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-/*----------------------------------------------------------------------------
- *
- * File:
- * eas_math.h
- *
- * Contents and purpose:
- * Contains common math routines for the various audio engines.
- *
- *
- * Copyright Sonic Network Inc. 2005
-
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- *
- *----------------------------------------------------------------------------
- * Revision Control:
- * $Revision: 584 $
- * $Date: 2007-03-08 09:49:24 -0800 (Thu, 08 Mar 2007) $
- *----------------------------------------------------------------------------
-*/
-
-#ifndef _EAS_MATH_H
-#define _EAS_MATH_H
-
-
-/** coefs for pan, generates sin, cos */
-#define COEFF_PAN_G2 -27146 /* -0.82842712474619 = 2 - 4/sqrt(2) */
-#define COEFF_PAN_G0 23170 /* 0.707106781186547 = 1/sqrt(2) */
-
-/*
-coefficients for approximating
-2^x = gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3
-where x is a int.frac number representing number of octaves.
-Actually, we approximate only the 2^(frac) using the power series
-and implement the 2^(int) as a shift, so that
-2^x == 2^(int.frac) == 2^(int) * 2^(fract)
- == (gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3) << (int)
-
-The gn2toX.. were generated using a best fit for a 3rd
-order polynomial, instead of taking the coefficients from
-a truncated Taylor (or Maclaurin?) series.
-*/
-
-#define GN2_TO_X0 32768 /* 1 */
-#define GN2_TO_X1 22833 /* 0.696807861328125 */
-#define GN2_TO_X2 7344 /* 0.22412109375 */
-#define GN2_TO_X3 2588 /* 0.0789794921875 */
-
-/*----------------------------------------------------------------------------
- * Fixed Point Math
- *----------------------------------------------------------------------------
- * These macros are used for fixed point multiplies. If the processor
- * supports fixed point multiplies, replace these macros with inline
- * assembly code to improve performance.
- *----------------------------------------------------------------------------
-*/
-
-/* Fixed point multiply 0.15 x 0.15 = 0.15 returned as 32-bits */
-#define FMUL_15x15(a,b) \
- /*lint -e(704) <avoid multiply for performance>*/ \
- (((EAS_I32)(a) * (EAS_I32)(b)) >> 15)
-
-/* Fixed point multiply 0.7 x 0.7 = 0.15 returned as 32-bits */
-#define FMUL_7x7(a,b) \
- /*lint -e(704) <avoid multiply for performance>*/ \
- (((EAS_I32)(a) * (EAS_I32)(b) ) << 1)
-
-/* Fixed point multiply 0.8 x 0.8 = 0.15 returned as 32-bits */
-#define FMUL_8x8(a,b) \
- /*lint -e(704) <avoid multiply for performance>*/ \
- (((EAS_I32)(a) * (EAS_I32)(b) ) >> 1)
-
-/* Fixed point multiply 0.8 x 1.15 = 0.15 returned as 32-bits */
-#define FMUL_8x15(a,b) \
- /*lint -e(704) <avoid divide for performance>*/ \
- (((EAS_I32)((a) << 7) * (EAS_I32)(b)) >> 15)
-
-/* macros for fractional phase accumulator */
-/*
-Note: changed the _U32 to _I32 on 03/14/02. This should not
-affect the phase calculations, and should allow us to reuse these
-macros for other audio sample related math.
-*/
-#define HARDWARE_BIT_WIDTH 32
-
-#define NUM_PHASE_INT_BITS 1
-#define NUM_PHASE_FRAC_BITS 15
-
-#define PHASE_FRAC_MASK (EAS_U32) ((0x1L << NUM_PHASE_FRAC_BITS) -1)
-
-#define GET_PHASE_INT_PART(x) (EAS_U32)((EAS_U32)(x) >> NUM_PHASE_FRAC_BITS)
-#define GET_PHASE_FRAC_PART(x) (EAS_U32)((EAS_U32)(x) & PHASE_FRAC_MASK)
-
-#define DEFAULT_PHASE_FRAC 0
-#define DEFAULT_PHASE_INT 0
-
-/*
-Linear interpolation calculates:
-output = (1-frac) * sample[n] + (frac) * sample[n+1]
-
-where conceptually 0 <= frac < 1
-
-For a fixed point implementation, frac is actually an integer value
-with an implied binary point one position to the left. The value of
-one (unity) is given by PHASE_ONE
-one half and one quarter are useful for 4-point linear interp.
