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/resampler/push_sinc_resampler.h

@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2013 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_RESAMPLER_PUSH_SINC_RESAMPLER_H_
+#define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/resampler/sinc_resampler.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// A thin wrapper over SincResampler to provide a push-based interface as
+// required by WebRTC. SincResampler uses a pull-based interface, and will
+// use SincResamplerCallback::Run() to request data upon a call to Resample().
+// These Run() calls will happen on the same thread Resample() is called on.
+class PushSincResampler : public SincResamplerCallback {
+ public:
+  // Provide the size of the source and destination blocks in samples. These
+  // must correspond to the same time duration (typically 10 ms) as the sample
+  // ratio is inferred from them.
+  PushSincResampler(size_t source_frames, size_t destination_frames);
+  ~PushSincResampler() override;
+
+  // Perform the resampling. |source_frames| must always equal the
+  // |source_frames| provided at construction. |destination_capacity| must be
+  // at least as large as |destination_frames|. Returns the number of samples
+  // provided in destination (for convenience, since this will always be equal
+  // to |destination_frames|).
+  size_t Resample(const int16_t* source, size_t source_frames,
+                  int16_t* destination, size_t destination_capacity);
+  size_t Resample(const float* source,
+                  size_t source_frames,
+                  float* destination,
+                  size_t destination_capacity);
+
+  // Delay due to the filter kernel. Essentially, the time after which an input
+  // sample will appear in the resampled output.
+  static float AlgorithmicDelaySeconds(int source_rate_hz) {
+    return 1.f / source_rate_hz * SincResampler::kKernelSize / 2;
+  }
+
+ protected:
+  // Implements SincResamplerCallback.
+  void Run(size_t frames, float* destination) override;
+
+ private:
+  friend class PushSincResamplerTest;
+  SincResampler* get_resampler_for_testing() { return resampler_.get(); }
+
+  rtc::scoped_ptr<SincResampler> resampler_;
+  rtc::scoped_ptr<float[]> float_buffer_;
+  const float* source_ptr_;
+  const int16_t* source_ptr_int_;
+  const size_t destination_frames_;
+
+  // True on the first call to Resample(), to prime the SincResampler buffer.
+  bool first_pass_;
+
+  // Used to assert we are only requested for as much data as is available.
+  size_t source_available_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_