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buffer: use an accessor to read struct audio_stream::frame_fmt 

This patch adds audio_stream_get_frm_fmt() to read
struct audio_stream::frame_fmt and converts all users. Apart from
hiding internals of the structure, it also enforces the use of the
__sparse_cache annotation.

Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
This commit is contained in:
Guennadi Liakhovetski 2023-04-21 14:53:03 +02:00 committed by Liam Girdwood
parent 3ab21638cf
commit 017b8ebcde
36 changed files with 98 additions and 84 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
src/audio/asrc/asrc.c
View File

@ -754,12 +754,12 @@ static int asrc_prepare(struct comp_dev *dev)
sink_c = buffer_acquire(sinkb);
/* get source data format and period bytes */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
source_period_bytes = audio_stream_period_bytes(&source_c->stream,
cd->source_frames);
/* get sink data format and period bytes */
cd->sink_format = sink_c->stream.frame_fmt;
cd->sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
sink_period_bytes = audio_stream_period_bytes(&sink_c->stream,
cd->sink_frames);
@ -785,7 +785,7 @@ static int asrc_prepare(struct comp_dev *dev)
}
/* ASRC supports S16_LE, S24_4LE and S32_LE formats */
switch (source_c->stream.frame_fmt) {
switch (audio_stream_get_frm_fmt(&source_c->stream)) {
case SOF_IPC_FRAME_S16_LE:
cd->asrc_func = src_copy_s16;
break;

2
src/audio/copier/copier.c
View File

@ -1410,7 +1410,7 @@ static void copier_dma_cb(void *arg, enum notify_id type, void *data)
else
sink = buffer_acquire(cd->hd->dma_buffer);
frames = bytes / get_sample_bytes(sink->stream.frame_fmt);
frames = bytes / get_sample_bytes(audio_stream_get_frm_fmt(&sink->stream));
frames = frames / sink->stream.channels;
ret = apply_attenuation(dev, cd, sink, frames);

5
src/audio/copier/copier_generic.c
View File

@ -30,7 +30,7 @@ int apply_attenuation(struct comp_dev *dev, struct copier_data *cd,
int32_t *dst = audio_stream_get_rptr(&sink->stream);
/* only support attenuation in format of 32bit */
switch (sink->stream.frame_fmt) {
switch (audio_stream_get_frm_fmt(&sink->stream)) {
case SOF_IPC_FRAME_S16_LE:
comp_err(dev, "16bit sample isn't supported by attenuation");
return -EINVAL;
@ -49,7 +49,8 @@ int apply_attenuation(struct comp_dev *dev, struct copier_data *cd,
return 0;
default:
comp_err(dev, "unsupported format %d for attenuation", sink->stream.frame_fmt);
comp_err(dev, "unsupported format %d for attenuation",
audio_stream_get_frm_fmt(&sink->stream));
return -EINVAL;
}
}

5
src/audio/copier/copier_hifi.c
View File

@ -36,7 +36,7 @@ int apply_attenuation(struct comp_dev *dev, struct copier_data *cd,
ae_int32x2 *out = (ae_int32x2 *)dst;
/* only support attenuation in format of 32bit */
switch (sink->stream.frame_fmt) {
switch (audio_stream_get_frm_fmt(&sink->stream)) {
case SOF_IPC_FRAME_S16_LE:
comp_err(dev, "16bit sample isn't supported by attenuation");
return -EINVAL;
@ -66,7 +66,8 @@ int apply_attenuation(struct comp_dev *dev, struct copier_data *cd,
return 0;
default:
comp_err(dev, "unsupported format %d for attenuation", sink->stream.frame_fmt);
comp_err(dev, "unsupported format %d for attenuation",
audio_stream_get_frm_fmt(&sink->stream));
return -EINVAL;
}
}

