AecDump implementation.

webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc

+/*
+ *  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_factory.h"
+
+#include "webrtc/base/task_queue.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/test/gtest.h"
+
+TEST(AecDumper, APICallsDoNotCrash) {
+  // Note order of initialization: factory first (has event pool),

[peah-webrtc, 2017/05/17 05:36:33]

I don't really follow the comments about the event pools?

[aleloi, 2017/05/17 10:49:56]

That was some kind of merge issue. The version of aec_dump_unittest.cc in the
first patch set is updated, while this one is not. There hasn't been an event
pool for a long while...

+  // then task queue (looks at event pool), then aec_dumper (posts
+  // stuff to task queue).
+  rtc::TaskQueue file_writer_queue("file_writer_queue");
+  auto aec_dump =
+      webrtc::AecDumpFactory::Create("file1", -1, &file_writer_queue);
+  EXPECT_TRUE(aec_dump);
+
+  // TODO(aleloi): wait a while and check that file1 is

[peah-webrtc, 2017/05/17 05:36:33]

Something to address before landing this?

[aleloi, 2017/05/17 10:49:56]

Same as above, this file somehow reverted to an earlier version.

+  // opened. Perhaps add a Flush/join method? Or call d-tor?
+  aec_dump = webrtc::AecDumpFactory::Create("file2", -1, &file_writer_queue);
+
+  aec_dump = webrtc::AecDumpFactory::Create("file3", 10, &file_writer_queue);
+
+  // TODO(aleloi): open from handle.

[peah-webrtc, 2017/05/17 05:36:33]

Something to address before landing this?

[aleloi, 2017/05/17 10:49:56]

Acknowledged.

+
+  const webrtc::AudioFrame frame;
+  aec_dump->WriteRenderStreamMessage(frame);
+
+  aec_dump->AddCaptureStreamInput(frame);
+  aec_dump->AddCaptureStreamOutput(frame);
+
+  aec_dump->WriteCaptureStreamMessage();
+
+  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);
+}