/*
 * Copyright (C) 2010, Google Inc. All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1.  Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2.  Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
 * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
 * ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE
 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
 * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
 * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
 * DAMAGE.
 */

#include "third_party/blink/renderer/modules/webaudio/realtime_analyser.h"

#include <limits.h>

#include <algorithm>
#include <bit>
#include <complex>

#include "base/compiler_specific.h"
#include "media/base/audio_bus.h"
#include "third_party/blink/renderer/platform/audio/audio_bus.h"
#include "third_party/blink/renderer/platform/audio/audio_utilities.h"
#include "third_party/blink/renderer/platform/audio/vector_math.h"
#include "third_party/blink/renderer/platform/wtf/math_extras.h"

namespace blink {

namespace {

constexpr unsigned kDefaultFFTSize = 2048;
constexpr unsigned kInputBufferSize = RealtimeAnalyser::kMaxFFTSize * 2;

void ApplyWindow(base::span<float> p) {
  DCHECK(IsMainThread());

  // Blackman window
  constexpr double alpha = 0.16;
  constexpr double a0 = 0.5 * (1 - alpha);
  constexpr double a1 = 0.5;
  constexpr double a2 = 0.5 * alpha;

  const size_t n = p.size();
  for (unsigned i = 0; i < n; ++i) {
    const double x = static_cast<double>(i) / static_cast<double>(n);
    const double window =
        a0 - a1 * cos(kTwoPiDouble * x) + a2 * cos(kTwoPiDouble * 2.0 * x);
    p[i] *= static_cast<float>(window);
  }
}

// Returns x if x is finite (not NaN or infinite), otherwise returns
// default_value
float EnsureFinite(float x, float default_value) {
  return std::isfinite(x) ? x : default_value;
}

}  // namespace

RealtimeAnalyser::RealtimeAnalyser() : fft_size_(kDefaultFFTSize) {
  analysis_frame_ = std::make_unique<FFTFrame>(kDefaultFFTSize);
}

bool RealtimeAnalyser::InitializeBuffers(unsigned render_quantum_frames) {
  down_mix_bus_ = AudioBus::TryCreate(1, render_quantum_frames);
  if (!down_mix_bus_) {
    return false;
  }
  return input_buffer_.TryAllocate(kInputBufferSize) &&
         magnitude_buffer_.TryAllocate(kDefaultFFTSize / 2);
}

bool RealtimeAnalyser::SetFftSize(uint32_t size) {
  DCHECK(IsMainThread());

  // Only allow powers of two within the allowed range.
  if (size > kMaxFFTSize || size < kMinFFTSize || !std::has_single_bit(size)) {
    return false;
  }

  if (fft_size_ != size) {
    analysis_frame_ = std::make_unique<FFTFrame>(size);
    // m_magnitudeBuffer has size = fftSize / 2 because it contains floats
    // reduced from complex values in m_analysisFrame.
    magnitude_buffer_.Allocate(size / 2);
    fft_size_ = size;
  }

  return true;
}

void RealtimeAnalyser::GetFloatFrequencyData(DOMFloat32Array* destination_array,
                                             double current_time) {
  DCHECK(IsMainThread());
  DCHECK(destination_array);

  if (current_time > last_analysis_time_) {
    // Time has advanced since the last call; update the FFT data.
    last_analysis_time_ = current_time;
    DoFFTAnalysis();
  }

  // Convert from linear magnitude to floating-point decibels.
  const size_t source_length = magnitude_buffer_.size();
  const size_t len = std::min(source_length, destination_array->length());
  if (len > 0) {
    base::span<float> destination = destination_array->AsSpan();
    for (unsigned i = 0; i < len; ++i) {
      const float linear_value = magnitude_buffer_[i];
      const double db_mag = audio_utilities::LinearToDecibels(linear_value);
      destination[i] = static_cast<float>(db_mag);
    }
  }
}

void RealtimeAnalyser::GetByteFrequencyData(DOMUint8Array* destination_array,
                                            double current_time) {
  DCHECK(IsMainThread());
  DCHECK(destination_array);

  if (current_time > last_analysis_time_) {
    // Time has advanced since the last call; update the FFT data.
    last_analysis_time_ = current_time;
    DoFFTAnalysis();
  }

  // FIXME: Is it worth caching the data so we don't have to do the conversion
  // every time?  Perhaps not, since we expect many calls in the same
  // rendering quantum.

  // Convert from linear magnitude to unsigned-byte decibels.
  const size_t source_length = magnitude_buffer_.size();
  const size_t len = std::min(source_length, destination_array->length());
  if (len > 0) {
    const double range_scale_factor =
        max_decibels_ == min_decibels_ ? 1.0
                                       : 1.0 / (max_decibels_ - min_decibels_);
    const double min_decibels = min_decibels_;

    base::span<unsigned char> destination = destination_array->AsSpan();
    for (unsigned i = 0; i < len; ++i) {
      const float linear_value = magnitude_buffer_[i];
      const double db_mag = audio_utilities::LinearToDecibels(linear_value);

      // The range m_minDecibels to m_maxDecibels will be scaled to byte values
      // from 0 to UCHAR_MAX.
      const double scaled_value =
          UCHAR_MAX * (db_mag - min_decibels) * range_scale_factor;