6
src/audio/crossover/crossover.c
View File

@ -716,16 +716,16 @@ static int crossover_prepare(struct comp_dev *dev)
source_c = buffer_acquire(source);
/* Get source data format */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
/* Validate frame format and buffer size of sinks */
list_for_item(sink_list, &dev->bsink_list) {
sink = container_of(sink_list, struct comp_buffer, source_list);
sink_c = buffer_acquire(sink);
if (cd->source_format != sink_c->stream.frame_fmt) {
if (cd->source_format != audio_stream_get_frm_fmt(&sink_c->stream)) {
comp_err(dev, "crossover_prepare(): Source fmt %d and sink fmt %d are different for sink %d.",
cd->source_format, sink_c->stream.frame_fmt,
cd->source_format, audio_stream_get_frm_fmt(&sink_c->stream),
sink_c->pipeline_id);
ret = -EINVAL;
} else {

10
src/audio/dai-legacy.c
View File

@ -332,8 +332,8 @@ static int dai_playback_params(struct comp_dev *dev, uint32_t period_bytes,
struct dma_sg_config *config = &dd->config;
struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer),
*local_buf = buffer_acquire(dd->local_buffer);
uint32_t local_fmt = local_buf->stream.frame_fmt;
uint32_t dma_fmt = dma_buf->stream.frame_fmt;
uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream);
uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream);
uint32_t fifo;
int err = 0;
@ -394,8 +394,8 @@ static int dai_capture_params(struct comp_dev *dev, uint32_t period_bytes,
struct dma_sg_config *config = &dd->config;
struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer),
*local_buf = buffer_acquire(dd->local_buffer);
uint32_t local_fmt = local_buf->stream.frame_fmt;
uint32_t dma_fmt = dma_buf->stream.frame_fmt;
uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream);
uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream);
uint32_t fifo;
int err = 0;
@ -941,7 +941,7 @@ static int dai_copy(struct comp_dev *dev)
buf_c = buffer_acquire(dd->dma_buffer);
dma_fmt = buf_c->stream.frame_fmt;
dma_fmt = audio_stream_get_frm_fmt(&buf_c->stream);
sampling = get_sample_bytes(dma_fmt);
buffer_release(buf_c);

10
src/audio/dai-zephyr.c
View File

@ -441,8 +441,8 @@ static int dai_playback_params(struct comp_dev *dev, uint32_t period_bytes,
struct dma_block_config *prev = NULL;
struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer),
*local_buf = buffer_acquire(dd->local_buffer);
uint32_t local_fmt = local_buf->stream.frame_fmt;
uint32_t dma_fmt = dma_buf->stream.frame_fmt;
uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream);
uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream);
uint32_t fifo, max_block_count, buf_size;
int i, err = 0;
@ -579,8 +579,8 @@ static int dai_capture_params(struct comp_dev *dev, uint32_t period_bytes,
struct dma_block_config *prev = NULL;
struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer),
*local_buf = buffer_acquire(dd->local_buffer);
uint32_t local_fmt = local_buf->stream.frame_fmt;
uint32_t dma_fmt = dma_buf->stream.frame_fmt;
uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream);
uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream);
uint32_t fifo, max_block_count, buf_size;
int i, err = 0;
@ -1241,7 +1241,7 @@ static int dai_copy(struct comp_dev *dev)
buf_c = buffer_acquire(dd->dma_buffer);
dma_fmt = buf_c->stream.frame_fmt;
dma_fmt = audio_stream_get_frm_fmt(&buf_c->stream);
sampling = get_sample_bytes(dma_fmt);
buffer_release(buf_c);

4
src/audio/dcblock/dcblock.c
View File

@ -359,10 +359,10 @@ static int dcblock_prepare(struct comp_dev *dev)
sink_c = buffer_acquire(sinkb);
/* get source data format */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
/* get sink data format and period bytes */
cd->sink_format = sink_c->stream.frame_fmt;
cd->sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
sink_period_bytes = audio_stream_period_bytes(&sink_c->stream, dev->frames);
if (audio_stream_get_size(&sink_c->stream) < sink_period_bytes) {

