/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* This Source Code Form is subject to the terms of the Mozilla Public
 * License, v. 2.0. If a copy of the MPL was not distributed with this
 * file, You can obtain one at http://mozilla.org/MPL/2.0/. */

#ifndef WebAudioUtils_h_
#define WebAudioUtils_h_

#include <cmath>
#include <limits>
#include "mozilla/TypeTraits.h"
#include "mozilla/FloatingPoint.h"
#include "MediaSegment.h"

// Forward declaration
typedef struct SpeexResamplerState_ SpeexResamplerState;

namespace mozilla {

class AudioNodeStream;

namespace dom {

class AudioParamTimeline;

namespace WebAudioUtils {
  // 32 is the minimum required by the spec for createBuffer() and
  // createScriptProcessor() and matches what is used by Blink.  The limit
  // protects against large memory allocations.
  const size_t MaxChannelCount = 32;
  // AudioContext::CreateBuffer() "must support sample-rates in at least the
  // range 22050 to 96000."
  const uint32_t MinSampleRate = 8000;
  const uint32_t MaxSampleRate = 192000;

  inline bool FuzzyEqual(float v1, float v2)
  {
    using namespace std;
    return fabsf(v1 - v2) < 1e-7f;
  }
  inline bool FuzzyEqual(double v1, double v2)
  {
    using namespace std;
    return fabs(v1 - v2) < 1e-7;
  }

  /**
   * Computes an exponential smoothing rate for a time based variable
   * over aDuration seconds.
   */
  inline double ComputeSmoothingRate(double aDuration, double aSampleRate)
  {
    return 1.0 - std::exp(-1.0 / (aDuration * aSampleRate));
  }

  /**
   * Converts AudioParamTimeline floating point time values to tick values
   * with respect to a source and a destination AudioNodeStream.
   *
   * This needs to be called for each AudioParamTimeline that gets sent to an
   * AudioNodeEngine on the engine side where the AudioParamTimeline is
   * received.  This means that such engines need to be aware of their source
   * and destination streams as well.
   */
  void ConvertAudioParamToTicks(AudioParamTimeline& aParam,
                                AudioNodeStream* aSource,
                                AudioNodeStream* aDest);

  /**
   * Converts a linear value to decibels.  Returns aMinDecibels if the linear
   * value is 0.
   */
  inline float ConvertLinearToDecibels(float aLinearValue, float aMinDecibels)
  {
    return aLinearValue ? 20.0f * std::log10(aLinearValue) : aMinDecibels;
  }

  /**
   * Converts a decibel value to a linear value.
   */
  inline float ConvertDecibelsToLinear(float aDecibels)
  {
    return std::pow(10.0f, 0.05f * aDecibels);
  }

  /**
   * Converts a decibel to a linear value.
   */
  inline float ConvertDecibelToLinear(float aDecibel)
  {
    return std::pow(10.0f, 0.05f * aDecibel);
  }

  inline void FixNaN(double& aDouble)
  {
    if (IsNaN(aDouble) || IsInfinite(aDouble)) {
      aDouble = 0.0;
    }
  }

  inline double DiscreteTimeConstantForSampleRate(double timeConstant, double sampleRate)
  {
    return 1.0 - std::exp(-1.0 / (sampleRate * timeConstant));
  }

  inline bool IsTimeValid(double aTime)
  {
    return aTime >= 0 && aTime <= (MEDIA_TIME_MAX >> TRACK_RATE_MAX_BITS);
  }

  /**
   * Converts a floating point value to an integral type in a safe and
   * platform agnostic way.  The following program demonstrates the kinds
   * of ways things can go wrong depending on the CPU architecture you're
   * compiling for:
   *
   * #include <stdio.h>
   * volatile float r;
   * int main()
   * {
   *   unsigned int q;
   *   r = 1e100;
   *   q = r;
   *   printf("%f %d\n", r, q);
   *   r = -1e100;
   *   q = r;
   *   printf("%f %d\n", r, q);
   *   r = 1e15;
   *   q = r;
   *   printf("%f %x\n", r, q);
   *   r = 0/0.;
   *   q = r;
   *   printf("%f %d\n", r, q);
   * }
   *
   * This program, when compiled for unsigned int, generates the following
   * results depending on the architecture:
   *
   * x86 and x86-64
   * ---
   *  inf 0
   *  -inf 0
   *  999999995904.000000 -727384064 d4a50000
   *  nan 0
   *
   * ARM
   * ---
   *  inf -1
   *  -inf 0
   *  999999995904.000000 -1
   *  nan 0
   *
   * When compiled for int, this program generates the following results:
   *
   * x86 and x86-64
   * ---
   *  inf -2147483648
   *  -inf -2147483648
   *  999999995904.000000 -2147483648
   *  nan -2147483648
   *
   * ARM
   * ---
   *  inf 2147483647
   *  -inf -2147483648
   *  999999995904.000000 2147483647
   *  nan 0
   *
   * Note that the caller is responsible to make sure that the value
   * passed to this function is not a NaN.  This function will abort if
   * it sees a NaN.
   */
  template <typename IntType, typename FloatType>
  IntType TruncateFloatToInt(FloatType f)
  {
    using namespace std;

    static_assert(mozilla::IsIntegral<IntType>::value == true,
                  "IntType must be an integral type");
    static_assert(mozilla::IsFloatingPoint<FloatType>::value == true,
                  "FloatType must be a floating point type");

    if (mozilla::IsNaN(f)) {
      // It is the responsibility of the caller to deal with NaN values.
      // If we ever get to this point, we have a serious bug to fix.
      NS_RUNTIMEABORT("We should never see a NaN here");
    }

    if (f > FloatType(numeric_limits<IntType>::max())) {
      // If the floating point value is outside of the range of maximum
      // integral value for this type, just clamp to the maximum value.
      return numeric_limits<IntType>::max();
    }

    if (f < FloatType(numeric_limits<IntType>::min())) {
      // If the floating point value is outside of the range of minimum
      // integral value for this type, just clamp to the minimum value.
      return numeric_limits<IntType>::min();
    }

    // Otherwise, this conversion must be well defined.
    return IntType(f);
  }

  void Shutdown();

  int
  SpeexResamplerProcess(SpeexResamplerState* aResampler,
                        uint32_t aChannel,
                        const float* aIn, uint32_t* aInLen,
                        float* aOut, uint32_t* aOutLen);

  int
  SpeexResamplerProcess(SpeexResamplerState* aResampler,
                        uint32_t aChannel,
                        const int16_t* aIn, uint32_t* aInLen,
                        float* aOut, uint32_t* aOutLen);

  int
  SpeexResamplerProcess(SpeexResamplerState* aResampler,
                        uint32_t aChannel,
                        const int16_t* aIn, uint32_t* aInLen,
                        int16_t* aOut, uint32_t* aOutLen);
  }

}
}

#endif

