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[DNM] POC: ] Direct use of cached / not-cached shared memory alias instead of coherent.h in case of buffers #8013

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[DNM] POC: ] Direct use of cached / not-cached shared memory alias instead of coherent.h in case of buffers #8013

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fc92971
buf: remove coherent.h
marcinszkudlinski Aug 8, 2023
0a96e6a
remove buffer_acquire from asrc.c
marcinszkudlinski Aug 21, 2023
f040f6d
sparse: Remove buffer_acquire() usage
tobonex Aug 23, 2023
543d106
sparse: remove all sparse_cache from struct sof_source
marcinszkudlinski Aug 24, 2023
0b4eef1
sparse: remove all sparse_cache from struct sof_sink
marcinszkudlinski Aug 24, 2023
7b6aaa2
sparse: remove sparse from access to audio_stream
marcinszkudlinski Aug 24, 2023
7d01b51
sparse: remove sparse from audio_stream
marcinszkudlinski Aug 24, 2023
0a9a3b4
sparse: remove __sparse from struct audio_stream references
tobonex Aug 28, 2023
d1b5793
sparse: remove __sparse from struct sof_source
tobonex Aug 28, 2023
c286bc3
sparse: remove remaining __sparse marks
tobonex Aug 29, 2023
d5d02b4
coherent.h change
tobonex Aug 31, 2023
db91ac5
coherent.h change 2
tobonex Aug 31, 2023
63fe1f5
merge buf
tobonex Aug 30, 2023
d0253c1
dla tobonex: selector.c forgotten buffer_c
marcinszkudlinski Aug 31, 2023
0bc2a7c
dla tobonex: helper.c i audio/audio_stream.h
marcinszkudlinski Aug 30, 2023
aa89d44
tobonex: missing changes
marcinszkudlinski Aug 31, 2023
39ee86b
tobonex: dai-zephyr.c incorrect line to be removed
marcinszkudlinski Aug 31, 2023
dcc4363
buf: remove buffer_acquire/buffer_release
marcinszkudlinski Aug 30, 2023
7e92994
te nie mogą być
marcinszkudlinski Sep 1, 2023
be92e40
tez do wywalenia
marcinszkudlinski Sep 1, 2023
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19 changes: 7 additions & 12 deletions src/audio/aria/aria.c
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Original file line number Diff line number Diff line change
Expand Up @@ -72,8 +72,8 @@ static int aria_algo_init(struct aria_data *cd, void *buffer_desc,
}

static inline void aria_process_data(struct processing_module *mod,
struct audio_stream __sparse_cache *source,
struct audio_stream __sparse_cache *sink,
struct audio_stream *source,
struct audio_stream *sink,
size_t frames)
{
struct aria_data *cd = module_get_private_data(mod);
Expand Down Expand Up @@ -153,24 +153,19 @@ static int aria_free(struct processing_module *mod)
static void aria_set_stream_params(struct comp_buffer *buffer,
struct processing_module *mod)
{
struct comp_buffer __sparse_cache *buffer_c;
const struct ipc4_audio_format *audio_fmt = &mod->priv.cfg.base_cfg.audio_fmt;

buffer_c = buffer_acquire(buffer);

ipc4_update_buffer_format(buffer_c, audio_fmt);
ipc4_update_buffer_format(buffer, audio_fmt);
#ifdef ARIA_GENERIC
audio_stream_init_alignment_constants(1, 1, &buffer_c->stream);
audio_stream_init_alignment_constants(1, 1, &buffer->stream);
#else
audio_stream_init_alignment_constants(8, 1, &buffer_c->stream);
audio_stream_init_alignment_constants(8, 1, &buffer->stream);
#endif

buffer_release(buffer_c);
}

static int aria_prepare(struct processing_module *mod,
struct sof_source __sparse_cache **sources, int num_of_sources,
struct sof_sink __sparse_cache **sinks, int num_of_sinks)
struct sof_source **sources, int num_of_sources,
struct sof_sink **sinks, int num_of_sinks)
{
int ret;
struct comp_buffer *source, *sink;
Expand Down
4 changes: 2 additions & 2 deletions src/audio/aria/aria_generic.c
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Original file line number Diff line number Diff line change
Expand Up @@ -20,7 +20,7 @@ const uint8_t INDEX_TAB[] = {
};