6
src/audio/drc/drc.c
View File

@ -391,7 +391,7 @@ static int drc_prepare(struct comp_dev *dev)
source_c = buffer_acquire(sourceb);
/* get source data format */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
/* Initialize DRC */
comp_info(dev, "drc_prepare(), source_format=%d, sink_format=%d",
@ -419,9 +419,9 @@ static int drc_prepare(struct comp_dev *dev)
sink_c = buffer_acquire(sinkb);
/* validate sink data format and period bytes */
if (cd->source_format != sink_c->stream.frame_fmt) {
if (cd->source_format != audio_stream_get_frm_fmt(&sink_c->stream)) {
comp_err(dev, "drc_prepare(): Source fmt %d and sink fmt %d are different.",
cd->source_format, sink_c->stream.frame_fmt);
cd->source_format, audio_stream_get_frm_fmt(&sink_c->stream));
ret = -EINVAL;
goto out_sink;
}

4
src/audio/eq_fir/eq_fir.c
View File

@ -529,7 +529,7 @@ static int eq_fir_process(struct processing_module *mod,
return ret;
} else if (cd->fir_delay_size) {
comp_dbg(mod->dev, "eq_fir_process(), active");
ret = set_fir_func(mod, source->frame_fmt);
ret = set_fir_func(mod, audio_stream_get_frm_fmt(source));
if (ret < 0)
return ret;
} else {
@ -593,7 +593,7 @@ static int eq_fir_prepare(struct processing_module *mod)
sink_c = buffer_acquire(sinkb);
eq_fir_set_alignment(&source_c->stream, &sink_c->stream);
channels = sink_c->stream.channels;
frame_fmt = source_c->stream.frame_fmt;
frame_fmt = audio_stream_get_frm_fmt(&source_c->stream);
buffer_release(sink_c);
buffer_release(source_c);

12
src/audio/eq_iir/eq_iir.c
View File

@ -720,8 +720,8 @@ static int eq_iir_verify_params(struct comp_dev *dev,
* pcm frame_fmt and will not make any conversion (sink and source
* frame_fmt will be equal).
*/
buffer_flag = eq_iir_find_func(source_c->stream.frame_fmt,
sink_c->stream.frame_fmt, fm_configured,
buffer_flag = eq_iir_find_func(audio_stream_get_frm_fmt(&source_c->stream),
audio_stream_get_frm_fmt(&sink_c->stream), fm_configured,
ARRAY_SIZE(fm_configured)) ?
BUFF_PARAMS_FRAME_FMT : 0;
@ -808,8 +808,8 @@ static int eq_iir_process(struct processing_module *mod,
/* Check for changed configuration */
if (comp_is_new_data_blob_available(cd->model_handler)) {
cd->config = comp_get_data_blob(cd->model_handler, NULL, NULL);
ret = eq_iir_new_blob(mod, cd, source->frame_fmt,
sink->frame_fmt, source->channels);
ret = eq_iir_new_blob(mod, cd, audio_stream_get_frm_fmt(source),
audio_stream_get_frm_fmt(sink), source->channels);
if (ret)
return ret;
}
@ -919,8 +919,8 @@ static int eq_iir_prepare(struct processing_module *mod)
/* get source and sink data format */
channels = sink_c->stream.channels;
source_format = source_c->stream.frame_fmt;
sink_format = sink_c->stream.frame_fmt;
source_format = audio_stream_get_frm_fmt(&source_c->stream);
sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
buffer_release(sink_c);
buffer_release(source_c);

2
src/audio/google/google_hotword_detect.c
View File

@ -181,7 +181,7 @@ static int ghd_params(struct comp_dev *dev,
if (source_c->stream.channels != 1) {
comp_err(dev, "ghd_params(): Only single-channel supported");
ret = -EINVAL;
} else if (source_c->stream.frame_fmt != SOF_IPC_FRAME_S16_LE) {
} else if (audio_stream_get_frm_fmt(&source_c->stream) != SOF_IPC_FRAME_S16_LE) {
comp_err(dev, "ghd_params(): Only S16_LE supported");
ret = -EINVAL;
} else if (source_c->stream.rate != KPB_SAMPLNG_FREQUENCY) {

