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())); |
+} |