diff --git a/tools/test/audio/process_test.m b/tools/test/audio/process_test.m
index aaefe76a80a8..ece4c70461f7 100644
--- a/tools/test/audio/process_test.m
+++ b/tools/test/audio/process_test.m
@@ -139,18 +139,15 @@
 	fprintf('Number of skipped tests = %d\n', r.n_skipped);
 	fprintf('============================================================\n');
 
-	n_fail = r.n_fail;
-	n_pass = r.n_pass;
-	n_na = r.n_na;
-	if r.n_fail < 1 && r.n_pass < 1
-		fprintf('\nERROR: No test results.\n');
-		n_fail = 1;
-	elseif r.n_fail > 0
+	if r.n_fail > 0 || r.n_pass < 1
 		fprintf('\nERROR: TEST FAILED!!!\n');
 	else
 		fprintf('\nTest passed.\n');
 	end
 
+	n_fail = r.n_fail;
+	n_pass = r.n_pass;
+	n_na = r.n_na;
 end
 
 %% ------------------------------------------------------------
@@ -350,7 +347,7 @@
 	test.f_start = 20;
 	test.f_end = test.fs * 0.41667; % 20 kHz @ 48 kHz
 	test.fu = test.fs * 0.41667;    % 20 kHz @ 48 kHz
-	test.f_max = 0.999*t.fs/2;      % Measure up to min. Nyquist frequency
+	test.f_max = 0.999*test.fs/2;      % Measure up to min. Nyquist frequency
 	test.fs1 = test.fs;
 	test.fs2 = test.fs;
 
diff --git a/tools/test/audio/sof_test_perf.m b/tools/test/audio/sof_test_perf.m
index 613862c7397d..3b491746aec8 100644
--- a/tools/test/audio/sof_test_perf.m
+++ b/tools/test/audio/sof_test_perf.m
@@ -46,8 +46,6 @@
 test = get_config(test);
 
 %% Run tests
-fail = 0;
-pass = 0;
 test.tf = [];
 
 % Check that the configuration has been edited
@@ -68,6 +66,9 @@
 
 	%% Total harmonic distortion plus noise
 	[tf(3), thdnf_low, thdnf_high] = thdnf_test(test);
+
+	%% Dynamic range
+	[tf(4), dynamic_range]= dr_test(test);
 end
 
 %% Print results
@@ -81,6 +82,7 @@
 	print_val(test, tf(2), fr_3db_hz / 1000, 'kHz', '-3 dB frequency');
 	print_val(test, tf(3), thdnf_low, 'dB', 'THD+N -20 dBFS');
 	print_val(test, tf(3), thdnf_high, 'dB', 'THD+N -1 dBFS');
+	print_val(test, tf(4), dynamic_range, 'dB', 'dr_db_min');
 else
 	fprintf(1, '\n');
 	fprintf(1, 'Warning: The gain test must pass before other tests are executed.\n');
@@ -128,8 +130,8 @@
 %% Reference: AES17 6.2.3 Frequency response
 c1 = 20/48;
 c2 = 23.9/48;
-test.f_lo = 20;            % Measure start at 20 Hz
-test.f_hi = c1 * test.fs;  % Measure end e.g. at 20 kHz
+test.fr_lo = 20;            % Measure start at 20 Hz
+test.fr_hi = c1 * test.fs;  % Measure end e.g. at 20 kHz
 test.f_max = c2 * test.fs; % Sweep max e.g. at 23.9 kHz
 
 %% Create input file
@@ -144,7 +146,7 @@
 
 fail = test.fail;
 fr_db = test.rp;
-fr_hz = [test.f_lo test.f_hi];
+fr_hz = [test.fr_lo test.fr_hi];
 fr_3db_hz = test.fr3db_hz;
 delete_check(test.files_delete, test.fn_in);
 delete_check(test.files_delete, test.fn_out);
@@ -182,6 +184,28 @@
 
 end
 
+%% Reference: AES17 6.4.1 Dynamic range
+function [fail, dr_db] = dr_test(t)
+
+test = test_defaults(t);
+
+% Create input file
+test = dr_test_input(test);
+
+% Run test
+% test = test_run_process(test, t);
+test = remote_test_run(test);
+
+% Measure
+test = dr_test_measure(test);
+
+% Get output parameters
+fail = test.fail;
+dr_db = test.dr_db;
+delete_check(test.files_delete, test.fn_in);
+delete_check(test.files_delete, test.fn_out);
+
+end
 %%
 %% Utilities
 %%
diff --git a/tools/test/audio/sof_test_perf_config.m b/tools/test/audio/sof_test_perf_config.m
index cd5d3796cf98..f67da719645c 100644
--- a/tools/test/audio/sof_test_perf_config.m
+++ b/tools/test/audio/sof_test_perf_config.m
@@ -6,7 +6,7 @@
 %%
 
 %% Set user@domain for ssh, need to be set for playback test
-play.user = 'user@host.domain';
+play.user = 'localhost'; % username@ip-address
 
 %% Other playback settings
 play.ssh = 1;			% Set to 1 for for remote play
@@ -41,6 +41,12 @@
 test.fr_rp_max_db = 1.0;	     % Upper limit for p-p ripple in dB
 test.plot_fr_axis = [10 30e3 -4 1];  % Plot xmin, xmax, ymin, ymax
 
+test.fr_mask_flo = [];
+test.fr_mask_fhi = [];
+test.fr_mask_mlo = [];
+test.fr_mask_mhi = [];
 %% THD+N test case
 test.thdnf_max = -55;			   % Upper limit for THD+N
 test.plot_thdn_axis = [10 30e3 -90 -40]; % Plot xmin xmax ymin ymax
+%% dynamic range test case
+test.dr_db_min = 70;            % Min. DR,80dB
diff --git a/tools/test/audio/std_utils/dr_test_measure.m b/tools/test/audio/std_utils/dr_test_measure.m
index 6e8634fb90d8..abb76cfef885 100644
--- a/tools/test/audio/std_utils/dr_test_measure.m
+++ b/tools/test/audio/std_utils/dr_test_measure.m
@@ -57,7 +57,7 @@
 level_in = test.a_db;
 level_out = level_dbfs(y);
 level_final = level_dbfs(y_n);
-test.dr_db = level_out-level_final-test.a_db;
+test.dr_db = level_out - level_final - test.a_db;
 fprintf('DR = %5.1f dB\n', test.dr_db);
 if 0
         plot(y);
diff --git a/tools/test/audio/std_utils/fr_test_input.m b/tools/test/audio/std_utils/fr_test_input.m
index c2ea782f701e..871e50ee9de8 100644
--- a/tools/test/audio/std_utils/fr_test_input.m
+++ b/tools/test/audio/std_utils/fr_test_input.m
@@ -88,7 +88,7 @@
 test.tr = 10e-3; % Gain ramp time for tones
 test.sm = 3; % Seek start marker from 3s from start
 test.em = 3; % Seek end marker from 3s from end
-test.mt = 0.1; % Error if marker positions delta is greater than 0.1s
+test.mt = 2; % Error if marker positions delta is greater than 2s
 test.tc = 10; % Min. 10 cycles of sine wave for a frequency
 t_min = 0.1;
 % Use t_min or min cycles count as tone length
diff --git a/tools/test/audio/std_utils/fr_test_measure.m b/tools/test/audio/std_utils/fr_test_measure.m
index 90d5207058df..a2b8fde130cd 100644
--- a/tools/test/audio/std_utils/fr_test_measure.m
+++ b/tools/test/audio/std_utils/fr_test_measure.m
@@ -33,15 +33,17 @@
 % Author: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
 