2
src/audio/google/google_rtc_audio_processing.c
View File

@ -555,7 +555,7 @@ static int google_rtc_audio_processing_prepare(struct comp_dev *dev)
}
output_c = buffer_acquire(cd->output);
frame_fmt = output_c->stream.frame_fmt;
frame_fmt = audio_stream_get_frm_fmt(&output_c->stream);
rate = output_c->stream.rate;
buffer_release(output_c);

4
src/audio/host-legacy.c
View File

@ -851,8 +851,8 @@ int host_zephyr_params(struct host_data *hd, struct comp_dev *dev,
host_copy_normal;
/* set processing function */
hd->process = pcm_get_conversion_function(host_buf_c->stream.frame_fmt,
host_buf_c->stream.frame_fmt);
hd->process = pcm_get_conversion_function(audio_stream_get_frm_fmt(&host_buf_c->stream),
audio_stream_get_frm_fmt(&host_buf_c->stream));
out:
buffer_release(host_buf_c);

10
src/audio/host-zephyr.c
View File

@ -860,11 +860,13 @@ int host_zephyr_params(struct host_data *hd, struct comp_dev *dev,
/* set processing function */
if (params->direction == SOF_IPC_STREAM_CAPTURE)
hd->process = pcm_get_conversion_function(host_buf_c->stream.frame_fmt,
dma_buf_c->stream.frame_fmt);
hd->process = pcm_get_conversion_function(
audio_stream_get_frm_fmt(&host_buf_c->stream),
audio_stream_get_frm_fmt(&dma_buf_c->stream));
else
hd->process = pcm_get_conversion_function(dma_buf_c->stream.frame_fmt,
host_buf_c->stream.frame_fmt);
hd->process = pcm_get_conversion_function(
audio_stream_get_frm_fmt(&dma_buf_c->stream),
audio_stream_get_frm_fmt(&host_buf_c->stream));
config->src_width = audio_stream_sample_bytes(&dma_buf_c->stream);
config->dest_width = config->src_width;

2
src/audio/igo_nr/igo_nr.c
View File

@ -249,7 +249,7 @@ static inline int32_t set_capture_func(struct comp_dev *dev)
sink_list);
/* The igo_nr supports S16_LE data. Format converter is needed. */
switch (sourceb->stream.frame_fmt) {
switch (audio_stream_get_frm_fmt(&sourceb->stream)) {
#if CONFIG_FORMAT_S16LE
case SOF_IPC_FRAME_S16_LE:
comp_info(dev, "set_capture_func(), SOF_IPC_FRAME_S16_LE");

4
src/audio/mfcc/mfcc.c
View File

@ -209,13 +209,13 @@ static int mfcc_prepare(struct processing_module *mod)
sink_c = buffer_acquire(sinkb);
/* get source data format */
source_format = source_c->stream.frame_fmt;
source_format = audio_stream_get_frm_fmt(&source_c->stream);
/* set align requirements */
mfcc_set_alignment(&source_c->stream, &sink_c->stream);
/* get sink data format and period bytes */
sink_format = sink_c->stream.frame_fmt;
sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
sink_period_bytes = audio_stream_period_bytes(&sink_c->stream, dev->frames);
comp_info(dev, "mfcc_prepare(), source_format = %d, sink_format = %d",
source_format, sink_format);

2
src/audio/module_adapter/module/dts.c
View File

@ -93,7 +93,7 @@ static int dts_effect_populate_buffer_configuration(struct comp_dev *dev,
buffer_fmt = source_c->buffer_fmt;
stream = &source_c->stream;
frame_fmt = stream->frame_fmt;
frame_fmt = audio_stream_get_frm_fmt(stream);
rate = stream->rate;
channels = stream->channels;

