module: audio: circ_buf: return sample count as size_t

Change the return type of cir_buf_samples_without_wrap_s16() and
cir_buf_samples_without_wrap_s32() from int to size_t, and update the
sample-count variables in all callers (volume, mixer, aria, asrc).

The returned value is a non-negative element count derived from pointer
arithmetic, so size_t matches the rest of the circular-buffer API.

Signed-off-by: Adrian Warecki <adrian.warecki@intel.com>

Author: softwarecki

src/audio/aria/aria_generic.c

@@ -47,19 +47,20 @@ static void aria_algo_get_data(struct processing_module *mod,
 	int32_t gain_begin = cd->gains[sof_aria_index_tab[gain_state_add_2]];
 	/* do linear approximation between points gain_begin and gain_end */
 	int32_t gain_end = cd->gains[sof_aria_index_tab[gain_state_add_3]];
-	int32_t m, n, i, ch;
-	int32_t samples = frames * audio_stream_get_channels(sink);
+	size_t m, n, i;
+	int ch;
+	size_t samples = frames * audio_stream_get_channels(sink);
 	int32_t *out = audio_stream_get_wptr(sink);
 	int32_t *in = cd->data_ptr;
 	int32_t gain;
 	const int ch_n = cd->chan_cnt;
 	const int shift = 31 - cd->att;
 
-	for (i = 1; i < ARIA_MAX_GAIN_STATES - 1; i++) {
-		if (cd->gains[sof_aria_index_tab[gain_state_add_2 + i]] < gain_begin)
-			gain_begin = cd->gains[sof_aria_index_tab[gain_state_add_2 + i]];
-		if (cd->gains[sof_aria_index_tab[gain_state_add_3 + i]] < gain_end)
-			gain_end = cd->gains[sof_aria_index_tab[gain_state_add_3 + i]];
+	for (ch = 1; ch < ARIA_MAX_GAIN_STATES - 1; ch++) {
+		if (cd->gains[sof_aria_index_tab[gain_state_add_2 + ch]] < gain_begin)
+			gain_begin = cd->gains[sof_aria_index_tab[gain_state_add_2 + ch]];
+		if (cd->gains[sof_aria_index_tab[gain_state_add_3 + ch]] < gain_end)
+			gain_end = cd->gains[sof_aria_index_tab[gain_state_add_3 + ch]];
 	}
 	step = (gain_end - gain_begin) / frames;
 	gain = gain_begin;

src/audio/asrc/asrc.c

@@ -46,8 +46,8 @@ static void src_copy_s32(struct processing_module *mod,
 	int32_t *buf;
 	const int32_t *src = source->ptr;
 	int32_t *snk = sink->ptr;
-	int n_wrap_src;
-	int n_wrap_snk;
+	size_t n_wrap_src;
+	size_t n_wrap_snk;
 	int n_copy;
 	unsigned int n;
 	int ret;
@@ -118,8 +118,8 @@ static void src_copy_s16(struct processing_module *mod,
 	const int16_t *src = source->ptr;
 	int16_t *snk = sink->ptr;
 	int16_t *buf;
-	int n_wrap_src;
-	int n_wrap_snk;
+	size_t n_wrap_src;
+	size_t n_wrap_snk;
 	int n_copy;
 	int s_copy;
 	int ret;

src/audio/dcblock/dcblock_generic.c

@@ -53,12 +53,12 @@ static int dcblock_s16_default(struct comp_data *cd,
 {
 	const int16_t *x = source->ptr;
 	int16_t *y = sink->ptr;
-	int samples_without_wrap;
+	size_t samples_without_wrap;
 	int nch = cd->channels;
-	int remaining_samples = frames * nch;
+	size_t remaining_samples = frames * nch;
 	int32_t tmp;
 	int ch = 0;
-	int i;
+	size_t i;
 
 	while (remaining_samples) {
 		samples_without_wrap = cir_buf_samples_without_wrap_s16(x, source->buf_end);
@@ -100,12 +100,12 @@ static int dcblock_s24_default(struct comp_data *cd,
 {
 	const int32_t *x = source->ptr;
 	int32_t *y = sink->ptr;
-	int samples_without_wrap;
+	size_t samples_without_wrap;
 	int nch = cd->channels;
-	int remaining_samples = frames * nch;
+	size_t remaining_samples = frames * nch;
 	int32_t tmp;
 	int ch = 0;
-	int i;
+	size_t i;
 