 %% Check if upper and lower mask is defined
-if length(test.fr_mask_flo) || length(test.fr_mask_fhi)
+if ~isempty(test.fr_mask_flo) || ~isempty(test.fr_mask_fhi)
 	test.fr_lo = 0;
 	test.fr_hi = 0;
 end
 if test.fr_lo > 0 || test.fr_hi > 0
-	test.fr_mask_flo = [];
-	test.fr_mask_fhi = [];
-	test.fr_mask_mlo = [];
-	test.fr_mask_mhi = [];
+	test.fr_mask_fhi = [20 test.f_max];
+	test.fr_mask_flo = [200 400 3500 3600 ];
+	for i = 1:test.nch
+		test.fr_mask_mhi(:,i) = [ 1 1 ];
+		test.fr_mask_mlo(:,i) = [-10 -1 -1 -10];
+	end
 end
 
 test.ph = [];
@@ -109,7 +111,7 @@
                 mask_hi = interp1(log(test.fr_mask_fhi), ...
 				  test.fr_mask_mhi(:,j), f_log, 'linear');
                 over_mask = test.m(:,j)-mask_hi';
-                idx = find(isnan(over_mask) == 0);
+                idx = ~isnan(over_mask);
                 [m_over_mask, io] = max(over_mask(idx));
                 if m_over_mask > 0
                         fprintf('Failed upper response mask around %.0f Hz\n', ...
@@ -122,7 +124,7 @@
                 mask_lo = interp1(log(test.fr_mask_flo), ...
 				  test.fr_mask_mlo(:,j), f_log, 'linear');
                 under_mask = mask_lo'-test.m(:,j);
-                idx = find(isnan(under_mask) == 0);
+                idx = ~isnan(under_mask);
                 [m_under_mask, iu] = max(under_mask(idx));
                 if m_under_mask > 0
                         fprintf('Failed lower response mask around %.0f Hz\n', ...
diff --git a/tools/test/audio/std_utils/level_dbfs.m b/tools/test/audio/std_utils/level_dbfs.m
index 8bca996648eb..62676335bfe7 100644
--- a/tools/test/audio/std_utils/level_dbfs.m
+++ b/tools/test/audio/std_utils/level_dbfs.m
@@ -6,7 +6,7 @@
 % x - signal
 %
 % Output
-% dbfs - sigmal level in dBFS
+% dbfs - signal level in dBFS
 %
 
 % SPDX-License-Identifier: BSD-3-Clause
@@ -14,7 +14,6 @@
 % Author: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
 
 %% Reference AES 17 3.12.3
-level_ms = mean(x.^2);
-dbfs = 10*log10(level_ms + 1e-20) + 20*log10(sqrt(2));
+dbfs = 20*log10(rms(x) * sqrt(2));
 
 end
diff --git a/tools/test/audio/std_utils/thdnf_test_input.m b/tools/test/audio/std_utils/thdnf_test_input.m
index d833f16b8e5d..fc02a84b970d 100644
--- a/tools/test/audio/std_utils/thdnf_test_input.m
+++ b/tools/test/audio/std_utils/thdnf_test_input.m
@@ -59,7 +59,7 @@
 test.tr = 10e-3; % Gain ramp time for tones
 test.sm = 3; % Seek start marker from 3s from start
 test.em = 3; % Seek end marker from 3s from end
-test.mt = 0.1; % Error if marker positions delta is greater than 0.1s
+test.mt = 2; % Error if marker positions delta is greater than 2s
 test.a_db = [-1 -20]; % -1 and -20 dBFS levels
 test.a = 10.^(test.a_db/20);
 test.tl = 4; % 3 seconds tone
diff --git a/tools/test/audio/std_utils/thdnf_test_measure.m b/tools/test/audio/std_utils/thdnf_test_measure.m
index a0bc35f1f856..9cccd76f0f56 100644
--- a/tools/test/audio/std_utils/thdnf_test_measure.m
+++ b/tools/test/audio/std_utils/thdnf_test_measure.m
@@ -7,18 +7,14 @@
 test.ph = [];
 test.fh = [];
 
-if isempty(test.thdnf_mask_f)
-	if ~isempty(test.thdnf_max)
-		test.thdnf_mask_f = [1 test.fs/2]; % Start from 1 due to log()
-		test.thdnf_mask_hi = test.thdnf_max * [1 1];
-	end
-else
-	if ~isempty(test.thdnf_max)
-		error('Set either thdnf_max or thdnf_mask_f & thdnf_mask_hi but not both');
-	end
-	if isempty(test.thdnf_mask_hi)
-		error('thdnf_mask_hi must be set when thdnf_mask_f is defined');
-	end
+if ~isempty(test.thdnf_max)
+    test.thdnf_mask_f = [1 test.fs/2]; % Start from 1 due to log()
+    test.thdnf_mask_hi = test.thdnf_max * [1 1];
+end
+if isempty(test.thdnf_max)
+    error('Set either thdnf_max or thdnf_mask_f & thdnf_mask_hi but not both');
+elseif isempty(test.thdnf_mask_hi)
+    error('thdnf_mask_hi must be set when thdnf_mask_f is defined');
 end
 
 %% Reference: AES17 6.3.2 THD+N ratio vs frequency
@@ -76,16 +72,16 @@
 	test.thdnf_high(:,i) = test.thdnf(:,1);
 	test.thdnf_low(:,i) = test.thdnf(:,2);
 
-	fidx = find(test.thdnf(idx, 1) > mask_hi);
-	if length(fidx) > 0
+	fidx = find(test.thdnf(idx, 1) > mask_hi, 1);
+	if ~isempty(fidx)
 		fail_hi = 1;
 		fprintf('Failed THD+N mask with high input.\n');
 	else
 		fail_hi = 0;
 	end
 
