Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h

Index: webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
diff --git a/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
new file mode 100644
index 0000000000000000000000000000000000000000..a33ae528e060b603f390cd0c1f13065261c4a065
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
@@ -0,0 +1,48 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/modules/audio_processing/include/aec_dump.h"
+
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
+namespace webrtc {
+
+class AecDumpFactory {
+ public:
+  // TODO(aleloi): update comments to new creation scheme.
+  // If called when a recording is active, that file is closed, and a
+  // new file is opened. Messages waiting to be written asynchronously
+  // to the old file may be lost. Returns true iff opening file for
+  // writing succeeded.
+
+  // Closes associated file. Messages waiting to be written to file
+  // asynchronously may be lost. This method is safe to call when no
+  // recording is active. A recording does not have to be closed
+  // manually with this method; instead the AecDump instance may be
+  // destroyed.
+  static std::unique_ptr<AecDump> Create(std::string file_name,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+  static std::unique_ptr<AecDump> Create(FILE* handle,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_