text: /*
   *  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
  