cubeb_resampler_internal.h

/*
 * Copyright © 2016 Mozilla Foundation
 *
 * This program is made available under an ISC-style license.  See the
 * accompanying file LICENSE for details.
 */

#if !defined(CUBEB_RESAMPLER_INTERNAL)
#define CUBEB_RESAMPLER_INTERNAL

#include <algorithm>
#include <cassert>
#include <cmath>
#include <memory>
#ifdef CUBEB_GECKO_BUILD
#include "mozilla/UniquePtr.h"
// In libc++, symbols such as std::unique_ptr may be defined in std::__1.
// The _LIBCPP_BEGIN_NAMESPACE_STD and _LIBCPP_END_NAMESPACE_STD macros
// will expand to the correct namespace.
#ifdef _LIBCPP_BEGIN_NAMESPACE_STD
#define MOZ_BEGIN_STD_NAMESPACE _LIBCPP_BEGIN_NAMESPACE_STD
#define MOZ_END_STD_NAMESPACE _LIBCPP_END_NAMESPACE_STD
#else
#define MOZ_BEGIN_STD_NAMESPACE namespace std {
#define MOZ_END_STD_NAMESPACE }
#endif
MOZ_BEGIN_STD_NAMESPACE
using mozilla::DefaultDelete;
using mozilla::UniquePtr;
#define default_delete DefaultDelete
#define unique_ptr UniquePtr
MOZ_END_STD_NAMESPACE
#endif
#include "cubeb-speex-resampler.h"
#include "cubeb/cubeb.h"
#include "cubeb_log.h"
#include "cubeb_resampler.h"
#include "cubeb_utils.h"
#include <stdio.h>

/* This header file contains the internal C++ API of the resamplers, for
 * testing. */

// When dropping audio input frames to prevent building
// an input delay, this function returns the number of frames
// to keep in the buffer.
// @parameter sample_rate The sample rate of the stream.
// @return A number of frames to keep.
uint32_t
min_buffered_audio_frame(uint32_t sample_rate);

int
to_speex_quality(cubeb_resampler_quality q);

struct cubeb_resampler {
  virtual long fill(void * input_buffer, long * input_frames_count,
                    void * output_buffer, long frames_needed) = 0;
  virtual long latency() = 0;
  virtual ~cubeb_resampler() {}
};

/** Base class for processors. This is just used to share methods for now. */
class processor {
public:
  explicit processor(uint32_t channels) : channels(channels) {}

protected:
  size_t frames_to_samples(size_t frames) const { return frames * channels; }
  size_t samples_to_frames(size_t samples) const
  {
    assert(!(samples % channels));
    return samples / channels;
  }
  /** The number of channel of the audio buffers to be resampled. */
  const uint32_t channels;
};

template <typename T>
class passthrough_resampler : public cubeb_resampler, public processor {
public:
  passthrough_resampler(cubeb_stream * s, cubeb_data_callback cb, void * ptr,
                        uint32_t input_channels, uint32_t sample_rate);

  virtual long fill(void * input_buffer, long * input_frames_count,
                    void * output_buffer, long output_frames);

  virtual long latency() { return 0; }

  void drop_audio_if_needed()
  {
    uint32_t to_keep = min_buffered_audio_frame(sample_rate);
    uint32_t available = samples_to_frames(internal_input_buffer.length());
    if (available > to_keep) {
      ALOGV("Dropping %u frames", available - to_keep);
      internal_input_buffer.pop(nullptr,
                                frames_to_samples(available - to_keep));
    }
  }

private:
  cubeb_stream * const stream;
  const cubeb_data_callback data_callback;
  void * const user_ptr;
  /* This allows to buffer some input to account for the fact that we buffer
   * some inputs. */
  auto_array<T> internal_input_buffer;
  uint32_t sample_rate;
};

/** Bidirectional resampler, can resample an input and an output stream, or just
 * an input stream or output stream. In this case a delay is inserted in the
 * opposite direction to keep the streams synchronized. */
template <typename T, typename InputProcessing, typename OutputProcessing>
class cubeb_resampler_speex : public cubeb_resampler {
public:
  cubeb_resampler_speex(InputProcessing * input_processor,
                        OutputProcessing * output_processor, cubeb_stream * s,
                        cubeb_data_callback cb, void * ptr);

  virtual ~cubeb_resampler_speex();

  virtual long fill(void * input_buffer, long * input_frames_count,
                    void * output_buffer, long output_frames_needed);

