Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these

webrtc/modules/audio_processing/beamformer/beamformer.h

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 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_BEAMFORMER_BEAMFORMER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_
+
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_processing/beamformer/array_util.h"
+
+namespace webrtc {
+
+template<typename T>
+class Beamformer {
+ public:
+  virtual ~Beamformer() {}
+
+  // Process one time-domain chunk of audio. The audio is expected to be split
+  // into frequency bands inside the ChannelBuffer. The number of frames and
+  // channels must correspond to the constructor parameters. The same
+  // ChannelBuffer can be passed in as |input| and |output|.
+  virtual void ProcessChunk(const ChannelBuffer<T>& input,
+                            ChannelBuffer<T>* output) = 0;
+
+  // Sample rate corresponds to the lower band.
+  // Needs to be called before the the Beamformer can be used.
+  virtual void Initialize(int chunk_size_ms, int sample_rate_hz) = 0;
+
+  // Indicates whether a given point is inside of the beam.
+  virtual bool IsInBeam(const SphericalPointf& spherical_point) { return true; }
+
+  // Returns true if the current data contains the target signal.
+  // Which signals are considered "targets" is implementation dependent.
+  virtual bool is_target_present() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_