10
src/audio/module_adapter/module/waves.c
View File

@ -251,9 +251,9 @@ static int waves_effect_check(struct comp_dev *dev)
}
/* different frame format not supported */
if (src_fmt->frame_fmt != snk_fmt->frame_fmt) {
if (src_fmt->frame_fmt != audio_stream_get_frm_fmt(snk_fmt)) {
comp_err(dev, "waves_effect_check() source %d sink %d sample format mismatch",
src_fmt->frame_fmt, snk_fmt->frame_fmt);
audio_stream_get_frm_fmt(src_fmt), audio_stream_get_frm_fmt(snk_fmt));
ret = -EINVAL;
goto out;
}
@ -265,7 +265,7 @@ static int waves_effect_check(struct comp_dev *dev)
goto out;
}
if (!format_is_supported(src_fmt->frame_fmt)) {
if (!format_is_supported(audio_stream_get_frm_fmt(src_fmt))) {
comp_err(dev, "waves_effect_check() float samples not supported");
ret = -EINVAL;
goto out;
@ -318,10 +318,10 @@ static int waves_effect_init(struct processing_module *mod)
comp_dbg(dev, "waves_effect_init() start");
sample_format = format_convert_sof_to_me(src_fmt->frame_fmt);
sample_format = format_convert_sof_to_me(audio_stream_get_frm_fmt(src_fmt));
if (sample_format < 0) {
comp_err(dev, "waves_effect_init() sof sample format %d not supported",
src_fmt->frame_fmt);
audio_stream_get_frm_fmt(src_fmt));
ret = -EINVAL;
goto out;
}

6
src/audio/multiband_drc/multiband_drc.c
View File

@ -534,7 +534,7 @@ static int multiband_drc_prepare(struct comp_dev *dev)
source_c = buffer_acquire(sourceb);
/* get source data format */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
/* Initialize DRC */
comp_dbg(dev, "multiband_drc_prepare(), source_format=%d, sink_format=%d",
@ -574,10 +574,10 @@ static int multiband_drc_prepare(struct comp_dev *dev)
sink_c = buffer_acquire(sinkb);
/* validate sink data format and period bytes */
if (cd->source_format != sink_c->stream.frame_fmt) {
if (cd->source_format != audio_stream_get_frm_fmt(&sink_c->stream)) {
comp_err(dev,
"multiband_drc_prepare(): Source fmt %d and sink fmt %d are different.",
cd->source_format, sink_c->stream.frame_fmt);
cd->source_format, audio_stream_get_frm_fmt(&sink_c->stream));
ret = -EINVAL;
goto out_sink;
}

2
src/audio/mux/mux.c
View File

@ -306,7 +306,7 @@ static void set_mux_params(struct processing_module *mod)
sink_c->stream.valid_sample_fmt = valid_fmt;
sink_c->buffer_fmt = out_fmt.interleaving_style;
params->frame_fmt = sink_c->stream.frame_fmt;
params->frame_fmt = audio_stream_get_frm_fmt(&sink_c->stream);
for (i = 0; i < SOF_IPC_MAX_CHANNELS; i++)
sink_c->chmap[i] = (out_fmt.ch_map >> i * 4) & 0xf;

4
src/audio/mux/mux_generic.c
View File

@ -535,7 +535,7 @@ mux_func mux_get_processing_function(struct processing_module *mod)
for (i = 0; i < ARRAY_SIZE(mux_func_map); i++) {
struct comp_buffer __sparse_cache *sink_c = buffer_acquire(sinkb);
enum sof_ipc_frame fmt = sink_c->stream.frame_fmt;
enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&sink_c->stream);
if (fmt == mux_func_map[i].frame_format)
@ -559,7 +559,7 @@ demux_func demux_get_processing_function(struct processing_module *mod)
for (i = 0; i < ARRAY_SIZE(mux_func_map); i++) {
struct comp_buffer __sparse_cache *source_c = buffer_acquire(sourceb);
enum sof_ipc_frame fmt = source_c->stream.frame_fmt;
enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&source_c->stream);
buffer_release(source_c);

