Style: Enforce braces around if blocks and loops

Using clang-tidy's `readability-braces-around-statements`.
https://clang.llvm.org/extra/clang-tidy/checks/readability-braces-around-statements.html

14 files changed, 129 insertions, 64 deletions

diff --git a/servers/audio/audio_driver_dummy.cpp b/servers/audio/audio_driver_dummy.cpp
index 5560385fd3..70d5ebbded 100644
--- a/servers/audio/audio_driver_dummy.cpp
+++ b/servers/audio/audio_driver_dummy.cpp
@@ -86,20 +86,23 @@ AudioDriver::SpeakerMode AudioDriverDummy::get_speaker_mode() const {
 };
 
 void AudioDriverDummy::lock() {
-	if (!thread)
+	if (!thread) {
 		return;
+	}
 	mutex.lock();
 };
 
 void AudioDriverDummy::unlock() {
-	if (!thread)
+	if (!thread) {
 		return;
+	}
 	mutex.unlock();
 };
 
 void AudioDriverDummy::finish() {
-	if (!thread)
+	if (!thread) {
 		return;
+	}
 
 	exit_thread = true;
 	Thread::wait_to_finish(thread);
diff --git a/servers/audio/audio_filter_sw.cpp b/servers/audio/audio_filter_sw.cpp
index b6759649cc..f5eafb7e60 100644
--- a/servers/audio/audio_filter_sw.cpp
+++ b/servers/audio/audio_filter_sw.cpp
@@ -54,8 +54,9 @@ void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) {
 	int sr_limit = (sampling_rate / 2) + 512;
 
 	double final_cutoff = (cutoff > sr_limit) ? sr_limit : cutoff;
-	if (final_cutoff < 1)
+	if (final_cutoff < 1) {
 		final_cutoff = 1; //don't allow less than this
+	}
 
 	double omega = 2.0 * Math_PI * final_cutoff / sampling_rate;
 
@@ -67,15 +68,17 @@ void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) {
 		Q = 0.0001;
 	}
 
-	if (mode == BANDPASS)
+	if (mode == BANDPASS) {
 		Q *= 2.0;
-	else if (mode == PEAK)
+	} else if (mode == PEAK) {
 		Q *= 3.0;
+	}
 
 	double tmpgain = gain;
 
-	if (tmpgain < 0.001)
+	if (tmpgain < 0.001) {
 		tmpgain = 0.001;
+	}
 
 	if (stages > 1) {
 		Q = (Q > 1.0 ? Math::pow(Q, 1.0 / stages) : Q);
@@ -142,8 +145,9 @@ void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) {
 		} break;
 		case LOWSHELF: {
 			double tmpq = Math::sqrt(Q);
-			if (tmpq <= 0)
+			if (tmpq <= 0) {
 				tmpq = 0.001;
+			}
 			double beta = Math::sqrt(tmpgain) / tmpq;
 
 			a0 = (tmpgain + 1.0) + (tmpgain - 1.0) * cos_v + beta * sin_v;
@@ -156,8 +160,9 @@ void AudioFilterSW::prepare_coefficients(Coeffs *p_coeffs) {
 		} break;
 		case HIGHSHELF: {
 			double tmpq = Math::sqrt(Q);
-			if (tmpq <= 0)
+			if (tmpq <= 0) {
 				tmpq = 0.001;
+			}
 			double beta = Math::sqrt(tmpgain) / tmpq;
 
 			a0 = (tmpgain + 1.0) - (tmpgain - 1.0) * cos_v + beta * sin_v;
@@ -235,8 +240,9 @@ void AudioFilterSW::Processor::set_filter(AudioFilterSW *p_filter, bool p_clear_
 }
 
 void AudioFilterSW::Processor::update_coeffs(int p_interp_buffer_len) {
-	if (!filter)
+	if (!filter) {
 		return;
+	}
 
 	if (p_interp_buffer_len) { //interpolate
 		Coeffs old_coeffs = coeffs;
@@ -253,8 +259,9 @@ void AudioFilterSW::Processor::update_coeffs(int p_interp_buffer_len) {
 }
 
