From 7df30109963092559d3760c0661a020f9daf1030 Mon Sep 17 00:00:00 2001 From: The Android Open Source Project Date: Tue, 3 Mar 2009 19:30:38 -0800 Subject: auto import from //depot/cupcake/@135843 --- arm-fm-22k/lib_src/eas_math.h | 412 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 412 insertions(+) create mode 100644 arm-fm-22k/lib_src/eas_math.h (limited to 'arm-fm-22k/lib_src/eas_math.h') diff --git a/arm-fm-22k/lib_src/eas_math.h b/arm-fm-22k/lib_src/eas_math.h new file mode 100644 index 0000000..719270b --- /dev/null +++ b/arm-fm-22k/lib_src/eas_math.h @@ -0,0 +1,412 @@ +/*---------------------------------------------------------------------------- + * + * 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) */ \ + (((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) */ \ + (((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) */ \ + (((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) */ \ + (((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 */ \ + (EAS_I32)( \ + ( \ + ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \ + ) \ + >> NUM_PHASE_FRAC_BITS \ + ) \ + /* lint +704 */ + +/* 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) */ \ + (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) */ \ + (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) */ \ + (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) */ \ + (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 */ \ + (EAS_I32)( \ + ( \ + ((EAS_I32)(dents)) * ((EAS_I32)(coef)) \ + ) \ + >> NUM_DENTS_FRAC_BITS \ + ) \ + /* lint +e704 */ + +/* 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) */ \ + (EAS_I32) ( \ + ( \ + ((EAS_I32)(audio)) * ((EAS_I32)(drive)) \ + ) \ + >> NUM_DRIVE_COEF_FRAC_BITS \ + ) + +#define MULT_AUDIO_AUDIO(audio1,audio2) /*lint -e(702) */ \ + (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 + -- cgit v1.2.3