2
src/audio/pipeline/pipeline-params.c
View File

@ -138,7 +138,7 @@ static void pipeline_update_buffer_pcm_params(struct comp_buffer __sparse_cache
int i;
params->buffer_fmt = buffer->buffer_fmt;
params->frame_fmt = buffer->stream.frame_fmt;
params->frame_fmt = audio_stream_get_frm_fmt(&buffer->stream);
params->rate = buffer->stream.rate;
params->channels = buffer->stream.channels;
for (i = 0; i < SOF_IPC_MAX_CHANNELS; i++)

4
src/audio/rtnr/rtnr.c
View File

@ -873,8 +873,8 @@ static int rtnr_prepare(struct comp_dev *dev)
/* Get sink data format */
sinkb = list_first_item(&dev->bsink_list, struct comp_buffer, source_list);
sink_c = buffer_acquire(sinkb);
cd->sink_format = sink_c->stream.frame_fmt;
cd->sink_stream.frame_fmt = sink_c->stream.frame_fmt;
cd->sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
cd->sink_stream.frame_fmt = audio_stream_get_frm_fmt(&sink_c->stream);
buffer_release(sink_c);
/* Check source and sink PCM format and get processing function */

8
src/audio/selector/selector.c
View File

@ -463,11 +463,11 @@ static int selector_prepare(struct comp_dev *dev)
sink_c = buffer_acquire(sinkb);
/* get source data format and period bytes */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
cd->source_period_bytes = audio_stream_period_bytes(&source_c->stream, dev->frames);
/* get sink data format and period bytes */
cd->sink_format = sink_c->stream.frame_fmt;
cd->sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
cd->sink_period_bytes = audio_stream_period_bytes(&sink_c->stream, dev->frames);
/* There is an assumption that sink component will report out
@ -920,11 +920,11 @@ static int selector_prepare(struct processing_module *mod)
audio_stream_init_alignment_constants(4, 1, &sink_c->stream);
/* get source data format and period bytes */
cd->source_format = source_c->stream.frame_fmt;
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
cd->source_period_bytes = audio_stream_period_bytes(&source_c->stream, dev->frames);
/* get sink data format and period bytes */
cd->sink_format = sink_c->stream.frame_fmt;
cd->sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
cd->sink_period_bytes = audio_stream_period_bytes(&sink_c->stream, dev->frames);
/* There is an assumption that sink component will report out

10
src/audio/smart_amp/smart_amp.c
View File

@ -558,8 +558,12 @@ static int smart_amp_process(struct comp_dev *dev,
static smart_amp_proc get_smart_amp_process(struct comp_dev *dev)
{
struct smart_amp_data *sad = comp_get_drvdata(dev);
struct comp_buffer __sparse_cache *source_buf = buffer_acquire(sad->source_buf);
enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&source_buf->stream);
switch (sad->source_buf->stream.frame_fmt) {
buffer_release(source_buf);
switch (fmt) {
case SOF_IPC_FRAME_S16_LE:
case SOF_IPC_FRAME_S24_4LE:
case SOF_IPC_FRAME_S32_LE:
@ -710,7 +714,7 @@ static int smart_amp_prepare(struct comp_dev *dev)
}
}
switch (source_c->stream.frame_fmt) {
switch (audio_stream_get_frm_fmt(&source_c->stream)) {
case SOF_IPC_FRAME_S16_LE:
bitwidth = 16;
break;
@ -722,7 +726,7 @@ static int smart_amp_prepare(struct comp_dev *dev)
break;
default:
comp_err(dev, "[DSM] smart_amp_process() error: not supported frame format %d",
source_c->stream.frame_fmt);
audio_stream_get_frm_fmt(&source_c->stream));
goto error;
}

