diff --git a/src/audio/aria/aria.c b/src/audio/aria/aria.c index 9d9c16c9c1d9..a20afc6e365f 100644 --- a/src/audio/aria/aria.c +++ b/src/audio/aria/aria.c @@ -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); @@ -153,7 +153,7 @@ 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; + struct comp_buffer *buffer_c; const struct ipc4_audio_format *audio_fmt = &mod->priv.cfg.base_cfg.audio_fmt; buffer_c = buffer_acquire(buffer); @@ -169,8 +169,8 @@ static void aria_set_stream_params(struct comp_buffer *buffer, } 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; diff --git a/src/audio/aria/aria_generic.c b/src/audio/aria/aria_generic.c index cc0f90b2de2e..09daee86c5d9 100644 --- a/src/audio/aria/aria_generic.c +++ b/src/audio/aria/aria_generic.c @@ -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; @@ -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; diff --git a/src/audio/aria/aria_hifi3.c b/src/audio/aria/aria_hifi3.c index 17e356b968bc..363ba583b0d3 100644 --- a/src/audio/aria/aria_hifi3.c +++ b/src/audio/aria/aria_hifi3.c @@ -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; @@ -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); @@ -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); diff --git a/src/audio/asrc/asrc.c b/src/audio/asrc/asrc.c index 6f25e15129c5..dbefb7ba14c6 100644 --- a/src/audio/asrc/asrc.c +++ b/src/audio/asrc/asrc.c @@ -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); @@ -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); @@ -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); @@ -534,7 +534,7 @@ 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; + struct comp_buffer *source_c, *sink_c; int err; comp_info(dev, "asrc_params()"); @@ -604,7 +604,7 @@ 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; + struct comp_buffer *source_c, *sink_c; int pid; /* Get current pipeline ID and walk to find the DAI */ @@ -722,7 +722,7 @@ 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; + struct comp_buffer *source_c, *sink_c; uint32_t source_period_bytes; uint32_t sink_period_bytes; int sample_bytes; @@ -978,8 +978,8 @@ 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; @@ -1005,7 +1005,7 @@ 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; + struct comp_buffer *source_c, *sink_c; int frames_src; int frames_snk; int ret; diff --git a/src/audio/audio_stream.c b/src/audio/audio_stream.c index 96f02cff8b90..5fa8fa805c10 100644 --- a/src/audio/audio_stream.c +++ b/src/audio/audio_stream.c @@ -6,20 +6,20 @@ #include #include -static size_t audio_stream_get_free_size(struct sof_sink __sparse_cache *sink) +static size_t audio_stream_get_free_size(struct sof_sink *sink) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(sink, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(sink, struct audio_stream, sink_api, __sparse_cache); return audio_stream_get_free_bytes(audio_stream); } -static int audio_stream_get_buffer(struct sof_sink __sparse_cache *sink, size_t req_size, +static int audio_stream_get_buffer(struct sof_sink *sink, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(sink, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(sink, struct audio_stream, sink_api, __sparse_cache); if (req_size > audio_stream_get_free_size(sink)) @@ -32,13 +32,13 @@ static int audio_stream_get_buffer(struct sof_sink __sparse_cache *sink, size_t return 0; } -static int audio_stream_commit_buffer(struct sof_sink __sparse_cache *sink, size_t commit_size) +static int audio_stream_commit_buffer(struct sof_sink *sink, size_t commit_size) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(sink, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(sink, struct audio_stream, sink_api, __sparse_cache); - struct comp_buffer __sparse_cache *buffer_c = - attr_container_of(audio_stream, struct comp_buffer __sparse_cache, + struct comp_buffer *buffer_c = + attr_container_of(audio_stream, struct comp_buffer, stream, __sparse_cache); if (commit_size) { @@ -49,24 +49,24 @@ static int audio_stream_commit_buffer(struct sof_sink __sparse_cache *sink, size return 0; } -static size_t audio_stream_get_data_available(struct sof_source __sparse_cache *source) +static size_t audio_stream_get_data_available(struct sof_source *source) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(source, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(source, struct audio_stream, source_api, __sparse_cache); return audio_stream_get_avail_bytes(audio_stream); } -static int audio_stream_get_data(struct sof_source __sparse_cache *source, size_t req_size, +static int audio_stream_get_data(struct sof_source *source, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(source, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(source, struct audio_stream, source_api, __sparse_cache); - struct comp_buffer __sparse_cache *buffer_c = - attr_container_of(audio_stream, struct comp_buffer __sparse_cache, + struct comp_buffer *buffer_c = + attr_container_of(audio_stream, struct comp_buffer, stream, __sparse_cache); if (req_size > audio_stream_get_data_available(source)) @@ -81,10 +81,10 @@ static int audio_stream_get_data(struct sof_source __sparse_cache *source, size_ return 0; } -static int audio_stream_release_data(struct sof_source __sparse_cache *source, size_t free_size) +static int audio_stream_release_data(struct sof_source *source, size_t free_size) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(source, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(source, struct audio_stream, source_api, __sparse_cache); if (free_size) @@ -93,40 +93,40 @@ static int audio_stream_release_data(struct sof_source __sparse_cache *source, s return 0; } -static int audio_stream_set_ipc_params_source(struct sof_source __sparse_cache *source, +static int audio_stream_set_ipc_params_source(struct sof_source *source, struct sof_ipc_stream_params *params, bool force_update) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(source, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(source, struct audio_stream, source_api, __sparse_cache); - struct comp_buffer __sparse_cache *buffer = - attr_container_of(audio_stream, struct comp_buffer __sparse_cache, + struct comp_buffer *buffer = + attr_container_of(audio_stream, struct comp_buffer, stream, __sparse_cache); return buffer_set_params(buffer, params, force_update); } -static int audio_stream_set_ipc_params_sink(struct sof_sink __sparse_cache *sink, +static int audio_stream_set_ipc_params_sink(struct sof_sink *sink, struct sof_ipc_stream_params *params, bool force_update) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(sink, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(sink, struct audio_stream, sink_api, __sparse_cache); - struct comp_buffer __sparse_cache *buffer = - attr_container_of(audio_stream, struct comp_buffer __sparse_cache, + struct comp_buffer *buffer = + attr_container_of(audio_stream, struct comp_buffer, stream, __sparse_cache); return buffer_set_params(buffer, params, force_update); } -static int audio_stream_source_set_alignment_constants(struct sof_source __sparse_cache *source, +static int audio_stream_source_set_alignment_constants(struct sof_source *source, const uint32_t byte_align, const uint32_t frame_align_req) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(source, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(source, struct audio_stream, source_api, __sparse_cache); audio_stream_init_alignment_constants(byte_align, frame_align_req, audio_stream); @@ -134,12 +134,12 @@ static int audio_stream_source_set_alignment_constants(struct sof_source __spars return 0; } -static int audio_stream_sink_set_alignment_constants(struct sof_sink __sparse_cache *sink, +static int audio_stream_sink_set_alignment_constants(struct sof_sink *sink, const uint32_t byte_align, const uint32_t frame_align_req) { - struct audio_stream __sparse_cache *audio_stream = - attr_container_of(sink, struct audio_stream __sparse_cache, + struct audio_stream *audio_stream = + attr_container_of(sink, struct audio_stream, sink_api, __sparse_cache); audio_stream_init_alignment_constants(byte_align, frame_align_req, audio_stream); @@ -163,7 +163,7 @@ static const struct sink_ops audio_stream_sink_ops = { .set_alignment_constants = audio_stream_sink_set_alignment_constants }; -void audio_stream_init(struct audio_stream __sparse_cache *audio_stream, +void audio_stream_init(struct audio_stream *audio_stream, void *buff_addr, uint32_t size) { audio_stream->size = size; diff --git a/src/audio/buffer.c b/src/audio/buffer.c index 78b36158e50e..1519671ff4e6 100644 --- a/src/audio/buffer.c +++ b/src/audio/buffer.c @@ -31,7 +31,7 @@ struct comp_buffer *buffer_alloc(uint32_t size, uint32_t caps, uint32_t flags, u bool is_shared) { struct comp_buffer *buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; void *stream_addr; tr_dbg(&buffer_tr, "buffer_alloc()"); @@ -79,7 +79,7 @@ struct comp_buffer *buffer_alloc(uint32_t size, uint32_t caps, uint32_t flags, u return buffer; } -void buffer_zero(struct comp_buffer __sparse_cache *buffer) +void buffer_zero(struct comp_buffer *buffer) { buf_dbg(buffer, "stream_zero()"); CORE_CHECK_STRUCT(buffer); @@ -91,7 +91,7 @@ void buffer_zero(struct comp_buffer __sparse_cache *buffer) audio_stream_get_size(&buffer->stream)); } -int buffer_set_size(struct comp_buffer __sparse_cache *buffer, uint32_t size, uint32_t alignment) +int buffer_set_size(struct comp_buffer *buffer, uint32_t size, uint32_t alignment) { void *new_ptr = NULL; @@ -129,7 +129,7 @@ int buffer_set_size(struct comp_buffer __sparse_cache *buffer, uint32_t size, ui return 0; } -int buffer_set_params(struct comp_buffer __sparse_cache *buffer, +int buffer_set_params(struct comp_buffer *buffer, struct sof_ipc_stream_params *params, bool force_update) { int ret; @@ -160,7 +160,7 @@ int buffer_set_params(struct comp_buffer __sparse_cache *buffer, return 0; } -bool buffer_params_match(struct comp_buffer __sparse_cache *buffer, +bool buffer_params_match(struct comp_buffer *buffer, struct sof_ipc_stream_params *params, uint32_t flag) { assert(params); @@ -217,7 +217,7 @@ void buffer_free(struct comp_buffer *buffer) * choice but to use our knowledge of the local notifier behaviour and pass * locked buffers to notification recipients. */ -void comp_update_buffer_produce(struct comp_buffer __sparse_cache *buffer, uint32_t bytes) +void comp_update_buffer_produce(struct comp_buffer *buffer, uint32_t bytes) { struct buffer_cb_transact cb_data = { .buffer = buffer, @@ -256,7 +256,7 @@ void comp_update_buffer_produce(struct comp_buffer __sparse_cache *buffer, uint3 #endif } -void comp_update_buffer_consume(struct comp_buffer __sparse_cache *buffer, uint32_t bytes) +void comp_update_buffer_consume(struct comp_buffer *buffer, uint32_t bytes) { struct buffer_cb_transact cb_data = { .buffer = buffer, diff --git a/src/audio/chain_dma.c b/src/audio/chain_dma.c index f2fbf3cdbfbb..41c28fa0c3e0 100644 --- a/src/audio/chain_dma.c +++ b/src/audio/chain_dma.c @@ -238,7 +238,7 @@ static enum task_state chain_task_run(void *data) * mode task will update read position based on transferred data size to avoid * overwriting valid data and write position by half buffer size. */ - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(cd->dma_buffer); + struct comp_buffer *buffer_c = buffer_acquire(cd->dma_buffer); const size_t buff_size = audio_stream_get_size(&buffer_c->stream); const size_t half_buff_size = buff_size / 2; @@ -510,7 +510,7 @@ static int chain_task_init(struct comp_dev *dev, uint8_t host_dma_id, uint8_t li uint32_t fifo_size) { struct chain_dma_data *cd = comp_get_drvdata(dev); - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t addr_align; size_t buff_size; void *buff_addr; diff --git a/src/audio/component.c b/src/audio/component.c index 51e6b4250f9b..1d30801221dd 100644 --- a/src/audio/component.c +++ b/src/audio/component.c @@ -158,8 +158,8 @@ void sys_comp_init(struct sof *sof) k_spinlock_init(&sof->comp_drivers->lock); } -void comp_get_copy_limits(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +void comp_get_copy_limits(struct comp_buffer *source, + struct comp_buffer *sink, struct comp_copy_limits *cl) { cl->frames = audio_stream_avail_frames(&source->stream, &sink->stream); @@ -169,8 +169,8 @@ void comp_get_copy_limits(struct comp_buffer __sparse_cache *source, cl->sink_bytes = cl->frames * cl->sink_frame_bytes; } -void comp_get_copy_limits_frame_aligned(const struct comp_buffer __sparse_cache *source, - const struct comp_buffer __sparse_cache *sink, +void comp_get_copy_limits_frame_aligned(const struct comp_buffer *source, + const struct comp_buffer *sink, struct comp_copy_limits *cl) { cl->frames = audio_stream_avail_frames_aligned(&source->stream, &sink->stream); @@ -182,8 +182,8 @@ void comp_get_copy_limits_frame_aligned(const struct comp_buffer __sparse_cache #ifdef STREAMCOPY_HIFI3 -int audio_stream_copy(const struct audio_stream __sparse_cache *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, uint32_t ooffset, uint32_t samples) +int audio_stream_copy(const struct audio_stream *source, uint32_t ioffset, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int ssize = audio_stream_sample_bytes(source); /* src fmt == sink fmt */ ae_int16x4 *src = (ae_int16x4 *)((int8_t *)audio_stream_get_rptr(source) + ioffset * ssize); @@ -227,8 +227,8 @@ int audio_stream_copy(const struct audio_stream __sparse_cache *source, uint32_t #else -int audio_stream_copy(const struct audio_stream __sparse_cache *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, uint32_t ooffset, uint32_t samples) +int audio_stream_copy(const struct audio_stream *source, uint32_t ioffset, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int ssize = audio_stream_sample_bytes(source); /* src fmt == sink fmt */ uint8_t *src = audio_stream_wrap(source, (uint8_t *)audio_stream_get_rptr(source) + @@ -279,7 +279,7 @@ void cir_buf_copy(void *src, void *src_addr, void *src_end, void *dst, } void audio_stream_copy_from_linear(const void *linear_source, int ioffset, - struct audio_stream __sparse_cache *sink, int ooffset, + struct audio_stream *sink, int ooffset, unsigned int samples) { int ssize = audio_stream_sample_bytes(sink); /* src fmt == sink fmt */ @@ -300,7 +300,7 @@ void audio_stream_copy_from_linear(const void *linear_source, int ioffset, } } -void audio_stream_copy_to_linear(const struct audio_stream __sparse_cache *source, int ioffset, +void audio_stream_copy_to_linear(const struct audio_stream *source, int ioffset, void *linear_sink, int ooffset, unsigned int samples) { int ssize = audio_stream_sample_bytes(source); /* src fmt == sink fmt */ diff --git a/src/audio/copier/copier.c b/src/audio/copier/copier.c index ae03226be8c1..2235d3e01c54 100644 --- a/src/audio/copier/copier.c +++ b/src/audio/copier/copier.c @@ -178,8 +178,8 @@ static int copier_free(struct processing_module *mod) static int copier_params(struct processing_module *mod); static int copier_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) { struct copier_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; @@ -266,7 +266,7 @@ static int copier_comp_trigger(struct comp_dev *dev, int cmd) struct sof_ipc_stream_posn posn; struct comp_dev *dai_copier; struct comp_buffer *buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t latency; int ret; @@ -396,8 +396,8 @@ static int copier_comp_trigger(struct comp_dev *dev, int cmd) static int do_conversion_copy(struct comp_dev *dev, struct copier_data *cd, - struct comp_buffer __sparse_cache *src, - struct comp_buffer __sparse_cache *sink, + struct comp_buffer *src, + struct comp_buffer *sink, struct comp_copy_limits *processed_data) { int i; @@ -423,10 +423,10 @@ static int do_conversion_copy(struct comp_dev *dev, } static int copier_copy_to_sinks(struct copier_data *cd, struct comp_dev *dev, - struct comp_buffer __sparse_cache *src_c, + struct comp_buffer *src_c, struct comp_copy_limits *processed_data) { - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct list_item *sink_list; struct comp_buffer *sink; int ret = 0; @@ -466,25 +466,25 @@ static int copier_module_copy(struct processing_module *mod, struct output_stream_buffer *output_buffers, int num_output_buffers) { struct copier_data *cd = module_get_private_data(mod); - struct comp_buffer __sparse_cache *src_c; + struct comp_buffer *src_c; struct comp_copy_limits processed_data; int i; if (!num_input_buffers || !num_output_buffers) return 0; - src_c = attr_container_of(input_buffers[0].data, struct comp_buffer __sparse_cache, + src_c = attr_container_of(input_buffers[0].data, struct comp_buffer, stream, __sparse_cache); processed_data.source_bytes = 0; /* convert format and copy to each active sink */ for (i = 0; i < num_output_buffers; i++) { - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct comp_dev *sink_dev; sink_c = attr_container_of(output_buffers[i].data, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); sink_dev = sink_c->sink; processed_data.sink_bytes = 0; @@ -518,7 +518,7 @@ static int copier_module_copy(struct processing_module *mod, static int copier_multi_endpoint_dai_copy(struct copier_data *cd, struct comp_dev *dev) { - struct comp_buffer __sparse_cache *src_c, *sink_c; + struct comp_buffer *src_c, *sink_c; struct comp_copy_limits processed_data; struct comp_buffer *src; int ret; @@ -649,7 +649,7 @@ static int copier_set_sink_fmt(struct comp_dev *dev, const void *data, const struct ipc4_copier_config_set_sink_format *sink_fmt = data; struct processing_module *mod = comp_get_drvdata(dev); struct copier_data *cd = module_get_private_data(mod); - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct list_item *sink_list; struct comp_buffer *sink; diff --git a/src/audio/copier/copier_dai.c b/src/audio/copier/copier_dai.c index 31b832b81d01..d5996848bdcd 100644 --- a/src/audio/copier/copier_dai.c +++ b/src/audio/copier/copier_dai.c @@ -310,9 +310,9 @@ int copier_dai_prepare(struct comp_dev *dev, struct copier_data *cd) return 0; } -static int copy_single_channel_c16(const struct audio_stream __sparse_cache *src, +static int copy_single_channel_c16(const struct audio_stream *src, unsigned int src_channel, - struct audio_stream __sparse_cache *dst, + struct audio_stream *dst, unsigned int dst_channel, unsigned int frame_count) { int16_t *r_ptr = (int16_t *)audio_stream_get_rptr(src) + src_channel; @@ -351,9 +351,9 @@ static int copy_single_channel_c16(const struct audio_stream __sparse_cache *src return 0; } -static int copy_single_channel_c32(const struct audio_stream __sparse_cache *src, +static int copy_single_channel_c32(const struct audio_stream *src, unsigned int src_channel, - struct audio_stream __sparse_cache *dst, + struct audio_stream *dst, unsigned int dst_channel, unsigned int frame_count) { int32_t *r_ptr = (int32_t *)audio_stream_get_rptr(src) + src_channel; @@ -395,7 +395,7 @@ static int copy_single_channel_c32(const struct audio_stream __sparse_cache *src int copier_dai_params(struct copier_data *cd, struct comp_dev *dev, struct sof_ipc_stream_params *params, int dai_index) { - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; struct sof_ipc_stream_params demuxed_params = *params; const struct ipc4_audio_format *in_fmt = &cd->config.base.audio_fmt; const struct ipc4_audio_format *out_fmt = &cd->config.out_fmt; diff --git a/src/audio/copier/copier_generic.c b/src/audio/copier/copier_generic.c index 13794b19bccd..67c0e23af636 100644 --- a/src/audio/copier/copier_generic.c +++ b/src/audio/copier/copier_generic.c @@ -22,7 +22,7 @@ LOG_MODULE_DECLARE(copier, CONFIG_SOF_LOG_LEVEL); #include int apply_attenuation(struct comp_dev *dev, struct copier_data *cd, - struct comp_buffer __sparse_cache *sink, int frame) + struct comp_buffer *sink, int frame) { int i; int n; @@ -62,7 +62,7 @@ void copier_update_params(struct copier_data *cd, struct comp_dev *dev, struct sof_ipc_stream_params *params) { struct comp_buffer *sink, *source; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; struct list_item *sink_list; memset(params, 0, sizeof(*params)); @@ -140,7 +140,7 @@ int create_endpoint_buffer(struct comp_dev *dev, enum sof_ipc_frame valid_fmt; struct sof_ipc_buffer ipc_buf; struct comp_buffer *buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t buf_size; uint32_t chan_map; int i; diff --git a/src/audio/copier/copier_hifi.c b/src/audio/copier/copier_hifi.c index de6a8606aa7b..a419a4aee431 100644 --- a/src/audio/copier/copier_hifi.c +++ b/src/audio/copier/copier_hifi.c @@ -21,7 +21,7 @@ LOG_MODULE_REGISTER(copier_hifi, CONFIG_SOF_LOG_LEVEL); int apply_attenuation(struct comp_dev *dev, struct copier_data *cd, - struct comp_buffer __sparse_cache *sink, int frame) + struct comp_buffer *sink, int frame) { int i; int n; diff --git a/src/audio/copier/copier_host.c b/src/audio/copier/copier_host.c index 35456ed4dec1..ab4c2e422c9e 100644 --- a/src/audio/copier/copier_host.c +++ b/src/audio/copier/copier_host.c @@ -246,7 +246,7 @@ void copier_host_dma_cb(struct comp_dev *dev, size_t bytes) { struct processing_module *mod = comp_get_drvdata(dev); struct copier_data *cd = module_get_private_data(mod); - struct comp_buffer __sparse_cache *sink, *source; + struct comp_buffer *sink, *source; int ret, frames; comp_dbg(dev, "copier_host_dma_cb() %p", dev); diff --git a/src/audio/copier/copier_ipcgtw.c b/src/audio/copier/copier_ipcgtw.c index 967467f098ad..642a934bbfe8 100644 --- a/src/audio/copier/copier_ipcgtw.c +++ b/src/audio/copier/copier_ipcgtw.c @@ -37,7 +37,7 @@ static struct comp_dev *find_ipcgtw_by_node_id(union ipc4_connector_node_id node } static inline void audio_stream_copy_bytes_from_linear(const void *linear_source, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, unsigned int bytes) { const uint8_t *src = (const uint8_t *)linear_source; @@ -55,7 +55,7 @@ static inline void audio_stream_copy_bytes_from_linear(const void *linear_source } static inline -void audio_stream_copy_bytes_to_linear(const struct audio_stream __sparse_cache *source, +void audio_stream_copy_bytes_to_linear(const struct audio_stream *source, void *linear_sink, unsigned int bytes) { uint8_t *src = audio_stream_wrap(source, audio_stream_get_rptr(source)); @@ -93,7 +93,7 @@ int copier_ipcgtw_process(const struct ipc4_ipcgtw_cmd *cmd, const struct ipc4_ipc_gateway_cmd_data *in; struct comp_dev *dev; struct comp_buffer *buf; - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; uint32_t data_size; struct ipc4_ipc_gateway_cmd_data_reply *out; @@ -184,7 +184,7 @@ int copier_ipcgtw_params(struct ipcgtw_data *ipcgtw_data, struct comp_dev *dev, struct sof_ipc_stream_params *params) { struct comp_buffer *buf; - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; int err; comp_dbg(dev, "ipcgtw_params()"); @@ -214,7 +214,7 @@ void copier_ipcgtw_reset(struct comp_dev *dev) struct comp_buffer *buf = get_buffer(dev); if (buf) { - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(buf); + struct comp_buffer *buf_c = buffer_acquire(buf); audio_stream_reset(&buf_c->stream); buffer_release(buf_c); diff --git a/src/audio/crossover/crossover.c b/src/audio/crossover/crossover.c index 029e2fcbf74c..dd1fb0468eda 100644 --- a/src/audio/crossover/crossover.c +++ b/src/audio/crossover/crossover.c @@ -133,7 +133,7 @@ static int crossover_assign_sinks(struct processing_module *mod, struct sof_crossover_config *config = cd->config; struct comp_dev *dev = mod->dev; struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct list_item *sink_list; int num_sinks = 0; int i; @@ -486,7 +486,7 @@ static int crossover_check_sink_assign(struct processing_module *mod, { struct comp_dev *dev = mod->dev; struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct list_item *sink_list; int num_assigned_sinks = 0; uint8_t assigned_sinks[SOF_CROSSOVER_MAX_STREAMS] = {0}; @@ -623,7 +623,7 @@ static int crossover_process_audio_stream(struct processing_module *mod, bool enabled_buffers[PLATFORM_MAX_STREAMS] = { false }; struct comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; - struct audio_stream __sparse_cache *source = input_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; uint32_t num_sinks; uint32_t num_assigned_sinks = 0; /* The frames count to process from module adapter applies for source buffer and @@ -691,7 +691,7 @@ static int crossover_process_audio_stream(struct processing_module *mod, static void crossover_params(struct processing_module *mod) { struct sof_ipc_stream_params *params = mod->stream_params; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; struct comp_buffer *sinkb, *sourceb; struct list_item *sink_list; struct comp_dev *dev = mod->dev; @@ -721,13 +721,13 @@ static void crossover_params(struct processing_module *mod) * \return Error code. */ static int crossover_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) { struct comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; struct comp_buffer *source, *sink; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct list_item *sink_list; int channels; int ret = 0; diff --git a/src/audio/crossover/crossover_generic.c b/src/audio/crossover/crossover_generic.c index ad8b6b930ea9..0823ce85920f 100644 --- a/src/audio/crossover/crossover_generic.c +++ b/src/audio/crossover/crossover_generic.c @@ -93,8 +93,8 @@ static void crossover_s16_default_pass(struct comp_data *cd, int32_t num_sinks, uint32_t frames) { - const struct audio_stream __sparse_cache *sink_stream; - const struct audio_stream __sparse_cache *source_stream = bsource->data; + const struct audio_stream *sink_stream; + const struct audio_stream *source_stream = bsource->data; int16_t *x; int32_t *y; int i, j; @@ -121,8 +121,8 @@ static void crossover_s32_default_pass(struct comp_data *cd, int32_t num_sinks, uint32_t frames) { - const struct audio_stream __sparse_cache *sink_stream; - const struct audio_stream __sparse_cache *source_stream = bsource->data; + const struct audio_stream *sink_stream; + const struct audio_stream *source_stream = bsource->data; int32_t *x, *y; int i, j; int n = audio_stream_get_channels(source_stream) * frames; @@ -149,8 +149,8 @@ static void crossover_s16_default(struct comp_data *cd, uint32_t frames) { struct crossover_state *state; - const struct audio_stream __sparse_cache *source_stream = bsource->data; - struct audio_stream __sparse_cache *sink_stream; + const struct audio_stream *source_stream = bsource->data; + struct audio_stream *sink_stream; int16_t *x, *y; int ch, i, j; int idx; @@ -187,8 +187,8 @@ static void crossover_s24_default(struct comp_data *cd, uint32_t frames) { struct