From 6ab94e0b318884bbcb95e2ea3835f951502e1d99 Mon Sep 17 00:00:00 2001 From: jaseg Date: Wed, 14 Oct 2020 12:47:28 +0200 Subject: Move firmware into subdirectory --- .../FilteringFunctions/arm_fir_interpolate_f32.c | 569 +++++++++++++++++++++ 1 file changed, 569 insertions(+) create mode 100644 fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c (limited to 'fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c') diff --git a/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c b/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c new file mode 100644 index 0000000..5f9d19c --- /dev/null +++ b/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c @@ -0,0 +1,569 @@ +/* ---------------------------------------------------------------------- + * Project: CMSIS DSP Library + * Title: arm_fir_interpolate_f32.c + * Description: Floating-point FIR interpolation sequences + * + * $Date: 27. January 2017 + * $Revision: V.1.5.1 + * + * Target Processor: Cortex-M cores + * -------------------------------------------------------------------- */ +/* + * Copyright (C) 2010-2017 ARM Limited or its affiliates. All rights reserved. + * + * SPDX-License-Identifier: Apache-2.0 + * + * 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 + * + * 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. + */ + +#include "arm_math.h" + +/** + * @defgroup FIR_Interpolate Finite Impulse Response (FIR) Interpolator + * + * These functions combine an upsampler (zero stuffer) and an FIR filter. + * They are used in multirate systems for increasing the sample rate of a signal without introducing high frequency images. + * Conceptually, the functions are equivalent to the block diagram below: + * \image html FIRInterpolator.gif "Components included in the FIR Interpolator functions" + * After upsampling by a factor of L, the signal should be filtered by a lowpass filter with a normalized + * cutoff frequency of 1/L in order to eliminate high frequency copies of the spectrum. + * The user of the function is responsible for providing the filter coefficients. + * + * The FIR interpolator functions provided in the CMSIS DSP Library combine the upsampler and FIR filter in an efficient manner. + * The upsampler inserts L-1 zeros between each sample. + * Instead of multiplying by these zero values, the FIR filter is designed to skip them. + * This leads to an efficient implementation without any wasted effort. + * The functions operate on blocks of input and output data. + * pSrc points to an array of blockSize input values and + * pDst points to an array of blockSize*L output values. + * + * The library provides separate functions for Q15, Q31, and floating-point data types. + * + * \par Algorithm: + * The functions use a polyphase filter structure: + * 
+ *    y[n] = b[0] * x[n] + b[L]   * x[n-1] + ... + b[L*(phaseLength-1)] * x[n-phaseLength+1]
+ *    y[n+1] = b[1] * x[n] + b[L+1] * x[n-1] + ... + b[L*(phaseLength-1)+1] * x[n-phaseLength+1]
+ *    ...
+ *    y[n+(L-1)] = b[L-1] * x[n] + b[2*L-1] * x[n-1] + ....+ b[L*(phaseLength-1)+(L-1)] * x[n-phaseLength+1]
+ *
+ * This approach is more efficient than straightforward upsample-then-filter algorithms. + * With this method the computation is reduced by a factor of 1/L when compared to using a standard FIR filter. + * \par + * pCoeffs points to a coefficient array of size numTaps. + * numTaps must be a multiple of the interpolation factor L and this is checked by the + * initialization functions. + * Internally, the function divides the FIR filter's impulse response into shorter filters of length + * phaseLength=numTaps/L. + * Coefficients are stored in time reversed order. + * \par + * 
+ *    {b[numTaps-1], b[numTaps-2], b[N-2], ..., b[1], b[0]}
+ *
+ * \par + * pState points to a state array of size blockSize + phaseLength - 1. + * Samples in the state buffer are stored in the order: + * \par + * 
+ *    {x[n-phaseLength+1], x[n-phaseLength], x[n-phaseLength-1], x[n-phaseLength-2]....x[0], x[1], ..., x[blockSize-1]}
+ *
