From 8ac9b2f5365cfc68eec7e51616939d20fed1edd7 Mon Sep 17 00:00:00 2001 From: boblark Date: Wed, 9 Mar 2022 12:00:36 -0800 Subject: [PATCH] Corrected comments, no code change --- AudioFilterBiquad_F32.h | 14 +++++++------- AudioFilterFIRGeneral_F32.h | 22 +++++++++++----------- 2 files changed, 18 insertions(+), 18 deletions(-) diff --git a/AudioFilterBiquad_F32.h b/AudioFilterBiquad_F32.h index 5d86346..9e15a02 100644 --- a/AudioFilterBiquad_F32.h +++ b/AudioFilterBiquad_F32.h @@ -1,5 +1,5 @@ /* - * AudioFilterBiquad_F32.cpp + * AudioFilterBiquad_F32.h * Chip Audette, OpenAudio, Apr 2017 * MIT License, Use at your own risk. * @@ -20,11 +20,11 @@ * begin of the ARM CMSIS. We can't do that with multiple stages. If you * encouter this, add myBiquad.begin(); to your INO after the * coefficients have been set. Feb 2021 - * + * * The sign of the coefficients for feedback, the a[], here use the * convention of the ARM CMSIS library. Matlab reverses the signs of these. * I believe these are treated per those rules!! Bob - * + * * Algorithm for CMSIS library * Each Biquad stage implements a second order filter using the difference equation: * y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] + a1 * y[n-1] + a2 * y[n-2] @@ -100,11 +100,11 @@ class AudioFilterBiquad_F32 : public AudioStream_F32 } // ARM DSP Math library filter instance. - // Does the initialization of ARM CMSIS DSP BiQuad structure. This MUST follow the - // setting of coefficients to catch the max number of stages and do the + // Does the initialization of ARM CMSIS DSP BiQuad structure. This MUST follow the + // setting of coefficients to catch the max number of stages and do the // double to float conversion for the CMSIS routine. void begin(void) { - // Initialize FIR instance (ARM DSP Math Library) + // Initialize BiQuad instance (ARM DSP Math Library) //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html arm_biquad_cascade_df1_init_f32(&iir_inst, numStagesUsed, &coeff32[0], &StateF32[0]); } @@ -115,7 +115,7 @@ class AudioFilterBiquad_F32 : public AudioStream_F32 void setSampleRate_Hz(float _fs_Hz) { sampleRate_Hz = _fs_Hz; } - // Deprecated + // Deprecated void setBlockDC(void) { // https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5 // Use matlab to compute the coeff for HP at 40Hz: [b,a]=butter(2,40/(44100/2),'high'); %assumes fs_Hz = 44100 diff --git a/AudioFilterFIRGeneral_F32.h b/AudioFilterFIRGeneral_F32.h index b49879f..89d6000 100644 --- a/AudioFilterFIRGeneral_F32.h +++ b/AudioFilterFIRGeneral_F32.h @@ -9,7 +9,7 @@ * for the library structures and wonderful Teensy products. * * There are enough different FIR filter Audio blocks to need a summary. Here goes: - * + * * AudioFilterFIR (Teensy Audio Library by PJRC) handles 16-bit integer data, * and a maximum of 200 FIR coefficients, even only. (taps). For Teensy Audio. * AudioFilterFIR_F32 (OpenAudio_ArduinoLibrary by Chip Audette) handles 32-bit floating point @@ -32,21 +32,21 @@ * FIR filters suffer from needing considerable computation of the multiply-and-add * sort. This limits the number of taps that can be used, but less so as time goes by. * In particular, the Teensy 4.x, if it *did nothing but* FIR calculations, could - * use about 6000 taps inmonaural, which is a huge number. But, this also + * use about 6000 taps inmonaural, which is a huge number. But, this also * suggests that if the filtering task is an important function of a project, * using, say 2000 taps is practical. - * + * * FIR filters can be (and are here) implemented to have symmetrical coefficients. This * results in constant delay at all frequencies (linear phase). For some applications this can * be an important feature. Sometimes it is suggested that the FIR should not be * used because of the latency it creates. Note that if constant delay is needed, the FIR * implementation does this with minimum latency. - * + * * For this block, AudioFilterFIRGeneral_F32, memory storage for the FIR * coefficiients as well as working storage for the ARM FIR routine is provided * by the calling .INO. This allows large FIR sizes without always tying up a * big memory block. - * + * * This block normally calculates the FIR coefficients using a Fourier transform * of the desired amplitude response and a Kaiser window. This flexability requires * the calling .INO to provide an array of response levels, in relative dB, @@ -57,17 +57,17 @@ * else dbA[i] = -140.0f; * } * firg1.FIRGeneralNew(&dbA[0], 300, &equalizeCoeffs[0], 50.0f, &workSpace[0]); - * + * * As an alternate to inputting the response function, the FIR coefficients can be * entered directly using LoadCoeffs(nFIR, cf32f, *pStateArray). This is a very quick * operation as only pointers to coefficients are involved. Several filters can be * stored in arrays and switched quickly this way. If this is done, pStateArray[] * as initially setup should be large enough for all filters. There will be "clicks" - * associated with filter changes and these may need to be muted. + * associated with filter changes and these may need to be muted. * * How well the desired response is achieved depends on the number of FIR coefficients * being used. As noted above, for some applications it may be desired to use - * large numbers of taps. The achieved response can be evaluated + * large numbers of taps. The achieved response can be evaluated * by the function getResponse(nPoints, pResponse) which fills an .INO-supplied array * pResponse[nPoints] with the frequency response of the equalizer in dB. The nPoints * are spread evenly between 0.0 and half of the sample frequency. @@ -99,11 +99,11 @@ * frequency response. * LoadCoeffs(nFIR, cf32f, *pStateArray); // To directly load FIR coefficients cf32f[] * getResponse(nFreq, *rdb); // To obtain the amplitude response in dB, rdb[] - * + * * Status: Tested T3.6, T4.0 No known bugs - * + * * Examples: TestFIRGeneralLarge4.ino TestFIRGeneralLarge5.ino - * + * * Copyright (c) 2020 Bob Larkin * Any snippets of code from PJRC or Chip Audette used here brings with it * the associated license.