 void AudioFilterSW::Processor::process(float *p_samples, int p_amount, int p_stride, bool p_interpolate) {
-	if (!filter)
+	if (!filter) {
 		return;
+	}
 
 	if (p_interpolate) {
 		for (int i = 0; i < p_amount; i++) {
diff --git a/servers/audio/audio_rb_resampler.cpp b/servers/audio/audio_rb_resampler.cpp
index a7542fab03..7613e70e64 100644
--- a/servers/audio/audio_rb_resampler.cpp
+++ b/servers/audio/audio_rb_resampler.cpp
@@ -34,8 +34,9 @@
 #include "servers/audio_server.h"
 
 int AudioRBResampler::get_channel_count() const {
-	if (!rb)
+	if (!rb) {
 		return 0;
+	}
 
 	return channels;
 }
@@ -101,8 +102,9 @@ uint32_t AudioRBResampler::_resample(AudioFrame *p_dest, int p_todo, int32_t p_i
 }
 
 bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {
-	if (!rb)
+	if (!rb) {
 		return false;
+	}
 
 	int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
 	int read_space = get_reader_space();
@@ -125,8 +127,9 @@ bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {
 				break;
 		}
 
-		if (src_read > read_space)
+		if (src_read > read_space) {
 			src_read = read_space;
+		}
 
 		rb_read_pos = (rb_read_pos + src_read) & rb_mask;
 
@@ -147,8 +150,9 @@ bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {
 }
 
 int AudioRBResampler::get_num_of_ready_frames() {
-	if (!is_ready())
+	if (!is_ready()) {
 		return 0;
+	}
 	int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
 	int read_space = get_reader_space();
 	return (int64_t(read_space) << MIX_FRAC_BITS) / increment;
@@ -192,8 +196,9 @@ Error AudioRBResampler::setup(int p_channels, int p_src_mix_rate, int p_target_m
 }
 
 void AudioRBResampler::clear() {
-	if (!rb)
+	if (!rb) {
 		return;
+	}
 
 	//should be stopped at this point but just in case
 	memdelete_arr(rb);
diff --git a/servers/audio/audio_stream.cpp b/servers/audio/audio_stream.cpp
index 11b96edb8d..2cc2f5c291 100644
--- a/servers/audio/audio_stream.cpp
+++ b/servers/audio/audio_stream.cpp
@@ -244,8 +244,9 @@ Ref<AudioStream> AudioStreamRandomPitch::get_audio_stream() const {
 }
 
 void AudioStreamRandomPitch::set_random_pitch(float p_pitch) {
-	if (p_pitch < 1)
+	if (p_pitch < 1) {
 		p_pitch = 1;
+	}
 	random_pitch = p_pitch;
 }
 
@@ -256,8 +257,9 @@ float AudioStreamRandomPitch::get_random_pitch() const {
 Ref<AudioStreamPlayback> AudioStreamRandomPitch::instance_playback() {
 	Ref<AudioStreamPlaybackRandomPitch> playback;
 	playback.instance();
-	if (audio_stream.is_valid())
+	if (audio_stream.is_valid()) {
 		playback->playback = audio_stream->instance_playback();
+	}
 
 	playbacks.insert(playback.ptr());
 	playback->random_pitch = Ref<AudioStreamRandomPitch>((AudioStreamRandomPitch *)this);
diff --git a/servers/audio/effects/audio_effect_chorus.cpp b/servers/audio/effects/audio_effect_chorus.cpp
index 48f050c5ab..2b530475f0 100644
--- a/servers/audio/effects/audio_effect_chorus.cpp
+++ b/servers/audio/effects/audio_effect_chorus.cpp
@@ -81,8 +81,9 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames, A
 		}
 
 		//low pass filter
-		if (v.cutoff == 0)
+		if (v.cutoff == 0) {
 			continue;
+		}
 		float auxlp = expf(-2.0 * Math_PI * v.cutoff / mix_rate);
 		float c1 = 1.0 - auxlp;
 		float c2 = auxlp;
diff --git a/servers/audio/effects/audio_effect_compressor.cpp b/servers/audio/effects/audio_effect_compressor.cpp
index b0f829b679..4b0b4dabea 100644
--- a/servers/audio/effects/audio_effect_compressor.cpp
+++ b/servers/audio/effects/audio_effect_compressor.cpp
@@ -66,11 +66,13 @@ void AudioEffectCompressorInstance::process(const AudioFrame *p_src_frames, Audi
 
 		float overdb = 2.08136898f * Math::linear2db(peak / threshold);
 