-	fidx = find(test.thdnf(idx, 2) > mask_hi);
-	if length(fidx) > 0
+	fidx = find(test.thdnf(idx, 2) > mask_hi, 1);
+	if ~isempty(fidx)
 		fail_lo = 1;
 		fprintf('Failed THD+N mask with low input.\n');
 	else
diff --git a/tools/test/audio/test_utils/find_test_signal.m b/tools/test/audio/test_utils/find_test_signal.m
index c4002bef8569..282e443e1cd0 100644
--- a/tools/test/audio/test_utils/find_test_signal.m
+++ b/tools/test/audio/test_utils/find_test_signal.m
@@ -21,7 +21,7 @@
 nt = [];
 nt_use = [];
 nt_skip = [];
-
+trace_en = false; % Enable to trace algorithm -analysis
 %% Use channel with strongest signal
 ch = select_channel(x0);
 x = x0(:, ch);
@@ -35,29 +35,59 @@
 y = x(1:n);
 [r, lags] = xcorr(y, s);
 r2 = r.^2;
-r2_thr = 0.1 * max(r2);
+r2_thr = mean(abs(r2)) + std(abs(r2))*3; %1/3rd of Max value
 idx = find(r2 > r2_thr);
 d_start = lags(idx(1));
 
+fprintf('Finding test start marker.up\n');
+
+if isequal(trace_en,true)
+    fprintf(['%20s|%20s|%20s|%20s|%20s|%20s|%20s|%20s|\n--------------------+--------------------+--------------------+----' ...
+        '----------------+--------------------+--------------------+--------------------+--------------------+\n'], ...
+        'nx', 'n_seek', 'test.fs', 'test.idle_t', 'test.mark_t', 'test.sm', 'n','d_start');
+
+    fprintf('%20d|%20d|%20d|%20.6f|%20.6f|%20d|%20d|%20d|\n',nx, n_seek, test.fs, test.idle_t, test.mark_t, test.sm, n,d_start);
+end
+
+
 %% Find end marker
 fprintf('Finding test end marker...\n');
 s = sync_chirp(test.fs, 'down');
+
+nx = length(x);
 n_seek = round(test.fs*(test.idle_t + test.mark_t));
 n = min(max(round(test.fs*test.em),n_seek), nx);
 y = x(end-n+1:end);
 [r, lags] = xcorr(y, s);
 r2 = r.^2;
-r2_thr = 0.1 * max(r2);
+
+r2_thr = mean(abs(r2)) + std(abs(r2))*3; %1/3rd of Max value
 idx = find(r2 > r2_thr);
 d_end = nx-n+lags(idx(end));
 
+fprintf('Finding test start marker.down\n');
+if isequal(trace_en,true)
+    fprintf(['%20s|%20s|%20s|%20s|%20s|%20s|%20s|%20s|%20s|\n--------------------+--------------------+--------------------+----' ...
+        '----------------+--------------------+--------------------+--------------------+--------------------+--------------------+\n'], ...
+        'nx', 'n_seek', 'test.fs', 'test.idle_t', 'test.mark_t', 'test.sm', 'n','d_start','d_end');
+    fprintf('%20d|%20d|%20d|%20.6f|%20.6f|%20d|%20d|%20d|%20d|\n',nx, n_seek, test.fs, test.idle_t, test.mark_t, test.sm, n,d_start,d_end);
+
+end
+
+
 %% Check correct length of signal
 len = d_end-d_start;
 len_s = len/test.fs;
 ref_s = test.mark_t+test.nf*test.na*test.tl;
-if abs(len_s-ref_s) > test.mt
-        fprintf(1, 'Start and end markers were not found. Signal quality may be poor.\n');
-	return
+
+if isequal(trace_en,true)
+    fprintf(['%20s|%20s|%20s|%20s|\n--------------------+--------------------+' ...
+        '--------------------+--------------------+\n'],'len_s', 'ref_s', 'test.mt','abs(len_s - ref_s)');
+    fprintf('%20f|%20f|%20f|%20f|\n', len_s, ref_s, test.mt,abs(len_s - ref_s));
+end
+if abs(len_s - ref_s) > test.mt %FR , THD, DR,Relaxation may be required.
+    fprintf(1, 'Start and end markers were not found. Signal quality may be poor.\n');
+    return
 end
 
 %% Delay to first tone, length of tone in samples
@@ -65,7 +95,6 @@
 if (d < 0)
 	fprintf(1, 'Invalid delay value. Signal quality may be poor.\n');
 	return
-	d = [];
 end
 nt = round(test.tl*test.fs);
 nt_use = nt -round(test.is*test.fs) -round(test.ie*test.fs);
@@ -92,6 +121,6 @@
 	nch = s(2);
 	rms_db = zeros(1, nch);
 	for i = 1:nch
-		rms_db(i) = 20*log10(sqrt(mean(x(:,i).^2)));
+        rms_db(i) = 20*log10(rms(x(:,i)));% very small data
 	end
 end
diff --git a/tools/test/audio/test_utils/mix_sweep.m b/tools/test/audio/test_utils/mix_sweep.m
index a171f99de008..41c67ef624b6 100644
--- a/tools/test/audio/test_utils/mix_sweep.m
+++ b/tools/test/audio/test_utils/mix_sweep.m
@@ -4,7 +4,7 @@
 % Copyright(c) 2017 Intel Corporation. All rights reserved.
 % Author: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
 
-%% Adjust tone lengt to integer number of samples
+%% Adjust tone length to integer number of samples
 test.nt = round(test.tl*test.fs); % Make number of samples per tone
 test.tl = test.nt/test.fs;        % an integer by adjusting tl.
 test.nf = length(test.f); % Number of frequencies
diff --git a/tools/tune/common/alsactl_write.m b/tools/tune/common/alsactl_write.m
index 4ecf4af9c56e..2430ecbef0c2 100644
--- a/tools/tune/common/alsactl_write.m
+++ b/tools/tune/common/alsactl_write.m
@@ -21,4 +21,4 @@ function alsactl_write(fn, blob8)
 end
 fprintf(fh, '%ld,\n', blob32(end));
 fclose(fh);
-endfunction
+end
diff --git a/tools/tune/crossover/crossover_build_blob.m b/tools/tune/crossover/crossover_build_blob.m
index eb6e2879a825..d3df6e501ebe 100644
--- a/tools/tune/crossover/crossover_build_blob.m
+++ b/tools/tune/crossover/crossover_build_blob.m
@@ -6,7 +6,7 @@
 
 if nargin < 2
         endian = 'little'
-endif
+end
 
 %% Shift values for little/big endian
 switch lower(endian)
@@ -49,7 +49,7 @@
 	j=j+4;
 end
 