inline void aria_algo_calc_gain(struct aria_data *cd, size_t gain_idx,
struct audio_stream __sparse_cache *source, int frames)
struct audio_stream *source, int frames)
{
int32_t max_data = 0;
int32_t sample_abs;
Expand Down Expand Up @@ -50,7 +50,7 @@ inline void aria_algo_calc_gain(struct aria_data *cd, size_t gain_idx,
}

void aria_algo_get_data(struct processing_module *mod,
struct audio_stream __sparse_cache *sink, int frames)
struct audio_stream *sink, int frames)
{
struct aria_data *cd = module_get_private_data(mod);
int32_t step, in_sample;
Expand Down
6 changes: 3 additions & 3 deletions src/audio/aria/aria_hifi3.c
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -21,7 +21,7 @@ const uint8_t INDEX_TAB[] = {
};

inline void aria_algo_calc_gain(struct aria_data *cd, size_t gain_idx,
struct audio_stream __sparse_cache *source, int frames)
struct audio_stream *source, int frames)
{
/* detecting maximum value in data chunk */
ae_int32x2 in_sample;
Expand Down Expand Up @@ -61,7 +61,7 @@ inline void aria_algo_calc_gain(struct aria_data *cd, size_t gain_idx,
}

void aria_algo_get_data_odd_channel(struct processing_module *mod,
struct audio_stream __sparse_cache *sink,
struct audio_stream *sink,
int frames)
{
struct aria_data *cd = module_get_private_data(mod);
Expand Down Expand Up @@ -118,7 +118,7 @@ void aria_algo_get_data_odd_channel(struct processing_module *mod,
}

void aria_algo_get_data_even_channel(struct processing_module *mod,
struct audio_stream __sparse_cache *sink,
struct audio_stream *sink,
int frames)
{
struct aria_data *cd = module_get_private_data(mod);
Expand Down
88 changes: 30 additions & 58 deletions src/audio/asrc/asrc.c
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Original file line number Diff line number Diff line change
Expand Up @@ -48,8 +48,8 @@
#define COEF_C2 Q_CONVERT_FLOAT(0.99, 30)

typedef void (*asrc_proc_func)(struct comp_dev *dev,
const struct audio_stream __sparse_cache *source,
struct audio_stream __sparse_cache *sink,
const struct audio_stream *source,
struct audio_stream *sink,
int *consumed,
int *produced);

Expand Down Expand Up @@ -122,8 +122,8 @@ static inline void src_inc_wrap_s16(int16_t **ptr, int16_t *end, size_t size)

/* A fast copy function for same in and out rate */
static void src_copy_s32(struct comp_dev *dev,
const struct audio_stream __sparse_cache *source,
struct audio_stream __sparse_cache *sink,
const struct audio_stream *source,
struct audio_stream *sink,
int *n_read, int *n_written)
{
struct comp_data *cd = comp_get_drvdata(dev);
Expand Down Expand Up @@ -194,8 +194,8 @@ static void src_copy_s32(struct comp_dev *dev,
}

static void src_copy_s16(struct comp_dev *dev,
const struct audio_stream __sparse_cache *source,
struct audio_stream __sparse_cache *sink,
const struct audio_stream *source,
struct audio_stream *sink,
int *n_read, int *n_written)
{
struct comp_data *cd = comp_get_drvdata(dev);
Expand Down Expand Up @@ -534,7 +534,6 @@ static int asrc_params(struct comp_dev *dev,
{
struct comp_data *cd = comp_get_drvdata(dev);
struct comp_buffer *sourceb, *sinkb;
struct comp_buffer __sparse_cache *source_c, *sink_c;
int err;

comp_info(dev, "asrc_params()");
Expand All @@ -554,25 +553,19 @@ static int asrc_params(struct comp_dev *dev,
sinkb = list_first_item(&dev->bsink_list, struct comp_buffer,
source_list);

source_c = buffer_acquire(sourceb);
sink_c = buffer_acquire(sinkb);

#if CONFIG_IPC_MAJOR_4
/* update the source/sink buffer formats. Sink rate will be modified below */
ipc4_update_buffer_format(source_c, &cd->ipc_config.base.audio_fmt);
ipc4_update_buffer_format(sink_c, &cd->ipc_config.base.audio_fmt);
ipc4_update_buffer_format(sourceb, &cd->ipc_config.base.audio_fmt);
ipc4_update_buffer_format(sinkb, &cd->ipc_config.base.audio_fmt);
#endif