-		if (overdb < 0.0) //we only care about what goes over to compress
+		if (overdb < 0.0) { //we only care about what goes over to compress
 			overdb = 0.0;
+		}
 
-		if (overdb - rundb > 5) // diffeence is too large
+		if (overdb - rundb > 5) { // diffeence is too large
 			averatio = 4;
+		}
 
 		if (overdb > rundb) {
 			rundb = overdb + atcoef * (rundb - overdb);
@@ -101,8 +103,9 @@ void AudioEffectCompressorInstance::process(const AudioFrame *p_src_frames, Audi
 			gr_meter = grv;
 		} else {
 			gr_meter *= gr_meter_decay;
-			if (gr_meter > 1)
+			if (gr_meter > 1) {
 				gr_meter = 1;
+			}
 		}
 
 		p_dst_frames[i] = p_src_frames[i] * grv * makeup * mix + p_src_frames[i] * (1.0 - mix);
diff --git a/servers/audio/effects/audio_effect_delay.cpp b/servers/audio/effects/audio_effect_delay.cpp
index 00cf7a0e70..d6ab14be89 100644
--- a/servers/audio/effects/audio_effect_delay.cpp
+++ b/servers/audio/effects/audio_effect_delay.cpp
@@ -105,8 +105,9 @@ void AudioEffectDelayInstance::_process_chunk(const AudioFrame *p_src_frames, Au
 
 		ring_buffer_pos++;
 
-		if ((++feedback_buffer_pos) >= feedback_delay_frames)
+		if ((++feedback_buffer_pos) >= feedback_delay_frames) {
 			feedback_buffer_pos = 0;
+		}
 	}
 }
 
diff --git a/servers/audio/effects/audio_effect_distortion.cpp b/servers/audio/effects/audio_effect_distortion.cpp
index da4c34ce82..dc5c2cc16f 100644
--- a/servers/audio/effects/audio_effect_distortion.cpp
+++ b/servers/audio/effects/audio_effect_distortion.cpp
@@ -59,10 +59,11 @@ void AudioEffectDistortionInstance::process(const AudioFrame *p_src_frames, Audi
 		switch (base->mode) {
 			case AudioEffectDistortion::MODE_CLIP: {
 				a = powf(a, 1.0001 - drive_f);
-				if (a > 1.0)
+				if (a > 1.0) {
 					a = 1.0;
-				else if (a < (-1.0))
+				} else if (a < (-1.0)) {
 					a = -1.0;
+				}
 
 			} break;
 			case AudioEffectDistortion::MODE_ATAN: {
diff --git a/servers/audio/effects/audio_effect_filter.cpp b/servers/audio/effects/audio_effect_filter.cpp
index cf6d0fa896..a5135ee1a6 100644
--- a/servers/audio/effects/audio_effect_filter.cpp
+++ b/servers/audio/effects/audio_effect_filter.cpp
@@ -36,12 +36,15 @@ void AudioEffectFilterInstance::_process_filter(const AudioFrame *p_src_frames,
 	for (int i = 0; i < p_frame_count; i++) {
 		float f = p_src_frames[i].l;
 		filter_process[0][0].process_one(f);
-		if (S > 1)
+		if (S > 1) {
 			filter_process[0][1].process_one(f);
-		if (S > 2)
+		}
+		if (S > 2) {
 			filter_process[0][2].process_one(f);
-		if (S > 3)
+		}
+		if (S > 3) {
 			filter_process[0][3].process_one(f);
+		}
 
