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/* ----------------------------------------------------------------------
* 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 <code>L</code>, the signal should be filtered by a lowpass filter with a normalized
* cutoff frequency of <code>1/L</code> 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 <code>L-1</code> 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.
* <code>pSrc</code> points to an array of <code>blockSize</code> input values and
* <code>pDst</code> points to an array of <code>blockSize*L</code> output values.
*
* The library provides separate functions for Q15, Q31, and floating-point data types.
*
* \par Algorithm:
* The functions use a polyphase filter structure:
* <pre>
* 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]
* </pre>
* This approach is more efficient than straightforward upsample-then-filter algorithms.
* With this method the computation is reduced by a factor of <code>1/L</code> when compared to using a standard FIR filter.
* \par
* <code>pCoeffs</code> points to a coefficient array of size <code>numTaps</code>.
* <code>numTaps</code> must be a multiple of the interpolation factor <code>L</code> and this is checked by the
* initialization functions.
* Internally, the function divides the FIR filter's impulse response into shorter filters of length
* <code>phaseLength=numTaps/L</code>.
* Coefficients are stored in time reversed order.
* \par
* <pre>
* {b[numTaps-1], b[numTaps-2], b[N-2], ..., b[1], b[0]}
* </pre>
* \par
* <code>pState</code> points to a state array of size <code>blockSize + phaseLength - 1</code>.
* Samples in the state buffer are stored in the order:
* \par
* <pre>
* {x[n-phaseLength+1], x[n-phaseLength], x[n-phaseLength-1], x[n-phaseLength-2]....x[0], x[1], ..., x[blockSize-1]}
* </pre>
* 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
* <pre>
* 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};
* </pre>
* where <code>L</code> is the interpolation factor; <code>phaseLength=numTaps/L</code> is the
* length of each of the shorter FIR filters used internally,
* <code>pCoeffs</code> is the address of the coefficient buffer;
* <code>pState</code> 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
*/