/* Don't change sink rate if value from IPC is 0 (auto detect) */
if (asrc_get_sink_rate(&cd->ipc_config))
audio_stream_set_rate(&sink_c->stream, asrc_get_sink_rate(&cd->ipc_config));
audio_stream_set_rate(&sinkb->stream, asrc_get_sink_rate(&cd->ipc_config));

/* set source/sink_frames/rate */
cd->source_rate = audio_stream_get_rate(&source_c->stream);
cd->sink_rate = audio_stream_get_rate(&sink_c->stream);

buffer_release(sink_c);
buffer_release(source_c);
cd->source_rate = audio_stream_get_rate(&sourceb->stream);
cd->sink_rate = audio_stream_get_rate(&sinkb->stream);

if (!cd->sink_rate) {
comp_err(dev, "asrc_params(), zero sink rate");
Expand Down Expand Up @@ -604,7 +597,6 @@ static int asrc_params(struct comp_dev *dev,
static int asrc_dai_find(struct comp_dev *dev, struct comp_data *cd)
{
struct comp_buffer *sourceb, *sinkb;
struct comp_buffer __sparse_cache *source_c, *sink_c;
int pid;

/* Get current pipeline ID and walk to find the DAI */
Expand All @@ -615,9 +607,7 @@ static int asrc_dai_find(struct comp_dev *dev, struct comp_data *cd)
do {
sinkb = list_first_item(&dev->bsink_list, struct comp_buffer, source_list);

sink_c = buffer_acquire(sinkb);
dev = sink_c->sink;
buffer_release(sink_c);
dev = sinkb->sink;

if (!dev) {
comp_cl_err(&comp_asrc, "At end, no DAI found.");
Expand All @@ -635,9 +625,7 @@ static int asrc_dai_find(struct comp_dev *dev, struct comp_data *cd)
do {
sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list);

source_c = buffer_acquire(sourceb);
dev = source_c->source;
buffer_release(source_c);
dev = sourceb->source;

if (!dev) {
comp_cl_err(&comp_asrc, "At beginning, no DAI found.");
Expand Down Expand Up @@ -722,7 +710,6 @@ static int asrc_prepare(struct comp_dev *dev)
{
struct comp_data *cd = comp_get_drvdata(dev);
struct comp_buffer *sourceb, *sinkb;
struct comp_buffer __sparse_cache *source_c, *sink_c;
uint32_t source_period_bytes;
uint32_t sink_period_bytes;
int sample_bytes;
Expand All @@ -748,23 +735,20 @@ static int asrc_prepare(struct comp_dev *dev)
sinkb = list_first_item(&dev->bsink_list,
struct comp_buffer, source_list);

source_c = buffer_acquire(sourceb);
sink_c = buffer_acquire(sinkb);

/* get source data format and period bytes */
cd->source_format = audio_stream_get_frm_fmt(&source_c->stream);
source_period_bytes = audio_stream_period_bytes(&source_c->stream,
cd->source_format = audio_stream_get_frm_fmt(&sourceb->stream);
source_period_bytes = audio_stream_period_bytes(&sourceb->stream,
cd->source_frames);

/* get sink data format and period bytes */
cd->sink_format = audio_stream_get_frm_fmt(&sink_c->stream);
sink_period_bytes = audio_stream_period_bytes(&sink_c->stream,
cd->sink_format = audio_stream_get_frm_fmt(&sinkb->stream);
sink_period_bytes = audio_stream_period_bytes(&sinkb->stream,
cd->sink_frames);

if (audio_stream_get_size(&sink_c->stream) <
if (audio_stream_get_size(&sinkb->stream) <
dev->ipc_config.periods_sink * sink_period_bytes) {
comp_err(dev, "asrc_prepare(): sink buffer size %d is insufficient < %d * %d",
audio_stream_get_size(&sink_c->stream), dev->ipc_config.periods_sink,
audio_stream_get_size(&sinkb->stream), dev->ipc_config.periods_sink,
sink_period_bytes);
ret = -ENOMEM;
goto err;
Expand All @@ -783,7 +767,7 @@ static int asrc_prepare(struct comp_dev *dev)
}

/* ASRC supports S16_LE, S24_4LE and S32_LE formats */
switch (audio_stream_get_frm_fmt(&source_c->stream)) {
switch (audio_stream_get_frm_fmt(&sourceb->stream)) {
case SOF_IPC_FRAME_S16_LE:
cd->asrc_func = src_copy_s16;
break;
Expand All @@ -801,7 +785,7 @@ static int asrc_prepare(struct comp_dev *dev)
}