crossover_state *state; - const struct audio_stream __sparse_cache *source_stream = bsource->data; - struct audio_stream __sparse_cache *sink_stream; + const struct audio_stream *source_stream = bsource->data; + struct audio_stream *sink_stream; int32_t *x, *y; int ch, i, j; int idx; @@ -225,8 +225,8 @@ static void crossover_s32_default(struct comp_data *cd, uint32_t frames) { struct crossover_state *state; - const struct audio_stream __sparse_cache *source_stream = bsource->data; - struct audio_stream __sparse_cache *sink_stream; + const struct audio_stream *source_stream = bsource->data; + struct audio_stream *sink_stream; int32_t *x, *y; int ch, i, j; int idx; diff --git a/src/audio/dai-legacy.c b/src/audio/dai-legacy.c index 446ae0c83c08..933d23d43fdb 100644 --- a/src/audio/dai-legacy.c +++ b/src/audio/dai-legacy.c @@ -97,7 +97,7 @@ static void dai_dma_cb(void *arg, enum notify_id type, void *data) struct comp_dev *dev = arg; struct dai_data *dd = comp_get_drvdata(dev); uint32_t bytes = next->elem.size; - struct comp_buffer __sparse_cache *local_buf, *dma_buf; + struct comp_buffer *local_buf, *dma_buf; int ret; comp_dbg(dev, "dai_dma_cb()"); @@ -138,7 +138,7 @@ static void dai_dma_cb(void *arg, enum notify_id type, void *data) /* assert dma_buffer_copy succeed */ if (ret < 0) { - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; source_c = dev->direction == SOF_IPC_STREAM_PLAYBACK ? local_buf : dma_buf; @@ -357,7 +357,7 @@ static int dai_playback_params(struct comp_dev *dev, uint32_t period_bytes, { struct dai_data *dd = comp_get_drvdata(dev); struct dma_sg_config *config = &dd->config; - struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer), + struct comp_buffer *dma_buf = buffer_acquire(dd->dma_buffer), *local_buf = buffer_acquire(dd->local_buffer); uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream); uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream); @@ -419,7 +419,7 @@ static int dai_capture_params(struct comp_dev *dev, uint32_t period_bytes, { struct dai_data *dd = comp_get_drvdata(dev); struct dma_sg_config *config = &dd->config; - struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer), + struct comp_buffer *dma_buf = buffer_acquire(dd->dma_buffer), *local_buf = buffer_acquire(dd->local_buffer); uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream); uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream); @@ -491,7 +491,7 @@ int dai_common_params(struct dai_data *dd, struct comp_dev *dev, struct sof_ipc_stream_params *params) { struct sof_ipc_stream_params hw_params = *params; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t frame_size; uint32_t period_count; uint32_t period_bytes; @@ -685,7 +685,7 @@ int dai_common_config_prepare(struct dai_data *dd, struct comp_dev *dev) int dai_common_prepare(struct dai_data *dd, struct comp_dev *dev) { - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; int ret; dd->total_data_processed = 0; @@ -815,7 +815,7 @@ static int dai_comp_trigger_internal(struct dai_data *dd, struct comp_dev *dev, * this is only supported at capture mode. */ if (dev->direction == SOF_IPC_STREAM_CAPTURE) { - struct comp_buffer __sparse_cache *buffer_c = + struct comp_buffer *buffer_c = buffer_acquire(dd->dma_buffer); buffer_zero(buffer_c); @@ -947,7 +947,7 @@ static int dai_comp_trigger(struct comp_dev *dev, int cmd) static void dai_report_xrun(struct comp_dev *dev, uint32_t bytes) { struct dai_data *dd = comp_get_drvdata(dev); - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(dd->local_buffer); + struct comp_buffer *buf_c = buffer_acquire(dd->local_buffer); if (dev->direction == SOF_IPC_STREAM_PLAYBACK) { comp_err(dev, "dai_report_xrun(): underrun due to no data available"); @@ -965,7 +965,7 @@ int dai_common_copy(struct dai_data *dd, struct comp_dev *dev, pcm_converter_fun { uint32_t dma_fmt; uint32_t sampling; - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; uint32_t avail_bytes = 0; uint32_t free_bytes = 0; uint32_t copy_bytes = 0; diff --git a/src/audio/dai-zephyr.c b/src/audio/dai-zephyr.c index f83bf3d27f94..6f72a68f26d9 100644 --- a/src/audio/dai-zephyr.c +++ b/src/audio/dai-zephyr.c @@ -226,7 +226,7 @@ static enum dma_cb_status dai_dma_cb(struct dai_data *dd, struct comp_dev *dev, uint32_t bytes, pcm_converter_func *converter) { - struct comp_buffer __sparse_cache *local_buf, *dma_buf; + struct comp_buffer *local_buf, *dma_buf; enum dma_cb_status dma_status = DMA_CB_STATUS_RELOAD; int ret; @@ -276,7 +276,7 @@ dai_dma_cb(struct dai_data *dd, struct comp_dev *dev, uint32_t bytes, * function */ list_for_item(sink_list, &dev->bsink_list) { - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct comp_dev *sink_dev; struct comp_buffer *sink; int j; @@ -318,7 +318,7 @@ dai_dma_cb(struct dai_data *dd, struct comp_dev *dev, uint32_t bytes, /* assert dma_buffer_copy succeed */ if (ret < 0) { - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; source_c = dev->direction == SOF_IPC_STREAM_PLAYBACK ? local_buf : dma_buf; @@ -346,7 +346,7 @@ static enum dma_cb_status dai_dma_multi_endpoint_cb(struct dai_data *dd, struct comp_dev *dev, uint32_t frames, struct comp_buffer *multi_endpoint_buffer) { - struct comp_buffer __sparse_cache *multi_buf_c, *dma_buf; + struct comp_buffer *multi_buf_c, *dma_buf; enum dma_cb_status dma_status = DMA_CB_STATUS_RELOAD; uint32_t i, bytes; @@ -592,7 +592,7 @@ static int dai_playback_params(struct dai_data *dd, struct comp_dev *dev, uint32 struct dma_config *dma_cfg; struct dma_block_config *dma_block_cfg; struct dma_block_config *prev = NULL; - struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer), + struct comp_buffer *dma_buf = buffer_acquire(dd->dma_buffer), *local_buf = buffer_acquire(dd->local_buffer); uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream); uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream); @@ -734,7 +734,7 @@ static int dai_capture_params(struct dai_data *dd, struct comp_dev *dev, uint32_ struct dma_config *dma_cfg; struct dma_block_config *dma_block_cfg; struct dma_block_config *prev = NULL; - struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer), + struct comp_buffer *dma_buf = buffer_acquire(dd->dma_buffer), *local_buf = buffer_acquire(dd->local_buffer); uint32_t local_fmt = audio_stream_get_frm_fmt(&local_buf->stream); uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream); @@ -885,7 +885,7 @@ int dai_common_params(struct dai_data *dd, struct comp_dev *dev, struct sof_ipc_stream_params *params) { struct sof_ipc_stream_params hw_params = *params; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t frame_size; uint32_t period_count; uint32_t period_bytes; @@ -1064,7 +1064,7 @@ int dai_common_config_prepare(struct dai_data *dd, struct comp_dev *dev) int dai_common_prepare(struct dai_data *dd, struct comp_dev *dev) { - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; int ret; dd->total_data_processed = 0; @@ -1193,7 +1193,7 @@ static int dai_comp_trigger_internal(struct dai_data *dd, struct comp_dev *dev, * this is only supported at capture mode. */ if (dev->direction == SOF_IPC_STREAM_CAPTURE) { - struct comp_buffer __sparse_cache *buffer_c = + struct comp_buffer *buffer_c = buffer_acquire(dd->dma_buffer); buffer_zero(buffer_c); @@ -1339,7 +1339,7 @@ static int dai_comp_trigger(struct comp_dev *dev, int cmd) /* report xrun occurrence */ static void dai_report_xrun(struct dai_data *dd, struct comp_dev *dev, uint32_t bytes) { - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(dd->local_buffer); + struct comp_buffer *buf_c = buffer_acquire(dd->local_buffer); if (dev->direction == SOF_IPC_STREAM_PLAYBACK) { comp_err(dev, "dai_report_xrun(): underrun due to no data available"); @@ -1357,7 +1357,7 @@ int dai_zephyr_multi_endpoint_copy(struct dai_data **dd, struct comp_dev *dev, struct comp_buffer *multi_endpoint_buffer, int num_endpoints) { - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; uint32_t avail_bytes = UINT32_MAX; uint32_t free_bytes = UINT32_MAX; uint32_t frames; @@ -1476,8 +1476,8 @@ int dai_zephyr_multi_endpoint_copy(struct dai_data **dd, struct comp_dev *dev, static void set_new_local_buffer(struct dai_data *dd, struct comp_dev *dev) { - struct comp_buffer __sparse_cache *dma_buf = buffer_acquire(dd->dma_buffer); - struct comp_buffer __sparse_cache *local_buf; + struct comp_buffer *dma_buf = buffer_acquire(dd->dma_buffer); + struct comp_buffer *local_buf; uint32_t dma_fmt = audio_stream_get_frm_fmt(&dma_buf->stream); uint32_t local_fmt; @@ -1509,7 +1509,7 @@ static void set_new_local_buffer(struct dai_data *dd, struct comp_dev *dev) int dai_common_copy(struct dai_data *dd, struct comp_dev *dev, pcm_converter_func *converter) { uint32_t sampling = dd->sampling; - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; struct dma_status stat; uint32_t avail_bytes; uint32_t free_bytes; @@ -1792,7 +1792,7 @@ static uint64_t dai_get_processed_data(struct comp_dev *dev, uint32_t stream_no, #ifdef CONFIG_IPC_MAJOR_4 int dai_zephyr_unbind(struct dai_data *dd, struct comp_dev *dev, void *data) { - struct comp_buffer __sparse_cache *local_buf_c; + struct comp_buffer *local_buf_c; struct ipc4_module_bind_unbind *bu; int buf_id; diff --git a/src/audio/dcblock/dcblock.c b/src/audio/dcblock/dcblock.c index 956181149900..3a476933299e 100644 --- a/src/audio/dcblock/dcblock.c +++ b/src/audio/dcblock/dcblock.c @@ -194,8 +194,8 @@ static int dcblock_process(struct processing_module *mod, int num_output_buffers) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = input_buffers[0].data; - struct audio_stream __sparse_cache *sink = output_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; + struct audio_stream *sink = output_buffers[0].data; uint32_t frames = input_buffers[0].size; comp_dbg(mod->dev, "dcblock_process()"); @@ -207,8 +207,8 @@ static int dcblock_process(struct processing_module *mod, } /* init and calculate the aligned setting for available frames and free frames retrieve*/ -static inline void dcblock_set_frame_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static inline void dcblock_set_frame_alignment(struct audio_stream *source, + struct audio_stream *sink) { const uint32_t byte_align = 1; const uint32_t frame_align_req = 1; @@ -221,7 +221,7 @@ static inline void dcblock_set_frame_alignment(struct audio_stream __sparse_cach static void dcblock_params(struct processing_module *mod) { struct sof_ipc_stream_params *params = mod->stream_params; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; struct comp_buffer *sinkb, *sourceb; struct comp_dev *dev = mod->dev; @@ -248,12 +248,12 @@ static void dcblock_params(struct processing_module *mod) * \return Error code. */ static int dcblock_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) { struct comp_data *cd = module_get_private_data(mod); struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct comp_dev *dev = mod->dev; comp_info(dev, "dcblock_prepare()"); diff --git a/src/audio/dcblock/dcblock_generic.c b/src/audio/dcblock/dcblock_generic.c index 3fb6b476d5e6..459d51993f09 100644 --- a/src/audio/dcblock/dcblock_generic.c +++ b/src/audio/dcblock/dcblock_generic.c @@ -36,8 +36,8 @@ static int32_t dcblock_generic(struct dcblock_state *state, #if CONFIG_FORMAT_S16LE static void dcblock_s16_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { struct dcblock_state *state; @@ -76,8 +76,8 @@ static void dcblock_s16_default(struct comp_data *cd, #if CONFIG_FORMAT_S24LE static void dcblock_s24_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { struct dcblock_state *state; @@ -116,8 +116,8 @@ static void dcblock_s24_default(struct comp_data *cd, #if CONFIG_FORMAT_S32LE static void dcblock_s32_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { struct dcblock_state *state; diff --git a/src/audio/dcblock/dcblock_hifi3.c b/src/audio/dcblock/dcblock_hifi3.c index e8ef80032bb6..30d8aa4d32a8 100644 --- a/src/audio/dcblock/dcblock_hifi3.c +++ b/src/audio/dcblock/dcblock_hifi3.c @@ -29,7 +29,7 @@ static inline ae_int32x2 dcblock_cal(ae_int32x2 R, ae_int32x2 state_x, ae_int32 } /* Setup circular for component source */ -static inline void dcblock_set_circular(const struct audio_stream __sparse_cache *source) +static inline void dcblock_set_circular(const struct audio_stream *source) { /* Set source as circular buffer 0 */ AE_SETCBEGIN0(audio_stream_get_addr(source)); @@ -38,8 +38,8 @@ static inline void dcblock_set_circular(const struct audio_stream __sparse_cache #if CONFIG_FORMAT_S16LE static void dcblock_s16_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { ae_int16 *src = audio_stream_get_rptr(source); @@ -85,8 +85,8 @@ static void dcblock_s16_default(struct comp_data *cd, #if CONFIG_FORMAT_S24LE static void dcblock_s24_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { ae_int32 *src = audio_stream_get_rptr(source); @@ -132,8 +132,8 @@ static void dcblock_s24_default(struct comp_data *cd, #if CONFIG_FORMAT_S32LE static void dcblock_s32_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { ae_int32 *src = audio_stream_get_rptr(source); diff --git a/src/audio/dcblock/dcblock_hifi4.c b/src/audio/dcblock/dcblock_hifi4.c index b537b3dce7b0..98d8edf6345e 100644 --- a/src/audio/dcblock/dcblock_hifi4.c +++ b/src/audio/dcblock/dcblock_hifi4.c @@ -29,8 +29,8 @@ static inline ae_int32x2 