1 files changed, 123 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/resampler/push_resampler.cc b/third_party/libwebrtc/common_audio/resampler/push_resampler.cc
new file mode 100644
index 0000000000..810d778993
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/resampler/push_resampler.cc
@@ -0,0 +1,123 @@
+/*
+ *  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 "common_audio/resampler/include/push_resampler.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <memory>
+
+#include "common_audio/include/audio_util.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "rtc_base/checks.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,
+                                         size_t num_channels) {
+  // These checks used to be factored out of this template function due to
+  // Windows debug build issues with clang. http://crbug.com/615050
+  RTC_DCHECK_GT(src_sample_rate_hz, 0);
+  RTC_DCHECK_GT(dst_sample_rate_hz, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+
+  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) {
+    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);
+  channel_resamplers_.clear();
+  for (size_t i = 0; i < num_channels; ++i) {
+    channel_resamplers_.push_back(ChannelResampler());
+    auto channel_resampler = channel_resamplers_.rbegin();
+    channel_resampler->resampler = std::make_unique<PushSincResampler>(
+        src_size_10ms_mono, dst_size_10ms_mono);
+    channel_resampler->source.resize(src_size_10ms_mono);
+    channel_resampler->destination.resize(dst_size_10ms_mono);
+  }
+
+  channel_data_array_.resize(num_channels_);
+
+  return 0;
+}
+
+template <typename T>
+int PushResampler<T>::Resample(const T* src,
+                               size_t src_length,
+                               T* dst,
+                               size_t dst_capacity) {
+  // These checks used to be factored out of this template function due to
+  // Windows debug build issues with clang. http://crbug.com/615050
+  const size_t src_size_10ms = (src_sample_rate_hz_ / 100) * num_channels_;
+  const size_t dst_size_10ms = (dst_sample_rate_hz_ / 100) * num_channels_;
+  RTC_DCHECK_EQ(src_length, src_size_10ms);
+  RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
+
+  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);
+  }
+
+  const size_t src_length_mono = src_length / num_channels_;
+  const size_t dst_capacity_mono = dst_capacity / num_channels_;
+
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    channel_data_array_[ch] = channel_resamplers_[ch].source.data();
+  }
+
+  Deinterleave(src, src_length_mono, num_channels_, channel_data_array_.data());
+
+  size_t dst_length_mono = 0;
+
+  for (auto& resampler : channel_resamplers_) {
+    dst_length_mono = resampler.resampler->Resample(
+        resampler.source.data(), src_length_mono, resampler.destination.data(),
+        dst_capacity_mono);
+  }
+
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    channel_data_array_[ch] = channel_resamplers_[ch].destination.data();
+  }
+
+  Interleave(channel_data_array_.data(), dst_length_mono, num_channels_, dst);
+  return static_cast<int>(dst_length_mono * num_channels_);
+}
+
+// Explictly generate required instantiations.
+template class PushResampler<int16_t>;
+template class PushResampler<float>;
+
+}  // namespace webrtc