 		p_dst_frames[i].l = f;
 	}
@@ -49,12 +52,15 @@ void AudioEffectFilterInstance::_process_filter(const AudioFrame *p_src_frames,
 	for (int i = 0; i < p_frame_count; i++) {
 		float f = p_src_frames[i].r;
 		filter_process[1][0].process_one(f);
-		if (S > 1)
+		if (S > 1) {
 			filter_process[1][1].process_one(f);
-		if (S > 2)
+		}
+		if (S > 2) {
 			filter_process[1][2].process_one(f);
-		if (S > 3)
+		}
+		if (S > 3) {
 			filter_process[1][3].process_one(f);
+		}
 
 		p_dst_frames[i].r = f;
 	}
diff --git a/servers/audio/effects/audio_effect_filter.h b/servers/audio/effects/audio_effect_filter.h
index c439c5a5b5..b11a4e3623 100644
--- a/servers/audio/effects/audio_effect_filter.h
+++ b/servers/audio/effects/audio_effect_filter.h
@@ -99,8 +99,9 @@ class AudioEffectLowPassFilter : public AudioEffectFilter {
 	GDCLASS(AudioEffectLowPassFilter, AudioEffectFilter);
 
 	void _validate_property(PropertyInfo &property) const {
-		if (property.name == "gain")
+		if (property.name == "gain") {
 			property.usage = 0;
+		}
 	}
 
 public:
@@ -111,8 +112,9 @@ public:
 class AudioEffectHighPassFilter : public AudioEffectFilter {
 	GDCLASS(AudioEffectHighPassFilter, AudioEffectFilter);
 	void _validate_property(PropertyInfo &property) const {
-		if (property.name == "gain")
+		if (property.name == "gain") {
 			property.usage = 0;
+		}
 	}
 
 public:
@@ -123,8 +125,9 @@ public:
 class AudioEffectBandPassFilter : public AudioEffectFilter {
 	GDCLASS(AudioEffectBandPassFilter, AudioEffectFilter);
 	void _validate_property(PropertyInfo &property) const {
-		if (property.name == "gain")
+		if (property.name == "gain") {
 			property.usage = 0;
+		}
 	}
 
 public:
diff --git a/servers/audio/effects/audio_effect_pitch_shift.cpp b/servers/audio/effects/audio_effect_pitch_shift.cpp
index a8f25ac325..fb6b56d984 100644
--- a/servers/audio/effects/audio_effect_pitch_shift.cpp
+++ b/servers/audio/effects/audio_effect_pitch_shift.cpp
@@ -94,7 +94,9 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
 	freqPerBin = sampleRate/(double)fftFrameSize;
 	expct = 2.*Math_PI*(double)stepSize/(double)fftFrameSize;
 	inFifoLatency = fftFrameSize-stepSize;
-	if (gRover == 0) gRover = inFifoLatency;
+	if (gRover == 0) { gRover = inFifoLatency;
+
+}
 
 	/* initialize our static arrays */
 
@@ -142,8 +144,10 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
 
 				/* map delta phase into +/- Pi interval */
 				qpd = tmp/Math_PI;
-				if (qpd >= 0) qpd += qpd&1;
-				else qpd -= qpd&1;
+				if (qpd >= 0) { qpd += qpd&1;
+				} else { qpd -= qpd&1;
+
+}
 				tmp -= Math_PI*(double)qpd;
 
 				/* get deviation from bin frequency from the +/- Pi interval */
@@ -200,7 +204,9 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
 			}
 
 			/* zero negative frequencies */
-			for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) gFFTworksp[k] = 0.;
+			for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) { gFFTworksp[k] = 0.;
+
+}
 
 			/* do inverse transform */
 			smbFft(gFFTworksp, fftFrameSize, 1);
@@ -210,13 +216,17 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
 				window = -.5*cos(2.*Math_PI*(double)k/(double)fftFrameSize)+.5;
 				gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
 			}
-			for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
+			for (k = 0; k < stepSize; k++) { gOutFIFO[k] = gOutputAccum[k];
+
+}
 
 			/* shift accumulator */
 			memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float));
 
 			/* move input FIFO */
-			for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize];
+			for (k = 0; k < inFifoLatency; k++) { gInFIFO[k] = gInFIFO[k+stepSize];
+
+}
 		}
 	}
 