dcblock_cal(ae_int32x2 R, ae_int32x2 state_x, ae_int32 } /* Setup circular for component sink and source */ -static inline void dcblock_set_circular(const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink) +static inline void dcblock_set_circular(const struct audio_stream *source, + const struct audio_stream *sink) { /* Set source as circular buffer 0 */ AE_SETCBEGIN0(audio_stream_get_addr(source)); @@ -43,8 +43,8 @@ static inline void dcblock_set_circular(const struct audio_stream __sparse_cache #if CONFIG_FORMAT_S16LE static void dcblock_s16_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { ae_int16 *in; @@ -80,8 +80,8 @@ static void dcblock_s16_default(struct comp_data *cd, #if CONFIG_FORMAT_S24LE static void dcblock_s24_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { ae_int32 *in; @@ -118,8 +118,8 @@ static void dcblock_s24_default(struct comp_data *cd, #if CONFIG_FORMAT_S32LE static void dcblock_s32_default(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames) { ae_int32 *in; diff --git a/src/audio/drc/drc.c b/src/audio/drc/drc.c index a435e347a608..47462b3f79ec 100644 --- a/src/audio/drc/drc.c +++ b/src/audio/drc/drc.c @@ -227,8 +227,8 @@ static int drc_get_config(struct processing_module *mod, return comp_data_blob_get_cmd(cd->model_handler, cdata, fragment_size); } -static void drc_set_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static void drc_set_alignment(struct audio_stream *source, + struct audio_stream *sink) { /* Currently no optimizations those would use wider loads and stores */ audio_stream_init_alignment_constants(1, 1, source); @@ -243,8 +243,8 @@ static int drc_process(struct processing_module *mod, { struct drc_comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; - struct audio_stream __sparse_cache *source = input_buffers[0].data; - struct audio_stream __sparse_cache *sink = output_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; + struct audio_stream *sink = output_buffers[0].data; int frames = input_buffers[0].size; int ret; @@ -272,7 +272,7 @@ static int drc_process(struct processing_module *mod, static void drc_params(struct processing_module *mod) { struct sof_ipc_stream_params *params = mod->stream_params; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; struct comp_buffer *sinkb, *sourceb; struct comp_dev *dev = mod->dev; @@ -294,12 +294,12 @@ static void drc_params(struct processing_module *mod) #endif /* CONFIG_IPC_MAJOR_4 */ static int drc_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) { struct drc_comp_data *cd = module_get_private_data(mod); struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct comp_dev *dev = mod->dev; int channels; int rate; diff --git a/src/audio/drc/drc_generic.c b/src/audio/drc/drc_generic.c index fb7978c54cdc..472431a0e408 100644 --- a/src/audio/drc/drc_generic.c +++ b/src/audio/drc/drc_generic.c @@ -466,8 +466,8 @@ static void drc_process_one_division(struct drc_state *state, } void drc_default_pass(struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, uint32_t frames) + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { audio_stream_copy(source, 0, sink, 0, frames * audio_stream_get_channels(source)); } @@ -479,8 +479,8 @@ static inline void drc_pre_delay_index_inc(int *idx, int increment) #if CONFIG_FORMAT_S16LE static void drc_delay_input_sample_s16(struct drc_state *state, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int16_t **x, int16_t **y, int samples) { int16_t *x1; @@ -528,8 +528,8 @@ static void drc_delay_input_sample_s16(struct drc_state *state, } static void drc_s16_default(struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { int16_t *x = audio_stream_get_rptr(source); @@ -574,8 +574,8 @@ static void drc_s16_default(struct processing_module *mod, #if CONFIG_FORMAT_S24LE || CONFIG_FORMAT_S32LE static void drc_delay_input_sample_s32(struct drc_state *state, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int32_t **x, int32_t **y, int samples) { int32_t *x1; @@ -625,8 +625,8 @@ static void drc_delay_input_sample_s32(struct drc_state *state, #if CONFIG_FORMAT_S24LE static void drc_delay_input_sample_s24(struct drc_state *state, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int32_t **x, int32_t **y, int samples) { int32_t *x1; @@ -674,8 +674,8 @@ static void drc_delay_input_sample_s24(struct drc_state *state, } static void drc_s24_default(struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { int32_t *x = audio_stream_get_rptr(source); @@ -722,8 +722,8 @@ static void drc_s24_default(struct processing_module *mod, #if CONFIG_FORMAT_S32LE static void drc_s32_default(struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { int32_t *x = audio_stream_get_rptr(source); diff --git a/src/audio/eq_fir/eq_fir.c b/src/audio/eq_fir/eq_fir.c index 5fad4db4babf..5d304d15c368 100644 --- a/src/audio/eq_fir/eq_fir.c +++ b/src/audio/eq_fir/eq_fir.c @@ -174,7 +174,7 @@ static int eq_fir_params(struct processing_module *mod) struct sof_ipc_stream_params comp_params; struct comp_dev *dev = mod->dev; struct comp_buffer *sinkb; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; enum sof_ipc_frame valid_fmt, frame_fmt; int i, ret; @@ -213,8 +213,8 @@ static void eq_fir_passthrough(struct fir_state_32x16 fir[], struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; audio_stream_copy(source, 0, sink, 0, frames * audio_stream_get_channels(source)); } @@ -512,7 +512,7 @@ static int eq_fir_process(struct processing_module *mod, int num_output_buffers) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = input_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; uint32_t frame_count = input_buffers[0].size; int ret; @@ -554,8 +554,8 @@ static int eq_fir_process(struct processing_module *mod, return 0; } -static void eq_fir_set_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static void eq_fir_set_alignment(struct audio_stream *source, + struct audio_stream *sink) { const uint32_t byte_align = 1; const uint32_t frame_align_req = 2; /* Process multiples of 2 frames */ @@ -565,12 +565,12 @@ static void eq_fir_set_alignment(struct audio_stream __sparse_cache *source, } static int eq_fir_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) { struct comp_data *cd = module_get_private_data(mod); struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct comp_dev *dev = mod->dev; int channels; enum sof_ipc_frame frame_fmt; diff --git a/src/audio/eq_fir/eq_fir_generic.c b/src/audio/eq_fir/eq_fir_generic.c index ff763d325864..7edea7b21057 100644 --- a/src/audio/eq_fir/eq_fir_generic.c +++ b/src/audio/eq_fir/eq_fir_generic.c @@ -23,8 +23,8 @@ LOG_MODULE_DECLARE(eq_fir, CONFIG_SOF_LOG_LEVEL); void eq_fir_s16(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *filter; int32_t z; int16_t *x0, *y0; @@ -61,8 +61,8 @@ void eq_fir_s16(struct fir_state_32x16 fir[], struct input_stream_buffer *bsourc void eq_fir_s24(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *filter; int32_t z; int32_t *x0, *y0; @@ -99,8 +99,8 @@ void eq_fir_s24(struct fir_state_32x16 fir[], struct input_stream_buffer *bsourc void eq_fir_s32(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *filter; int32_t *x0, *y0; int32_t *x = audio_stream_get_rptr(source); diff --git a/src/audio/eq_fir/eq_fir_hifi2ep.c b/src/audio/eq_fir/eq_fir_hifi2ep.c index 29d12587ab1f..fc5ce500b9de 100644 --- a/src/audio/eq_fir/eq_fir_hifi2ep.c +++ b/src/audio/eq_fir/eq_fir_hifi2ep.c @@ -28,8 +28,8 @@ LOG_MODULE_DECLARE(eq_fir, CONFIG_SOF_LOG_LEVEL); void eq_fir_2x_s32(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *f; int32_t *src = audio_stream_get_rptr(source); int32_t *snk = audio_stream_get_wptr(sink); @@ -72,8 +72,8 @@ void eq_fir_2x_s32(struct fir_state_32x16 fir[], struct input_stream_buffer *bso void eq_fir_2x_s24(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *f; int32_t *src = audio_stream_get_rptr(source); int32_t *snk = audio_stream_get_wptr(sink); @@ -120,8 +120,8 @@ void eq_fir_2x_s24(struct fir_state_32x16 fir[], struct input_stream_buffer *bso void eq_fir_2x_s16(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *f; int16_t *src = audio_stream_get_rptr(source); int16_t *snk = audio_stream_get_wptr(sink); diff --git a/src/audio/eq_fir/eq_fir_hifi3.c b/src/audio/eq_fir/eq_fir_hifi3.c index b9b5f3b1ec63..00ce836741f5 100644 --- a/src/audio/eq_fir/eq_fir_hifi3.c +++ b/src/audio/eq_fir/eq_fir_hifi3.c @@ -27,8 +27,8 @@ LOG_MODULE_DECLARE(eq_fir, CONFIG_SOF_LOG_LEVEL); void eq_fir_2x_s32(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *f; ae_int32x2 d0 = 0; ae_int32x2 d1 = 0; @@ -84,8 +84,8 @@ void eq_fir_2x_s32(struct fir_state_32x16 fir[], struct input_stream_buffer *bso void eq_fir_2x_s24(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *f; ae_int32x2 d0 = 0; ae_int32x2 d1 = 0; @@ -156,8 +156,8 @@ void eq_fir_2x_s24(struct fir_state_32x16 fir[], struct input_stream_buffer *bso void eq_fir_2x_s16(struct fir_state_32x16 fir[], struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct fir_state_32x16 *f; ae_int16x4 d0 = AE_ZERO16(); ae_int16x4 d1 = AE_ZERO16(); diff --git a/src/audio/eq_iir/eq_iir.c b/src/audio/eq_iir/eq_iir.c index 1ff8ed6879e6..cf9c6fdbf6c1 100644 --- a/src/audio/eq_iir/eq_iir.c +++ b/src/audio/eq_iir/eq_iir.c @@ -64,8 +64,8 @@ static void eq_iir_s16_default(struct processing_module *mod, struct input_strea struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct iir_state_df1 *filter; int16_t *x0; int16_t *y0; @@ -112,8 +112,8 @@ static void eq_iir_s24_default(struct processing_module *mod, struct input_strea struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct iir_state_df1 *filter; int32_t *x0; int32_t *y0; @@ -160,8 +160,8 @@ static void eq_iir_s32_default(struct processing_module *mod, struct input_strea struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct iir_state_df1 *filter; int32_t *x0; int32_t *y0; @@ -209,8 +209,8 @@ static void eq_iir_s32_16_default(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct iir_state_df1 *filter; int32_t *x0; int16_t *y0; @@ -257,8 +257,8 @@ static void eq_iir_s32_24_default(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct iir_state_df1 *filter; int32_t *x0; int32_t *y0; @@ -303,8 +303,8 @@ static void eq_iir_s32_24_default(struct processing_module *mod, static void eq_iir_pass(struct processing_module *mod, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, uint32_t frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; audio_stream_copy(source, 0, sink, 0, frames * audio_stream_get_channels(source)); } @@ -314,8 +314,8 @@ static void eq_iir_pass(struct processing_module *mod, struct input_stream_buffe static void eq_iir_s32_s16_pass(struct processing_module *mod, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, uint32_t frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x = audio_stream_get_rptr(source); int16_t *y = audio_stream_get_wptr(sink); int nmax; @@ -344,8 +344,8 @@ static void eq_iir_s32_s16_pass(struct processing_module *mod, struct input_stre static void eq_iir_s32_s24_pass(struct processing_module *mod, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, uint32_t frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x = audio_stream_get_rptr(source); int32_t *y = audio_stream_get_wptr(sink); int nmax; @@ -694,7 +694,7 @@ static int eq_iir_verify_params(struct comp_dev *dev, struct sof_ipc_stream_params *params) { struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; uint32_t buffer_flag; int ret; @@ -794,8 +794,8 @@ static int eq_iir_process(struct processing_module *mod, struct output_stream_buffer *output_buffers, int num_output_buffers) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = input_buffers[0].data; - struct audio_stream __sparse_cache *sink = output_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; + struct audio_stream *sink = output_buffers[0].data; uint32_t frame_count = input_buffers[0].size; int ret; @@ -821,8 +821,8 @@ static int eq_iir_process(struct processing_module *mod, * \param[in,out] source Structure pointer of source. * \param[in,out] sink Structure pointer of sink. */ -static void eq_iir_set_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static void eq_iir_set_alignment(struct audio_stream *source, + struct audio_stream *sink) { const uint32_t byte_align = 8; const uint32_t frame_align_req = 2; @@ -838,7 +838,7 @@ static int eq_iir_params(struct processing_module *mod) struct sof_ipc_stream_params comp_params; struct comp_dev *dev = mod->dev; struct comp_buffer *sinkb; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; enum sof_ipc_frame valid_fmt, frame_fmt; int i, ret; @@ -880,12 +880,12 @@ static void eq_iir_set_passthrough_func(struct comp_data *cd, } static int eq_iir_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) { struct comp_data *cd = module_get_private_data(mod); struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct comp_dev *dev = mod->dev; enum sof_ipc_frame source_format; enum sof_ipc_frame sink_format; diff --git a/src/audio/google/google_hotword_detect.c b/src/audio/google/google_hotword_detect.c index b948420ec36b..766a26b0616e 100644 --- a/src/audio/google/google_hotword_detect.c +++ b/src/audio/google/google_hotword_detect.c @@ -159,7 +159,7 @@ static int ghd_params(struct comp_dev *dev, struct sof_ipc_stream_params *params) { struct comp_buffer *sourceb; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; int ret; /* Detector is used only in KPB topology. It always requires channels @@ -383,8 +383,8 @@ static int ghd_copy(struct comp_dev *dev) { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *source; - struct comp_buffer __sparse_cache *source_c; - struct audio_stream __sparse_cache *stream; + struct comp_buffer *source_c; + struct audio_stream *stream; uint32_t bytes, tail_bytes, head_bytes = 0; int ret; diff --git a/src/audio/google/google_rtc_audio_processing.c b/src/audio/google/google_rtc_audio_processing.c index 513f8dbb3334..6a3d038dbd4e 100644 --- a/src/audio/google/google_rtc_audio_processing.c +++ b/src/audio/google/google_rtc_audio_processing.c @@ -89,7 +89,7 @@ static int google_rtc_audio_processing_params( int ret; #if CONFIG_IPC_MAJOR_4 struct google_rtc_audio_processing_comp_data *cd = comp_get_drvdata(dev); - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct comp_buffer *sink; /* update sink buffer format */ @@ -502,7 +502,7 @@ static int google_rtc_audio_processing_prepare(struct comp_dev *dev) { struct google_rtc_audio_processing_comp_data *cd = comp_get_drvdata(dev); struct list_item *source_buffer_list_item; - struct comp_buffer __sparse_cache *output_c; + struct comp_buffer *output_c; unsigned int aec_channels = 0, frame_fmt, rate; int ret; @@ -512,7 +512,7 @@ static int google_rtc_audio_processing_prepare(struct comp_dev *dev) list_for_item(source_buffer_list_item, &dev->bsource_list) { struct comp_buffer *source = container_of(source_buffer_list_item, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); + struct comp_buffer *source_c = buffer_acquire(source); #if CONFIG_IPC_MAJOR_4 if (IPC4_SINK_QUEUE_ID(source_c->id) == SOF_AEC_FEEDBACK_QUEUE_ID) { @@ -592,7 +592,7 @@ static int google_rtc_audio_processing_reset(struct comp_dev *dev) static int google_rtc_audio_processing_copy(struct comp_dev *dev) { struct google_rtc_audio_processing_comp_data *cd = comp_get_drvdata(dev); - struct comp_buffer __sparse_cache *buffer_c, *mic_buf, *output_buf; + struct comp_buffer *buffer_c, *mic_buf, *output_buf; struct comp_copy_limits cl; int16_t *src, *dst, *ref; uint32_t num_aec_reference_frames; diff --git a/src/audio/host-legacy.c b/src/audio/host-legacy.c index 546cc6edbfc1..2574e3b54467 100644 --- a/src/audio/host-legacy.c +++ b/src/audio/host-legacy.c @@ -105,7 +105,7 @@ static int host_dma_set_config_and_copy(struct host_data *hd, struct comp_dev *d */ static uint32_t host_get_copy_bytes_one_shot(struct host_data *hd, struct comp_dev *dev) { - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(hd->local_buffer); + struct comp_buffer *buffer_c = buffer_acquire(hd->local_buffer); uint32_t copy_bytes; /* calculate minimum size to copy */ @@ -168,7 +168,7 @@ static int host_copy_one_shot(struct host_data *hd, struct comp_dev *dev, copy_c static uint32_t host_get_copy_bytes_one_shot(struct host_data *hd, struct comp_dev *dev) { struct dma_sg_elem *local_elem = hd->config.elem_array.elems; - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(hd->local_buffer); + struct comp_buffer *buffer_c = buffer_acquire(hd->local_buffer); uint32_t copy_bytes; uint32_t split_value; @@ -232,8 +232,8 @@ static int host_copy_one_shot(struct host_data *hd, struct comp_dev *dev, copy_c void host_common_update(struct host_data *hd, struct comp_dev *dev, uint32_t bytes) { - struct comp_buffer __sparse_cache *source; - struct comp_buffer __sparse_cache *sink; + struct comp_buffer *source; + struct comp_buffer *sink; int ret; bool update_mailbox = false; bool send_ipc = false; @@ -372,7 +372,7 @@ static void host_dma_cb(void *arg, enum notify_id type, void *data) */ static uint32_t host_get_copy_bytes_normal(struct host_data *hd, struct comp_dev *dev) { - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t avail_bytes = 0; uint32_t free_bytes = 0; uint32_t copy_bytes = 0; @@ -446,7 +446,7 @@ static int host_copy_normal(struct host_data *hd, struct comp_dev *dev, copy_cal static int create_local_elems(struct host_data *hd, struct comp_dev *dev, uint32_t buffer_count, uint32_t buffer_bytes) { - struct comp_buffer __sparse_cache *dma_buf_c; + struct comp_buffer *dma_buf_c; struct dma_sg_elem_array *elem_array; uint32_t dir; int err; @@ -690,8 +690,8 @@ int host_common_params(struct host_data *hd, struct comp_dev *dev, struct sof_ipc_stream_params *params, notifier_callback_t cb) { struct dma_sg_config *config = &hd->config; - struct comp_buffer __sparse_cache *host_buf_c; - struct comp_buffer __sparse_cache *dma_buf_c; + struct comp_buffer *host_buf_c; + struct comp_buffer *dma_buf_c; uint32_t period_count; uint32_t period_bytes; uint32_t buffer_size; @@ -886,7 +886,7 @@ static int host_params(struct comp_dev *dev, int host_common_prepare(struct host_data *hd) { - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(hd->dma_buffer); + struct comp_buffer *buf_c = buffer_acquire(hd->dma_buffer); buffer_zero(buf_c); buffer_release(buf_c); diff --git a/src/audio/host-zephyr.c b/src/audio/host-zephyr.c index 5a2a57438536..49f64e42179d 100644 --- a/src/audio/host-zephyr.c +++ b/src/audio/host-zephyr.c @@ -109,7 +109,7 @@ static int host_dma_set_config_and_copy(struct host_data *hd, struct comp_dev *d static uint32_t host_get_copy_bytes_one_shot(struct host_data *hd) { struct comp_buffer *buffer = hd->local_buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t copy_bytes; buffer_c = buffer_acquire(buffer); @@ -175,7 +175,7 @@ static uint32_t host_get_copy_bytes_one_shot(struct host_data *hd) { struct dma_sg_elem *local_elem = hd->config.elem_array.elems; struct comp_buffer *buffer = hd->local_buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t copy_bytes; uint32_t split_value; @@ -241,8 +241,8 @@ static int host_copy_one_shot(struct host_data *hd, struct comp_dev *dev, copy_c void host_common_update(struct host_data *hd, struct comp_dev *dev, uint32_t bytes) { - struct comp_buffer __sparse_cache *source; - struct comp_buffer __sparse_cache *sink; + struct comp_buffer *source; + struct comp_buffer *sink; int ret; bool update_mailbox = false; bool send_ipc = false; @@ -379,8 +379,8 @@ static void host_dma_cb(struct comp_dev *dev, size_t bytes) static uint32_t host_get_copy_bytes_normal(struct host_data *hd, struct comp_dev *dev) { struct comp_buffer *buffer = hd->local_buffer; - struct comp_buffer __sparse_cache *buffer_c; - struct comp_buffer __sparse_cache *dma_buf_c; + struct comp_buffer *buffer_c; + struct comp_buffer *dma_buf_c; struct dma_status dma_stat; uint32_t avail_samples; uint32_t free_samples; @@ -464,7 +464,7 @@ static inline bool stream_sync(struct host_data *hd, struct comp_dev *dev) */ static int host_copy_normal(struct host_data *hd, struct comp_dev *dev, copy_callback_t cb) { - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t copy_bytes; const unsigned int threshold = #if CONFIG_HOST_DMA_RELOAD_DELAY_ENABLE @@ -512,7 +512,7 @@ static int host_copy_normal(struct host_data *hd, struct comp_dev *dev, copy_cal static int create_local_elems(struct host_data *hd, struct comp_dev *dev, uint32_t buffer_count, uint32_t buffer_bytes, uint32_t direction) { - struct comp_buffer __sparse_cache *dma_buf_c; + struct comp_buffer *dma_buf_c; struct dma_sg_elem_array *elem_array; uint32_t dir; int err; @@ -759,8 +759,8 @@ int host_common_params(struct host_data *hd, struct comp_dev *dev, struct dma_sg_elem *sg_elem; struct dma_config *dma_cfg = &hd->z_config; struct dma_block_config dma_block_cfg; - struct comp_buffer __sparse_cache *host_buf_c; - struct comp_buffer __sparse_cache *dma_buf_c; + struct comp_buffer *host_buf_c; + struct comp_buffer *dma_buf_c; uint32_t period_count; uint32_t period_bytes; uint32_t buffer_size; @@ -1007,7 +1007,7 @@ static int host_params(struct comp_dev *dev, int host_common_prepare(struct host_data *hd) { - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(hd->dma_buffer); + struct comp_buffer *buf_c = buffer_acquire(hd->dma_buffer); buffer_zero(buf_c); buffer_release(buf_c); diff --git a/src/audio/igo_nr/igo_nr.c b/src/audio/igo_nr/igo_nr.c index 7ee3884bd023..aaea52ed2178 100644 --- a/src/audio/igo_nr/igo_nr.c +++ b/src/audio/igo_nr/igo_nr.c @@ -66,8 +66,8 @@ static void igo_nr_lib_process(struct comp_data *cd) #if CONFIG_FORMAT_S16LE static void igo_nr_capture_s16(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int32_t frames) { int32_t nch = audio_stream_get_channels(source); @@ -125,8 +125,8 @@ static void igo_nr_capture_s16(struct comp_data *cd, #if CONFIG_FORMAT_S24LE static void igo_nr_capture_s24(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int32_t frames) { int32_t nch = audio_stream_get_channels(source); @@ -184,8 +184,8 @@ static void igo_nr_capture_s24(struct comp_data *cd, #if CONFIG_FORMAT_S32LE static void igo_nr_capture_s32(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int32_t frames) { int32_t nch = audio_stream_get_channels(source); @@ -379,7 +379,7 @@ static int32_t igo_nr_params(struct comp_dev *dev, { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *sinkb, *sourceb; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; int32_t err; comp_info(dev, "igo_nr_params()"); @@ -619,8 +619,8 @@ static int32_t igo_nr_cmd(struct comp_dev *dev, } static void igo_nr_process(struct comp_dev *dev, - struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, + struct comp_buffer *source, + struct comp_buffer *sink, struct comp_copy_limits *cl, int32_t frames) @@ -701,7 +701,7 @@ static int32_t igo_nr_copy(struct comp_dev *dev) { struct comp_copy_limits cl; struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct comp_data *cd = comp_get_drvdata(dev); int32_t src_frames; int32_t sink_frames; diff --git a/src/audio/kpb.c b/src/audio/kpb.c index dee693d6fd9d..45a8b7f9b3cc 100644 --- a/src/audio/kpb.c +++ b/src/audio/kpb.c @@ -120,18 +120,18 @@ static int kpb_register_client(struct comp_data *kpb, struct kpb_client *cli); static void kpb_init_draining(struct comp_dev *dev, struct kpb_client *cli); static enum task_state kpb_draining_task(void *arg); static int kpb_buffer_data(struct comp_dev *dev, - const struct comp_buffer __sparse_cache *source, size_t size); + const struct comp_buffer *source, size_t size); static size_t kpb_allocate_history_buffer(struct comp_data *kpb, size_t hb_size_req); static void kpb_clear_history_buffer(struct history_buffer *buff); static void kpb_free_history_buffer(struct history_buffer *buff); static inline bool kpb_is_sample_width_supported(uint32_t sampling_width); -static void kpb_copy_samples(struct comp_buffer __sparse_cache *sink, - struct comp_buffer __sparse_cache *source, size_t size, +static void kpb_copy_samples(struct comp_buffer *sink, + struct comp_buffer *source, size_t size, size_t sample_width, uint32_t channels); -static void kpb_drain_samples(void *source, struct audio_stream __sparse_cache *sink, +static void kpb_drain_samples(void *source, struct audio_stream *sink, size_t size, size_t sample_width); -static void kpb_buffer_samples(const struct audio_stream __sparse_cache *source, +static void kpb_buffer_samples(const struct audio_stream *source, int offset, void *sink, size_t size, size_t sample_width); static void kpb_reset_history_buffer(struct history_buffer *buff); @@ -330,7 +330,7 @@ static int kpb_bind(struct comp_dev *dev, void *data) list_for_item(blist, &dev->bsink_list) { struct comp_buffer *sink = container_of(blist, struct comp_buffer, source_list); - struct comp_buffer __sparse_cache *sink_c = buffer_acquire(sink); + struct comp_buffer *sink_c = buffer_acquire(sink); int sink_buf_id; if (!sink_c->sink) { @@ -848,7 +848,7 @@ static int kpb_prepare(struct comp_dev *dev) list_for_item(blist, &dev->bsink_list) { struct comp_buffer *sink = container_of(blist, struct comp_buffer, source_list); - struct comp_buffer __sparse_cache *sink_c = buffer_acquire(sink); + struct comp_buffer *sink_c = buffer_acquire(sink); enum sof_comp_type type; if (!sink_c->sink) { @@ -886,7 +886,7 @@ static int kpb_prepare(struct comp_dev *dev) list_for_item(sink_list, &dev->bsink_list) { struct comp_buffer *sink = container_of(sink_list, struct comp_buffer, source_list); - struct comp_buffer __sparse_cache *sink_c = buffer_acquire(sink); + struct comp_buffer *sink_c = buffer_acquire(sink); audio_stream_init_alignment_constants(byte_align, frame_align_req, &sink_c->stream); @@ -991,12 +991,12 @@ static int kpb_reset(struct comp_dev *dev) #ifdef KPB_HIFI3 #if CONFIG_FORMAT_S16LE -static void kpb_micselect_copy16(struct comp_buffer __sparse_cache *sink, - struct comp_buffer __sparse_cache *source, size_t size, +static void kpb_micselect_copy16(struct comp_buffer *sink, + struct comp_buffer *source, size_t size, uint32_t in_channels, uint32_t micsel_channels, uint32_t *offsets) { - struct audio_stream __sparse_cache *istream = &source->stream; - struct audio_stream __sparse_cache *ostream = &sink->stream; + struct audio_stream *istream = &source->stream; + struct audio_stream *ostream = &sink->stream; uint16_t ch; size_t i; @@ -1024,12 +1024,12 @@ static void kpb_micselect_copy16(struct comp_buffer __sparse_cache *sink, } #endif #if CONFIG_FORMAT_S24LE || CONFIG_FORMAT_S32LE -static void kpb_micselect_copy32(struct comp_buffer __sparse_cache *sink, - struct comp_buffer __sparse_cache *source, size_t size, +static void kpb_micselect_copy32(struct comp_buffer *sink, + struct comp_buffer *source, size_t size, uint32_t in_channels, uint32_t micsel_channels, uint32_t *offsets) { - struct audio_stream __sparse_cache *istream = &source->stream; - struct audio_stream __sparse_cache *ostream = &sink->stream; + struct audio_stream *istream = &source->stream; + struct audio_stream *ostream = &sink->stream; uint16_t ch; size_t i; @@ -1058,12 +1058,12 @@ static void kpb_micselect_copy32(struct comp_buffer __sparse_cache *sink, } #endif #else -static void kpb_micselect_copy16(struct comp_buffer __sparse_cache *sink, - struct comp_buffer __sparse_cache *source, size_t size, +static void kpb_micselect_copy16(struct comp_buffer *sink, + struct comp_buffer *source, size_t size, uint32_t in_channels, uint32_t micsel_channels, uint32_t *offsets) { - struct audio_stream __sparse_cache *istream = &source->stream; - struct audio_stream __sparse_cache *ostream = &sink->stream; + struct audio_stream *istream = &source->stream; + struct audio_stream *ostream = &sink->stream; buffer_stream_invalidate(source, size); size_t out_samples; @@ -1090,12 +1090,12 @@ static void kpb_micselect_copy16(struct comp_buffer __sparse_cache *sink, } } -static void kpb_micselect_copy32(struct comp_buffer __sparse_cache *sink, - struct comp_buffer __sparse_cache *source, size_t size, +static void kpb_micselect_copy32(struct comp_buffer *sink, + struct comp_buffer *source, size_t size, uint32_t in_channels, uint32_t micsel_channels, uint32_t *offsets) { - struct audio_stream __sparse_cache *istream = &source->stream; - struct audio_stream __sparse_cache *ostream = &sink->stream; + struct audio_stream *istream = &source->stream; + struct audio_stream *ostream = &sink->stream; buffer_stream_invalidate(source, size); size_t out_samples; @@ -1121,8 +1121,8 @@ static void kpb_micselect_copy32(struct comp_buffer __sparse_cache *sink, } } #endif -static void kpb_micselect_copy(struct comp_dev *dev, struct comp_buffer __sparse_cache *sink_c, - struct comp_buffer __sparse_cache *source_c, size_t copy_bytes, +static void kpb_micselect_copy(struct comp_dev *dev, struct comp_buffer *sink_c, + struct comp_buffer *source_c, size_t copy_bytes, uint32_t channels) { struct comp_data *kpb = comp_get_drvdata(dev); @@ -1167,7 +1167,7 @@ static int kpb_copy(struct comp_dev *dev) int ret = 0; struct comp_data *kpb = comp_get_drvdata(dev); struct comp_buffer *source, *sink; - struct comp_buffer __sparse_cache *source_c, *sink_c = NULL; + struct comp_buffer *source_c, *sink_c = NULL; size_t copy_bytes = 0, produced_bytes = 0; size_t sample_width = kpb->config.sampling_width; struct draining_data *dd = &kpb->draining_task_data; @@ -1365,7 +1365,7 @@ static int kpb_copy(struct comp_dev *dev) * */ static int kpb_buffer_data(struct comp_dev *dev, - const struct comp_buffer __sparse_cache *source, size_t size) + const struct comp_buffer *source, size_t size) { int ret = 0; size_t size_to_copy = size; @@ -1729,7 +1729,7 @@ static void kpb_init_draining(struct comp_dev *dev, struct kpb_client *cli) static enum task_state kpb_draining_task(void *arg) { struct draining_data *draining_data = (struct draining_data *)arg; - struct comp_buffer __sparse_cache *sink = buffer_acquire(draining_data->sink); + struct comp_buffer *sink = buffer_acquire(draining_data->sink); struct history_buffer *buff = draining_data->hb; size_t drain_req = draining_data->drain_req; size_t sample_width = draining_data->sample_width; @@ -1880,7 +1880,7 @@ static enum task_state kpb_draining_task(void *arg) #ifdef KPB_HIFI3 static void kpb_convert_24b_to_32b(const void *linear_source, int ioffset, - struct audio_stream __sparse_cache *sink, int ooffset, + struct audio_stream *sink, int ooffset, unsigned int n_samples) { int ssize = audio_stream_sample_bytes(sink); @@ -1928,7 +1928,7 @@ static void kpb_convert_24b_to_32b(const void *linear_source, int ioffset, } #else static void kpb_convert_24b_to_32b(const void *source, int ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, int ooffset, unsigned int samples) { int ssize = audio_stream_sample_bytes(sink); @@ -1960,7 +1960,7 @@ static void kpb_convert_24b_to_32b(const void *source, int ioffset, * * \return none. */ -static void kpb_drain_samples(void *source, struct audio_stream __sparse_cache *sink, +static void kpb_drain_samples(void *source, struct audio_stream *sink, size_t size, size_t sample_width) { unsigned int samples; @@ -1989,7 +1989,7 @@ static void kpb_drain_samples(void *source, struct audio_stream __sparse_cache * } #ifdef KPB_HIFI3 -static void kpb_convert_32b_to_24b(const struct audio_stream __sparse_cache *source, int ioffset, +static void kpb_convert_32b_to_24b(const struct audio_stream *source, int ioffset, void *linear_sink, int ooffset, unsigned int n_samples) { int ssize = audio_stream_sample_bytes(source); @@ -2027,7 +2027,7 @@ static void kpb_convert_32b_to_24b(const struct audio_stream __sparse_cache *sou } } #else -static void kpb_convert_32b_to_24b(const struct audio_stream __sparse_cache *source, int ioffset, +static void kpb_convert_32b_to_24b(const struct audio_stream *source, int ioffset, void *sink, int ooffset, unsigned int samples) { int ssize = audio_stream_sample_bytes(source); @@ -2060,7 +2060,7 @@ static void kpb_convert_32b_to_24b(const struct audio_stream __sparse_cache *sou * \param[in] size Requested copy size in bytes. * \param[in] sample_width Sample size. */ -static void kpb_buffer_samples(const struct audio_stream __sparse_cache *source, +static void kpb_buffer_samples(const struct audio_stream *source, int offset, void *sink, size_t size, size_t sample_width) { @@ -2147,8 +2147,8 @@ static inline bool kpb_is_sample_width_supported(uint32_t sampling_width) } #ifdef KPB_HIFI3 -static void kpb_copy_24b_in_32b(const struct audio_stream __sparse_cache *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, uint32_t ooffset, +static void kpb_copy_24b_in_32b(const struct audio_stream *source, uint32_t ioffset, + struct audio_stream *sink, uint32_t ooffset, uint32_t n_samples) { int ssize = audio_stream_sample_bytes(source); /* src fmt == sink fmt */ @@ -2181,8 +2181,8 @@ static void kpb_copy_24b_in_32b(const struct audio_stream __sparse_cache *source } } #else -static void kpb_copy_24b_in_32b(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static void kpb_copy_24b_in_32b(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -2217,12 +2217,12 @@ static void kpb_copy_24b_in_32b(const struct audio_stream __sparse_cache *source * * \return none. */ -static void kpb_copy_samples(struct comp_buffer __sparse_cache *sink, - struct comp_buffer __sparse_cache *source, size_t size, +static void kpb_copy_samples(struct comp_buffer *sink, + struct comp_buffer *source, size_t size, size_t sample_width, uint32_t channels) { - struct audio_stream __sparse_cache *istream = &source->stream; - struct audio_stream __sparse_cache *ostream = &sink->stream; + struct audio_stream *istream = &source->stream; + struct audio_stream *ostream = &sink->stream; unsigned int samples; buffer_stream_invalidate(source, size); diff --git a/src/audio/mfcc/mfcc.c b/src/audio/mfcc/mfcc.c index b93e49dab743..bc3803a104ff 100644 --- a/src/audio/mfcc/mfcc.c +++ b/src/audio/mfcc/mfcc.c @@ -161,8 +161,8 @@ static int mfcc_process(struct processing_module *mod, struct output_stream_buffer *output_buffers, int num_output_buffers) { struct mfcc_comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = input_buffers->data; - struct audio_stream __sparse_cache *sink = output_buffers->data; + struct audio_stream *source = input_buffers->data; + struct audio_stream *sink = output_buffers->data; int frames = input_buffers->size; comp_dbg(mod->dev, "mfcc_process(), start"); @@ -187,14 +187,14 @@ static void mfcc_set_alignment(struct audio_stream *source, struct audio_stream } static int mfcc_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) { struct mfcc_comp_data *cd = module_get_private_data(mod); struct comp_buffer *sourceb; struct comp_buffer *sinkb; - struct comp_buffer __sparse_cache *source_c; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *source_c; + struct comp_buffer *sink_c; struct comp_dev *dev = mod->dev; enum sof_ipc_frame source_format; enum sof_ipc_frame sink_format; diff --git a/src/audio/mfcc/mfcc_common.c b/src/audio/mfcc/mfcc_common.c index 8ea1d7db4184..63ab84943669 100644 --- a/src/audio/mfcc/mfcc_common.c +++ b/src/audio/mfcc/mfcc_common.c @@ -143,7 +143,7 @@ static int mfcc_stft_process(const struct comp_dev *dev, struct mfcc_state *stat void mfcc_s16_default(struct processing_module *mod, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *sink = bsink->data; struct mfcc_comp_data *cd = module_get_private_data(mod); struct mfcc_state *state = &cd->state; struct mfcc_buffer *buf = &cd->state.buf; diff --git a/src/audio/mfcc/mfcc_generic.c b/src/audio/mfcc/mfcc_generic.c index 447b1382ce11..9384b16e0cee 100644 --- a/src/audio/mfcc/mfcc_generic.c +++ b/src/audio/mfcc/mfcc_generic.c @@ -27,7 +27,7 @@ void mfcc_source_copy_s16(struct input_stream_buffer *bsource, struct mfcc_buffer *buf, struct mfcc_pre_emph *emph, int frames, int source_channel) { - struct audio_stream __sparse_cache *source = bsource->data; + struct audio_stream *source = bsource->data; int32_t s; int16_t *x0; int16_t *x = audio_stream_get_rptr(source); diff --git a/src/audio/mfcc/mfcc_hifi3.c b/src/audio/mfcc/mfcc_hifi3.c index 6b274a58ae84..41062e014893 100644 --- a/src/audio/mfcc/mfcc_hifi3.c +++ b/src/audio/mfcc/mfcc_hifi3.c @@ -36,7 +36,7 @@ static inline void set_circular_buf0(const void *start, const void *end) void mfcc_source_copy_s16(struct input_stream_buffer *bsource, struct mfcc_buffer *buf, struct mfcc_pre_emph *emph, int frames, int source_channel) { - struct audio_stream __sparse_cache *source = bsource->data; + struct audio_stream *source = bsource->data; int copied; int nmax; int n; diff --git a/src/audio/mfcc/mfcc_hifi4.c b/src/audio/mfcc/mfcc_hifi4.c index 4e8853a4903b..25df53b4ca06 100644 --- a/src/audio/mfcc/mfcc_hifi4.c +++ b/src/audio/mfcc/mfcc_hifi4.c @@ -42,7 +42,7 @@ static inline void set_circular_buf1(const void *start, const void *end) void mfcc_source_copy_s16(struct input_stream_buffer *bsource, struct mfcc_buffer *buf, struct mfcc_pre_emph *emph, int frames, int source_channel) { - struct audio_stream __sparse_cache *source = bsource->data; + struct audio_stream *source = bsource->data; int num_channels = audio_stream_get_channels(source); ae_int16 *in = (ae_int16 *)source->r_ptr + source_channel; ae_int16 *out = (ae_int16 *)buf->w_ptr; diff --git a/src/audio/mixer/mixer.c b/src/audio/mixer/mixer.c index 3dc6ee1407dc..b26a880d1d1f 100644 --- a/src/audio/mixer/mixer.c +++ b/src/audio/mixer/mixer.c @@ -82,7 +82,7 @@ static int mixer_process(struct processing_module *mod, { struct mixer_data *md = module_get_private_data(mod); struct comp_dev *dev = mod->dev; - const struct audio_stream __sparse_cache *sources_stream[PLATFORM_MAX_STREAMS]; + const struct audio_stream *sources_stream[PLATFORM_MAX_STREAMS]; int sources_indices[PLATFORM_MAX_STREAMS]; int32_t i = 0, j = 0; uint32_t frames = INT32_MAX; @@ -174,7 +174,7 @@ static int mixer_reset(struct processing_module *mod) /* FIXME: this is racy and implicitly protected by serialised IPCs */ struct comp_buffer *source = container_of(blist, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); + struct comp_buffer *source_c = buffer_acquire(source); bool stop = false; if (source_c->source && source_c->source->state > COMP_STATE_READY) @@ -194,7 +194,7 @@ static int mixer_reset(struct processing_module *mod) } /* init and calculate the aligned setting for available frames and free frames retrieve*/ -static inline void mixer_set_frame_alignment(struct audio_stream __sparse_cache *source) +static inline void mixer_set_frame_alignment(struct audio_stream *source) { #if XCHAL_HAVE_HIFI3 || XCHAL_HAVE_HIFI4 @@ -220,11 +220,11 @@ static inline void mixer_set_frame_alignment(struct audio_stream __sparse_cache } static int mixer_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) { struct mixer_data *md = module_get_private_data(mod); - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct comp_dev *dev = mod->dev; struct comp_buffer *sink; struct list_item *blist; @@ -239,7 +239,7 @@ static int mixer_prepare(struct processing_module *mod, /* check each mixer source state */ list_for_item(blist, &dev->bsource_list) { struct comp_buffer *source; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; bool stop; /* diff --git a/src/audio/mixer/mixer_generic.c b/src/audio/mixer/mixer_generic.c index b8be3ecf63a1..0deabe2093a7 100644 --- a/src/audio/mixer/mixer_generic.c +++ b/src/audio/mixer/mixer_generic.c @@ -11,8 +11,8 @@ #if CONFIG_FORMAT_S16LE /* Mix n 16 bit PCM source streams to one sink stream */ -static void mix_n_s16(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +static void mix_n_s16(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames) { int16_t *src[PLATFORM_MAX_CHANNELS]; @@ -57,8 +57,8 @@ static void mix_n_s16(struct comp_dev *dev, struct audio_stream __sparse_cache * #if CONFIG_FORMAT_S24LE /* Mix n 24 bit PCM source