10
src/audio/smart_amp/smart_amp_maxim_dsm.c
View File

@ -592,7 +592,7 @@ static int smart_amp_get_buffer(int32_t *buf, uint32_t frames,
output.buf16 = (int16_t *)buf;
output.buf32 = (int32_t *)buf;
switch (stream->frame_fmt) {
switch (audio_stream_get_frm_fmt(stream)) {
case SOF_IPC_FRAME_S16_LE:
for (idx = 0 ; idx < frames ; idx++) {
for (ch = 0 ; ch < num_ch; ch++) {
@ -642,7 +642,7 @@ static int smart_amp_put_buffer(int32_t *buf, uint32_t frames,
output.buf16 = audio_stream_get_wptr(stream);
output.buf32 = audio_stream_get_wptr(stream);
switch (stream->frame_fmt) {
switch (audio_stream_get_frm_fmt(stream)) {
case SOF_IPC_FRAME_S16_LE:
for (idx = 0 ; idx < frames ; idx++) {
for (ch = 0 ; ch < num_ch_out; ch++) {
@ -708,7 +708,7 @@ int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames,
if (ret)
goto err;
switch (source->frame_fmt) {
switch (audio_stream_get_frm_fmt(source)) {
case SOF_IPC_FRAME_S16_LE:
maxim_dsm_ff_proc(hspk, dev,
hspk->buf.frame_in,
@ -767,7 +767,7 @@ int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames,
if (ret)
goto err;
switch (source->frame_fmt) {
switch (audio_stream_get_frm_fmt(source)) {
case SOF_IPC_FRAME_S16_LE:
maxim_dsm_fb_proc(hspk, dev, hspk->buf.frame_iv,
frames * num_ch, sizeof(int16_t));
@ -784,6 +784,6 @@ int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames,
return 0;
err:
comp_err(dev, "[DSM] Not supported frame format : %d",
source->frame_fmt);
audio_stream_get_frm_fmt(source));
return ret;
}

8
src/audio/src/src.c
View File

@ -467,7 +467,7 @@ static void src_copy_sxx(struct comp_dev *dev, struct comp_data *cd,
{
int frames = cd->param.blk_in;
switch (sink->frame_fmt) {
switch (audio_stream_get_frm_fmt(sink)) {
case SOF_IPC_FRAME_S16_LE:
case SOF_IPC_FRAME_S24_4LE:
case SOF_IPC_FRAME_S32_LE:
@ -766,7 +766,7 @@ static int src_params_general(struct comp_dev *dev, struct comp_data *cd,
/* Allocate needed memory for delay lines */
comp_info(dev, "src_params(), source_rate = %u, sink_rate = %u, format = %d",
cd->source_rate, cd->sink_rate, source_c->stream.frame_fmt);
cd->source_rate, cd->sink_rate, audio_stream_get_frm_fmt(&source_c->stream));
comp_info(dev, "src_params(), sourceb->channels = %u, sinkb->channels = %u, dev->frames = %u",
source_c->stream.channels, sink_c->stream.channels, dev->frames);
err = src_buffer_lengths(dev, cd, source_c->stream.channels);
@ -861,8 +861,8 @@ static int src_prepare_general(struct comp_dev *dev, struct comp_data *cd)
#endif
/* get source/sink data format */
source_format = source_c->stream.frame_fmt;
sink_format = sink_c->stream.frame_fmt;
source_format = audio_stream_get_frm_fmt(&source_c->stream);
sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
ret = src_check_buffer_sizes(dev, cd, &source_c->stream, &sink_c->stream);
if (ret < 0)

2
src/audio/tdfb/tdfb.c
View File

@ -745,7 +745,7 @@ static int tdfb_prepare(struct processing_module *mod)
sink_c = buffer_acquire(sinkb);
tdfb_set_alignment(&source_c->stream, &sink_c->stream);
frame_fmt = source_c->stream.frame_fmt;
frame_fmt = audio_stream_get_frm_fmt(&source_c->stream);
source_channels = source_c->stream.channels;
sink_channels = sink_c->stream.channels;
rate = source_c->stream.rate;

6
src/include/sof/audio/audio_stream.h
View File

@ -112,6 +112,12 @@ static inline uint32_t audio_stream_get_free(const struct audio_stream __sparse_
return buf->free;
}
static inline enum sof_ipc_frame audio_stream_get_frm_fmt(
const struct audio_stream __sparse_cache *buf)
{
return buf->frame_fmt;
}
/**
* Retrieves readable address of a sample at specified index (see versions of
* this macro specialized for various sample types).