 	while (remaining_samples) {
 		samples_without_wrap = cir_buf_samples_without_wrap_s32(x, source->buf_end);
@@ -147,11 +147,11 @@ static int dcblock_s32_default(struct comp_data *cd,
 {
 	const int32_t *x = source->ptr;
 	int32_t *y = sink->ptr;
-	int samples_without_wrap;
+	size_t samples_without_wrap;
 	int nch = cd->channels;
-	int remaining_samples = frames * nch;
+	size_t remaining_samples = frames * nch;
 	int ch = 0;
-	int i;
+	size_t i;
 
 	while (remaining_samples) {
 		samples_without_wrap = cir_buf_samples_without_wrap_s32(x, source->buf_end);

src/audio/drc/drc_generic.c

@@ -508,18 +508,18 @@ static inline void drc_pre_delay_index_inc(int *idx, int increment)
 static void drc_delay_input_sample_s16(struct drc_state *state,
 				       const struct cir_buf_source *source,
 				       struct cir_buf_sink *sink,
-				       const int16_t **x, int16_t **y, int samples, int nch)
+				       const int16_t **x, int16_t **y, size_t samples, int nch)
 {
 	const int16_t *x1;
 	int16_t *y1;
 	int16_t *pd;
 	int pd_write_index, pd_read_index;
-	int nbuf, npcm, nfrm;
+	size_t nbuf, npcm, nfrm;
 	int ch;
-	int i;
+	size_t i;
 	const int16_t *x0 = *x;
 	int16_t *y0 = *y;
-	int remaining_samples = samples;
+	size_t remaining_samples = samples;
 
 	while (remaining_samples) {
 		nbuf = cir_buf_samples_without_wrap_s16(x0, source->buf_end);
@@ -562,11 +562,11 @@ static void drc_s16_default(struct processing_module *mod,
 	int nch = cd->channels;
 	const int16_t *x = (int16_t *)source->ptr;
 	int16_t *y = (int16_t *)sink->ptr;
-	int samples = frames * nch;
+	size_t samples = frames * nch;
 	struct drc_state *state = &cd->state;
 	const struct sof_drc_params *p = &cd->config->params; /* Read-only */
-	int fragment_samples;
-	int fragment;
+	size_t fragment_samples;
+	size_t fragment;
 
 	if (!cd->enabled) {
 		/* Delay the input sample only and don't do other processing. This is used when the
@@ -602,18 +602,18 @@ static void drc_s16_default(struct processing_module *mod,
 static void drc_delay_input_sample_s32(struct drc_state *state,
 				       const struct cir_buf_source *source,
 				       struct cir_buf_sink *sink,
-				       const int32_t **x, int32_t **y, int samples, int nch)
+				       const int32_t **x, int32_t **y, size_t samples, int nch)
 {
 	const int32_t *x1;
 	int32_t *y1;
 	int32_t *pd;
 	int pd_write_index, pd_read_index;
-	int nbuf, npcm, nfrm;
+	size_t nbuf, npcm, nfrm;
 	int ch;
-	int i;
+	size_t i;
 	const int32_t *x0 = *x;
 	int32_t *y0 = *y;
-	int remaining_samples = samples;
+	size_t remaining_samples = samples;
 
 	while (remaining_samples) {
 		nbuf = cir_buf_samples_without_wrap_s32(x0, source->buf_end);
@@ -652,18 +652,18 @@ static void drc_delay_input_sample_s32(struct drc_state *state,
 static void drc_delay_input_sample_s24(struct drc_state *state,
 				       const struct cir_buf_source *source,
 				       struct cir_buf_sink *sink,
-				       const int32_t **x, int32_t **y, int samples, int nch)
+				       const int32_t **x, int32_t **y, size_t samples, int nch)
 {
 	const int32_t *x1;
 	int32_t *y1;
 	int32_t *pd;
 	int pd_write_index, pd_read_index;
-	int nbuf, npcm, nfrm;
+	size_t nbuf, npcm, nfrm;
 	int ch;
-	int i;
+	size_t i;
 	const int32_t *x0 = *x;
 	int32_t *y0 = *y;
-	int remaining_samples = samples;
+	size_t remaining_samples = samples;
 