      // Clip to valid range.
      destination[i] =
          static_cast<unsigned char>(ClampTo(scaled_value, 0, UCHAR_MAX));
    }
  }
}

void RealtimeAnalyser::GetFloatTimeDomainData(
    DOMFloat32Array* destination_array) {
  DCHECK(IsMainThread());
  DCHECK(destination_array);

  const unsigned fft_size = FftSize();
  const size_t len =
      std::min(static_cast<size_t>(fft_size), destination_array->length());
  if (len > 0) {
    DCHECK_EQ(input_buffer_.size(), kInputBufferSize);
    DCHECK_GT(input_buffer_.size(), fft_size);

    const unsigned write_index = GetWriteIndex();

    base::span<float> destination = destination_array->AsSpan();
    for (unsigned i = 0; i < len; ++i) {
      // Buffer access is protected due to modulo operation.
      float value =
          input_buffer_[(i + write_index - fft_size + kInputBufferSize) %
                        kInputBufferSize];

      destination[i] = value;
    }
  }
}

void RealtimeAnalyser::GetByteTimeDomainData(DOMUint8Array* destination_array) {
  DCHECK(IsMainThread());
  DCHECK(destination_array);

  const unsigned fft_size = FftSize();
  const size_t len =
      std::min(static_cast<size_t>(fft_size), destination_array->length());
  if (len > 0) {
    DCHECK_EQ(input_buffer_.size(), kInputBufferSize);
    DCHECK_GT(input_buffer_.size(), fft_size);

    const unsigned write_index = GetWriteIndex();

    base::span<unsigned char> destination = destination_array->AsSpan();
    for (unsigned i = 0; i < len; ++i) {
      // Buffer access is protected due to modulo operation.
      const float value =
          input_buffer_[(i + write_index - fft_size + kInputBufferSize) %
                        kInputBufferSize];

      // Scale from nominal -1 -> +1 to unsigned byte.
      const double scaled_value = 128 * (value + 1);

      // Clip to valid range.
      destination[i] =
          static_cast<unsigned char>(ClampTo(scaled_value, 0, UCHAR_MAX));
    }
  }
}

void RealtimeAnalyser::WriteInput(AudioBus* bus, uint32_t frames_to_process) {
  DCHECK(bus);
  DCHECK_GT(bus->NumberOfChannels(), 0u);
  DCHECK_GE(bus->Channel(0)->length(), frames_to_process);

  const unsigned write_index = GetWriteIndex();
  DCHECK_LT(write_index, input_buffer_.size());

  // Perform real-time analysis

  // Clear the bus and downmix the input according to the down mixing rules.
  // Then save the result in the m_inputBuffer at the appropriate place.
  down_mix_bus_->Zero();
  down_mix_bus_->SumFrom(*bus);

  base::span<const float> src =
      down_mix_bus_->Channel(0)->Span().first(frames_to_process);
  if (src.size() > kInputBufferSize) {
    src = src.last(kInputBufferSize);
  }

  const uint32_t frames_to_end = kInputBufferSize - write_index;
  if (src.size() <= frames_to_end) {
    input_buffer_.as_span().subspan(write_index, src.size()).copy_from(src);
  } else {
    input_buffer_.as_span()
        .subspan(write_index, frames_to_end)
        .copy_from(src.first(frames_to_end));
    input_buffer_.as_span()
        .first(src.size() - frames_to_end)
        .copy_from(src.subspan(frames_to_end));
  }
  SetWriteIndex((write_index + src.size()) % kInputBufferSize);
}

void RealtimeAnalyser::DoFFTAnalysis() {
  DCHECK(IsMainThread());

  // Unroll the input buffer into a temporary buffer, where we'll apply an
  // analysis window followed by an FFT.
  const uint32_t fft_size = FftSize();

  AudioFloatArray temporary_buffer(fft_size);
  base::span<const float> input_buffer = input_buffer_.as_span();
  base::span<float> temp_span = temporary_buffer.as_span();

  // Take the previous fftSize values from the input buffer and copy into the
  // temporary buffer.
  const unsigned write_index = GetWriteIndex();
  if (write_index < fft_size) {
    temp_span.first(fft_size - write_index)
        .copy_from(input_buffer.subspan(
            write_index - fft_size + kInputBufferSize, fft_size - write_index));
    temp_span.subspan(fft_size - write_index)
        .copy_from(input_buffer.first(write_index));
  } else {
    temp_span.copy_from(input_buffer.subspan(write_index - fft_size, fft_size));
  }

  // Window the input samples.
  ApplyWindow(temp_span);

  // Do the analysis.
  analysis_frame_->DoFFT(temp_span);

  const AudioFloatArray& real = analysis_frame_->RealData();
  AudioFloatArray& imag = analysis_frame_->ImagData();

  // Blow away the packed nyquist component.
  imag[0] = 0;

  // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
  // scaling factor).
  const double magnitude_scale = 1.0 / fft_size;

  // A value of 0 does no averaging with the previous result.  Larger values
  // produce slower, but smoother changes.
  const double k = ClampTo(smoothing_time_constant_, 0.0, 1.0);

  // Convert the analysis data from complex to magnitude and average with the
  // previous result.
  base::span<float> destination = magnitude_buffer_.as_span();
  const size_t n = destination.size();
  DCHECK_GE(real.size(), n);
  DCHECK_GE(imag.size(), n);
  for (size_t i = 0; i < n; ++i) {
    std::complex<double> c(real[i], imag[i]);
    double scalar_magnitude = abs(c) * magnitude_scale;
    destination[i] = EnsureFinite(
        static_cast<float>(k * destination[i] + (1 - k) * scalar_magnitude), 0);
  }
}

}  // namespace blink