2
src/include/sof/audio/mixer.h
View File

@ -68,7 +68,7 @@ static inline mixer_func mixer_get_processing_function(struct comp_dev *dev,
/* map the volume function for source and sink buffers */
for (i = 0; i < mixer_func_count; i++) {
if (sinkb->stream.frame_fmt != mixer_func_map[i].frame_fmt)
if (audio_stream_get_frm_fmt(&sinkb->stream) != mixer_func_map[i].frame_fmt)
continue;
return mixer_func_map[i].func;

2
src/include/sof/audio/volume.h
View File

@ -190,7 +190,7 @@ static inline vol_scale_func vol_get_processing_function(struct comp_dev *dev,
/* map the volume function for source and sink buffers */
for (i = 0; i < volume_func_count; i++) {
if (sinkb->stream.frame_fmt != volume_func_map[i].frame_fmt)
if (audio_stream_get_frm_fmt(&sinkb->stream) != volume_func_map[i].frame_fmt)
continue;
return volume_func_map[i].func;

2
src/ipc/ipc-helper.c
View File

@ -87,7 +87,7 @@ static void comp_update_params(uint32_t flag,
struct comp_buffer __sparse_cache *buffer)
{
if (flag & BUFF_PARAMS_FRAME_FMT)
params->frame_fmt = buffer->stream.frame_fmt;
params->frame_fmt = audio_stream_get_frm_fmt(&buffer->stream);
if (flag & BUFF_PARAMS_BUFFER_FMT)
params->buffer_fmt = buffer->buffer_fmt;

2
src/probe/probe.c
View File

@ -829,7 +829,7 @@ static void probe_cb_produce(void *arg, enum notify_id type, void *data)
}
if (_probe->probe_points[i].purpose == PROBE_PURPOSE_EXTRACTION) {
format = probe_gen_format(buffer->stream.frame_fmt,
format = probe_gen_format(audio_stream_get_frm_fmt(&buffer->stream),
buffer->stream.rate,
buffer->stream.channels);
ret = probe_gen_header(buffer_id,

2
src/samples/audio/detect_test.c
View File

@ -767,7 +767,7 @@ static int test_keyword_params(struct comp_dev *dev,
sink_list);
source_c = buffer_acquire(sourceb);
channels = source_c->stream.channels;
frame_fmt = source_c->stream.frame_fmt;
frame_fmt = audio_stream_get_frm_fmt(&source_c->stream);
rate = source_c->stream.rate;
buffer_release(source_c);

4
src/samples/audio/smart_amp_test.c
View File

@ -162,7 +162,7 @@ static void smart_amp_set_params(struct comp_dev *dev,
sink_c->stream.valid_sample_fmt = valid_fmt;
sink_c->buffer_fmt = out_fmt.interleaving_style;
params->frame_fmt = sink_c->stream.frame_fmt;
params->frame_fmt = audio_stream_get_frm_fmt(&sink_c->stream);
sink_c->hw_params_configured = true;
@ -636,7 +636,7 @@ static int smart_amp_process_s32(struct comp_dev *dev,
static smart_amp_proc get_smart_amp_process(struct comp_dev *dev,
struct comp_buffer __sparse_cache *buf)
{
switch (buf->stream.frame_fmt) {
switch (audio_stream_get_frm_fmt(&buf->stream)) {
case SOF_IPC_FRAME_S16_LE:
return smart_amp_process_s16;
case SOF_IPC_FRAME_S24_4LE: