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( |
+ webrtc::test::TempFilename(webrtc::test::OutputPath(), prefix)); |
+ } |
+ |
+ auto aec_dump = |
+ webrtc::AecDumpFactory::Create(file_names[0], -1, &file_writer_queue); |
+ EXPECT_TRUE(aec_dump); |
+ |
+ aec_dump = |
+ webrtc::AecDumpFactory::Create(file_names[1], -1, &file_writer_queue); |
+ |
+ aec_dump = |
+ 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()); |
+ } |
+} |