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-rw-r--r--third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.cc102
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diff --git a/third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.cc b/third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.cc
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+++ b/third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.cc
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+/*
+ * 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/push_sinc_resampler.h"
+
+#include <cstring>
+
+#include "common_audio/include/audio_util.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+PushSincResampler::PushSincResampler(size_t source_frames,
+ size_t destination_frames)
+ : resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
+ source_frames,
+ this)),
+ source_ptr_(nullptr),
+ source_ptr_int_(nullptr),
+ destination_frames_(destination_frames),
+ first_pass_(true),
+ source_available_(0) {}
+
+PushSincResampler::~PushSincResampler() {}
+
+size_t PushSincResampler::Resample(const int16_t* source,
+ size_t source_length,
+ int16_t* destination,
+ size_t destination_capacity) {
+ if (!float_buffer_.get())
+ float_buffer_.reset(new float[destination_frames_]);
+
+ source_ptr_int_ = source;
+ // Pass nullptr as the float source to have Run() read from the int16 source.
+ Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
+ FloatS16ToS16(float_buffer_.get(), destination_frames_, destination);
+ source_ptr_int_ = nullptr;
+ return destination_frames_;
+}
+
+size_t PushSincResampler::Resample(const float* source,
+ size_t source_length,
+ float* destination,
+ size_t destination_capacity) {
+ RTC_CHECK_EQ(source_length, resampler_->request_frames());
+ RTC_CHECK_GE(destination_capacity, destination_frames_);
+ // Cache the source pointer. Calling Resample() will immediately trigger
+ // the Run() callback whereupon we provide the cached value.
+ source_ptr_ = source;
+ source_available_ = source_length;
+
+ // On the first pass, we call Resample() twice. During the first call, we
+ // provide dummy input and discard the output. This is done to prime the
+ // SincResampler buffer with the correct delay (half the kernel size), thereby
+ // ensuring that all later Resample() calls will only result in one input
+ // request through Run().
+ //
+ // If this wasn't done, SincResampler would call Run() twice on the first
+ // pass, and we'd have to introduce an entire `source_frames` of delay, rather
+ // than the minimum half kernel.
+ //
+ // It works out that ChunkSize() is exactly the amount of output we need to
+ // request in order to prime the buffer with a single Run() request for
+ // `source_frames`.
+ if (first_pass_)
+ resampler_->Resample(resampler_->ChunkSize(), destination);
+
+ resampler_->Resample(destination_frames_, destination);
+ source_ptr_ = nullptr;
+ return destination_frames_;
+}
+
+void PushSincResampler::Run(size_t frames, float* destination) {
+ // Ensure we are only asked for the available samples. This would fail if
+ // Run() was triggered more than once per Resample() call.
+ RTC_CHECK_EQ(source_available_, frames);
+
+ if (first_pass_) {
+ // Provide dummy input on the first pass, the output of which will be
+ // discarded, as described in Resample().
+ std::memset(destination, 0, frames * sizeof(*destination));
+ first_pass_ = false;
+ return;
+ }
+
+ if (source_ptr_) {
+ std::memcpy(destination, source_ptr_, frames * sizeof(*destination));
+ } else {
+ for (size_t i = 0; i < frames; ++i)
+ destination[i] = static_cast<float>(source_ptr_int_[i]);
+ }
+ source_available_ -= frames;
+}
+
+} // namespace webrtc