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+/* ----------------------------------------------------------------------
+ * Project: CMSIS DSP Library
+ * Title: arm_lms_f32.c
+ * Description: Processing function for the floating-point LMS filter
+ *
+ * $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"
+
+/**
+ * @ingroup groupFilters
+ */
+
+/**
+ * @defgroup LMS Least Mean Square (LMS) Filters
+ *
+ * LMS filters are a class of adaptive filters that are able to "learn" an unknown transfer functions.
+ * LMS filters use a gradient descent method in which the filter coefficients are updated based on the instantaneous error signal.
+ * Adaptive filters are often used in communication systems, equalizers, and noise removal.
+ * The CMSIS DSP Library contains LMS filter functions that operate on Q15, Q31, and floating-point data types.
+ * The library also contains normalized LMS filters in which the filter coefficient adaptation is indepedent of the level of the input signal.
+ *
+ * An LMS filter consists of two components as shown below.
+ * The first component is a standard transversal or FIR filter.
+ * The second component is a coefficient update mechanism.
+ * The LMS filter has two input signals.
+ * The "input" feeds the FIR filter while the "reference input" corresponds to the desired output of the FIR filter.
+ * That is, the FIR filter coefficients are updated so that the output of the FIR filter matches the reference input.
+ * The filter coefficient update mechanism is based on the difference between the FIR filter output and the reference input.
+ * This "error signal" tends towards zero as the filter adapts.
+ * The LMS processing functions accept the input and reference input signals and generate the filter output and error signal.
+ * \image html LMS.gif "Internal structure of the Least Mean Square filter"
+ *
+ * The functions operate on blocks of data and each call to the function processes
+ * <code>blockSize</code> samples through the filter.
+ * <code>pSrc</code> points to input signal, <code>pRef</code> points to reference signal,
+ * <code>pOut</code> points to output signal and <code>pErr</code> points to error signal.
+ * All arrays contain <code>blockSize</code> values.
+ *
+ * The functions operate on a block-by-block basis.
+ * Internally, the filter coefficients <code>b[n]</code> are updated on a sample-by-sample basis.
+ * The convergence of the LMS filter is slower compared to the normalized LMS algorithm.
+ *
+ * \par Algorithm:
+ * The output signal <code>y[n]</code> is computed by a standard FIR filter:
+ * <pre>
+ * y[n] = b[0] * x[n] + b[1] * x[n-1] + b[2] * x[n-2] + ...+ b[numTaps-1] * x[n-numTaps+1]
+ * </pre>
+ *
+ * \par
+ * The error signal equals the difference between the reference signal <code>d[n]</code> and the filter output:
+ * <pre>
+ * e[n] = d[n] - y[n].
+ * </pre>
+ *
+ * \par
+ * After each sample of the error signal is computed, the filter coefficients <code>b[k]</code> are updated on a sample-by-sample basis:
+ * <pre>
+ * b[k] = b[k] + e[n] * mu * x[n-k], for k=0, 1, ..., numTaps-1
+ * </pre>
+ * where <code>mu</code> is the step size and controls the rate of coefficient convergence.
+ *\par
+ * In the APIs, <code>pCoeffs</code> points to a coefficient array of size <code>numTaps</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>numTaps + blockSize - 1</code>.
+ * Samples in the state buffer are stored in the order:
+ * \par
+ * <pre>
+ * {x[n-numTaps+1], x[n-numTaps], x[n-numTaps-1], x[n-numTaps-2]....x[0], x[1], ..., x[blockSize-1]}
+ * </pre>
+ * \par
+ * Note that the length of the state buffer exceeds the length of the coefficient array by <code>blockSize-1</code> samples.
+ * The increased state buffer length allows circular addressing, which is traditionally used in FIR filters,
+ * to be avoided and yields a significant speed improvement.
+ * The state variables are updated after each block of data is processed.
+ * \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 and
+ * coefficient and state arrays cannot be shared among instances.
+ * 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.
+ * To do this manually without calling the init function, assign the follow subfields of the instance structure:
+ * numTaps, pCoeffs, mu, postShift (not for f32), 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.
+ * Set the values in the state buffer to zeros before static initialization.
+ * The code below statically initializes each of the 3 different data type filter instance structures
+ * <pre>
+ * arm_lms_instance_f32 S = {numTaps, pState, pCoeffs, mu};
+ * arm_lms_instance_q31 S = {numTaps, pState, pCoeffs, mu, postShift};
+ * arm_lms_instance_q15 S = {numTaps, pState, pCoeffs, mu, postShift};
+ * </pre>
+ * where <code>numTaps</code> is the number of filter coefficients in the filter; <code>pState</code> is the address of the state buffer;
+ * <code>pCoeffs</code> is the address of the coefficient buffer; <code>mu</code> is the step size parameter; and <code>postShift</code> is the shift applied to coefficients.
+ *
+ * \par Fixed-Point Behavior:
+ * Care must be taken when using the Q15 and Q31 versions of the LMS filter.
+ * The following issues must be considered:
+ * - Scaling of coefficients
+ * - Overflow and saturation
+ *
+ * \par Scaling of Coefficients:
+ * Filter coefficients are represented as fractional values and
+ * coefficients are restricted to lie in the range <code>[-1 +1)</code>.
+ * The fixed-point functions have an additional scaling parameter <code>postShift</code>.
+ * At the output of the filter's accumulator is a shift register which shifts the result by <code>postShift</code> bits.
+ * This essentially scales the filter coefficients by <code>2^postShift</code> and
+ * allows the filter coefficients to exceed the range <code>[+1 -1)</code>.
+ * The value of <code>postShift</code> is set by the user based on the expected gain through the system being modeled.
+ *
+ * \par Overflow and Saturation:
+ * Overflow and saturation behavior of the fixed-point Q15 and Q31 versions are
+ * described separately as part of the function specific documentation below.
+ */
+
+/**
+ * @addtogroup LMS
+ * @{
+ */
+
+/**
+ * @details
+ * This function operates on floating-point data types.
+ *
+ * @brief Processing function for floating-point LMS filter.
+ * @param[in] *S points to an instance of the floating-point LMS filter structure.
+ * @param[in] *pSrc points to the block of input data.
+ * @param[in] *pRef points to the block of reference data.
+ * @param[out] *pOut points to the block of output data.
+ * @param[out] *pErr points to the block of error data.
+ * @param[in] blockSize number of samples to process.
+ * @return none.
+ */
+
+void arm_lms_f32(
+ const arm_lms_instance_f32 * S,
+ float32_t * pSrc,
+ float32_t * pRef,
+ float32_t * pOut,
+ float32_t * pErr,
+ 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 *px, *pb; /* Temporary pointers for state and coefficient buffers */
+ float32_t mu = S->mu; /* Adaptive factor */
+ uint32_t numTaps = S->numTaps; /* Number of filter coefficients in the filter */
+ uint32_t tapCnt, blkCnt; /* Loop counters */
+ float32_t sum, e, d; /* accumulator, error, reference data sample */
+ float32_t w = 0.0f; /* weight factor */
+
+ e = 0.0f;
+ d = 0.0f;
+
+ /* S->pState points to state array which contains previous frame (numTaps - 1) samples */
+ /* pStateCurnt points to the location where the new input data should be written */
+ pStateCurnt = &(S->pState[(numTaps - 1U)]);
+
+ blkCnt = blockSize;
+
+
+#if defined (ARM_MATH_DSP)
+
+ /* Run the below code for Cortex-M4 and Cortex-M3 */
+
+ while (blkCnt > 0U)
+ {
+ /* Copy the new input sample into the state buffer */
+ *pStateCurnt++ = *pSrc++;
+
+ /* Initialize pState pointer */
+ px = pState;
+
+ /* Initialize coeff pointer */
+ pb = (pCoeffs);
+
+ /* Set the accumulator to zero */
+ sum = 0.0f;
+
+ /* Loop unrolling. Process 4 taps at a time. */
+ tapCnt = numTaps >> 2;
+
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ sum += (*px++) * (*pb++);
+ sum += (*px++) * (*pb++);
+ sum += (*px++) * (*pb++);
+ sum += (*px++) * (*pb++);
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* If the filter length is not a multiple of 4, compute the remaining filter taps */
+ tapCnt = numTaps % 0x4U;
+
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ sum += (*px++) * (*pb++);
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* The result in the accumulator, store in the destination buffer. */
+ *pOut++ = sum;
+
+ /* Compute and store error */
+ d = (float32_t) (*pRef++);
+ e = d - sum;
+ *pErr++ = e;
+
+ /* Calculation of Weighting factor for the updating filter coefficients */
+ w = e * mu;
+
+ /* Initialize pState pointer */
+ px = pState;
+
+ /* Initialize coeff pointer */
+ pb = (pCoeffs);
+
+ /* Loop unrolling. Process 4 taps at a time. */
+ tapCnt = numTaps >> 2;
+
+ /* Update filter coefficients */
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ *pb = *pb + (w * (*px++));
+ pb++;
+
+ *pb = *pb + (w * (*px++));
+ pb++;
+
+ *pb = *pb + (w * (*px++));
+ pb++;
+
+ *pb = *pb + (w * (*px++));
+ pb++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* If the filter length is not a multiple of 4, compute the remaining filter taps */
+ tapCnt = numTaps % 0x4U;
+
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ *pb = *pb + (w * (*px++));
+ pb++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* Advance state pointer by 1 for the next sample */
+ pState = pState + 1;
+
+ /* Decrement the loop counter */
+ blkCnt--;
+ }
+
+
+ /* Processing is complete. Now copy the last numTaps - 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 pState buffer */
+ pStateCurnt = S->pState;
+
+ /* Loop unrolling for (numTaps - 1U) samples copy */
+ tapCnt = (numTaps - 1U) >> 2U;
+
+ /* copy data */
+ while (tapCnt > 0U)
+ {
+ *pStateCurnt++ = *pState++;
+ *pStateCurnt++ = *pState++;
+ *pStateCurnt++ = *pState++;
+ *pStateCurnt++ = *pState++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* Calculate remaining number of copies */
+ tapCnt = (numTaps - 1U) % 0x4U;
+
+ /* Copy the remaining q31_t data */
+ while (tapCnt > 0U)
+ {
+ *pStateCurnt++ = *pState++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+#else
+
+ /* Run the below code for Cortex-M0 */
+
+ while (blkCnt > 0U)
+ {
+ /* Copy the new input sample into the state buffer */
+ *pStateCurnt++ = *pSrc++;
+
+ /* Initialize pState pointer */
+ px = pState;
+
+ /* Initialize pCoeffs pointer */
+ pb = pCoeffs;
+
+ /* Set the accumulator to zero */
+ sum = 0.0f;
+
+ /* Loop over numTaps number of values */
+ tapCnt = numTaps;
+
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ sum += (*px++) * (*pb++);
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* The result is stored in the destination buffer. */
+ *pOut++ = sum;
+
+ /* Compute and store error */
+ d = (float32_t) (*pRef++);
+ e = d - sum;
+ *pErr++ = e;
+
+ /* Weighting factor for the LMS version */
+ w = e * mu;
+
+ /* Initialize pState pointer */
+ px = pState;
+
+ /* Initialize pCoeffs pointer */
+ pb = pCoeffs;
+
+ /* Loop over numTaps number of values */
+ tapCnt = numTaps;
+
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ *pb = *pb + (w * (*px++));
+ pb++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ /* Advance state pointer by 1 for the next sample */
+ pState = pState + 1;
+
+ /* Decrement the loop counter */
+ blkCnt--;
+ }
+
+
+ /* Processing is complete. Now copy the last numTaps - 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 pState buffer */
+ pStateCurnt = S->pState;
+
+ /* Copy (numTaps - 1U) samples */
+ tapCnt = (numTaps - 1U);
+
+ /* Copy the data */
+ while (tapCnt > 0U)
+ {
+ *pStateCurnt++ = *pState++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+#endif /* #if defined (ARM_MATH_DSP) */
+
+}
+
+/**
+ * @} end of LMS group
+ */