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/common_audio/audio_converter.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2014 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_COMMON_AUDIO_AUDIO_CONVERTER_H_
+#define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+// Format conversion (remixing and resampling) for audio. Only simple remixing
+// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
+// upmix from mono (i.e. |src_channels == 1|).
+//
+// The source and destination chunks have the same duration in time; specifying
+// the number of frames is equivalent to specifying the sample rates.
+class AudioConverter {
+ public:
+  // Returns a new AudioConverter, which will use the supplied format for its
+  // lifetime. Caller is responsible for the memory.
+  static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
+                                                size_t src_frames,
+                                                int dst_channels,
+                                                size_t dst_frames);
+  virtual ~AudioConverter() {};
+
+  // Convert |src|, containing |src_size| samples, to |dst|, having a sample
+  // capacity of |dst_capacity|. Both point to a series of buffers containing
+  // the samples for each channel. The sizes must correspond to the format
+  // passed to Create().
+  virtual void Convert(const float* const* src, size_t src_size,
+                       float* const* dst, size_t dst_capacity) = 0;
+
+  int src_channels() const { return src_channels_; }
+  size_t src_frames() const { return src_frames_; }
+  int dst_channels() const { return dst_channels_; }
+  size_t dst_frames() const { return dst_frames_; }
+
+ protected:
+  AudioConverter();
+  AudioConverter(int src_channels, size_t src_frames, int dst_channels,
+                 size_t dst_frames);
+
+  // Helper to RTC_CHECK that inputs are correctly sized.
+  void CheckSizes(size_t src_size, size_t dst_capacity) const;
+
+ private:
+  const int src_channels_;
+  const size_t src_frames_;
+  const int dst_channels_;
+  const size_t dst_frames_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_