Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_processing/aec3/block_processor.h

@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2016 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_BLOCK_PROCESSOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/modules/audio_processing/aec3/echo_remover.h"
+#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
+#include "webrtc/modules/audio_processing/aec3/render_delay_controller.h"
+
+namespace webrtc {
+
+// Class for performing echo cancellation on 64 sample blocks of audio data.
+class BlockProcessor {
+ public:
+  static BlockProcessor* Create(
+      const AudioProcessing::Config::EchoCanceller3& config,
+      int sample_rate_hz);
+  // Only used for testing purposes.
+  static BlockProcessor* Create(
+      const AudioProcessing::Config::EchoCanceller3& config,
+      int sample_rate_hz,
+      std::unique_ptr<RenderDelayBuffer> render_buffer);
+  static BlockProcessor* Create(
+      const AudioProcessing::Config::EchoCanceller3& config,
+      int sample_rate_hz,
+      std::unique_ptr<RenderDelayBuffer> render_buffer,
+      std::unique_ptr<RenderDelayController> delay_controller,
+      std::unique_ptr<EchoRemover> echo_remover);
+
+  virtual ~BlockProcessor() = default;
+
+  // Processes a block of capture data.
+  virtual void ProcessCapture(
+      bool echo_path_gain_change,
+      bool capture_signal_saturation,
+      std::vector<std::vector<float>>* capture_block) = 0;
+
+  // Buffers a block of render data supplied by a FrameBlocker object.
+  virtual void BufferRender(
+      const std::vector<std::vector<float>>& render_block) = 0;
+
+  // Reports whether echo leakage has been detected in the echo canceller
+  // output.
+  virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_