  virtual long latency()
  {
    if (input_processor && output_processor) {
      assert(input_processor->latency() == output_processor->latency());
      return input_processor->latency();
    } else if (input_processor) {
      return input_processor->latency();
    } else {
      return output_processor->latency();
    }
  }

private:
  typedef long (cubeb_resampler_speex::*processing_callback)(
      T * input_buffer, long * input_frames_count, T * output_buffer,
      long output_frames_needed);

  long fill_internal_duplex(T * input_buffer, long * input_frames_count,
                            T * output_buffer, long output_frames_needed);
  long fill_internal_input(T * input_buffer, long * input_frames_count,
                           T * output_buffer, long output_frames_needed);
  long fill_internal_output(T * input_buffer, long * input_frames_count,
                            T * output_buffer, long output_frames_needed);

  std::unique_ptr<InputProcessing> input_processor;
  std::unique_ptr<OutputProcessing> output_processor;
  processing_callback fill_internal;
  cubeb_stream * const stream;
  const cubeb_data_callback data_callback;
  void * const user_ptr;
  bool draining = false;
};

/** Handles one way of a (possibly) duplex resampler, working on interleaved
 * audio buffers of type T. This class is designed so that the number of frames
 * coming out of the resampler can be precisely controled. It manages its own
 * input buffer, and can use the caller's output buffer, or allocate its own. */
template <typename T> class cubeb_resampler_speex_one_way : public processor {
public:
  /** The sample type of this resampler, either 16-bit integers or 32-bit
   * floats. */
  typedef T sample_type;
  /** Construct a resampler resampling from #source_rate to #target_rate, that
   * can be arbitrary, strictly positive number.
   * @parameter channels The number of channels this resampler will resample.
   * @parameter source_rate The sample-rate of the audio input.
   * @parameter target_rate The sample-rate of the audio output.
   * @parameter quality A number between 0 (fast, low quality) and 10 (slow,
   * high quality). */
  cubeb_resampler_speex_one_way(uint32_t channels, uint32_t source_rate,
                                uint32_t target_rate, int quality)
      : processor(channels),
        resampling_ratio(static_cast<float>(source_rate) / target_rate),
        source_rate(source_rate), additional_latency(0), leftover_samples(0)
  {
    int r;
    speex_resampler =
        speex_resampler_init(channels, source_rate, target_rate, quality, &r);
    assert(r == RESAMPLER_ERR_SUCCESS && "resampler allocation failure");

    uint32_t input_latency = speex_resampler_get_input_latency(speex_resampler);
    const size_t LATENCY_SAMPLES = 8192;
    T input_buffer[LATENCY_SAMPLES] = {};
    T output_buffer[LATENCY_SAMPLES] = {};
    uint32_t input_frame_count = input_latency;
    uint32_t output_frame_count = LATENCY_SAMPLES;
    assert(input_latency * channels <= LATENCY_SAMPLES);
    speex_resample(input_buffer, &input_frame_count, output_buffer,
                   &output_frame_count);
  }

  /** Destructor, deallocate the resampler */
  virtual ~cubeb_resampler_speex_one_way()
  {
    speex_resampler_destroy(speex_resampler);
  }

  /* Fill the resampler with `input_frame_count` frames. */
  void input(T * input_buffer, size_t input_frame_count)
  {
    resampling_in_buffer.push(input_buffer,
                              frames_to_samples(input_frame_count));
  }

  /** Outputs exactly `output_frame_count` into `output_buffer`.
   * `output_buffer` has to be at least `output_frame_count` long. */
  size_t output(T * output_buffer, size_t output_frame_count)
  {
    uint32_t in_len = samples_to_frames(resampling_in_buffer.length());
    uint32_t out_len = output_frame_count;

    speex_resample(resampling_in_buffer.data(), &in_len, output_buffer,
                   &out_len);

    /* This shifts back any unresampled samples to the beginning of the input
       buffer. */
    resampling_in_buffer.pop(nullptr, frames_to_samples(in_len));

    return out_len;
  }

  size_t output_for_input(uint32_t input_frames)
  {
    return (size_t)floorf(
        (input_frames + samples_to_frames(resampling_in_buffer.length())) /
        resampling_ratio);
  }

  /** Returns a buffer containing exactly `output_frame_count` resampled frames.
   * The consumer should not hold onto the pointer. */
  T * output(size_t output_frame_count, size_t * input_frames_used)
  {
    if (resampling_out_buffer.capacity() <
        frames_to_samples(output_frame_count)) {
      resampling_out_buffer.reserve(frames_to_samples(output_frame_count));
    }

    uint32_t in_len = samples_to_frames(resampling_in_buffer.length());
    uint32_t out_len = output_frame_count;

    speex_resample(resampling_in_buffer.data(), &in_len,
                   resampling_out_buffer.data(), &out_len);

    if (out_len < output_frame_count) {
      LOGV("underrun during resampling: got %u frames, expected %zu",
           (unsigned)out_len, output_frame_count);
      // silence the rightmost part
      T * data = resampling_out_buffer.data();
      for (uint32_t i = frames_to_samples(out_len);
           i < frames_to_samples(output_frame_count); i++) {
        data[i] = 0;
      }
    }

    /* This shifts back any unresampled samples to the beginning of the input
       buffer. */
    resampling_in_buffer.pop(nullptr, frames_to_samples(in_len));
    *input_frames_used = in_len;

    return resampling_out_buffer.data();
  }

  /** Get the latency of the resampler, in output frames. */
  uint32_t latency() const
  {
    /* The documentation of the resampler talks about "samples" here, but it
     * only consider a single channel here so it's the same number of frames. */
    int latency = 0;

    latency = speex_resampler_get_output_latency(speex_resampler) +
              additional_latency;

    assert(latency >= 0);

    return latency;
  }

  /** Returns the number of frames to pass in the input of the resampler to have
   * exactly `output_frame_count` resampled frames. This can return a number
   * slightly bigger than what is strictly necessary, but it guaranteed that the
   * number of output frames will be exactly equal. */
  uint32_t input_needed_for_output(int32_t output_frame_count) const
  {
    assert(output_frame_count >= 0); // Check overflow
    int32_t unresampled_frames_left =
        samples_to_frames(resampling_in_buffer.length());
    int32_t resampled_frames_left =
        samples_to_frames(resampling_out_buffer.length());
    float input_frames_needed =
        (output_frame_count - unresampled_frames_left) * resampling_ratio -
        resampled_frames_left;
    if (input_frames_needed < 0) {
      return 0;
    }
    return (uint32_t)ceilf(input_frames_needed);
  }

  /** Returns a pointer to the input buffer, that contains empty space for at
   * least `frame_count` elements. This is useful so that consumer can directly
   * write into the input buffer of the resampler. The pointer returned is
   * adjusted so that leftover data are not overwritten.
   */
  T * input_buffer(size_t frame_count)
  {
    leftover_samples = resampling_in_buffer.length();
    resampling_in_buffer.reserve(leftover_samples +
                                 frames_to_samples(frame_count));
    return resampling_in_buffer.data() + leftover_samples;
  }

  /** This method works with `input_buffer`, and allows to inform the processor
      how much frames have been written in the provided buffer. */
  void written(size_t written_frames)
  {
    resampling_in_buffer.set_length(leftover_samples +
                                    frames_to_samples(written_frames));
  }

  void drop_audio_if_needed()
  {
    // Keep at most 100ms buffered.
    uint32_t available = samples_to_frames(resampling_in_buffer.length());
    uint32_t to_keep = min_buffered_audio_frame(source_rate);
    if (available > to_keep) {
      ALOGV("Dropping %u frames", available - to_keep);
      resampling_in_buffer.pop(nullptr, frames_to_samples(available - to_keep));
    }
  }

private:
  /** Wrapper for the speex resampling functions to have a typed
   * interface. */
  void speex_resample(float * input_buffer, uint32_t * input_frame_count,
                      float * output_buffer, uint32_t * output_frame_count)
  {
#ifndef NDEBUG
    int rv;
    rv =
#endif
        speex_resampler_process_interleaved_float(
            speex_resampler, input_buffer, input_frame_count, output_buffer,
            output_frame_count);
    assert(rv == RESAMPLER_ERR_SUCCESS);
  }

  void speex_resample(short * input_buffer, uint32_t * input_frame_count,
                      short * output_buffer, uint32_t * output_frame_count)
  {
#ifndef NDEBUG
    int rv;
    rv =
#endif
        speex_resampler_process_interleaved_int(
            speex_resampler, input_buffer, input_frame_count, output_buffer,
            output_frame_count);
    assert(rv == RESAMPLER_ERR_SUCCESS);
  }
  /** The state for the speex resampler used internaly. */
  SpeexResamplerState * speex_resampler;
  /** Source rate / target rate. */
  const float resampling_ratio;
  const uint32_t source_rate;
  /** Storage for the input frames, to be resampled. Also contains
   * any unresampled frames after resampling. */
  auto_array<T> resampling_in_buffer;
  /* Storage for the resampled frames, to be passed back to the caller. */
  auto_array<T> resampling_out_buffer;
  /** Additional latency inserted into the pipeline for synchronisation. */
  uint32_t additional_latency;
  /** When `input_buffer` is called, this allows tracking the number of samples
      that were in the buffer. */
  uint32_t leftover_samples;
};