-*/
-#define PHASE_ONE (EAS_I32) (0x1L << NUM_PHASE_FRAC_BITS)
-
-/*
- Multiply the signed audio sample by the unsigned fraction.
-- a is the signed audio sample
-- b is the unsigned fraction (cast to signed int as long as coef
- uses (n-1) or less bits, where n == hardware bit width)
-*/
-#define MULT_AUDIO_COEF(audio,coef) /*lint -e704 <avoid divide for performance>*/ \
- (EAS_I32)( \
- ( \
- ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \
- ) \
- >> NUM_PHASE_FRAC_BITS \
- ) \
- /* lint +704 <restore checking>*/
-
-/* wet / dry calculation macros */
-#define NUM_WET_DRY_FRAC_BITS 7 // 15
-#define NUM_WET_DRY_INT_BITS 9 // 1
-
-/* define a 1.0 */
-#define WET_DRY_ONE (EAS_I32) ((0x1L << NUM_WET_DRY_FRAC_BITS))
-#define WET_DRY_MINUS_ONE (EAS_I32) (~WET_DRY_ONE)
-#define WET_DRY_FULL_SCALE (EAS_I32) (WET_DRY_ONE - 1)
-
-#define MULT_AUDIO_WET_DRY_COEF(audio,coef) /*lint -e(702) <avoid divide for performance>*/ \
- (EAS_I32)( \
- ( \
- ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \
- ) \
- >> NUM_WET_DRY_FRAC_BITS \
- )
-
-/* Envelope 1 (EG1) calculation macros */
-#define NUM_EG1_INT_BITS 1
-#define NUM_EG1_FRAC_BITS 15
-
-/* the max positive gain used in the synth for EG1 */
-/* SYNTH_FULL_SCALE_EG1_GAIN must match the value in the dls2eas
-converter, otherwise, the values we read from the .eas file are bogus. */
-#define SYNTH_FULL_SCALE_EG1_GAIN (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS) -1)
-
-/* define a 1.0 */
-#define EG1_ONE (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS))
-#define EG1_MINUS_ONE (EAS_I32) (~SYNTH_FULL_SCALE_EG1_GAIN)
-
-#define EG1_HALF (EAS_I32) (EG1_ONE/2)
-#define EG1_MINUS_HALF (EAS_I32) (EG1_MINUS_ONE/2)
-
-/*
-We implement the EG1 using a linear gain value, which means that the
-attack segment is handled by incrementing (adding) the linear gain.
-However, EG1 treats the Decay, Sustain, and Release differently than
-the Attack portion. For Decay, Sustain, and Release, the gain is
-linear on dB scale, which is equivalent to exponential damping on
-a linear scale. Because we use a linear gain for EG1, we implement
-the Decay and Release as multiplication (instead of incrementing
-as we did for the attack segment).
-Therefore, we need the following macro to implement the multiplication
-(i.e., exponential damping) during the Decay and Release segments of
-the EG1
-*/
-#define MULT_EG1_EG1(gain,damping) /*lint -e(704) <avoid divide for performance>*/ \
- (EAS_I32)( \
- ( \
- ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \
- ) \
- >> NUM_EG1_FRAC_BITS \
- )
-
-// Use the following macro specifically for the filter, when multiplying
-// the b1 coefficient. The 0 <= |b1| < 2, which therefore might overflow
-// in certain conditions because we store b1 as a 1.15 value.