+ * The state variables are updated after each block of data is processed, the coefficients are untouched. + * + * \par Instance Structure + * The coefficients and state variables for a filter are stored together in an instance data structure. + * A separate instance structure must be defined for each filter. + * Coefficient arrays may be shared among several instances while state variable array should be allocated separately. + * There are separate instance structure declarations for each of the 3 supported data types. + * + * \par Initialization Functions + * There is also an associated initialization function for each data type. + * The initialization function performs the following operations: + * - Sets the values of the internal structure fields. + * - Zeros out the values in the state buffer. + * - Checks to make sure that the length of the filter is a multiple of the interpolation factor. + * To do this manually without calling the init function, assign the follow subfields of the instance structure: + * L (interpolation factor), pCoeffs, phaseLength (numTaps / L), pState. Also set all of the values in pState to zero. + * + * \par + * Use of the initialization function is optional. + * However, if the initialization function is used, then the instance structure cannot be placed into a const data section. + * To place an instance structure into a const data section, the instance structure must be manually initialized. + * The code below statically initializes each of the 3 different data type filter instance structures + * 
+ * arm_fir_interpolate_instance_f32 S = {L, phaseLength, pCoeffs, pState};
+ * arm_fir_interpolate_instance_q31 S = {L, phaseLength, pCoeffs, pState};
+ * arm_fir_interpolate_instance_q15 S = {L, phaseLength, pCoeffs, pState};
+ *
+ * where L is the interpolation factor; phaseLength=numTaps/L is the + * length of each of the shorter FIR filters used internally, + * pCoeffs is the address of the coefficient buffer; + * pState is the address of the state buffer. + * Be sure to set the values in the state buffer to zeros when doing static initialization. + * + * \par Fixed-Point Behavior + * Care must be taken when using the fixed-point versions of the FIR interpolate filter functions. + * In particular, the overflow and saturation behavior of the accumulator used in each function must be considered. + * Refer to the function specific documentation below for usage guidelines. + */ + +/** + * @addtogroup FIR_Interpolate + * @{ + */ + +/** + * @brief Processing function for the floating-point FIR interpolator. + * @param[in] *S points to an instance of the floating-point FIR interpolator structure. + * @param[in] *pSrc points to the block of input data. + * @param[out] *pDst points to the block of output data. + * @param[in] blockSize number of input samples to process per call. + * @return none. + */ +#if defined (ARM_MATH_DSP) + + /* Run the below code for Cortex-M4 and Cortex-M3 */ + +void arm_fir_interpolate_f32( + const arm_fir_interpolate_instance_f32 * S, + float32_t * pSrc, + float32_t * pDst, + uint32_t blockSize) +{ + float32_t *pState = S->pState; /* State pointer */ + float32_t *pCoeffs = S->pCoeffs; /* Coefficient pointer */ + float32_t *pStateCurnt; /* Points to the current sample of the state */ + float32_t *ptr1, *ptr2; /* Temporary pointers for state and coefficient buffers */ + float32_t sum0; /* Accumulators */ + float32_t x0, c0; /* Temporary variables to hold state and coefficient values */ + uint32_t i, blkCnt, j; /* Loop counters */ + uint16_t phaseLen = S->phaseLength, tapCnt; /* Length of each polyphase filter component */ + float32_t acc0, acc1, acc2, acc3; + float32_t x1, x2, x3; + uint32_t blkCntN4; + float32_t c1, c2, c3; + + /* S->pState buffer contains previous frame (phaseLen - 1) samples */ + /* pStateCurnt points to the location where the new input data should be written */ + pStateCurnt = S->pState + (phaseLen - 1U); + + /* Initialise blkCnt */ + blkCnt = blockSize / 4; + blkCntN4 = blockSize - (4 * blkCnt); + + /* Samples loop unrolled by 4 */ + while (blkCnt > 0U) + { + /* Copy new input sample into the state buffer */ + *pStateCurnt++ = *pSrc++; + *pStateCurnt++ = *pSrc++; + *pStateCurnt++ = *pSrc++; + *pStateCurnt++ = *pSrc++; + + /* Address modifier index of coefficient buffer */ + j = 1U; + + /* Loop