/** This class allows delaying an audio stream by `frames` frames. */
template <typename T> class delay_line : public processor {
public:
  /** Constructor
   * @parameter frames the number of frames of delay.
   * @parameter channels the number of channels of this delay line.
   * @parameter sample_rate sample-rate of the audio going through this delay
   * line */
  delay_line(uint32_t frames, uint32_t channels, uint32_t sample_rate)
      : processor(channels), length(frames), leftover_samples(0),
        sample_rate(sample_rate)
  {
    /* Fill the delay line with some silent frames to add latency. */
    delay_input_buffer.push_silence(frames * channels);
  }
  /** Push some frames into the delay line.
   * @parameter buffer the frames to push.
   * @parameter frame_count the number of frames in #buffer. */
  void input(T * buffer, uint32_t frame_count)
  {
    delay_input_buffer.push(buffer, frames_to_samples(frame_count));
  }
  /** Pop some frames from the internal buffer, into a internal output buffer.
   * @parameter frames_needed the number of frames to be returned.
   * @return a buffer containing the delayed frames. The consumer should not
   * hold onto the pointer. */
  T * output(uint32_t frames_needed, size_t * input_frames_used)
  {
    if (delay_output_buffer.capacity() < frames_to_samples(frames_needed)) {
      delay_output_buffer.reserve(frames_to_samples(frames_needed));
    }

    delay_output_buffer.clear();
    delay_output_buffer.push(delay_input_buffer.data(),
                             frames_to_samples(frames_needed));
    delay_input_buffer.pop(nullptr, frames_to_samples(frames_needed));
    *input_frames_used = frames_needed;

    return delay_output_buffer.data();
  }
  /** Get a pointer to the first writable location in the input buffer>
   * @parameter frames_needed the number of frames the user needs to write into
   * the buffer.
   * @returns a pointer to a location in the input buffer where #frames_needed
   * can be writen. */
  T * input_buffer(uint32_t frames_needed)
  {
    leftover_samples = delay_input_buffer.length();
    delay_input_buffer.reserve(leftover_samples +
                               frames_to_samples(frames_needed));
    return delay_input_buffer.data() + leftover_samples;
  }
  /** This method works with `input_buffer`, and allows to inform the processor
      how much frames have been written in the provided buffer. */
  void written(size_t frames_written)
  {
    delay_input_buffer.set_length(leftover_samples +
                                  frames_to_samples(frames_written));
  }
  /** Drains the delay line, emptying the buffer.
   * @parameter output_buffer the buffer in which the frames are written.
   * @parameter frames_needed the maximum number of frames to write.
   * @return the actual number of frames written. */
  size_t output(T * output_buffer, uint32_t frames_needed)
  {
    uint32_t in_len = samples_to_frames(delay_input_buffer.length());
    uint32_t out_len = frames_needed;

    uint32_t to_pop = std::min(in_len, out_len);

    delay_input_buffer.pop(output_buffer, frames_to_samples(to_pop));

    return to_pop;
  }
  /** Returns the number of frames one needs to input into the delay line to get
   * #frames_needed frames back.
   * @parameter frames_needed the number of frames one want to write into the
   * delay_line
   * @returns the number of frames one will get. */
  uint32_t input_needed_for_output(int32_t frames_needed) const
  {
    assert(frames_needed >= 0); // Check overflow
    return frames_needed;
  }
  /** Returns the number of frames produces for `input_frames` frames in input
   */
  size_t output_for_input(uint32_t input_frames) { return input_frames; }
  /** The number of frames this delay line delays the stream by.
   * @returns The number of frames of delay. */
  size_t latency() { return length; }

  void drop_audio_if_needed()
  {
    size_t available = samples_to_frames(delay_input_buffer.length());
    uint32_t to_keep = min_buffered_audio_frame(sample_rate);
    if (available > to_keep) {
      ALOGV("Dropping %u frames", available - to_keep);
      delay_input_buffer.pop(nullptr, frames_to_samples(available - to_keep));
    }
  }

private:
  /** The length, in frames, of this delay line */
  uint32_t length;
  /** When `input_buffer` is called, this allows tracking the number of samples
      that where in the buffer. */
  uint32_t leftover_samples;
  /** The input buffer, where the delay is applied. */
  auto_array<T> delay_input_buffer;
  /** The output buffer. This is only ever used if using the ::output with a
   * single argument. */
  auto_array<T> delay_output_buffer;
  uint32_t sample_rate;
};