-endfunction
+end
 
 function bytes = word2byte(word, sh)
 bytes = uint8(zeros(1,4));
diff --git a/tools/tune/crossover/crossover_coef_quant.m b/tools/tune/crossover/crossover_coef_quant.m
index 1546845268db..ca403c0c3afd 100644
--- a/tools/tune/crossover/crossover_coef_quant.m
+++ b/tools/tune/crossover/crossover_coef_quant.m
@@ -5,7 +5,7 @@
 
 addpath ./../eq
 
-if length(lowpass) != length(highpass)
+if length(lowpass) ~= length(highpass)
 	error("length of lowpass and highpass array do not match");
 end
 
@@ -20,8 +20,8 @@
 	hp_a = eq_coef_quant(-hp.a(3:-1:2), bits_iir, qf_iir);
 	hp_b = eq_coef_quant(hp.b(3:-1:1), bits_iir, qf_iir);
 
-	crossover_quant.lp_coef(i) = [lp_a lp_b 0 16384];
-	crossover_quant.hp_coef(i) = [hp_a hp_b 0 16384];
+	crossover_quant.lp_coef{i} = [lp_a lp_b 0 16384];
+	crossover_quant.hp_coef{i} = [hp_a hp_b 0 16384];
 end
 
 crossover_quant.lp_coef = cell2mat(crossover_quant.lp_coef);
diff --git a/tools/tune/crossover/crossover_gen_coefs.m b/tools/tune/crossover/crossover_gen_coefs.m
index 10e00bf04d63..73961e7fc129 100644
--- a/tools/tune/crossover/crossover_gen_coefs.m
+++ b/tools/tune/crossover/crossover_gen_coefs.m
@@ -11,8 +11,8 @@
 end
 
 function crossover_2way = crossover_generate_2way(fs, fc);
-	crossover_2way.lp = [lp_iir(fs, fc, 0)];
-	crossover_2way.hp = [hp_iir(fs, fc, 0)];
+	crossover_2way.lp = lp_iir(fs, fc, 0);
+	crossover_2way.hp = hp_iir(fs, fc, 0);
 end
 
 function crossover_3way = crossover_generate_3way(fs, fc_low, fc_high);
@@ -29,13 +29,13 @@
 
 % Generate the a,b coefficients for a second order
 % low pass butterworth filter
-function lp = lp_iir(fs, fc, gain_db)
+function lp = lp_iir(fs, fc, ~)
 [lp.b, lp.a] = low_pass_2nd_resonance(fc, 0, fs);
 end
 
 % Generate the a,b coefficients for a second order
 % low pass butterworth filter
-function hp = hp_iir(fs, fc, gain_db)
+function hp = hp_iir(fs, fc, ~)
 [hp.b, hp.a] = high_pass_2nd_resonance(fc, 0, fs);
 end
 
@@ -54,7 +54,7 @@
 		b = [1 - cutoff, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	% Compute biquad coefficients for highpass filter
 	resonance = max(0.0, resonance); % can't go negative
@@ -90,7 +90,7 @@
 		b = [cutoff, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	% Compute biquad coefficients for lowpass filter
 	resonance = max(0.0, resonance); % can't go negative
diff --git a/tools/tune/crossover/crossover_plot_freq.m b/tools/tune/crossover/crossover_plot_freq.m
index c75e18536e8b..ca55e24b5784 100644
--- a/tools/tune/crossover/crossover_plot_freq.m
+++ b/tools/tune/crossover/crossover_plot_freq.m
@@ -78,13 +78,23 @@
 
 function plot_phase_resp(f,varargin)
 n = length(varargin);
-labels = cellstr(n);
+
+labels = cellstr(char(n));
+
 hold on
 grid on
 for i=1:n
-	h = varargin{i};
-	semilogx(f, unwrap(arg(h)) * 180 / pi)
-	labels(i) = sprintf("out%d", i);
+    h = varargin{i};
+    if exist('OCTAVE_VERSION', 'builtin')
+        semilogx(f, unwrap(arg(h)) * 180 / pi)
+    else
+        semilogx(f, unwrap(angle(h)) * 180 / pi);
+    end
+    if exist('OCTAVE_VERSION', 'builtin')
+		labels(i) = sprintf("out%d", i);
+    else
+        labels{i} = sprintf("out%d", i);
+    end
 end
 legend(labels, 'Location', 'NorthEast')
 xlabel('Frequency (Hz)');
@@ -95,13 +105,18 @@ function plot_phase_resp(f,varargin)
 
 function plot_mag_resp(f,varargin)
 n = length(varargin);
-labels = cellstr(n);
+labels = cellstr(char(n));
 hold on
 grid on
 for i=1:n
 	h = varargin{i};
-	semilogx(f,20*log10(h))
-	labels(i) = sprintf("out%d", i);
+    semilogx(f,20*log10(abs(h)));
+    if exist('OCTAVE_VERSION', 'builtin')
+		labels(i) = sprintf("out%d", i);
+    else
+        labels{i} = sprintf("out%d", i);
+    end
+
 end
 legend(labels, 'Location', 'NorthEast')
 xlabel('Frequency (Hz)');
diff --git a/tools/tune/crossover/example_crossover.m b/tools/tune/crossover/example_crossover.m
index d46be04a6f0b..8c86c795ace3 100644
--- a/tools/tune/crossover/example_crossover.m
+++ b/tools/tune/crossover/example_crossover.m
@@ -1,10 +1,10 @@
 function example_crossover();
 
 % Set the parameters here
-tplg_fn = "../../topology/topology1/m4/crossover_coef_default.m4" % Control Bytes File
+tplg_fn = "../../topology/topology1/m4/crossover_coef_default.m4"; % Control Bytes File
 % Use those files with sof-ctl to update the component's configuration
-blob_fn = "../../ctl/crossover_coef.blob" % Blob binary file
-alsa_fn = "../../ctl/crossover_coef.txt" % ALSA CSV format file
+blob_fn = "../../ctl/crossover_coef.blob"; % Blob binary file
+alsa_fn = "../../ctl/crossover_coef.txt"; % ALSA CSV format file
 
 endian = "little";
 
@@ -51,4 +51,4 @@
 crossover_plot_freq(crossover.lp, crossover.hp, fs, num_sinks);
 rmpath ./../common
 
-endfunction
+end
diff --git a/tools/tune/dcblock/dcblock_build_blob.m b/tools/tune/dcblock/dcblock_build_blob.m
index 43fa67f5c915..6719d96f1ccb 100644
--- a/tools/tune/dcblock/dcblock_build_blob.m
+++ b/tools/tune/dcblock/dcblock_build_blob.m
@@ -6,7 +6,7 @@
 
 if nargin < 2
         endian = 'little'
-endif
+end
 
 %% Shift values for little/big endian
 switch lower(endian)
@@ -40,7 +40,7 @@
 	j=j+4;
 end
 
