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Diffstat (limited to 'fw/hid-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c')
| -rw-r--r-- | fw/hid-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c | 569 |
1 files changed, 569 insertions, 0 deletions
diff --git a/fw/hid-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c b/fw/hid-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_fir_interpolate_f32.c new file mode 100644 index 0000000..5f9d19c --- /dev/null +++ b/fw/hid-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 <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
+ */
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