/* Allocate input and output data buffer for ASRC processing */
frame_bytes = audio_stream_frame_bytes(&source_c->stream);
frame_bytes = audio_stream_frame_bytes(&sourceb->stream);
cd->buf_size = (cd->source_frames_max + cd->sink_frames_max) *
frame_bytes;

Expand All @@ -815,16 +799,16 @@ static int asrc_prepare(struct comp_dev *dev)
goto err;
}

sample_bytes = frame_bytes / audio_stream_get_channels(&source_c->stream);
for (i = 0; i < audio_stream_get_channels(&source_c->stream); i++) {
sample_bytes = frame_bytes / audio_stream_get_channels(&sourceb->stream);
for (i = 0; i < audio_stream_get_channels(&sourceb->stream); i++) {
cd->ibuf[i] = cd->buf + i * sample_bytes;
cd->obuf[i] = cd->ibuf[i] + cd->source_frames_max * frame_bytes;
}

/* Get required size and allocate memory for ASRC */
sample_bits = sample_bytes * 8;
ret = asrc_get_required_size(dev, &cd->asrc_size,
audio_stream_get_channels(&source_c->stream),
audio_stream_get_channels(&sourceb->stream),
sample_bits);
if (ret) {
comp_err(dev, "asrc_prepare(), get_required_size_bytes failed");
Expand All @@ -850,7 +834,7 @@ static int asrc_prepare(struct comp_dev *dev)
fs_sec = cd->source_rate;
}

ret = asrc_initialise(dev, cd->asrc_obj, audio_stream_get_channels(&source_c->stream),
ret = asrc_initialise(dev, cd->asrc_obj, audio_stream_get_channels(&sourceb->stream),
fs_prim, fs_sec,
ASRC_IOF_INTERLEAVED, ASRC_IOF_INTERLEAVED,
ASRC_BM_LINEAR, cd->frames, sample_bits,
Expand Down Expand Up @@ -886,9 +870,6 @@ static int asrc_prepare(struct comp_dev *dev)
goto err_free_asrc;
}

buffer_release(sink_c);
buffer_release(source_c);

return 0;

err_free_asrc:
Expand All @@ -901,8 +882,6 @@ static int asrc_prepare(struct comp_dev *dev)
cd->buf = NULL;

err:
buffer_release(sink_c);
buffer_release(source_c);
comp_set_state(dev, COMP_TRIGGER_RESET);
return ret;
}
Expand Down Expand Up @@ -978,8 +957,8 @@ static int asrc_control_loop(struct comp_dev *dev, struct comp_data *cd)
return 0;
}

static void asrc_process(struct comp_dev *dev, struct comp_buffer __sparse_cache *source,
struct comp_buffer __sparse_cache *sink)
static void asrc_process(struct comp_dev *dev, struct comp_buffer *source,
struct comp_buffer *sink)
{
struct comp_data *cd = comp_get_drvdata(dev);
int consumed = 0;
Expand All @@ -1005,7 +984,6 @@ static int asrc_copy(struct comp_dev *dev)
{
struct comp_data *cd = comp_get_drvdata(dev);
struct comp_buffer *source, *sink;
struct comp_buffer __sparse_cache *source_c, *sink_c;
int frames_src;
int frames_snk;
int ret;
Expand All @@ -1022,11 +1000,8 @@ static int asrc_copy(struct comp_dev *dev)
sink = list_first_item(&dev->bsink_list, struct comp_buffer,
source_list);

source_c = buffer_acquire(source);
sink_c = buffer_acquire(sink);

frames_src = audio_stream_get_avail_frames(&source_c->stream);
frames_snk = audio_stream_get_free_frames(&sink_c->stream);
frames_src = audio_stream_get_avail_frames(&source->stream);
frames_snk = audio_stream_get_free_frames(&sink->stream);

if (cd->mode == ASRC_OM_PULL) {
/* Let ASRC access max number of source frames in pull mode.
Expand All @@ -1052,10 +1027,7 @@ static int asrc_copy(struct comp_dev *dev)
}

if (cd->source_frames && cd->sink_frames)
asrc_process(dev, source_c, sink_c);

buffer_release(sink_c);
buffer_release(source_c);
asrc_process(dev, source, sink);

return 0;
}
Expand Down
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