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_resampler.cc

@@ -0,0 +1,110 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
+
+#include <string.h>
+
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
+
+namespace webrtc {
+
+template <typename T>
+PushResampler<T>::PushResampler()
+    : src_sample_rate_hz_(0),
+      dst_sample_rate_hz_(0),
+      num_channels_(0) {
+}
+
+template <typename T>
+PushResampler<T>::~PushResampler() {
+}
+
+template <typename T>
+int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
+                                         int dst_sample_rate_hz,
+                                         int num_channels) {
+  if (src_sample_rate_hz == src_sample_rate_hz_ &&
+      dst_sample_rate_hz == dst_sample_rate_hz_ &&
+      num_channels == num_channels_)
+    // No-op if settings haven't changed.
+    return 0;
+
+  if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
+      num_channels <= 0 || num_channels > 2)
+    return -1;
+
+  src_sample_rate_hz_ = src_sample_rate_hz;
+  dst_sample_rate_hz_ = dst_sample_rate_hz;
+  num_channels_ = num_channels;
+
+  const size_t src_size_10ms_mono =
+      static_cast<size_t>(src_sample_rate_hz / 100);
+  const size_t dst_size_10ms_mono =
+      static_cast<size_t>(dst_sample_rate_hz / 100);
+  sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
+                                              dst_size_10ms_mono));
+  if (num_channels_ == 2) {
+    src_left_.reset(new T[src_size_10ms_mono]);
+    src_right_.reset(new T[src_size_10ms_mono]);
+    dst_left_.reset(new T[dst_size_10ms_mono]);
+    dst_right_.reset(new T[dst_size_10ms_mono]);
+    sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
+                                                      dst_size_10ms_mono));
+  }
+
+  return 0;
+}
+
+template <typename T>
+int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
+                               size_t dst_capacity) {
+  const size_t src_size_10ms =
+      static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
+  const size_t dst_size_10ms =
+      static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
+  if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
+    return -1;
+
+  if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
+    // The old resampler provides this memcpy facility in the case of matching
+    // sample rates, so reproduce it here for the sinc resampler.
+    memcpy(dst, src, src_length * sizeof(T));
+    return static_cast<int>(src_length);
+  }
+  if (num_channels_ == 2) {
+    const size_t src_length_mono = src_length / num_channels_;
+    const size_t dst_capacity_mono = dst_capacity / num_channels_;
+    T* deinterleaved[] = {src_left_.get(), src_right_.get()};
+    Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
+
+    size_t dst_length_mono =
+        sinc_resampler_->Resample(src_left_.get(), src_length_mono,
+                                  dst_left_.get(), dst_capacity_mono);
+    sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
+                                    dst_right_.get(), dst_capacity_mono);
+
+    deinterleaved[0] = dst_left_.get();
+    deinterleaved[1] = dst_right_.get();
+    Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
+    return static_cast<int>(dst_length_mono * num_channels_);
+  } else {
+    return static_cast<int>(
+        sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
+  }
+}
+
+// Explictly generate required instantiations.
+template class PushResampler<int16_t>;
+template class PushResampler<float>;
+
+}  // namespace webrtc