@@ -245,7 +255,9 @@ void SMBPitchShift::smbFft(float *fftBuffer, long fftFrameSize, long sign)
 
 	for (i = 2; i < 2*fftFrameSize-2; i += 2) {
 		for (bitm = 2, j = 0; bitm < 2*fftFrameSize; bitm <<= 1) {
-			if (i & bitm) j++;
+			if (i & bitm) { j++;
+
+}
 			j <<= 1;
 		}
 		if (i < j) {
diff --git a/servers/audio/effects/audio_effect_spectrum_analyzer.cpp b/servers/audio/effects/audio_effect_spectrum_analyzer.cpp
index a3fd11c6c0..e744dbf9b0 100644
--- a/servers/audio/effects/audio_effect_spectrum_analyzer.cpp
+++ b/servers/audio/effects/audio_effect_spectrum_analyzer.cpp
@@ -50,8 +50,9 @@ static void smbFft(float *fftBuffer, long fftFrameSize, long sign)
 
 	for (i = 2; i < 2 * fftFrameSize - 2; i += 2) {
 		for (bitm = 2, j = 0; bitm < 2 * fftFrameSize; bitm <<= 1) {
-			if (i & bitm)
+			if (i & bitm) {
 				j++;
+			}
 			j <<= 1;
 		}
 		if (i < j) {
diff --git a/servers/audio/effects/eq.cpp b/servers/audio/effects/eq.cpp
index 59b3ba2d0b..08a6cf55fa 100644
--- a/servers/audio/effects/eq.cpp
+++ b/servers/audio/effects/eq.cpp
@@ -42,22 +42,25 @@ static int solve_quadratic(double a, double b, double c, double *r1, double *r2)
 	//solves quadractic and returns number of roots
 
 	double base = 2 * a;
-	if (base == 0.0f)
+	if (base == 0.0f) {
 		return 0;
+	}
 
 	double squared = b * b - 4 * a * c;
-	if (squared < 0.0)
+	if (squared < 0.0) {
 		return 0;
+	}
 
 	squared = sqrt(squared);
 
 	*r1 = (-b + squared) / base;
 	*r2 = (-b - squared) / base;
 
-	if (*r1 == *r2)
+	if (*r1 == *r2) {
 		return 1;
-	else
+	} else {
 		return 2;
+	}
 }
 
 EQ::BandProcess::BandProcess() {
diff --git a/servers/audio/effects/reverb.cpp b/servers/audio/effects/reverb.cpp
index 99f60557e1..7c35d88ced 100644
--- a/servers/audio/effects/reverb.cpp
+++ b/servers/audio/effects/reverb.cpp
@@ -57,22 +57,27 @@ const float Reverb::allpass_tunings[MAX_ALLPASS] = {
 };
 
 void Reverb::process(float *p_src, float *p_dst, int p_frames) {
-	if (p_frames > INPUT_BUFFER_MAX_SIZE)
+	if (p_frames > INPUT_BUFFER_MAX_SIZE) {
 		p_frames = INPUT_BUFFER_MAX_SIZE;
+	}
 
 	int predelay_frames = lrint((params.predelay / 1000.0) * params.mix_rate);
-	if (predelay_frames < 10)
+	if (predelay_frames < 10) {
 		predelay_frames = 10;
-	if (predelay_frames >= echo_buffer_size)
+	}
+	if (predelay_frames >= echo_buffer_size) {
 		predelay_frames = echo_buffer_size - 1;
+	}
 
 	for (int i = 0; i < p_frames; i++) {
-		if (echo_buffer_pos >= echo_buffer_size)
+		if (echo_buffer_pos >= echo_buffer_size) {
 			echo_buffer_pos = 0;
+		}
 
 		int read_pos = echo_buffer_pos - predelay_frames;
-		while (read_pos < 0)
+		while (read_pos < 0) {
 			read_pos += echo_buffer_size;
+		}
 
 		float in = undenormalise(echo_buffer[read_pos] * params.predelay_fb + p_src[i]);
 
