| 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..b5736ec15b18c57c1296d49d3d004ad95d083510
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
|
| @@ -0,0 +1,73 @@
|
| +/*
|
| + * 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 <utility>
|
| +
|
| +#include "webrtc/modules/audio_processing/aec_dump/aec_dump.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: Task queue has to be initialized
|
| + // before AecDump.
|
| + rtc::TaskQueue file_writer_queue("file_writer_queue");
|
| +
|
| + const std::string filename =
|
| + webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
|
| +
|
| + {
|
| + webrtc::AecDump aec_dump(filename, -1, &file_writer_queue);
|
| +
|
| + const webrtc::AudioFrame frame;
|
| + aec_dump.WriteRenderStreamMessage(frame);
|
| +
|
| + webrtc::AecDump::CaptureStreamInfo capture_stream_info;
|
| +
|
| + capture_stream_info.AddInput(frame);
|
| + capture_stream_info.AddOutput(frame);
|
| +
|
| + aec_dump.WriteCaptureStreamMessage(&capture_stream_info);
|
| +
|
| + webrtc::InternalAPMConfig apm_config;
|
| + aec_dump.WriteConfig(apm_config, false);
|
| +
|
| + aec_dump.WriteConfig(apm_config, true);
|
| +
|
| + webrtc::InternalAPMStreamsConfig streams_config;
|
| + aec_dump.WriteInitMessage(streams_config);
|
| + }
|
| + // Remove file after the AecDump d-tor has finished.
|
| + ASSERT_EQ(0, remove(filename.c_str()));
|
| +}
|
| +
|
| +TEST(AecDumper, WriteToFile) {
|
| + rtc::TaskQueue file_writer_queue("file_writer_queue");
|
| +
|
| + const std::string filename =
|
| + webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
|
| +
|
| + {
|
| + webrtc::AecDump aec_dump(filename, -1, &file_writer_queue);
|
| + const webrtc::AudioFrame frame;
|
| + aec_dump.WriteRenderStreamMessage(frame);
|
| + }
|
| +
|
| + // Verify the file has been written after the AecDump d-tor has
|
| + // finished.
|
| + FILE* fid = fopen(filename.c_str(), "r");
|
| + ASSERT_TRUE(fid != NULL);
|
| +
|
| + // Clean it up.
|
| + ASSERT_EQ(0, fclose(fid));
|
| + ASSERT_EQ(0, remove(filename.c_str()));
|
| +}
|
|
|