From 03430013994214f6a990b2f8521461690e9291bc Mon Sep 17 00:00:00 2001
From: shastry <malladi.sastry@intel.com>
Date: Thu, 3 Nov 2022 13:14:03 +0530
Subject: [PATCH 1/3] Audio: BugFix : Enhancing audio quality tests for device
 testing

For the purpose of remote device testing, this check-in adds a
correction to the scripts that evaluate gain,frequency responsiveness,
total harmonic distortion plus noise

Signed-off-by: shastry <malladi.sastry@intel.com>
---
 tools/test/audio/process_test.m               | 13 +++++------
 tools/test/audio/sof_test_perf.m              |  6 ++---
 tools/test/audio/sof_test_perf_config.m       |  8 ++++++-
 .../test/audio/std_utils/thdnf_test_measure.m | 20 +++++++----------
 tools/tune/crossover/crossover_coef_quant.m   |  6 ++---
 tools/tune/eq/mls_freq_resp.m                 | 22 +++++++++----------
 6 files changed, 37 insertions(+), 38 deletions(-)

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..1cb3d86f5cc1 100644
--- a/tools/test/audio/sof_test_perf.m
+++ b/tools/test/audio/sof_test_perf.m
@@ -128,8 +128,8 @@
 %% 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 +144,7 @@
 
 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);
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/thdnf_test_measure.m b/tools/test/audio/std_utils/thdnf_test_measure.m
index a0bc35f1f856..041b6c794823 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
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/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

From f89036686426445896cdd199b31d8ebd6bd302e4 Mon Sep 17 00:00:00 2001
From: shastry <malladi.sastry@intel.com>
Date: Thu, 3 Nov 2022 18:47:22 +0530
Subject: [PATCH 2/3] Audio:Enhancements:Enhancing audio quality tests for
 device testing

The threshold scale has been defined and the decibel full scale computation
has been made easier using new scripts.
The dynamic range test is included in the test suite
Relax the marker position for fr and thdn.

Signed-off-by: shastry <malladi.sastry@intel.com>
---
 tools/test/audio/sof_test_perf.m              | 28 +++++++++++-
 tools/test/audio/std_utils/fr_test_input.m    |  2 +-
 tools/test/audio/std_utils/fr_test_measure.m  | 16 ++++---
 tools/test/audio/std_utils/level_dbfs.m       |  5 +--
 tools/test/audio/std_utils/thdnf_test_input.m |  2 +-
 .../test/audio/test_utils/find_test_signal.m  | 45 +++++++++++++++----
 6 files changed, 76 insertions(+), 22 deletions(-)

diff --git a/tools/test/audio/sof_test_perf.m b/tools/test/audio/sof_test_perf.m
index 1cb3d86f5cc1..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');
@@ -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/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/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

From 4060f441a42faa5634f2ed3070d7eda73dd8302c Mon Sep 17 00:00:00 2001
From: shastry <malladi.sastry@intel.com>
Date: Thu, 3 Nov 2022 19:35:28 +0530
Subject: [PATCH 3/3] Audio:Matlab compatibility : Enhancing audio quality
 tests for device testing

The check-in overcomes Octave and Matlab compatibility issues.
The performance of matlab and octave is now comparable.

Signed-off-by: shastry <malladi.sastry@intel.com>
---
 tools/test/audio/std_utils/dr_test_measure.m  |  2 +-
 .../test/audio/std_utils/thdnf_test_measure.m |  8 +--
 tools/test/audio/test_utils/mix_sweep.m       |  2 +-
 tools/tune/common/alsactl_write.m             |  2 +-
 tools/tune/crossover/crossover_build_blob.m   |  4 +-
 tools/tune/crossover/crossover_gen_coefs.m    | 12 ++--
 tools/tune/crossover/crossover_plot_freq.m    | 29 ++++++--
 tools/tune/crossover/example_crossover.m      |  8 +--
 tools/tune/dcblock/dcblock_build_blob.m       |  4 +-
 tools/tune/dcblock/dcblock_plot_stepfn.m      |  2 +-
 tools/tune/dcblock/dcblock_plot_transferfn.m  | 12 ++--
 tools/tune/dcblock/example_dcblock.m          |  2 +-
 tools/tune/dmic/dmic_fir.m                    |  2 +-
 tools/tune/drc/drc_build_blob.m               |  4 +-
 tools/tune/drc/drc_gen_coefs.m                |  8 +--
 tools/tune/drc/drc_plot_db_curve.m            | 10 ++-
 tools/tune/drc/example_drc.m                  |  2 +-
 tools/tune/eq/eq_compute.m                    | 34 +++++-----
 tools/tune/eq/eq_define_parametric_eq.m       | 66 +++++++++----------
 tools/tune/eq/example_spk_eq.m                | 18 ++---
 .../multiband_drc/example_multiband_drc.m     | 12 ++--
 .../tune/multiband_drc/iir_gen_quant_coefs.m  | 20 +++---
 22 files changed, 143 insertions(+), 120 deletions(-)

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/thdnf_test_measure.m b/tools/test/audio/std_utils/thdnf_test_measure.m
index 041b6c794823..9cccd76f0f56 100644
--- a/tools/test/audio/std_utils/thdnf_test_measure.m
+++ b/tools/test/audio/std_utils/thdnf_test_measure.m
@@ -72,16 +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/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_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/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