@@ -104,8 +109,9 @@ void Reverb::process(float *p_src, float *p_dst, int p_frames) {
 
 		int size_limit = c.size - lrintf((float)c.extra_spread_frames * (1.0 - params.extra_spread));
 		for (int j = 0; j < p_frames; j++) {
-			if (c.pos >= size_limit) //reset this now just in case
+			if (c.pos >= size_limit) { //reset this now just in case
 				c.pos = 0;
+			}
 
 			float out = undenormalise(c.buffer[c.pos] * c.feedback);
 			out = out * (1.0 - c.damp) + c.damp_h * c.damp; //lowpass
@@ -156,8 +162,9 @@ void Reverb::process(float *p_src, float *p_dst, int p_frames) {
 		int size_limit = a.size - lrintf((float)a.extra_spread_frames * (1.0 - params.extra_spread));
 
 		for (int j = 0; j < p_frames; j++) {
-			if (a.pos >= size_limit)
+			if (a.pos >= size_limit) {
 				a.pos = 0;
+			}
 
 			float aux = a.buffer[a.pos];
 			a.buffer[a.pos] = undenormalise(allpass_feedback * aux + p_dst[j]);
@@ -200,10 +207,12 @@ void Reverb::set_predelay_feedback(float p_predelay_fb) {
 }
 
 void Reverb::set_highpass(float p_frq) {
-	if (p_frq > 1)
+	if (p_frq > 1) {
 		p_frq = 1;
-	if (p_frq < 0)
+	}
+	if (p_frq < 0) {
 		p_frq = 0;
+	}
 	params.hpf = p_frq;
 }
 
@@ -230,13 +239,15 @@ void Reverb::configure_buffers() {
 		c.extra_spread_frames = lrint(params.extra_spread_base * params.mix_rate);
 
 		int len = lrint(comb_tunings[i] * params.mix_rate) + c.extra_spread_frames;
-		if (len < 5)
+		if (len < 5) {
 			len = 5; //may this happen?
+		}
 
 		c.buffer = memnew_arr(float, len);
 		c.pos = 0;
-		for (int j = 0; j < len; j++)
+		for (int j = 0; j < len; j++) {
 			c.buffer[j] = 0;
+		}
 		c.size = len;
 	}
 
@@ -246,13 +257,15 @@ void Reverb::configure_buffers() {
 		a.extra_spread_frames = lrint(params.extra_spread_base * params.mix_rate);
 
 		int len = lrint(allpass_tunings[i] * params.mix_rate) + a.extra_spread_frames;
-		if (len < 5)
+		if (len < 5) {
 			len = 5; //may this happen?
+		}
 
 		a.buffer = memnew_arr(float, len);
 		a.pos = 0;
-		for (int j = 0; j < len; j++)
+		for (int j = 0; j < len; j++) {
 			a.buffer[j] = 0;
+		}
 		a.size = len;
 	}
 
@@ -273,10 +286,11 @@ void Reverb::update_parameters() {
 	for (int i = 0; i < MAX_COMBS; i++) {
 		Comb &c = comb[i];
 		c.feedback = room_offset + params.room_size * room_scale;
-		if (c.feedback < room_offset)
+		if (c.feedback < room_offset) {
 			c.feedback = room_offset;
-		else if (c.feedback > (room_offset + room_scale))
+		} else if (c.feedback > (room_offset + room_scale)) {
 			c.feedback = (room_offset + room_scale);
+		}
 
 		float auxdmp = params.damp / 2.0 + 0.5; //only half the range (0.5 .. 1.0  is enough)
 		auxdmp *= auxdmp;
@@ -286,19 +300,22 @@ void Reverb::update_parameters() {
 }
 
 void Reverb::clear_buffers() {
-	if (echo_buffer)
+	if (echo_buffer) {
 		memdelete_arr(echo_buffer);
+	}
 
 	for (int i = 0; i < MAX_COMBS; i++) {
-		if (comb[i].buffer)
+		if (comb[i].buffer) {
 			memdelete_arr(comb[i].buffer);
+		}
 
 		comb[i].buffer = nullptr;
 	}
 
 	for (int i = 0; i < MAX_ALLPASS; i++) {
-		if (allpass[i].buffer)
+		if (allpass[i].buffer) {
 			memdelete_arr(allpass[i].buffer);
+		}
 
 		allpass[i].buffer = nullptr;
 	}