/** This sits behind the C API and is more typed. */
template <typename T>
cubeb_resampler *
cubeb_resampler_create_internal(cubeb_stream * stream,
                                cubeb_stream_params * input_params,
                                cubeb_stream_params * output_params,
                                unsigned int target_rate,
                                cubeb_data_callback callback, void * user_ptr,
                                cubeb_resampler_quality quality,
                                cubeb_resampler_reclock reclock)
{
  std::unique_ptr<cubeb_resampler_speex_one_way<T>> input_resampler = nullptr;
  std::unique_ptr<cubeb_resampler_speex_one_way<T>> output_resampler = nullptr;
  std::unique_ptr<delay_line<T>> input_delay = nullptr;
  std::unique_ptr<delay_line<T>> output_delay = nullptr;

  assert((input_params || output_params) &&
         "need at least one valid parameter pointer.");

  /* All the streams we have have a sample rate that matches the target
     sample rate, use a no-op resampler, that simply forwards the buffers to the
     callback. */
  if (((input_params && input_params->rate == target_rate) &&
       (output_params && output_params->rate == target_rate)) ||
      (input_params && !output_params && (input_params->rate == target_rate)) ||
      (output_params && !input_params &&
       (output_params->rate == target_rate))) {
    LOG("Input and output sample-rate match, target rate of %dHz", target_rate);
    return new passthrough_resampler<T>(
        stream, callback, user_ptr, input_params ? input_params->channels : 0,
        target_rate);
  }

  /* Determine if we need to resampler one or both directions, and create the
     resamplers. */
  if (output_params && (output_params->rate != target_rate)) {
    output_resampler.reset(new cubeb_resampler_speex_one_way<T>(
        output_params->channels, target_rate, output_params->rate,
        to_speex_quality(quality)));
    if (!output_resampler) {
      return NULL;
    }
  }

  if (input_params && (input_params->rate != target_rate)) {
    input_resampler.reset(new cubeb_resampler_speex_one_way<T>(
        input_params->channels, input_params->rate, target_rate,
        to_speex_quality(quality)));
    if (!input_resampler) {
      return NULL;
    }
  }

  /* If we resample only one direction but we have a duplex stream, insert a
   * delay line with a length equal to the resampler latency of the
   * other direction so that the streams are synchronized. */
  if (input_resampler && !output_resampler && input_params && output_params) {
    output_delay.reset(new delay_line<T>(input_resampler->latency(),
                                         output_params->channels,
                                         output_params->rate));
    if (!output_delay) {
      return NULL;
    }
  } else if (output_resampler && !input_resampler && input_params &&
             output_params) {
    input_delay.reset(new delay_line<T>(output_resampler->latency(),
                                        input_params->channels,
                                        output_params->rate));
    if (!input_delay) {
      return NULL;
    }
  }

  if (input_resampler && output_resampler) {
    LOG("Resampling input (%d) and output (%d) to target rate of %dHz",
        input_params->rate, output_params->rate, target_rate);
    return new cubeb_resampler_speex<T, cubeb_resampler_speex_one_way<T>,
                                     cubeb_resampler_speex_one_way<T>>(
        input_resampler.release(), output_resampler.release(), stream, callback,
        user_ptr);
  } else if (input_resampler) {
    LOG("Resampling input (%d) to target and output rate of %dHz",
        input_params->rate, target_rate);
    return new cubeb_resampler_speex<T, cubeb_resampler_speex_one_way<T>,
                                     delay_line<T>>(input_resampler.release(),
                                                    output_delay.release(),
                                                    stream, callback, user_ptr);
  } else {
    LOG("Resampling output (%dHz) to target and input rate of %dHz",
        output_params->rate, target_rate);
    return new cubeb_resampler_speex<T, delay_line<T>,
                                     cubeb_resampler_speex_one_way<T>>(
        input_delay.release(), output_resampler.release(), stream, callback,
        user_ptr);
  }
}

#endif /* CUBEB_RESAMPLER_INTERNAL */