-endfunction
+end
 
 function bytes = word2byte(word, sh)
 bytes = uint8(zeros(1,4));
diff --git a/tools/tune/dcblock/dcblock_plot_stepfn.m b/tools/tune/dcblock/dcblock_plot_stepfn.m
index 724b4bcfcf40..6cc0cd40be35 100644
--- a/tools/tune/dcblock/dcblock_plot_stepfn.m
+++ b/tools/tune/dcblock/dcblock_plot_stepfn.m
@@ -13,4 +13,4 @@
 tstr = sprintf("DC Blocking Filter Step Response, R = %i", R);
 title(tstr);
 
-endfunction
+end
diff --git a/tools/tune/dcblock/dcblock_plot_transferfn.m b/tools/tune/dcblock/dcblock_plot_transferfn.m
index 8237222014a6..006d9b1ec26f 100644
--- a/tools/tune/dcblock/dcblock_plot_transferfn.m
+++ b/tools/tune/dcblock/dcblock_plot_transferfn.m
@@ -1,4 +1,4 @@
-function dcblock_plot_transferfn(R, fs);
+function dcblock_plot_transferfn(R, fs)
 % Plot the transfer function.
 % For a DC Blocking filter: H(z) = (1-1/z)/(1 - R/z)
 % Therefore the coefficients are b = [1 -1], a = [1 -R]
@@ -6,12 +6,14 @@
 a = [1 -R];
 
 f = linspace(1, fs/2, 500);
-
-semilogx(f, 20*log10(freqz(b, a, f, fs)));
+[h,w] = freqz(b, a, f, fs);
+% semilogx(f, 20*log10(freqz(b, a, f, fs)));
+semilogx(w, 20*log10(abs(h)));
 grid on
-xlabel('Frequency (Hz)');
+% xlabel('Frequency (Hz)');
+xlabel('Normalized Frequency (\times\pi rad/sample)');
 ylabel('Magnitude (dB)');
 tstr = sprintf("DC Blocking Filter Frequency Response, R = %i", R);
 title(tstr);
 
-endfunction
+end
diff --git a/tools/tune/dcblock/example_dcblock.m b/tools/tune/dcblock/example_dcblock.m
index 74490e5ab159..b7ada48df66c 100644
--- a/tools/tune/dcblock/example_dcblock.m
+++ b/tools/tune/dcblock/example_dcblock.m
@@ -27,4 +27,4 @@
 
 rmpath ./../common
 
-endfunction
+end
diff --git a/tools/tune/dmic/dmic_fir.m b/tools/tune/dmic/dmic_fir.m
index b82a3c0faafe..b70651c15526 100644
--- a/tools/tune/dmic/dmic_fir.m
+++ b/tools/tune/dmic/dmic_fir.m
@@ -46,7 +46,7 @@
 
 %% Iterate order and passband width
 iterate = 1;
-lo_ind = find(f < passhz/20);
+lo_ind = f < passhz/20;
 pb_ind = find(f < passhz);
 sb_ind = find(f > stophz);
 pb_ripple_min = 1000;
diff --git a/tools/tune/drc/drc_build_blob.m b/tools/tune/drc/drc_build_blob.m
index 1599c7715b2e..e9433c9991fd 100644
--- a/tools/tune/drc/drc_build_blob.m
+++ b/tools/tune/drc/drc_build_blob.m
@@ -2,7 +2,7 @@
 
 if nargin < 2
         endian = 'little'
-endif
+end
 
 %% Shift values for little/big endian
 switch lower(endian)
@@ -58,7 +58,7 @@
 blob8(j:j+3) = word2byte(blob_struct.kD, sh); j=j+4;
 blob8(j:j+3) = word2byte(blob_struct.kE, sh); j=j+4;
 
-endfunction
+end
 
 function bytes = word2byte(word, sh)
 bytes = uint8(zeros(1,4));
diff --git a/tools/tune/drc/drc_gen_coefs.m b/tools/tune/drc/drc_gen_coefs.m
index 632c5b5fc935..0edfa39d2513 100644
--- a/tools/tune/drc/drc_gen_coefs.m
+++ b/tools/tune/drc/drc_gen_coefs.m
@@ -2,7 +2,7 @@
 
 if exist('OCTAVE_VERSION', 'builtin')
 	pkg load control;
-endif
+end
 
 % Print out params
 params
@@ -82,7 +82,7 @@
 		max_k = k; % k is too high
 	else
 		min_k = k; % k is too low
-	endif
+    end
 
 	% Re-calculate based on geometric mean
 	k = sqrt(min_k * max_k);
@@ -103,7 +103,7 @@
 	y2_db = mag2db(knee_curve(linear_threshold, x2, k));
 
 	slope = (y2_db - y_db) / (x2_db - x_db);
-endif
+end
 
 end
 
@@ -114,6 +114,6 @@
 	y = x;
 else
 	y = linear_threshold + (1 - exp(-k * (x - linear_threshold))) / k;
-endif
+end
 
 end
diff --git a/tools/tune/drc/drc_plot_db_curve.m b/tools/tune/drc/drc_plot_db_curve.m
index ae3952973e4e..230b9049f8be 100644
--- a/tools/tune/drc/drc_plot_db_curve.m
+++ b/tools/tune/drc/drc_plot_db_curve.m
@@ -2,7 +2,7 @@
 
 if exist('OCTAVE_VERSION', 'builtin')
 	pkg load control;
-endif
+end
 
 db_thres = coefs.db_threshold;
 db_knee_thres = db_thres + coefs.db_knee;
@@ -18,21 +18,27 @@
 
 % Plot the x-y dB curve
 plot(db_xs, db_ys, "b");
+text(db_xs(end), db_ys(end), "dB curve");
 hold on
 grid on
 % Plot the reference line: x = db_thres
 plot([db_thres, db_thres], [db_ys(1), db_knee_thres], "g");
+text([db_thres, db_thres], [db_ys(1), db_knee_thres], "reference line: x = db thres");
 hold on
 % Plot the reference line: x = db_knee_thres
 plot([db_knee_thres, db_knee_thres], [db_ys(1), db_knee_thres], "g");
+text([db_knee_thres, db_knee_thres], [db_ys(1), db_knee_thres], "reference line: x = db knee thres");
 hold on
 % Plot the reference line: y = x
 plot([db_xs(1), db_knee_thres], [db_xs(1), db_knee_thres], "r");
+text([db_xs(1), db_knee_thres] + 0.5, [db_xs(1), db_knee_thres], "reference line: y = x",'HorizontalAlignment', 'Center', 'VerticalAlignment', 'Bottom');
 hold off
+title('Single band DRC range of db(x) determined by thres(db) and knee thres(db)');
 %pause();
 
 end
 
+
 function db_ys = db_curve(coefs, db_xs);
 
 db_thres = coefs.db_threshold;
@@ -50,7 +56,7 @@
 	else
 		% Among knee curve
 		db_ys(i) = db_knee_curve(coefs, db_x);
-	endif
+    end
 end
 
 end
diff --git a/tools/tune/drc/example_drc.m b/tools/tune/drc/example_drc.m
index 580aede93d8f..773f3b19db49 100644
--- a/tools/tune/drc/example_drc.m
+++ b/tools/tune/drc/example_drc.m
@@ -53,4 +53,4 @@
 
 rmpath ./../common
 
-endfunction
+end
diff --git a/tools/tune/eq/eq_compute.m b/tools/tune/eq/eq_compute.m
index f20863772261..266af9096422 100644
--- a/tools/tune/eq/eq_compute.m
+++ b/tools/tune/eq/eq_compute.m
@@ -41,7 +41,7 @@
 
 %% Define target (e.g. speaker) response as parametric filter. This could also
 %  be numerical data interpolated to the grid.
-if length(eq.parametric_target_response) > 0
+if ~isempty(eq.parametric_target_response)
         [eq.t_z, eq.t_p, eq.t_k] = eq_define_parametric_eq( ...
                 eq.parametric_target_response, eq.fs);
         eq.t_db = eq_compute_response(eq.t_z, eq.t_p, eq.t_k, eq.f, eq.fs);
@@ -121,7 +121,7 @@
 %% Smooth response with 1/N octave filter for plotting
 eq.m_db_s = logsmooth(eq.f, eq.m_db, eq.logsmooth_plot);
 
-if length(eq.target_m_db) > 0
+if ~isempty(eq.target_m_db)
         % Use target_m_db as dummy group delay, ignore other than magnitude
         [f0, m0, ~] = fix_response_dcnyquist_mult(eq.target_f, ...
                 eq.target_m_db, [], eq.fs);
@@ -139,14 +139,14 @@
 if min(f_hz) >  0
         f_hz = [0 f_hz];
         m_db = [m_db(1) m_db]; % DC the same as 1st measured point
-        if length(gd_s) > 0
+        if ~isempty(gd_s)
                 gd_s = [gd_s(1) gd_s]; % DC the same as 1st measured point
         end
 end
 if max(f_hz) <  fs/2
         f_hz = [f_hz fs/2];
         m_db = [m_db m_db(end)]; % Fs/2 the same as last measured point
-        if length(gd_s) > 0
+        if ~isempty(gd_s)
                 gd_s = [gd_s gd_s(end)]; % Fs/2 the same as last measured point
         end
 end
@@ -251,17 +251,17 @@
 m_db = err2db - err2db(idx);
 if auto
         cf = [1e3 6e3];
-        ind1 = find(fhz < cf(2));
+        ind1 = fhz < cf(2);
         ind2 = find(fhz(ind1) > cf(1));
         ipeak = find(m_db(ind2) == max(m_db(ind2))) + ind2(1);
-        ind1 = find(fhz < cf(1));
+        ind1 = fhz < cf(1);
         ind2 = find(m_db(ind1) > m_db(ipeak));
-        if length(ind2) > 0
+        if ~isempty(ind2)
                 fmin_fir = fhz(ind2(end));
         end
         ind1 = find(fhz > cf(2));
         ind2 = find(m_db(ind1) > m_db(ipeak)) + ind1(1);
-        if length(ind2) > 0
+        if ~isempty(ind2)
                 fmax_fir = fhz(ind2(1));
         end
 end
@@ -270,16 +270,16 @@
 ind1 = find(fhz < fmin_fir);
 ind2 = find(fhz > fmax_fir);
 p1 = ind1(end)+1;
-if length(ind2) > 0
+if ~isempty(ind2)
         p2 = ind2(1)-1;
 else
         p2 = length(fhz);
 end
 m_db(ind1) = m_db(p1);
 m_db(ind2) = m_db(p2);
-ind = find(m_db > amax_fir);
+ind = m_db > amax_fir;
 m_db(ind) = amax_fir;
-ind = find(m_db < amin_fir);
+ind = m_db < amin_fir;
 m_db(ind) = amin_fir;
 
 %% Smooth high frequency corner with spline
@@ -342,13 +342,13 @@
 %% Find zeros inside unit circle
 myeps = 1e-3;
 hdzeros = roots(blin);
-ind1 = find( abs(hdzeros) < (1-myeps) );
+ind1 =  abs(hdzeros) < (1-myeps) ;
 minzeros = hdzeros(ind1);
 
 %% Find double zeros at unit circle
-ind2 = find( abs(hdzeros) > (1-myeps) );
+ind2 =  abs(hdzeros) > (1-myeps) ;
 outzeros = hdzeros(ind2);
-ind3 = find( abs(outzeros) < (1+myeps) );
+ind3 =  abs(outzeros) < (1+myeps) ;
 circlezeros = outzeros(ind3);
 
 %% Get half of the unit circle zeros
@@ -359,7 +359,7 @@
         cangle = angle(circlezeros);
         [sorted_cangle, ind] = sort(cangle);
         sorted_czeros = circlezeros(ind);
-        pos = find(angle(sorted_czeros) > 0);
+        pos = angle(sorted_czeros) > 0;
         neg = find(angle(sorted_czeros) < 0);
         pos_czeros = sorted_czeros(pos);
         neg_czeros = sorted_czeros(neg(end:-1:1));
@@ -369,7 +369,7 @@
                 h1 = [h1' complex(cos(x),sin(x))]';
         end
         h2 = [];
-        for i = 1:2:length(neg_czeros)-1;
+        for i = 1:2:length(neg_czeros)-1
                 x=mean(angle(neg_czeros(i:i+1)));
                 h2 = [h2' complex(cos(x),sin(x))]';
         end
@@ -412,7 +412,7 @@
         i1 = zeros(1,n1);
         i2 = zeros(1,n1);
         i1(1) = 1;
-        for i = 1:n1-1;
+        for i = 1:n1-1
                 if rem > 0
                         i2(i) = i1(i)+rpb;
                         rem = rem-1;
diff --git a/tools/tune/eq/eq_define_parametric_eq.m b/tools/tune/eq/eq_define_parametric_eq.m
index 70a728e14c73..29e4684c5b82 100644
--- a/tools/tune/eq/eq_define_parametric_eq.m
+++ b/tools/tune/eq/eq_define_parametric_eq.m
@@ -73,10 +73,10 @@
                         otherwise
                                 error('Unknown parametric EQ type');
                 end
-                if length(a0) > 0
+                if ~isempty(a0)
                         [z0, p0, k0] = tf2zp(b0, a0);
                 end
-                if length(k0) > 0
+                if ~isempty(k0)
                         z = [z ; z0(:)];
                         p = [p ; p0(:)];
                         k = k * k0;
@@ -127,15 +127,15 @@
 	A = 10^(gdb/40); % Square root of linear gain
 	wc = 2*pi*fhz/fs;
 