streams to one sink stream */ -static void mix_n_s24(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +static void mix_n_s24(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames) { int32_t *src[PLATFORM_MAX_CHANNELS]; @@ -105,8 +105,8 @@ static void mix_n_s24(struct comp_dev *dev, struct audio_stream __sparse_cache * #if CONFIG_FORMAT_S32LE /* Mix n 32 bit PCM source streams to one sink stream */ -static void mix_n_s32(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +static void mix_n_s32(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames) { int32_t *src[PLATFORM_MAX_CHANNELS]; diff --git a/src/audio/mixer/mixer_hifi3.c b/src/audio/mixer/mixer_hifi3.c index 3147ce0a35b6..2bba6c213df6 100644 --- a/src/audio/mixer/mixer_hifi3.c +++ b/src/audio/mixer/mixer_hifi3.c @@ -13,8 +13,8 @@ #if CONFIG_FORMAT_S16LE /* Mix n 16 bit PCM source streams to one sink stream */ -static void mix_n_s16(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +static void mix_n_s16(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames) { ae_int16x4 * in[PLATFORM_MAX_CHANNELS]; @@ -69,8 +69,8 @@ static void mix_n_s16(struct comp_dev *dev, struct audio_stream __sparse_cache * #if CONFIG_FORMAT_S24LE /* Mix n 24 bit PCM source streams to one sink stream */ -static void mix_n_s24(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +static void mix_n_s24(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames) { ae_int32x2 *in[PLATFORM_MAX_CHANNELS]; @@ -114,8 +114,8 @@ static void mix_n_s24(struct comp_dev *dev, struct audio_stream __sparse_cache * #if CONFIG_FORMAT_S32LE /* Mix n 32 bit PCM source streams to one sink stream */ -static void mix_n_s32(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +static void mix_n_s32(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames) { ae_q32s * in[PLATFORM_MAX_CHANNELS]; diff --git a/src/audio/mixin_mixout/mixin_mixout.c b/src/audio/mixin_mixout/mixin_mixout.c index 78d69223de56..6538e8747fc3 100644 --- a/src/audio/mixin_mixout/mixin_mixout.c +++ b/src/audio/mixin_mixout/mixin_mixout.c @@ -167,9 +167,9 @@ static int mixout_free(struct processing_module *mod) } static int mix_and_remap(struct comp_dev *dev, const struct mixin_data *mixin_data, - uint16_t sink_index, struct audio_stream __sparse_cache *sink, + uint16_t sink_index, struct audio_stream *sink, uint32_t start_frame, uint32_t mixed_frames, - const struct audio_stream __sparse_cache *source, uint32_t frame_count) + const struct audio_stream *source, uint32_t frame_count) { const struct mixin_sink_config *sink_config; @@ -197,7 +197,7 @@ static int mix_and_remap(struct comp_dev *dev, const struct mixin_data *mixin_da } /* mix silence into stream, i.e. set not yet mixed data in stream to zero */ -static void silence(struct audio_stream __sparse_cache *stream, uint32_t start_frame, +static void silence(struct audio_stream *stream, uint32_t start_frame, uint32_t mixed_frames, uint32_t frame_count) { uint32_t skip_mixed_frames; @@ -274,19 +274,19 @@ static int mixin_process(struct processing_module *mod, * and frames free in each connected mixout sink buffer. */ for (i = 0; i < num_output_buffers; i++) { - struct comp_buffer __sparse_cache *unused_in_between_buf_c; + struct comp_buffer *unused_in_between_buf_c; struct comp_dev *mixout; uint16_t sink_id; struct comp_buffer *sink; struct mixout_data *mixout_data; struct processing_module *mixout_mod; struct module_source_info __sparse_cache *mod_source_info; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; uint32_t free_frames, pending_frames; /* unused buffer between mixin and mixout */ unused_in_between_buf_c = attr_container_of(output_buffers[i].data, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); mixout = unused_in_between_buf_c->sink; sink_id = IPC4_SRC_QUEUE_ID(unused_in_between_buf_c->id); @@ -334,7 +334,7 @@ static int mixin_process(struct processing_module *mod, } if (source_avail_frames > 0) { - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; frames_to_copy = MIN(source_avail_frames, sinks_free_frames); bytes_to_consume_from_source_buf = @@ -342,7 +342,7 @@ static int mixin_process(struct processing_module *mod, if (bytes_to_consume_from_source_buf > 0) { input_buffers[0].consumed = bytes_to_consume_from_source_buf; source_c = attr_container_of(input_buffers[0].data, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); buffer_stream_invalidate(source_c, bytes_to_consume_from_source_buf); } @@ -363,7 +363,7 @@ static int mixin_process(struct processing_module *mod, struct module_source_info __sparse_cache *mod_source_info; struct processing_module *mixout_mod; uint32_t start_frame; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; uint32_t writeback_size; mixout = active_mixouts[i]; @@ -455,14 +455,14 @@ static int mixout_process(struct processing_module *mod, * produced now. */ for (i = 0; i < num_input_buffers; i++) { - const struct audio_stream __sparse_cache *source_stream; - struct comp_buffer __sparse_cache *unused_in_between_buf; + const struct audio_stream *source_stream; + struct comp_buffer *unused_in_between_buf; struct comp_dev *source; int source_index; source_stream = input_buffers[i].data; unused_in_between_buf = attr_container_of(source_stream, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); source = unused_in_between_buf->source; @@ -480,15 +480,15 @@ static int mixout_process(struct processing_module *mod, if (frames_to_produce > 0 && frames_to_produce < INT32_MAX) { for (i = 0; i < num_input_buffers; i++) { - const struct audio_stream __sparse_cache *source_stream; - struct comp_buffer __sparse_cache *unused_in_between_buf; + const struct audio_stream *source_stream; + struct comp_buffer *unused_in_between_buf; struct comp_dev *source; int source_index; uint32_t pending_frames; source_stream = input_buffers[i].data; unused_in_between_buf = attr_container_of(source_stream, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); source = unused_in_between_buf->source; @@ -547,7 +547,7 @@ static int mixout_reset(struct processing_module *mod) if (dev->pipeline->source_comp->direction == SOF_IPC_STREAM_PLAYBACK) { list_for_item(blist, &dev->bsource_list) { struct comp_buffer *source; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; bool stop; /* FIXME: this is racy and implicitly protected by serialised IPCs */ @@ -584,7 +584,7 @@ static int mixin_params(struct processing_module *mod) */ list_for_item(blist, &dev->bsink_list) { struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; enum sof_ipc_frame frame_fmt, valid_fmt; uint16_t sink_id; @@ -641,13 +641,13 @@ static int mixin_params(struct processing_module *mod) * if downstream is not currently active. */ static int mixin_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) { struct mixin_data *md = module_get_private_data(mod); struct comp_dev *dev = mod->dev; struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; enum sof_ipc_frame fmt; int ret; @@ -687,7 +687,7 @@ static int mixout_params(struct processing_module *mod) { struct sof_ipc_stream_params *params = mod->stream_params; struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct comp_dev *dev = mod->dev; enum sof_ipc_frame frame_fmt, valid_fmt; uint32_t sink_period_bytes, sink_stream_size; @@ -737,8 +737,8 @@ static int mixout_params(struct processing_module *mod) } static int mixout_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) { struct module_source_info __sparse_cache *mod_source_info; struct comp_dev *dev = mod->dev; diff --git a/src/audio/mixin_mixout/mixin_mixout_generic.c b/src/audio/mixin_mixout/mixin_mixout_generic.c index 75468fdb4d67..a99788be32bc 100644 --- a/src/audio/mixin_mixout/mixin_mixout_generic.c +++ b/src/audio/mixin_mixout/mixin_mixout_generic.c @@ -17,9 +17,9 @@ * parameters: multichannel stream is treated as single channel and so the entire stream * contents is mixed. */ -static void normal_mix_channel_s16(struct audio_stream __sparse_cache *sink, int32_t start_frame, +static void normal_mix_channel_s16(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain) { int32_t frames_to_mix, frames_to_copy, left_frames; @@ -61,7 +61,7 @@ static void normal_mix_channel_s16(struct audio_stream __sparse_cache *sink, int } } -static void mute_channel_s16(struct audio_stream __sparse_cache *stream, int32_t channel_index, +static void mute_channel_s16(struct audio_stream *stream, int32_t channel_index, int32_t start_frame, int32_t mixed_frames, int32_t frame_count) { int32_t skip_mixed_frames, n, left_frames, i, channel_count, frames, samples; @@ -101,9 +101,9 @@ static void mute_channel_s16(struct audio_stream __sparse_cache *stream, int32_t * parameters: multichannel stream is treated as single channel and so the entire stream * contents is mixed. */ -static void normal_mix_channel_s24(struct audio_stream __sparse_cache *sink, int32_t start_frame, +static void normal_mix_channel_s24(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain) { int32_t frames_to_mix, frames_to_copy, left_frames; @@ -153,9 +153,9 @@ static void normal_mix_channel_s24(struct audio_stream __sparse_cache *sink, int * parameters: multichannel stream is treated as single channel and so the entire stream * contents is mixed. */ -static void normal_mix_channel_s32(struct audio_stream __sparse_cache *sink, int32_t start_frame, +static void normal_mix_channel_s32(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain) { int32_t frames_to_mix, frames_to_copy, left_frames; @@ -198,7 +198,7 @@ static void normal_mix_channel_s32(struct audio_stream __sparse_cache *sink, int #endif /* CONFIG_FORMAT_S32LE */ #if CONFIG_FORMAT_S32LE || CONFIG_FORMAT_S24LE -static void mute_channel_s32(struct audio_stream __sparse_cache *stream, int32_t channel_index, +static void mute_channel_s32(struct audio_stream *stream, int32_t channel_index, int32_t start_frame, int32_t mixed_frames, int32_t frame_count) { int32_t skip_mixed_frames, left_frames, n, channel_count, i, frames, samples; diff --git a/src/audio/mixin_mixout/mixin_mixout_hifi3.c b/src/audio/mixin_mixout/mixin_mixout_hifi3.c index 3fb46f2b74f6..93905a5d80af 100644 --- a/src/audio/mixin_mixout/mixin_mixout_hifi3.c +++ b/src/audio/mixin_mixout/mixin_mixout_hifi3.c @@ -16,9 +16,9 @@ * parameters: multichannel stream is treated as single channel and so the entire stream * contents is mixed. */ -static void normal_mix_channel_s16(struct audio_stream __sparse_cache *sink, int32_t start_frame, +static void normal_mix_channel_s16(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain) { int frames_to_mix, frames_to_copy, left_frames; @@ -104,7 +104,7 @@ static void normal_mix_channel_s16(struct audio_stream __sparse_cache *sink, int } } -static void mute_channel_s16(struct audio_stream __sparse_cache *stream, int32_t channel_index, +static void mute_channel_s16(struct audio_stream *stream, int32_t channel_index, int32_t start_frame, int32_t mixed_frames, int32_t frame_count) { int skip_mixed_frames, left_frames; @@ -140,9 +140,9 @@ static void mute_channel_s16(struct audio_stream __sparse_cache *stream, int32_t * parameters: multichannel stream is treated as single channel and so the entire stream * contents is mixed. */ -static void normal_mix_channel_s24(struct audio_stream __sparse_cache *sink, int32_t start_frame, +static void normal_mix_channel_s24(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain) { int frames_to_mix, frames_to_copy, left_frames; @@ -230,9 +230,9 @@ static void normal_mix_channel_s24(struct audio_stream __sparse_cache *sink, int * parameters: multichannel stream is treated as single channel and so the entire stream * contents is mixed. */ -static void normal_mix_channel_s32(struct audio_stream __sparse_cache *sink, int32_t start_frame, +static void normal_mix_channel_s32(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain) { int frames_to_mix, frames_to_copy, left_frames; @@ -315,7 +315,7 @@ static void normal_mix_channel_s32(struct audio_stream __sparse_cache *sink, int #endif /* CONFIG_FORMAT_S32LE */ #if CONFIG_FORMAT_S32LE || CONFIG_FORMAT_S24LE -static void mute_channel_s32(struct audio_stream __sparse_cache *stream, int32_t channel_index, +static void mute_channel_s32(struct audio_stream *stream, int32_t channel_index, int32_t start_frame, int32_t mixed_frames, int32_t frame_count) { int skip_mixed_frames, left_frames; diff --git a/src/audio/module_adapter/module/cadence.c b/src/audio/module_adapter/module/cadence.c index 9279ce3a430a..d3f9a9e407e7 100644 --- a/src/audio/module_adapter/module/cadence.c +++ b/src/audio/module_adapter/module/cadence.c @@ -633,8 +633,8 @@ static int cadence_codec_init_process(struct processing_module *mod) } static int cadence_codec_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 = 0, mem_tabs_size; struct comp_dev *dev = mod->dev; @@ -716,7 +716,7 @@ cadence_codec_process(struct processing_module *mod, struct output_stream_buffer *output_buffers, int num_output_buffers) { struct comp_buffer *local_buff; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct comp_dev *dev = mod->dev; struct module_data *codec = &mod->priv; struct cadence_codec_data *cd = codec->private; diff --git a/src/audio/module_adapter/module/dts/dts.c b/src/audio/module_adapter/module/dts/dts.c index 62415f38c660..b136b23f586e 