AecDump interface

webrtc/modules/audio_processing/aec_dumper/aec_dumper_unittest.cc

Index: webrtc/modules/audio_processing/aec_dumper/aec_dumper_unittest.cc
diff --git a/webrtc/modules/audio_processing/aec_dumper/aec_dumper_unittest.cc b/webrtc/modules/audio_processing/aec_dumper/aec_dumper_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..992c60dab8c57b65e7694e033260b15a6b305c7c
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec_dumper/aec_dumper_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ *  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_dumper/aec_dumper.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: factory first (has event pool),
+  // then task queue (looks at event pool), then aec_dumper (posts
+  // stuff to task queue).
+  rtc::TaskQueue file_writer_queue("file_writer_queue");
+
+  std::vector<std::string> file_names;
+  for (const std::string prefix : {"aecdumper0", "aecdumper1", "aecdumper2"}) {
+    file_names.push_back(
+        webrtc::test::TempFilename(webrtc::test::OutputPath(), prefix));
+  }
+
+  auto aec_dumper =
+      webrtc::AecDumper::Create(file_names[0], -1, &file_writer_queue);
+  EXPECT_TRUE(aec_dumper);
+
+  aec_dumper = webrtc::AecDumper::Create(file_names[1], -1, &file_writer_queue);
+
+  aec_dumper = webrtc::AecDumper::Create(file_names[2], 10, &file_writer_queue);
+
+  FILE* fid = nullptr;
+  aec_dumper = webrtc::AecDumper::Create(fid, 10, &file_writer_queue);
+
+  const webrtc::AudioFrame frame;
+  aec_dumper->WriteReverseStreamMessage(frame);
+
+  auto capture_stream_info = aec_dumper->GetCaptureStreamInfo();
+
+  capture_stream_info->AddInput(frame);

[peah-webrtc, 2017/03/31 07:24:42]

It would be good to have calls for 
AddInput/AddOutput(std::vector<rtc::ArrayView<const float>> src) in the test as
well.

[aleloi, 2017/04/06 15:46:11]

Done.

+  capture_stream_info->AddOutput(frame);
+
+  aec_dumper->WriteCaptureStreamMessage(std::move(capture_stream_info));
+
+  webrtc::InternalAPMConfig apm_config;
+  aec_dumper->WriteConfig(apm_config, false);
+
+  aec_dumper->WriteConfig(apm_config, true);
+
+  webrtc::ProcessingConfig processing_config;
+  aec_dumper->WriteInitMessage(processing_config);
+
+  for (const auto& filename : file_names) {
+    remove(filename.c_str());
+  }
+}