// Copyright 2016 The Fuchsia Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#ifndef GARNET_BIN_MEDIA_AUDIO_CORE_MIXER_MIXER_H_
#define GARNET_BIN_MEDIA_AUDIO_CORE_MIXER_MIXER_H_

#include <fuchsia/media/cpp/fidl.h>
#include <memory>

#include "garnet/bin/media/audio_core/mixer/constants.h"
#include "garnet/bin/media/audio_core/mixer/gain.h"
#include "lib/media/timeline/timeline_function.h"

namespace media {
namespace audio {

class Mixer;
using MixerPtr = std::unique_ptr<Mixer>;
struct Bookkeeping;

class Mixer {
 public:
  static constexpr uint32_t FRAC_ONE = 1u << kPtsFractionalBits;
  static constexpr uint32_t FRAC_MASK = FRAC_ONE - 1u;
  virtual ~Mixer();

  //
  // Resampler enum
  //
  // This enum lists Fuchsia's available resamplers. Callers of Mixer::Select
  // optionally use this enum to specify a resampler type. Default allows an
  // algorithm to select a resampler based on the ratio of incoming and outgoing
  // rates, using Linear for all except "Integer-to-One" resampling ratios.
  enum class Resampler {
    Default = 0,
    SampleAndHold,
    LinearInterpolation,
  };

  //
  // Select
  //
  // Select an appropriate mixer instance, based on an optionally-specified
  // resampler type, or else by the properties of source/destination formats.
  //
  // When calling Mixer::Select, resampler_type is optional. If caller specifies
  // a particular resampler, Mixer::Select will either instantiate exactly what
  // was requested, or return nullptr -- even if otherwise it could successfully
  // instantiate a different one. Setting this param to non-Default says "I know
  // exactly what I need: I want you to fail rather than give me anything else."
  //
  // If resampler_type is absent or indicates Default, the resampler type is
  // determined by algorithm (as has been the case before this CL).
  // For optimum system performance across changing conditions, callers should
  // take care when directly specifying a resampler type, if they do so at all.
  // The default should be allowed whenever possible.
  static MixerPtr Select(const fuchsia::media::AudioStreamType& src_format,
                         const fuchsia::media::AudioStreamType& dest_format,
                         Resampler resampler_type = Resampler::Default);

  //
  // Mix
  //
  // Perform a mixing operation from source buffer into destination buffer.
  //
  // @param dest
  // The pointer to the destination buffer, into which frames will be mixed.
  //
  // @param dest_frames
  // The total number of frames of audio which comprise the destination buffer.
  //
  // @param dest_offset
  // The pointer to the offset (in output frames) from start of dest buffer, at
  // which we should mix destination frames. Essentially this tells Mix how many
  // 'dest' frames to skip over, when determining where to place the first mixed
  // output frame. When Mix has finished, dest_offset is updated to indicate the
  // destination buffer offset of the next frame to be mixed.
  //
  // @param src
  // The pointer to the source buffer, containing input frames to be mixed into
  // the destination buffer.
  //
  // @param frac_src_frames
  // The total number (in 19.13 fixed) of input frames within the source buffer.
  //
  // @param frac_src_offset
  // A pointer to the offset (in fractional input frames) from start of src
  // buffer, at which the first input frame should be sampled. When Mix has
  // finished, frac_src_offset will be updated to indicate the offset of the
  // sampling position of the next frame to be sampled.
  //
  // @param accumulate
  // When true, Mix will accumulate into the destination buffer (sum the mix
  // results with existing values in the dest buffer). When false, Mix will
  // overwrite any existing destination buffer values with its mix output.
  //
  // @param info
  // The Bookkeeping struct (documented above) contains initial and ongoing
  // details about how this source relates to this destination, specifically in
  // the areas of timing/clocking and gain scaling.
  //
  // @return True if the mixer is finished with this source data and will not
  // need it in the future. False if the mixer has not consumed the entire
  // source buffer and will need more of it in the future.
  //
  // TODO(mpuryear): Change parameter frac_src_frames to src_frames (change
  // subframes to int frames), as this was never intended to be fractional.
  virtual bool Mix(float* dest, uint32_t dest_frames, uint32_t* dest_offset,
                   const void* src, uint32_t frac_src_frames,
                   int32_t* frac_src_offset, bool accumulate,
                   Bookkeeping* info) = 0;
  //
  // Reset
  //
  // Reset the internal state of the mixer. Will be called every time there is
  // a discontinuity in the source stream. Mixer implementations should reset
  // anything related to their internal filter state. If/when we include the
  // Bookkeeping struct into Mixer, this would include clearing src_pos_modulo.
  virtual void Reset() {}

  //
  // Filter widths
  //
  // The positive and negative widths of the filter for this mixer, expressed in
  // fractional input AudioRenderer units. These widths convey which input
  // frames will be referenced by the filter, when producing output for a
  // specific instant in time. Positive filter width refers to how far forward
  // (positively) the filter looks, from the PTS in question; negative filter
  // width refers to how far backward (negatively) the filter looks, from that
  // same PTS. Specifically...
  //
  // Let:
  // P = pos_filter_width()
  // N = neg_filter_width()
  // S = An arbitrary point in time at which the input stream will be sampled.
  // X = The PTS of an input frame.
  //
  // If (X >= (S - N)) && (X <= (S + P))
  // Then input frame X is within the filter and contributes to mix operation.
  //
  // Conversely, input frame X contributes to the output samples S where
  //  (S >= X - P)  and  (S <= X + N)
  //
  inline uint32_t pos_filter_width() const { return pos_filter_width_; }
  inline uint32_t neg_filter_width() const { return neg_filter_width_; }

 protected:
  Mixer(uint32_t pos_filter_width, uint32_t neg_filter_width);

 private:
  uint32_t pos_filter_width_;
  uint32_t neg_filter_width_;
};

// TODO(mpuryear): integrate Bookkeeping into the Mixer class: MTWN-129.
// TODO(mpuryear): Rationalize naming/usage of Bookkeeping and MixJob structs.
//
// Bookkeeping
// Maintained by an AudioLink object that connects a source stream to a
// destination mix stream, this struct represents the state of that mix
// operation from the source point-of-view. In a Mix, the relationship between
// sources and destinations is many-to-one, so this struct largely includes
// details about its source stream, and how it relates to the destination.
//
// When calling Mix(), we communicate resampling details with three parameters
// found in the Bookkeeping. To augment step_size, rate_modulo and denominator
// arguments capture any remaining aspects that are not expressed by the 19.13
// fixed-point step_size. Because frac_src_offset and step_size both use
// the 19.13 format, they exhibit the same precision limitations. These rate
// and position limitations are reiterated upon the start of each mix job.
//
// Just as we address *rate* with rate_modulo and denominator, likewise for
// *position* Bookkeeping uses src_pos_modulo to track initial and ongoing
// modulo of src subframes. This work is only partially complete; the
// remaining work (e.g., setting src_pos_modulo's initial value to anything
// other than 0) is tracked with MTWN-128.
//
// With *rate*, the effect of inaccuracy accumulates over time, causing
// measurable distortion that cripples larger mix jobs. For *position*, a
// change in mix job size affects distortion frequency but not distortion
// amplitude. Having added this to Bookkeeping, any residual effect seems to
// be below audible thresholds; for now we are deferring the remaining work.

// Bookkeeping struct members:
//
// mixer
// This is a pointer to the Mixer object that resamples the input. Currently the
// resampler types include SampleAndHold and LinearInterpolation.
//
// gain
// This object maintains gain values contained in the mix path. This includes
// source gain and a snapshot of destination gain (Gain objects correspond with
// source streams, so the definitive value for destination gain is naturally
// owned elsewhere). In the future, this object may include explicit Mute
// states for source and dest stages, a separately controlled Category gain
// stage, and/or the ability to ramp one or more of these gains over time.
// Gain accepts level in dB, and provides gainscale as float multiplier.
//
// step_size
// This 19.13 fixed-point value represents how much to increment our sampling
// position in the input (src) stream, for each output (dest) frame produced.
//
// rate_modulo
// If step_size cannot perfectly express the mix's resampling ratio, this
// parameter (along with subsequent denominator) expresses leftover precision.
// When non-zero, rate_modulo and denominator express a fractional value of
// step_size unit that src position should advance, for each dest frame.
//
// denominator
// If step_size cannot perfectly express the mix's resampling ratio, this
// parameter (along with precedent rate_modulo) expresses leftover precision.
// When non-zero, rate_modulo and denominator express a fractional value of
// step_size unit that src position should advance, for each dest frame.
//
// SnapshotDenominatorFromDestTrans
// This inline function returns a snapshot of the denominator as determined by
// the 'dest_frames_to_frac_source_frames' timeline transform.
//
// src_pos_modulo
// If src_offset cannot perfectly express the source's position, this
// parameter (along with denominator) expresses any leftover precision. When
// present, src_pos_modulo and denominator express a fractional value of
// src_offset unit that should be used when advancing src position.
//
// dest_frames_to_frac_source_frames
// This translates a destination frame value into a source subframe value.
//
// dest_trans_gen_id
// dest_frames_to_frac_source_frames may change over time; this value represents
// the current generation (which version), so any change can be detected.
//
// clock_mono_to_frac_source_frames
// This translates a CLOCK_MONOTONIC value into a source subframe value.
//
// source_trans_gen_id
// clock_mono_to_frac_source_frames may change over time; this value represents
// the current generation (which version), so any change can be detected.
//
struct Bookkeeping {
  Bookkeeping() = default;
  ~Bookkeeping() = default;

  MixerPtr mixer;
  Gain gain;

  uint32_t step_size = Mixer::FRAC_ONE;
  uint32_t rate_modulo = 0;
  uint32_t denominator = 0;
  uint32_t SnapshotDenominatorFromDestTrans() const {
    return dest_frames_to_frac_source_frames.rate().reference_delta();
  }
  uint32_t src_pos_modulo = 0;

  // The output values of these functions are in 19.13 input subframes.
  // TODO(mpuryear): should TLFunctions also produce a modulo?
  TimelineFunction dest_frames_to_frac_source_frames;
  uint32_t dest_trans_gen_id = kInvalidGenerationId;

  TimelineFunction clock_mono_to_frac_source_frames;
  uint32_t source_trans_gen_id = kInvalidGenerationId;

  void Reset() {
    mixer->Reset();
    src_pos_modulo = 0;
    gain.ClearSourceRamp();
  }

  // TODO(mpuryear): move this into the Mixer or Gain class, along with the
  // other Bookkeeping parameters.
  static constexpr uint32_t kScaleArrLen = 960;
  std::unique_ptr<Gain::AScale[]> scale_arr =
      std::make_unique<Gain::AScale[]>(kScaleArrLen);
};

}  // namespace audio
}  // namespace media

#endif  // GARNET_BIN_MEDIA_AUDIO_CORE_MIXER_MIXER_H_