summaryrefslogtreecommitdiff
path: root/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_lms_norm_f32.c
diff options
context:
space:
mode:
Diffstat (limited to 'fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_lms_norm_f32.c')
-rw-r--r--fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_lms_norm_f32.c454
1 files changed, 454 insertions, 0 deletions
diff --git a/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_lms_norm_f32.c b/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_lms_norm_f32.c
new file mode 100644
index 0000000..a365b33
--- /dev/null
+++ b/fw/midi-dials/Drivers/CMSIS/DSP/Source/FilteringFunctions/arm_lms_norm_f32.c
@@ -0,0 +1,454 @@
+/* ----------------------------------------------------------------------
+ * Project: CMSIS DSP Library
+ * Title: arm_lms_norm_f32.c
+ * Description: Processing function for the floating-point Normalised LMS
+ *
+ * $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_NORM Normalized LMS Filters
+ *
+ * This set of functions implements a commonly used adaptive filter.
+ * It is related to the Least Mean Square (LMS) adaptive filter and includes an additional normalization
+ * factor which increases the adaptation rate of the filter.
+ * The CMSIS DSP Library contains normalized LMS filter functions that operate on Q15, Q31, and floating-point data types.
+ *
+ * A normalized least mean square (NLMS) 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 NLMS 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 NLMS 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 NLMS adaptive 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 instanteous energy of the filter state variables is calculated:
+ * <pre>
+ * E = x[n]^2 + x[n-1]^2 + ... + x[n-numTaps+1]^2.
+ * </pre>
+ * The filter coefficients <code>b[k]</code> are then updated on a sample-by-sample basis:
+ * <pre>
+ * b[k] = b[k] + e[n] * (mu/E) * 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, energy, x0, pState. Also set all of the values in pState to zero.
+ * For Q7, Q15, and Q31 the following fields must also be initialized;
+ * recipTable, postShift
+ *
+ * \par
+ * Instance structure cannot be placed into a const data section and it is recommended to use the initialization function.
+ * \par Fixed-Point Behavior:
+ * Care must be taken when using the Q15 and Q31 versions of the normalised 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_NORM
+ * @{
+ */
+
+
+ /**
+ * @brief Processing function for floating-point normalized LMS filter.
+ * @param[in] *S points to an instance of the floating-point normalized 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_norm_f32(
+ arm_lms_norm_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 energy; /* Energy of the input */
+ float32_t sum, e, d; /* accumulator, error, reference data sample */
+ float32_t w, x0, in; /* weight factor, temporary variable to hold input sample and state */
+
+ /* Initializations of error, difference, Coefficient update */
+ e = 0.0f;
+ d = 0.0f;
+ w = 0.0f;
+
+ energy = S->energy;
+ x0 = S->x0;
+
+ /* S->pState points to buffer 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)]);
+
+ /* Loop over blockSize number of values */
+ 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);
+
+ /* Read the sample from input buffer */
+ in = *pSrc++;
+
+ /* Update the energy calculation */
+ energy -= x0 * x0;
+ energy += in * in;
+
+ /* 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 updating filter coefficients */
+ /* epsilon value 0.000000119209289f */
+ w = (e * mu) / (energy + 0.000000119209289f);
+
+ /* 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 += w * (*px++);
+ pb++;
+
+ *pb += w * (*px++);
+ pb++;
+
+ *pb += w * (*px++);
+ 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 += w * (*px++);
+ pb++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ x0 = *pState;
+
+ /* Advance state pointer by 1 for the next sample */
+ pState = pState + 1;
+
+ /* Decrement the loop counter */
+ blkCnt--;
+ }
+
+ S->energy = energy;
+ S->x0 = x0;
+
+ /* 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)/4 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;
+
+ /* Read the sample from input buffer */
+ in = *pSrc++;
+
+ /* Update the energy calculation */
+ energy -= x0 * x0;
+ energy += in * in;
+
+ /* 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 in the accumulator is stored in the destination buffer. */
+ *pOut++ = sum;
+
+ /* Compute and store error */
+ d = (float32_t) (*pRef++);
+ e = d - sum;
+ *pErr++ = e;
+
+ /* Calculation of Weighting factor for updating filter coefficients */
+ /* epsilon value 0.000000119209289f */
+ w = (e * mu) / (energy + 0.000000119209289f);
+
+ /* Initialize pState pointer */
+ px = pState;
+
+ /* Initialize pCcoeffs pointer */
+ pb = pCoeffs;
+
+ /* Loop over numTaps number of values */
+ tapCnt = numTaps;
+
+ while (tapCnt > 0U)
+ {
+ /* Perform the multiply-accumulate */
+ *pb += w * (*px++);
+ pb++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+ x0 = *pState;
+
+ /* Advance state pointer by 1 for the next sample */
+ pState = pState + 1;
+
+ /* Decrement the loop counter */
+ blkCnt--;
+ }
+
+ S->energy = energy;
+ S->x0 = x0;
+
+ /* 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;
+
+ /* Copy (numTaps - 1U) samples */
+ tapCnt = (numTaps - 1U);
+
+ /* Copy the remaining q31_t data */
+ while (tapCnt > 0U)
+ {
+ *pStateCurnt++ = *pState++;
+
+ /* Decrement the loop counter */
+ tapCnt--;
+ }
+
+#endif /* #if defined (ARM_MATH_DSP) */
+
+}
+
+/**
+ * @} end of LMS_NORM group
+ */