webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

71 lines
2.2 KiB
C++

/*
* 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 "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/task_queue.h"
#include "test/gtest.h"
#include "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");
{
std::unique_ptr<webrtc::AecDump> aec_dump =
webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
const webrtc::AudioFrame frame;
aec_dump->WriteRenderStreamMessage(frame);
aec_dump->AddCaptureStreamInput(frame);
aec_dump->AddCaptureStreamOutput(frame);
aec_dump->WriteCaptureStreamMessage();
webrtc::InternalAPMConfig apm_config;
aec_dump->WriteConfig(apm_config);
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");
{
std::unique_ptr<webrtc::AecDump> aec_dump =
webrtc::AecDumpFactory::Create(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()));
}