 	while (remaining_samples) {
 		nbuf = cir_buf_samples_without_wrap_s32(x0, source->buf_end);
@@ -706,11 +706,11 @@ static void drc_s24_default(struct processing_module *mod,
 	int nch = cd->channels;
 	const int32_t *x = (int32_t *)source->ptr;
 	int32_t *y = (int32_t *)sink->ptr;
-	int samples = frames * nch;
+	size_t samples = frames * nch;
 	struct drc_state *state = &cd->state;
 	const struct sof_drc_params *p = &cd->config->params; /* Read-only */
-	int fragment_samples;
-	int fragment;
+	size_t fragment_samples;
+	size_t fragment;
 
 	if (!cd->enabled) {
 		/* Delay the input sample only and don't do other processing. This is used when the
@@ -754,11 +754,11 @@ static void drc_s32_default(struct processing_module *mod,
 	int nch = cd->channels;
 	const int32_t *x = (int32_t *)source->ptr;
 	int32_t *y = (int32_t *)sink->ptr;
-	int samples = frames * nch;
+	size_t samples = frames * nch;
 	struct drc_state *state = &cd->state;
 	const struct sof_drc_params *p = &cd->config->params; /* Read-only */
-	int fragment_samples;
-	int fragment;
+	size_t fragment_samples;
+	size_t fragment;
 
 	if (!cd->enabled) {
 		/* Delay the input sample only and don't do other processing. This is used when the

src/audio/mixer/mixer_generic.c

@@ -19,11 +19,9 @@ static void mix_n_s16(struct comp_dev *dev, struct audio_stream *sink,
 	int16_t *src[PLATFORM_MAX_CHANNELS];
 	int16_t *dest;
 	int32_t val;
-	int nmax;
-	int i, j, n, ns;
-	int processed = 0;
-	int nch = audio_stream_get_channels(sink);
-	int samples = frames * nch;
+	size_t nmax, ns, n, i, processed = 0;
+	uint32_t j;
+	int samples = frames * audio_stream_get_channels(sink);
 
 	dest = audio_stream_get_wptr(sink);
 	for (j = 0; j < num_sources; j++)
@@ -33,8 +31,8 @@ static void mix_n_s16(struct comp_dev *dev, struct audio_stream *sink,
 		nmax = samples - processed;
 		n = audio_stream_samples_without_wrap_s16(sink, dest);
 		n = MIN(n, nmax);
-		for (i = 0; i < num_sources; i++) {
-			ns = audio_stream_samples_without_wrap_s16(sources[i], src[i]);
+		for (j = 0; j < num_sources; j++) {
+			ns = audio_stream_samples_without_wrap_s16(sources[j], src[j]);
 			n = MIN(n, ns);
 		}
 		for (i = 0; i < n; i++) {
@@ -50,8 +48,8 @@ static void mix_n_s16(struct comp_dev *dev, struct audio_stream *sink,
 		}
 		processed += n;
 		dest = audio_stream_wrap(sink, dest);
-		for (i = 0; i < num_sources; i++)
-			src[i] = audio_stream_wrap(sources[i], src[i]);
+		for (j = 0; j < num_sources; j++)
+			src[j] = audio_stream_wrap(sources[j], src[j]);
 	}
 }
 #endif /* CONFIG_FORMAT_S16LE */
@@ -63,14 +61,12 @@ static void mix_n_s24(struct comp_dev *dev, struct audio_stream *sink,
 		      uint32_t frames)
 {
 	int32_t *src[PLATFORM_MAX_CHANNELS];
+	size_t nmax, ns, n, i, processed = 0;
 	int32_t *dest;
+	uint32_t j;
 	int32_t val;
 	int32_t x;
-	int nmax;
-	int i, j, n, ns;
-	int processed = 0;
-	int nch = audio_stream_get_channels(sink);
-	int samples = frames * nch;
+	int samples = frames * audio_stream_get_channels(sink);
 
