repitch_stretcher.hpp

#pragma once
#include <ossia/detail/config.hpp>
 
#if defined(OSSIA_ENABLE_LIBSAMPLERATE)
#if __has_include(<samplerate.h>)
#include <ossia/dataflow/audio_port.hpp>
#include <ossia/dataflow/audio_stretch_mode.hpp>
#include <ossia/dataflow/graph_node.hpp>
#include <ossia/dataflow/nodes/media.hpp>
#include <ossia/dataflow/token_request.hpp>
 
#include <boost/circular_buffer.hpp>
 
#include <samplerate.h>
 
#include <cinttypes>
#include <cmath>
namespace ossia
{
static constexpr auto get_samplerate_preset(ossia::audio_stretch_mode mode)
{
  int qmode = SRC_SINC_BEST_QUALITY;
  if(mode == audio_stretch_mode::RepitchMediumQ)
    qmode = SRC_SINC_FASTEST;
  if(mode == audio_stretch_mode::RepitchFastestQ)
    qmode = SRC_LINEAR;
  return qmode;
}
 
struct repitch_stretcher
{
  struct resample_channel
  {
    explicit resample_channel(int preset, int buffersize) noexcept
        : resampler{src_new(preset, 1, nullptr)}
        , data(10 * buffersize)
    {
    }
    resample_channel(resample_channel&& other) noexcept
        : resampler{other.resampler}
        , data{std::move(other.data)}
    {
      other.resampler = nullptr;
    }
    resample_channel& operator=(resample_channel&& other) noexcept
    {
      resampler = other.resampler;
      data = std::move(other.data);
      other.resampler = nullptr;
      return *this;
    }
 
    resample_channel(const resample_channel&) = delete;
    resample_channel& operator=(const resample_channel&) = delete;
 
    ~resample_channel()
    {
      if(resampler)
        src_delete(resampler);
    }
 
    std::vector<float> input_buffer;
    SRC_STATE* resampler{};
    boost::circular_buffer<float> data;
  };
 
  repitch_stretcher(int preset, int channels, int bufferSize, int64_t pos)
      : next_sample_to_read{pos}
      , preset{preset}
  {
    repitchers.reserve(channels);
    while(int(repitchers.size()) < channels)
    {
      repitchers.emplace_back(preset, bufferSize);
    }
  }
 
  std::vector<float*> input_channels;
  std::vector<float> output_buffer;
  std::vector<resample_channel> repitchers;
  int64_t next_sample_to_read{};
  int preset{};
 
  // libsamplerate auto-pads its FIR ring; see src_sinc.c:1175.
  [[nodiscard]] static constexpr int64_t start_delay() noexcept { return 0; }
 
  void transport(int64_t date) { next_sample_to_read = date; }
 
  template <typename T>
  void
  run(T& audio_fetcher, const ossia::token_request& t, ossia::exec_state_facade e,
      double tempo_ratio, const std::size_t chan, const int64_t len,
      int64_t samples_to_read, const int64_t samples_to_write,
      const int64_t samples_offset, const ossia::mutable_audio_span<double>& ap) noexcept
  {
    assert(chan > 0);
 
    input_channels.resize(chan);
    for(std::size_t i = 0; i < chan; i++)
    {
      repitchers[i].input_buffer.resize(std::max((int64_t)16, samples_to_read));
      input_channels[i] = repitchers[i].input_buffer.data();
    }
    output_buffer.resize(samples_to_write);
    auto output = output_buffer.data();
 
    int64_t num_samples_available = repitchers[0].data.size();
 
    if(t.forward())
    {
      while(num_samples_available < samples_to_write)
      {
        audio_fetcher.fetch_audio(
            next_sample_to_read, samples_to_read, input_channels.data());
 
        SRC_DATA data;
        for(std::size_t i = 0; i < chan; ++i)
        {
          data.data_in = repitchers[i].input_buffer.data();
          data.data_out = output;
          data.input_frames = samples_to_read;
          data.output_frames = samples_to_write - num_samples_available;
          data.input_frames_used = 0;
          data.output_frames_gen = 0;
          data.src_ratio = std::min(70., tempo_ratio);
          data.end_of_input = 0;
 
          // Resample
          src_process(repitchers[i].resampler, &data);
 
          for(int j = 0; j < data.output_frames_gen; j++)
            repitchers[i].data.push_back(output[j]);
        }
        next_sample_to_read += data.input_frames_used;
        samples_to_read = 16;
        num_samples_available = repitchers[0].data.size();
      }
 
      for(std::size_t i = 0; i < chan; ++i)
      {
        auto it = repitchers[i].data.begin();
        for(int j = 0; j < samples_to_write; j++)
        {
          ap[i][j + samples_offset] = double(*it);
          ++it;
        }
 
        repitchers[i].data.erase_begin(samples_to_write);
      }
    }
    else
    {
      // Backward playback
      while(num_samples_available < samples_to_write)
      {
        audio_fetcher.fetch_audio_backward(
            next_sample_to_read, samples_to_read, input_channels.data());
 
        SRC_DATA data;
        for(std::size_t i = 0; i < chan; ++i)
        {
          data.data_in = repitchers[i].input_buffer.data();
          data.data_out = output;
          data.input_frames = samples_to_read;
          data.output_frames = samples_to_write - num_samples_available;
          data.input_frames_used = 0;
          data.output_frames_gen = 0;
          data.src_ratio = std::min(70., std::abs(tempo_ratio));
          data.end_of_input = 0;
 
          src_process(repitchers[i].resampler, &data);
 
          for(int j = 0; j < data.output_frames_gen; j++)
            repitchers[i].data.push_back(output[j]);
        }
        next_sample_to_read -= data.input_frames_used;
        samples_to_read = 16;
        num_samples_available = repitchers[0].data.size();
      }
 
      for(std::size_t i = 0; i < chan; ++i)
      {
        auto it = repitchers[i].data.begin();
        for(int j = 0; j < samples_to_write; j++)
        {
          ap[i][j + samples_offset] = double(*it);
          ++it;
        }
 
        repitchers[i].data.erase_begin(samples_to_write);
      }
    }
  }
};
}
 
#else
 
#include <ossia/dataflow/nodes/timestretch/raw_stretcher.hpp>
 
namespace ossia
{
static constexpr int get_samplerate_preset(ossia::audio_stretch_mode mode)
{
  return 0;
}
using repitch_stretcher = raw_stretcher;
}
#endif
 
#else
 
#include <ossia/dataflow/nodes/timestretch/raw_stretcher.hpp>
 
namespace ossia
{
static constexpr int get_samplerate_preset(ossia::audio_stretch_mode mode)
{
  return 0;
}
using repitch_stretcher = raw_stretcher;
}
#endif