
|
| +
|
| + aec_dump->WriteConfig(apm_config, true);
|
| +
|
| + webrtc::ProcessingConfig processing_config;
|
| + aec_dump->WriteInitMessage(processing_config);
|
| +
|
| + for (const auto& filename : file_names) {
|
| + remove(filename.c_str());
|
| + }
|
| +}
|
|
|