-// Instead, we could store b1 as b1p (b1' == b1 "prime") where
-// b1p == b1/2, thus ensuring no potential overflow for b1p because
-// 0 <= |b1p| < 1
-// However, during the filter calculation, we must account for the fact
-// that we are using b1p instead of b1, and thereby multiply by
-// an extra factor of 2. Rather than multiply by an extra factor of 2,
-// we can instead shift the result right by one less, hence the
-// modified shift right value of (NUM_EG1_FRAC_BITS -1)
-#define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) <avoid divide for performance>*/ \
- (EAS_I32)( \
- ( \
- ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \
- ) \
- >> (NUM_EG1_FRAC_BITS -1) \
- )
-
-#define SATURATE_EG1(x) /*lint -e{734} saturation operation */ \
- ((EAS_I32)(x) > SYNTH_FULL_SCALE_EG1_GAIN) ? (SYNTH_FULL_SCALE_EG1_GAIN) : \
- ((EAS_I32)(x) < EG1_MINUS_ONE) ? (EG1_MINUS_ONE) : (x);
-
-
-/* use "digital cents" == "dents" instead of cents */
-/* we coudl re-use the phase frac macros, but if we do,
-we must change the phase macros to cast to _I32 instead of _U32,
-because using a _U32 cast causes problems when shifting the exponent
-for the 2^x calculation, because right shift a negative values MUST
-be sign extended, or else the 2^x calculation is wrong */
-
-/* use "digital cents" == "dents" instead of cents */
-#define NUM_DENTS_FRAC_BITS 12
-#define NUM_DENTS_INT_BITS (HARDWARE_BIT_WIDTH - NUM_DENTS_FRAC_BITS)
-
-#define DENTS_FRAC_MASK (EAS_I32) ((0x1L << NUM_DENTS_FRAC_BITS) -1)
-
-#define GET_DENTS_INT_PART(x) /*lint -e(704) <avoid divide for performance>*/ \
- (EAS_I32)((EAS_I32)(x) >> NUM_DENTS_FRAC_BITS)
-
-#define GET_DENTS_FRAC_PART(x) (EAS_I32)((EAS_I32)(x) & DENTS_FRAC_MASK)
-
-#define DENTS_ONE (EAS_I32) (0x1L << NUM_DENTS_FRAC_BITS)
-
-/* use CENTS_TO_DENTS to convert a value in cents to dents */
-#define CENTS_TO_DENTS (EAS_I32) (DENTS_ONE * (0x1L << NUM_EG1_FRAC_BITS) / 1200L) \
-
-
-/*
-For gain, the LFO generates a value that modulates in terms
-of dB. However, we use a linear gain value, so we must convert
-the LFO value in dB to a linear gain. Normally, we would use
-linear gain = 10^x, where x = LFO value in dB / 20.
-Instead, we implement 10^x using our 2^x approximation.
-because
-
- 10^x = 2^(log2(10^x)) = 2^(x * log2(10))
-
-so we need to multiply by log2(10) which is just a constant.
-Ah, but just wait -- our 2^x actually doesn't exactly implement
-2^x, but it actually assumes that the input is in cents, and within
-the 2^x approximation converts its input from cents to octaves
-by dividing its input by 1200.
-
-So, in order to convert the LFO gain value in dB to something
-that our existing 2^x approximation can use, multiply the LFO gain
-by log2(10) * 1200 / 20
-
-The divide by 20 helps convert dB to linear gain, and we might
-as well incorporate that operation into this conversion.