-	if Q <= 0
-		% To fix gui edge cases, comment from CRAS code:
-		% When Q = 0, the above formulas have problems. If we
-		% look at the z-transform, we can see that the limit
-		% as Q->0 is A^2, so set the filter that way.
-		b = [A * A, 0, 0]
-		a = [1, 0, 0];
-		return;
-	endif
+    if Q <= 0
+        % To fix gui edge cases, comment from CRAS code:
+        % When Q = 0, the above formulas have problems. If we
+        % look at the z-transform, we can see that the limit
+        % as Q->0 is A^2, so set the filter that way.
+        b = [A * A, 0, 0];
+        a = [1, 0, 0];
+        return;
+    end
 
 	alpha = sin(wc)/(2*Q);
 	b0 = 1 + alpha * A;
@@ -163,7 +163,7 @@
 		b = [1 - cutoff, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	% Compute biquad coefficients for highpass filter
 	resonance = max(0.0, resonance); % can't go negative
@@ -199,7 +199,7 @@
 		b = [cutoff, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	% Compute biquad coefficients for lowpass filter
 	resonance = max(0.0, resonance); % can't go negative
@@ -239,7 +239,7 @@
 		b = [0, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 	if (Q <= 0)
 		% When Q = 0, the above formulas have problems. If we
 		% look at the z-transform, we can see that the limit
@@ -247,7 +247,7 @@
 		b = [1, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	w0 = pi * frequency;
 	alpha = sin(w0) / (2 * Q);
@@ -271,20 +271,20 @@
 	% Don't let Q go negative, which causes an unstable filter.
 	Q = max(0.0, Q);
 
-	if frequency <= 0 || frequency >= 1
-		% When frequency is 0 or 1, the z-transform is 1.
-		b = [1, 0, 0];
-		a = [1, 0, 0];
-		return;
-	endif
-	if Q <= 0
-		% When Q = 0, the above formulas have problems. If we
-		% look at the z-transform, we can see that the limit
-		% as Q->0 is 0, so set the filter that way.
-		b = [0, 0, 0];
-		a = [1, 0, 0];
-		return;
-	endif
+    if frequency <= 0 || frequency >= 1
+        % When frequency is 0 or 1, the z-transform is 1.
+        b = [1, 0, 0];
+        a = [1, 0, 0];
+        return;
+    end
+    if Q <= 0
+        % When Q = 0, the above formulas have problems. If we
+        % look at the z-transform, we can see that the limit
+        % as Q->0 is 0, so set the filter that way.
+        b = [0, 0, 0];
+        a = [1, 0, 0];
+        return;
+    end
 
 	w0 = pi * frequency;
 	alpha = sin(w0) / (2 * Q);
@@ -313,13 +313,13 @@
 		b = [A * A, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 	if (frequency <= 0)
 		% When frequency is 0, the z-transform is 1.
 		b = [1, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	w0 = pi * frequency;
 	S = 1; % filter slope (1 is max value)
@@ -353,13 +353,13 @@
 		b = [1, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 	if (frequency <= 0)
 		% When frequency = 0, the filter is just a gain, A^2.
 		b = [A * A, 0, 0];
 		a = [1, 0, 0];
 		return;
-	endif
+    end
 
 	w0 = pi * frequency;
 	S = 1; % filter slope (1 is max value)
diff --git a/tools/tune/eq/example_spk_eq.m b/tools/tune/eq/example_spk_eq.m
index fdc1f096ac42..37b7f557098a 100644
--- a/tools/tune/eq/example_spk_eq.m
+++ b/tools/tune/eq/example_spk_eq.m
@@ -90,7 +90,7 @@ function example_spk_eq()
 %  The lowest and highest frequeciens below and above speaker
 %  capability are attenuated with high-pass and low-pass
 %  filtering.
-if eq.enable_iir;
+if eq.enable_iir
 	eq.peq = [ ...
 			 eq.PEQ_HP2   100 NaN NaN; ...
 			 eq.PEQ_PN2  1480  -7 2.0; ...
@@ -102,12 +102,12 @@ function example_spk_eq()
 %% With FIR EQ try to flatten frequency response within
 %  1 - 13 kHz frequency band.
 if eq.enable_fir
-        eq.fir_minph = 1;
-	eq.fir_beta = 4;
-	eq.fir_length = 63;
-	eq.fir_autoband = 0;
-	eq.fmin_fir = 900;
-	eq.fmax_fir = 10700;
+    eq.fir_minph = 1;
+    eq.fir_beta = 4;
+    eq.fir_length = 63;
+    eq.fir_autoband = 0;
+    eq.fmin_fir = 900;
+    eq.fmax_fir = 10700;
 end
 
 %% Design EQ
@@ -125,7 +125,7 @@ function example_spk_eq()
         bm_fir = eq_fir_blob_merge(channels_in_config, ...
                 num_responses, ...
                 assign_response, ...
-                [ bq_fir ]);
+                 bq_fir );
         bp_fir = eq_fir_blob_pack(bm_fir);
         eq_alsactl_write(fn_fir_sofctl, bp_fir);
         eq_blob_write(fn_fir_blob, bp_fir);
@@ -138,7 +138,7 @@ function example_spk_eq()
         bm_iir = eq_iir_blob_merge(channels_in_config, ...
                 num_responses, ...
                 assign_response, ...
-                [ bq_iir ]);
+                ( bq_iir ));
         bp_iir = eq_iir_blob_pack(bm_iir);
         eq_alsactl_write(fn_iir_sofctl, bp_iir);
         eq_blob_write(fn_iir_blob, bp_iir);
diff --git a/tools/tune/eq/mls_freq_resp.m b/tools/tune/eq/mls_freq_resp.m
index a6fbdbdb0497..aabddc584aaf 100644
--- a/tools/tune/eq/mls_freq_resp.m
+++ b/tools/tune/eq/mls_freq_resp.m
@@ -57,8 +57,8 @@
 else
 	selftest = 0;
 end
-measfn = sprintf('mls-%s.wav', id);
-csvfn = sprintf('mls-%s.txt', id);
+measfn = sprintf('mls-%d.wav', id);
+csvfn = sprintf('mls-%d.txt', id);
 
