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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
70 lines
2.2 KiB
C++
70 lines
2.2 KiB
C++
/*
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* Copyright (c) 2017 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 <utility>
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#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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TEST(AecDumper, APICallsDoNotCrash) {
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// Note order of initialization: Task queue has to be initialized
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// before AecDump.
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webrtc::TaskQueueForTest file_writer_queue("file_writer_queue");
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const std::string filename =
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webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
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{
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std::unique_ptr<webrtc::AecDump> aec_dump =
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webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
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const webrtc::AudioFrame frame;
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aec_dump->WriteRenderStreamMessage(frame);
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aec_dump->AddCaptureStreamInput(frame);
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aec_dump->AddCaptureStreamOutput(frame);
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aec_dump->WriteCaptureStreamMessage();
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webrtc::InternalAPMConfig apm_config;
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aec_dump->WriteConfig(apm_config);
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webrtc::ProcessingConfig api_format;
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constexpr int64_t kTimeNowMs = 123456789ll;
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aec_dump->WriteInitMessage(api_format, kTimeNowMs);
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}
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// Remove file after the AecDump d-tor has finished.
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ASSERT_EQ(0, remove(filename.c_str()));
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}
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TEST(AecDumper, WriteToFile) {
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webrtc::TaskQueueForTest file_writer_queue("file_writer_queue");
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const std::string filename =
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webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
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{
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std::unique_ptr<webrtc::AecDump> aec_dump =
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webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
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const webrtc::AudioFrame frame;
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aec_dump->WriteRenderStreamMessage(frame);
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}
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// Verify the file has been written after the AecDump d-tor has
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// finished.
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FILE* fid = fopen(filename.c_str(), "r");
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ASSERT_TRUE(fid != NULL);
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// Clean it up.
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ASSERT_EQ(0, fclose(fid));
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ASSERT_EQ(0, remove(filename.c_str()));
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}
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