-Of course, we need to keep some fractional bits, so multiply
-the constant by NUM_EG1_FRAC_BITS
-*/
-
-/* use LFO_GAIN_TO_CENTS to convert the LFO gain value to cents */
-#if 0
-#define DOUBLE_LOG2_10 (double) (3.32192809488736) /* log2(10) */
-
-#define DOUBLE_LFO_GAIN_TO_CENTS (double) \
- ( \
- (DOUBLE_LOG2_10) * \
- 1200.0 / \
- 20.0 \
- )
-
-#define LFO_GAIN_TO_CENTS (EAS_I32) \
- ( \
- DOUBLE_LFO_GAIN_TO_CENTS * \
- (0x1L << NUM_EG1_FRAC_BITS) \
- )
-#endif
-
-#define LFO_GAIN_TO_CENTS (EAS_I32) (1671981156L >> (23 - NUM_EG1_FRAC_BITS))
-
-
-#define MULT_DENTS_COEF(dents,coef) /*lint -e704 <avoid divide for performance>*/ \
- (EAS_I32)( \
- ( \
- ((EAS_I32)(dents)) * ((EAS_I32)(coef)) \
- ) \
- >> NUM_DENTS_FRAC_BITS \
- ) \
- /* lint +e704 <restore checking>*/
-
-/* we use 16-bits in the PC per audio sample */
-#define BITS_PER_AUDIO_SAMPLE 16
-
-/* we define 1 as 1.0 - 1 LSbit */
-#define DISTORTION_ONE (EAS_I32)((0x1L << (BITS_PER_AUDIO_SAMPLE-1)) -1)
-#define DISTORTION_MINUS_ONE (EAS_I32)(~DISTORTION_ONE)
-
-/* drive coef is given as int.frac */
-#define NUM_DRIVE_COEF_INT_BITS 1
-#define NUM_DRIVE_COEF_FRAC_BITS 4
-
-#define MULT_AUDIO_DRIVE(audio,drive) /*lint -e(702) <avoid divide for performance>*/ \
- (EAS_I32) ( \
- ( \
- ((EAS_I32)(audio)) * ((EAS_I32)(drive)) \
- ) \
- >> NUM_DRIVE_COEF_FRAC_BITS \
- )
-
-#define MULT_AUDIO_AUDIO(audio1,audio2) /*lint -e(702) <avoid divide for performance>*/ \
- (EAS_I32) ( \
- ( \
- ((EAS_I32)(audio1)) * ((EAS_I32)(audio2)) \
- ) \
- >> (BITS_PER_AUDIO_SAMPLE-1) \
- )
-
-#define SATURATE(x) \
- ((((EAS_I32)(x)) > DISTORTION_ONE) ? (DISTORTION_ONE) : \
- (((EAS_I32)(x)) < DISTORTION_MINUS_ONE) ? (DISTORTION_MINUS_ONE) : ((EAS_I32)(x)));
-
-
-
-/*----------------------------------------------------------------------------
- * EAS_Calculate2toX()
- *----------------------------------------------------------------------------
- * Purpose:
- * Calculate 2^x
- *
- * Inputs:
- * nCents - measured in cents
- *
- * Outputs:
- * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS)
- *
- * Side Effects:
- *
- *----------------------------------------------------------------------------
-*/
-EAS_I32 EAS_Calculate2toX (EAS_I32 nCents);
-
-/*----------------------------------------------------------------------------
- * EAS_LogToLinear16()
- *----------------------------------------------------------------------------
- * Purpose:
- * Transform log value to linear gain multiplier using piece-wise linear
- * approximation
- *
- * Inputs:
- * nGain - log scale value in 20.10 format. Even though gain is normally
- * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate
- * the need for saturation checking when combining gain values.
- *
- * Outputs:
- * Returns a 16-bit linear value approximately equal to 2^(nGain/1024)
- *
- * Side Effects:
- *
- *----------------------------------------------------------------------------
-*/
-EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain);
-
-/*----------------------------------------------------------------------------
- * EAS_VolumeToGain()
- *----------------------------------------------------------------------------
- * Purpose:
- * Transform volume control in 1dB increments to gain multiplier
- *
- * Inputs:
- * volume - 100 = 0dB, 99 = -1dB, 0 = -inf
- *
- * Outputs:
- * Returns a 16-bit linear value
- *----------------------------------------------------------------------------
-*/
-EAS_I16 EAS_VolumeToGain (EAS_INT volume);
-
-/*----------------------------------------------------------------------------
- * EAS_fsqrt()
- *----------------------------------------------------------------------------
- * Purpose:
- * Calculates the square root of a 32-bit fixed point value
- *
- * Inputs:
- * n = value of interest
- *
- * Outputs:
- * returns the square root of n
- *
- *----------------------------------------------------------------------------
-*/
-EAS_U16 EAS_fsqrt (EAS_U32 n);
-
-/*----------------------------------------------------------------------------
- * EAS_flog2()
- *----------------------------------------------------------------------------
- * Purpose:
- * Calculates the log2 of a 32-bit fixed point value
- *
- * Inputs:
- * n = value of interest
- *
- * Outputs:
- * returns the log2 of n
- *
- *----------------------------------------------------------------------------
-*/
-EAS_I32 EAS_flog2 (EAS_U32 n);
-
-#endif
-