 %% Paths
 addpath('../../test/audio/test_utils');
@@ -101,10 +101,10 @@
 recfn = 'recch.wav';
 y = [];
 if selftest
-	labels = cell(play_cfg.nch * rec_cfg.nch + 1, 1);
+	labels = cellstr(char(play_cfg.nch * rec_cfg.nch + 1, 1));
 	labels(end) = 'Reference';
 else
-	labels = cell(play_cfg.nch * rec_cfg.nch, 1);
+	labels = cellstr(char(play_cfg.nch * rec_cfg.nch, 1));
 end
 label_idx = 1;
 for i=1:play_cfg.nch
@@ -130,7 +130,7 @@
 		r = get_recording(recfn, rec_cfg);
 	end
 	for j = 1:rec_cfg.nch
-		labels(label_idx) = sprintf('p%d-r%d', i, j);
+		labels{label_idx} = sprintf('p%d-r%d', i, j);
 		label_idx = label_idx + 1;
 	end
 	[d, nt] = find_test_signal(r(:,1), fnd);
@@ -200,7 +200,7 @@
 legend(labels);
 
 if selftest
-        idx = find(f < f_hi);
+        idx = f < f_hi;
         idx = find(f(idx) > f_lo);
         m_lin = 10.^(m_db_align(idx)/20);
         ref_lin = 10.^(ref_db_align(idx)/20);
@@ -330,14 +330,14 @@ function remote_capture(fn, cfg, t)
 	sens =[];
 	desc = '';
 	str = fgets(fh);
-	idx = findstr(str, '"');
-	while length(idx) > 0
+	idx = strfind(str, '"');
+	while ~isempty(idx)
 		line = str(idx(1)+1:idx(2)-1);
 		desc = sprintf('%s%s ', desc, line);
 		str = fgets(fh);
-		idx = findstr(str, '"');
+		idx = strfind(str, '"');
 	end
-	if length(strfind(str, 'Sens'))
+	if contains(str, 'Sens')
 		desc = str;
 		str = fgets(fh);
 	end
@@ -403,7 +403,7 @@ function remote_capture(fn, cfg, t)
 
 function [cal_f, cal_m_db] = apply_mic_calibration(f, m_db, rec)
 
-	if length(rec.cm) > 0
+	if ~isempty(rec.cm)
 		if ~isvector(rec.cm)
 			error('Calibration can be for one channel only');
 		end
diff --git a/tools/tune/multiband_drc/example_multiband_drc.m b/tools/tune/multiband_drc/example_multiband_drc.m
index 2594c3b86682..a4772dc6caae 100644
--- a/tools/tune/multiband_drc/example_multiband_drc.m
+++ b/tools/tune/multiband_drc/example_multiband_drc.m
@@ -97,17 +97,17 @@
 [emp_coefs, deemp_coefs] = iir_gen_quant_coefs(iir_params);
 
 % Generate Crossover quantized coefs struct from parameters
-crossover_coefs = crossover_gen_quant_coefs(num_bands, sample_rate,
-					    drc_params(2).band_lower_freq,
-					    drc_params(3).band_lower_freq,
+crossover_coefs = crossover_gen_quant_coefs(num_bands, sample_rate,...
+					    drc_params(2).band_lower_freq, ...
+					    drc_params(3).band_lower_freq, ...
 					    drc_params(4).band_lower_freq);
 
 % Generate DRC quantized coefs struct from parameters
 drc_coefs = drc_gen_quant_coefs(num_bands, sample_rate, drc_params);
 
 % Convert quantized coefs structs to binary blob
-blob8 = multiband_drc_build_blob(num_bands, enable_emp_deemp, emp_coefs,
-				 deemp_coefs, crossover_coefs, drc_coefs,
+blob8 = multiband_drc_build_blob(num_bands, enable_emp_deemp, emp_coefs,...
+				 deemp_coefs, crossover_coefs, drc_coefs,...
 				 endian);
 
 % Generate output files
@@ -119,4 +119,4 @@
 
 rmpath ../common
 
-endfunction
+end
diff --git a/tools/tune/multiband_drc/iir_gen_quant_coefs.m b/tools/tune/multiband_drc/iir_gen_quant_coefs.m
index 425a7513bba5..b67727f4a833 100644
--- a/tools/tune/multiband_drc/iir_gen_quant_coefs.m
+++ b/tools/tune/multiband_drc/iir_gen_quant_coefs.m
@@ -1,4 +1,4 @@
-function [emp_coefs, deemp_coefs] = iir_gen_quant_coefs(params);
+function [emp_coefs, deemp_coefs] = iir_gen_quant_coefs(params)
 
 stage_gain = params.stage_gain;
 stage_ratio = params.stage_ratio;
@@ -7,13 +7,13 @@
 emp = cell(1, 2);
 deemp = cell(1, 2);
 % Generate the coefficients of (de)emphasis for the 1-st stage of biquads
-[emp(1), deemp(1)] = emp_deemp_stage_biquad(stage_gain, anchor_freq,
+[emp(1), deemp(1)] = emp_deemp_stage_biquad(stage_gain, anchor_freq,...
 					    anchor_freq / stage_ratio);
 
 anchor_freq = anchor_freq / (stage_ratio * stage_ratio);
 
 % Generate the coefficients of (de)emphasis for the 2-nd stage of biquads
-[emp(2), deemp(2)] = emp_deemp_stage_biquad(stage_gain, anchor_freq,
+[emp(2), deemp(2)] = emp_deemp_stage_biquad(stage_gain, anchor_freq,...
                                             anchor_freq / stage_ratio);
 
 % Adjust the stage gain (push gains to the last stage) of emphasis filter
@@ -29,7 +29,7 @@
 
 end
 
-function [emp, deemp] = emp_deemp_stage_biquad(gain, f1, f2);
+function [emp, deemp] = emp_deemp_stage_biquad(gain, f1, f2)
 
 [z1, p1] = emp_stage_roots(gain, f1);
 [z2, p2] = emp_stage_roots(gain, f2);
@@ -58,7 +58,7 @@
 
 end
 
-function [zero, pole] = emp_stage_roots(gain, normalized_freq);
+function [zero, pole] = emp_stage_roots(gain, normalized_freq)
 
 gk = 1 - gain / 20;
 f1 = normalized_freq * gk;
@@ -68,7 +68,7 @@
 
 end
 
-function quant_coefs = iir_coef_quant(coefs);
+function quant_coefs = iir_coef_quant(coefs)
 
 bits_iir = 32; % Q2.30
 qf_iir = 30;
@@ -92,7 +92,7 @@
 
 end
 
-function [bq1, bq2] = stage_gain_adjust(prev_bq1, prev_bq2);
+function [bq1, bq2] = stage_gain_adjust(prev_bq1, prev_bq2)
 
 prev_bq1 = cell2mat(prev_bq1);
 prev_bq2 = cell2mat(prev_bq2);
@@ -103,7 +103,7 @@
 
 end
 
-function [rshift, gain] = decompose_gain(prev_gain);
+function [rshift, gain] = decompose_gain(prev_gain)
 
 max_abs_val = 2; % Q2.14
 rshift = 0;
@@ -111,7 +111,7 @@
 
 while (abs(gain) >= max_abs_val)
 	gain = gain / 2;
-	rshift--; % left-shift in shift stage
-endwhile
+    rshift = rshift - 1;%left-shift in shift stage
+end
 
 end