over the Interpolation factor. */ + i = (S->L); + + while (i > 0U) + { + /* Set accumulator to zero */ + acc0 = 0.0f; + acc1 = 0.0f; + acc2 = 0.0f; + acc3 = 0.0f; + + /* Initialize state pointer */ + ptr1 = pState; + + /* Initialize coefficient pointer */ + ptr2 = pCoeffs + (S->L - j); + + /* Loop over the polyPhase length. Unroll by a factor of 4. + ** Repeat until we've computed numTaps-(4*S->L) coefficients. */ + tapCnt = phaseLen >> 2U; + + x0 = *(ptr1++); + x1 = *(ptr1++); + x2 = *(ptr1++); + + while (tapCnt > 0U) + { + + /* Read the input sample */ + x3 = *(ptr1++); + + /* Read the coefficient */ + c0 = *(ptr2); + + /* Perform the multiply-accumulate */ + acc0 += x0 * c0; + acc1 += x1 * c0; + acc2 += x2 * c0; + acc3 += x3 * c0; + + /* Read the coefficient */ + c1 = *(ptr2 + S->L); + + /* Read the input sample */ + x0 = *(ptr1++); + + /* Perform the multiply-accumulate */ + acc0 += x1 * c1; + acc1 += x2 * c1; + acc2 += x3 * c1; + acc3 += x0 * c1; + + /* Read the coefficient */ + c2 = *(ptr2 + S->L * 2); + + /* Read the input sample */ + x1 = *(ptr1++); + + /* Perform the multiply-accumulate */ + acc0 += x2 * c2; + acc1 += x3 * c2; + acc2 += x0 * c2; + acc3 += x1 * c2; + + /* Read the coefficient */ + c3 = *(ptr2 + S->L * 3); + + /* Read the input sample */ + x2 = *(ptr1++); + + /* Perform the multiply-accumulate */ + acc0 += x3 * c3; + acc1 += x0 * c3; + acc2 += x1 * c3; + acc3 += x2 * c3; + + + /* Upsampling is done by stuffing L-1 zeros between each sample. + * So instead of multiplying zeros with coefficients, + * Increment the coefficient pointer by interpolation factor times. */ + ptr2 += 4 * S->L; + + /* Decrement the loop counter */ + tapCnt--; + } + + /* If the polyPhase length is not a multiple of 4, compute the remaining filter taps */ + tapCnt = phaseLen % 0x4U; + + while (tapCnt > 0U) + { + + /* Read the input sample */ + x3 = *(ptr1++); + + /* Read the coefficient */ + c0 = *(ptr2); + + /* Perform the multiply-accumulate */ + acc0 += x0 * c0; + acc1 += x1 * c0; + acc2 += x2 * c0; + acc3 += x3 * c0; + + /* Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* update states for next sample processing */ + x0 = x1; + x1 = x2; + x2 = x3; + + /* Decrement the loop counter */ + tapCnt--; + } + + /* The result is in the accumulator, store in the destination buffer. */ + *pDst = acc0; + *(pDst + S->L) = acc1; + *(pDst + 2 * S->L) = acc2; + *(pDst + 3 * S->L) = acc3; + + pDst++; + + /* Increment the address modifier index of coefficient buffer */ + j++; + + /* Decrement the loop counter */ + i--; + } + + /* Advance the state pointer by 1 + * to process the next group of interpolation factor number samples */ + pState = pState + 4; + + pDst += S->L * 3; + + /* Decrement the loop counter */ + blkCnt--; + } + + /* If the blockSize is not a multiple of 4, compute any remaining output samples here. + ** No loop unrolling is used. */ + + while (blkCntN4 > 0U) + { + /* Copy new input sample into the state buffer */ + *pStateCurnt++ = *pSrc++; + + /* Address modifier index of coefficient buffer */ + j = 1U; + + /* Loop over the Interpolation factor. */ + i = S->L; + while (i > 0U) + { + /* Set accumulator to zero */ + sum0 = 0.0f; + + /* Initialize state pointer */ + ptr1 = pState; + + /* Initialize coefficient pointer */ + ptr2 = pCoeffs + (S->L - j); + + /* Loop over the polyPhase length. Unroll by a factor of 4. + ** Repeat until we've computed numTaps-(4*S->L) coefficients. */ + tapCnt = phaseLen >> 2U; + while (tapCnt > 0U) + { + + /* Read the coefficient */ + c0 = *(ptr2); + + /* Upsampling is done by stuffing L-1 zeros between each sample. + * So instead of multiplying zeros with coefficients, + * Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* Read the input sample */ + x0 = *(ptr1++); + + /* Perform the multiply-accumulate */ + sum0 += x0 * c0; + + /* Read the coefficient */ + c0 = *(ptr2); + + /* Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* Read the input sample */ + x0 = *(ptr1++); + + /* Perform the multiply-accumulate */ + sum0 += x0 * c0; + + /* Read the coefficient */ + c0 = *(ptr2); + + /* Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* Read the input sample */ + x0 = *(ptr1++); + + /* Perform the multiply-accumulate */ + sum0 += x0 * c0; + + /* Read the coefficient */ + c0 = *(ptr2); + + /* Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* Read the input sample */ + x0 = *(ptr1++); + + /* Perform the multiply-accumulate */ + sum0 += x0 * c0; + + /* Decrement the loop counter */ + tapCnt--; + } + + /* If the polyPhase length is not a multiple of 4, compute the remaining filter taps */ + tapCnt = phaseLen % 0x4U; + + while (tapCnt > 0U) + { + /* Perform the multiply-accumulate */ + sum0 += *(ptr1++) * (*ptr2); + + /* Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* Decrement the loop counter */ + tapCnt--; + } + + /* The result is in the accumulator, store in the destination buffer. */ + *pDst++ = sum0; + + /* Increment the address modifier index of coefficient buffer */ + j++; + + /* Decrement the loop counter */ + i--; + } + + /* Advance the state pointer by 1 + * to process the next group of interpolation factor number samples */ + pState = pState + 1; + + /* Decrement the loop counter */ + blkCntN4--; + } + + /* Processing is complete. + ** Now copy the last phaseLen - 1 samples to the satrt of the state buffer. + ** This prepares the state buffer for the next function call. */ + + /* Points to the start of the state buffer */ + pStateCurnt = S->pState; + + tapCnt = (phaseLen - 1U) >> 2U; + + /* copy data */ + while (tapCnt > 0U) + { + *pStateCurnt++ = *pState++; + *pStateCurnt++ = *pState++; + *pStateCurnt++ = *pState++; + *pStateCurnt++ = *pState++; + + /* Decrement the loop counter */ + tapCnt--; + } + + tapCnt = (phaseLen - 1U) % 0x04U; + + /* copy data */ + while (tapCnt > 0U) + { + *pStateCurnt++ = *pState++; + + /* Decrement the loop counter */ + tapCnt--; + } +} + +#else + + /* Run the below code for Cortex-M0 */ + +void arm_fir_interpolate_f32( + const arm_fir_interpolate_instance_f32 * S, + float32_t * pSrc, + float32_t * pDst, + uint32_t blockSize) +{ + float32_t *pState = S->pState; /* State pointer */ + float32_t *pCoeffs = S->pCoeffs; /* Coefficient pointer */ + float32_t *pStateCurnt; /* Points to the current sample of the state */ + float32_t *ptr1, *ptr2; /* Temporary pointers for state and coefficient buffers */ + + + float32_t sum; /* Accumulator */ + uint32_t i, blkCnt; /* Loop counters */ + uint16_t phaseLen = S->phaseLength, tapCnt; /* Length of each polyphase filter component */ + + + /* S->pState buffer contains previous frame (phaseLen - 1) samples */ + /* pStateCurnt points to the location where the new input data should be written */ + pStateCurnt = S->pState + (phaseLen - 1U); + + /* Total number of intput samples */ + blkCnt = blockSize; + + /* Loop over the blockSize. */ + while (blkCnt > 0U) + { + /* Copy new input sample into the state buffer */ + *pStateCurnt++ = *pSrc++; + + /* Loop over the Interpolation factor. */ + i = S->L; + + while (i > 0U) + { + /* Set accumulator to zero */ + sum = 0.0f; + + /* Initialize state pointer */ + ptr1 = pState; + + /* Initialize coefficient pointer */ + ptr2 = pCoeffs + (i - 1U); + + /* Loop over the polyPhase length */ + tapCnt = phaseLen; + + while (tapCnt > 0U) + { + /* Perform the multiply-accumulate */ + sum += *ptr1++ * *ptr2; + + /* Increment the coefficient pointer by interpolation factor times. */ + ptr2 += S->L; + + /* Decrement the loop counter */ + tapCnt--; + } + + /* The result is in the accumulator, store in the destination buffer. */ + *pDst++ = sum; + + /* Decrement the loop counter */ + i--; + } + + /* Advance the state pointer by 1 + * to process the next group of interpolation factor number samples */ + pState = pState + 1; + + /* Decrement the loop counter */ + blkCnt--; + } + + /* Processing is complete. + ** Now copy the last phaseLen - 1 samples to the start of the state buffer. + ** This prepares the state buffer for the next function call. */ + + /* Points to the start of the state buffer */ + pStateCurnt = S->pState; + + tapCnt = phaseLen - 1U; + + while (tapCnt > 0U) + { + *pStateCurnt++ = *pState++; + + /* Decrement the loop counter */ + tapCnt--; + } + +} + +#endif /* #if defined (ARM_MATH_DSP) */ + + + + /** + * @} end of FIR_Interpolate group + */ -- cgit