Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(30)

Unified Diff: webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc

Issue 2778783002: AecDump interface (Closed)
Patch Set: Next version; large changes to interface. Created 3 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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());
+ }
+}

Powered by Google App Engine
This is Rietveld 408576698