 	dest = audio_stream_get_wptr(sink);
 	for (j = 0; j < num_sources; j++)
@@ -80,8 +76,8 @@ static void mix_n_s24(struct comp_dev *dev, struct audio_stream *sink,
 		nmax = samples - processed;
 		n = audio_stream_samples_without_wrap_s24(sink, dest);
 		n = MIN(n, nmax);
-		for (i = 0; i < num_sources; i++) {
-			ns = audio_stream_samples_without_wrap_s24(sources[i], src[i]);
+		for (j = 0; j < num_sources; j++) {
+			ns = audio_stream_samples_without_wrap_s24(sources[j], src[j]);
 			n = MIN(n, ns);
 		}
 		for (i = 0; i < n; i++) {
@@ -98,8 +94,8 @@ static void mix_n_s24(struct comp_dev *dev, struct audio_stream *sink,
 		}
 		processed += n;
 		dest = audio_stream_wrap(sink, dest);
-		for (i = 0; i < num_sources; i++)
-			src[i] = audio_stream_wrap(sources[i], src[i]);
+		for (j = 0; j < num_sources; j++)
+			src[j] = audio_stream_wrap(sources[j], src[j]);
 	}
 }
 #endif /* CONFIG_FORMAT_S24LE */
@@ -111,13 +107,11 @@ static void mix_n_s32(struct comp_dev *dev, struct audio_stream *sink,
 		      uint32_t frames)
 {
 	int32_t *src[PLATFORM_MAX_CHANNELS];
+	size_t nmax, ns, i, n, processed = 0;
 	int32_t *dest;
 	int64_t val;
-	int nmax;
-	int i, j, n, ns;
-	int processed = 0;
-	int nch = audio_stream_get_channels(sink);
-	int samples = frames * nch;
+	uint32_t j;
+	size_t samples = frames * audio_stream_get_channels(sink);
 
 	dest = audio_stream_get_wptr(sink);
 	for (j = 0; j < num_sources; j++)
@@ -127,8 +121,8 @@ static void mix_n_s32(struct comp_dev *dev, struct audio_stream *sink,
 		nmax = samples - processed;
 		n = audio_stream_samples_without_wrap_s32(sink, dest);
 		n = MIN(n, nmax);
-		for (i = 0; i < num_sources; i++) {
-			ns = audio_stream_samples_without_wrap_s32(sources[i], src[i]);
+		for (j = 0; j < num_sources; j++) {
+			ns = audio_stream_samples_without_wrap_s32(sources[j], src[j]);
 			n = MIN(n, ns);
 		}
 		for (i = 0; i < n; i++) {
@@ -144,8 +138,8 @@ static void mix_n_s32(struct comp_dev *dev, struct audio_stream *sink,
 		}
 		processed += n;
 		dest = audio_stream_wrap(sink, dest);
-		for (i = 0; i < num_sources; i++)
-			src[i] = audio_stream_wrap(sources[i], src[i]);
+		for (j = 0; j < num_sources; j++)
+			src[j] = audio_stream_wrap(sources[j], src[j]);
 	}
 }
 #endif /* CONFIG_FORMAT_S32LE */

src/audio/mixer/mixer_hifi3.c

@@ -26,8 +26,9 @@ static void mix_n_s16(struct comp_dev *dev, struct audio_stream *sink,
 	ae_int32x2 val2;
 	ae_int32x2 sample_1;
 	ae_int32x2 sample_2;
-	unsigned int n, m, nmax, i, j, left_samples;
-	unsigned int samples = frames * audio_stream_get_channels(sink);
+	size_t n, m, nmax, i, left_samples;
+	uint32_t j;
+	size_t samples = frames * audio_stream_get_channels(sink);
 
 	for (j = 0; j < num_sources; j++)
 		in[j] = audio_stream_get_rptr(sources[j]);
@@ -78,8 +79,9 @@ static void mix_n_s24(struct comp_dev *dev, struct audio_stream *sink,
 	ae_int32x2 *out = audio_stream_get_wptr(sink);
 	ae_int32x2 val;
 	ae_int32x2 sample = AE_ZERO32();
-	unsigned int n, m, nmax, i, j, left_samples;
-	unsigned int samples = frames * audio_stream_get_channels(sink);
+	size_t n, m, nmax, i, left_samples;
+	uint32_t j;
+	size_t samples = frames * audio_stream_get_channels(sink);
 
 	for (j = 0; j < num_sources; j++)
 		in[j] = audio_stream_get_rptr(sources[j]);
@@ -124,9 +126,10 @@ static void mix_n_s32(struct comp_dev *dev, struct audio_stream *sink,
 	ae_int64 sample;
 	ae_int64 val;
 	ae_int32x2 res;
-	unsigned int n, nmax, i, j, left_samples;
-	unsigned int m = 0;
-	unsigned int samples = frames * audio_stream_get_channels(sink);
+	size_t n, nmax, i, left_samples;
+	size_t m = 0;
+	uint32_t j;
+	size_t samples = frames * audio_stream_get_channels(sink);
 
 	for (j = 0; j < num_sources; j++)
 		in[j] = audio_stream_get_rptr(sources[j]);