Adding second layer of the echo canceller 3 functionality.

webrtc/modules/audio_processing/aec3/render_delay_controller.h

Index: webrtc/modules/audio_processing/aec3/render_delay_controller.h
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
new file mode 100644
index 0000000000000000000000000000000000000000..d3f04ec8c2addaad67ecd0801408f452dd0dc5de
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
@@ -0,0 +1,41 @@
+/*
+ *  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_AEC3_RENDER_DELAY_CONTROLLER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// Class for aligning the render and capture signal using an RenderDelayBuffer.

[aleloi, 2017/01/20 14:45:37]

an -> a

[peah-webrtc, 2017/01/23 14:16:40]

Done.

+class RenderDelayController {
+ public:
+  static RenderDelayController* Create(
+      int sample_rate_hz,
+      const RenderDelayBuffer& render_delay_buffer);
+  virtual ~RenderDelayController() = default;
+
+  // Aligns the render buffer content with the capture signal.
+  virtual size_t GetDelay(rtc::ArrayView<const float> capture) = 0;
+
+  // Analyses the render signal and returns false if there is a buffer overrun.

[aleloi, 2017/01/20 14:45:37]

Nit: analyse/analyze inconsistent between method and comment.

[peah-webrtc, 2017/01/23 14:16:40]

Impressive catch!

Done.

+  virtual bool AnalyzeRender(rtc::ArrayView<const float> render) = 0;
+
+  // Returns an approximate value for the headroom in the buffer alignment
+  // expressed in samples if such a value is available.
+  virtual rtc::Optional<size_t> AlignmentHeadroom() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_