100644 --- a/src/audio/module_adapter/module/dts/dts.c +++ b/src/audio/module_adapter/module/dts/dts.c @@ -78,8 +78,8 @@ static int dts_effect_populate_buffer_configuration(struct comp_dev *dev, { struct comp_buffer *source = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c; - const struct audio_stream __sparse_cache *stream; + struct comp_buffer *source_c; + const struct audio_stream *stream; DtsSofInterfaceBufferLayout buffer_layout; DtsSofInterfaceBufferFormat buffer_format; unsigned int buffer_fmt, frame_fmt, rate, channels; @@ -184,8 +184,8 @@ static int dts_codec_init(struct processing_module *mod) } static int dts_codec_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_dev *dev = mod->dev; diff --git a/src/audio/module_adapter/module/generic.c b/src/audio/module_adapter/module/generic.c index 760652982245..709230b6639f 100644 --- a/src/audio/module_adapter/module/generic.c +++ b/src/audio/module_adapter/module/generic.c @@ -196,8 +196,8 @@ static int validate_config(struct module_config *cfg) } int module_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 module_data *md = &mod->priv; @@ -283,8 +283,8 @@ int module_process_legacy(struct processing_module *mod, } int module_process_sink_src(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) { struct comp_dev *dev = mod->dev; diff --git a/src/audio/module_adapter/module/modules.c b/src/audio/module_adapter/module/modules.c index 865aa945baea..1bb917fd36a4 100644 --- a/src/audio/module_adapter/module/modules.c +++ b/src/audio/module_adapter/module/modules.c @@ -155,8 +155,8 @@ static int modules_init(struct processing_module *mod) * There is one assumption - all IADK modules utilize IPC4 protocol. */ static int modules_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) { struct comp_dev *dev = mod->dev; int ret = 0; diff --git a/src/audio/module_adapter/module/passthrough.c b/src/audio/module_adapter/module/passthrough.c index a7e41c04ff65..ef86eadc896e 100644 --- a/src/audio/module_adapter/module/passthrough.c +++ b/src/audio/module_adapter/module/passthrough.c @@ -21,8 +21,8 @@ static int passthrough_codec_init(struct processing_module *mod) } static int passthrough_codec_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) { struct comp_dev *dev = mod->dev; struct module_data *codec = &mod->priv; diff --git a/src/audio/module_adapter/module/waves/waves.c b/src/audio/module_adapter/module/waves/waves.c index 570a68f27b82..1124bc5d3a69 100644 --- a/src/audio/module_adapter/module/waves/waves.c +++ b/src/audio/module_adapter/module/waves/waves.c @@ -220,10 +220,10 @@ static int waves_effect_check(struct comp_dev *dev) source_list); struct comp_buffer *source = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); - struct comp_buffer __sparse_cache *sink_c = buffer_acquire(sink); - const struct audio_stream __sparse_cache *src_fmt = &source_c->stream; - const struct audio_stream __sparse_cache *snk_fmt = &sink_c->stream; + struct comp_buffer *source_c = buffer_acquire(source); + struct comp_buffer *sink_c = buffer_acquire(sink); + const struct audio_stream *src_fmt = &source_c->stream; + const struct audio_stream *snk_fmt = &sink_c->stream; int ret = 0; /* Init sink & source buffers */ @@ -303,10 +303,10 @@ static int waves_effect_init(struct processing_module *mod) struct comp_dev *dev = mod->dev; struct comp_buffer *source = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); + struct comp_buffer *source_c = buffer_acquire(source); struct module_data *codec = &mod->priv; struct waves_codec_data *waves_codec = codec->private; - const struct audio_stream __sparse_cache *src_fmt = &source_c->stream; + const struct audio_stream *src_fmt = &source_c->stream; MaxxStatus_t status; MaxxBuffer_Format_t sample_format; MaxxBuffer_Layout_t buffer_format; @@ -686,8 +686,8 @@ static int waves_codec_init(struct processing_module *mod) } static int waves_codec_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) { struct comp_dev *dev = mod->dev; int ret; diff --git a/src/audio/module_adapter/module_adapter.c b/src/audio/module_adapter/module_adapter.c index 81b0cb890bf8..824fe9fa831b 100644 --- a/src/audio/module_adapter/module_adapter.c +++ b/src/audio/module_adapter/module_adapter.c @@ -175,10 +175,10 @@ struct comp_dev *module_adapter_new(const struct comp_driver *drv, static int module_adapter_sink_src_prepare(struct comp_dev *dev) { - struct comp_buffer __sparse_cache *source_buffers_c[PLATFORM_MAX_STREAMS]; - struct comp_buffer __sparse_cache *sinks_buffers_c[PLATFORM_MAX_STREAMS]; - struct sof_sink __sparse_cache *audio_sink[PLATFORM_MAX_STREAMS]; - struct sof_source __sparse_cache *audio_src[PLATFORM_MAX_STREAMS]; + struct comp_buffer *source_buffers_c[PLATFORM_MAX_STREAMS]; + struct comp_buffer *sinks_buffers_c[PLATFORM_MAX_STREAMS]; + struct sof_sink *audio_sink[PLATFORM_MAX_STREAMS]; + struct sof_source *audio_src[PLATFORM_MAX_STREAMS]; struct processing_module *mod = comp_get_drvdata(dev); struct list_item *blist; uint32_t num_of_sources = 0; @@ -236,8 +236,8 @@ int module_adapter_prepare(struct comp_dev *dev) int ret; struct processing_module *mod = comp_get_drvdata(dev); struct module_data *md = &mod->priv; - struct comp_buffer __sparse_cache *buffer_c; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *buffer_c; + struct comp_buffer *sink_c; struct comp_buffer *sink; struct list_item *blist, *_blist; uint32_t buff_periods; @@ -567,7 +567,7 @@ int module_adapter_params(struct comp_dev *dev, struct sof_ipc_stream_params *pa * @bytes: number of bytes available in the source buffer */ static void -ca_copy_from_source_to_module(const struct audio_stream __sparse_cache *source, +ca_copy_from_source_to_module(const struct audio_stream *source, void __sparse_cache *buff, uint32_t buff_size, size_t bytes) { /* head_size - available data until end of source buffer */ @@ -595,7 +595,7 @@ ca_copy_from_source_to_module(const struct audio_stream __sparse_cache *source, * @bytes: number of bytes available in the module output buffer */ static void -ca_copy_from_module_to_sink(const struct audio_stream __sparse_cache *sink, +ca_copy_from_module_to_sink(const struct audio_stream *sink, void __sparse_cache *buff, size_t bytes) { /* head_size - free space until end of sink buffer */ @@ -624,7 +624,7 @@ ca_copy_from_module_to_sink(const struct audio_stream __sparse_cache *sink, * * \return: none. */ -static void generate_zeroes(struct comp_buffer __sparse_cache *sink, uint32_t bytes) +static void generate_zeroes(struct comp_buffer *sink, uint32_t bytes) { uint32_t tmp, copy_bytes = bytes; void *ptr; @@ -639,8 +639,8 @@ static void generate_zeroes(struct comp_buffer __sparse_cache *sink, uint32_t by comp_update_buffer_produce(sink, bytes); } -static void module_copy_samples(struct comp_dev *dev, struct comp_buffer __sparse_cache *src_buffer, - struct comp_buffer __sparse_cache *sink_buffer, uint32_t produced) +static void module_copy_samples(struct comp_dev *dev, struct comp_buffer *src_buffer, + struct comp_buffer *sink_buffer, uint32_t produced) { struct processing_module *mod = comp_get_drvdata(dev); struct comp_copy_limits cl; @@ -681,7 +681,7 @@ static void module_adapter_process_output(struct comp_dev *dev) { struct processing_module *mod = comp_get_drvdata(dev); struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct list_item *blist; int i = 0; @@ -692,7 +692,7 @@ static void module_adapter_process_output(struct comp_dev *dev) list_for_item(blist, &mod->sink_buffer_list) { if (mod->output_buffers[i].size > 0) { struct comp_buffer *buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; buffer = container_of(blist, struct comp_buffer, sink_list); buffer_c = buffer_acquire(buffer); @@ -714,7 +714,7 @@ static void module_adapter_process_output(struct comp_dev *dev) list_for_item(_blist, &mod->sink_buffer_list) { if (i == j) { struct comp_buffer *source; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; sink = container_of(blist, struct comp_buffer, source_list); source = container_of(_blist, struct comp_buffer, sink_list); @@ -739,8 +739,8 @@ static void module_adapter_process_output(struct comp_dev *dev) static uint32_t module_single_sink_setup(struct comp_dev *dev, - struct comp_buffer __sparse_cache **source_c, - struct comp_buffer __sparse_cache **sinks_c) + struct comp_buffer **source_c, + struct comp_buffer **sinks_c) { struct processing_module *mod = comp_get_drvdata(dev); struct list_item *blist; @@ -780,8 +780,8 @@ module_single_sink_setup(struct comp_dev *dev, static uint32_t module_single_source_setup(struct comp_dev *dev, - struct comp_buffer __sparse_cache **source_c, - struct comp_buffer __sparse_cache **sinks_c) + struct comp_buffer **source_c, + struct comp_buffer **sinks_c) { struct processing_module *mod = comp_get_drvdata(dev); struct list_item *blist; @@ -824,8 +824,8 @@ module_single_source_setup(struct comp_dev *dev, static int module_adapter_audio_stream_copy_1to1(struct comp_dev *dev) { struct processing_module *mod = comp_get_drvdata(dev); - struct comp_buffer __sparse_cache *source_c; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *source_c; + struct comp_buffer *sink_c; uint32_t num_output_buffers = 0; uint32_t frames; int ret; @@ -872,8 +872,8 @@ static int module_adapter_audio_stream_copy_1to1(struct comp_dev *dev) static int module_adapter_audio_stream_type_copy(struct comp_dev *dev) { - struct comp_buffer __sparse_cache *source_c[PLATFORM_MAX_STREAMS]; - struct comp_buffer __sparse_cache *sinks_c[PLATFORM_MAX_STREAMS]; + struct comp_buffer *source_c[PLATFORM_MAX_STREAMS]; + struct comp_buffer *sinks_c[PLATFORM_MAX_STREAMS]; struct processing_module *mod = comp_get_drvdata(dev); struct list_item *blist; uint32_t num_input_buffers, num_output_buffers; @@ -949,10 +949,10 @@ static int module_adapter_audio_stream_type_copy(struct comp_dev *dev) /* consume from all active input buffers */ for (i = 0; i < num_input_buffers; i++) { - struct comp_buffer __sparse_cache *src_c; + struct comp_buffer *src_c; src_c = attr_container_of(mod->input_buffers[i].data, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); if (mod->input_buffers[i].consumed) audio_stream_consume(&src_c->stream, mod->input_buffers[i].consumed); @@ -972,10 +972,10 @@ static int module_adapter_audio_stream_type_copy(struct comp_dev *dev) /* produce data into all active output buffers */ for (i = 0; i < num_output_buffers; i++) { - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; sink_c = attr_container_of(mod->output_buffers[i].data, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); if (!mod->skip_sink_buffer_writeback) @@ -1010,10 +1010,10 @@ static int module_adapter_audio_stream_type_copy(struct comp_dev *dev) static int module_adapter_sink_source_copy(struct comp_dev *dev) { - struct comp_buffer __sparse_cache *source_buffers_c[PLATFORM_MAX_STREAMS]; - struct comp_buffer __sparse_cache *sinks_buffers_c[PLATFORM_MAX_STREAMS]; - struct sof_sink __sparse_cache *audio_sink[PLATFORM_MAX_STREAMS]; - struct sof_source __sparse_cache *audio_src[PLATFORM_MAX_STREAMS]; + struct comp_buffer *source_buffers_c[PLATFORM_MAX_STREAMS]; + struct comp_buffer *sinks_buffers_c[PLATFORM_MAX_STREAMS]; + struct sof_sink *audio_sink[PLATFORM_MAX_STREAMS]; + struct sof_source *audio_src[PLATFORM_MAX_STREAMS]; struct processing_module *mod = comp_get_drvdata(dev); struct list_item *blist; uint32_t num_of_sources = 0; @@ -1075,7 +1075,7 @@ static int module_adapter_raw_data_type_copy(struct comp_dev *dev) struct processing_module *mod = comp_get_drvdata(dev); struct module_data *md = &mod->priv; struct comp_buffer *source, *sink; - struct comp_buffer __sparse_cache *sink_c = NULL; + struct comp_buffer *sink_c = NULL; struct list_item *blist; size_t size = MAX(mod->deep_buff_bytes, mod->period_bytes); uint32_t min_free_frames = UINT_MAX; @@ -1094,7 +1094,7 @@ static int module_adapter_raw_data_type_copy(struct comp_dev *dev) /* copy source samples into input buffer */ list_for_item(blist, &dev->bsource_list) { - struct comp_buffer __sparse_cache *src_c; + struct comp_buffer *src_c; uint32_t bytes_to_process; int frames, source_frame_bytes; @@ -1139,7 +1139,7 @@ static int module_adapter_raw_data_type_copy(struct comp_dev *dev) i = 0; /* consume from all input buffers */ list_for_item(blist, &dev->bsource_list) { - struct comp_buffer __sparse_cache *src_c; + struct comp_buffer *src_c; source = container_of(blist, struct comp_buffer, sink_list); src_c = buffer_acquire(source); @@ -1334,7 +1334,7 @@ static int module_source_status_count(struct comp_dev *dev, uint32_t status) */ struct comp_buffer *source = container_of(blist, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); + struct comp_buffer *source_c = buffer_acquire(source); if (source_c->source && source_c->source->state == status) count++; @@ -1424,7 +1424,7 @@ int module_adapter_reset(struct comp_dev *dev) list_for_item(blist, &mod->sink_buffer_list) { struct comp_buffer *buffer = container_of(blist, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(buffer); + struct comp_buffer *buffer_c = buffer_acquire(buffer); buffer_zero(buffer_c); buffer_release(buffer_c); diff --git a/src/audio/multiband_drc/multiband_drc.c b/src/audio/multiband_drc/multiband_drc.c index a23a6eebc830..9a836b3e5644 100644 --- a/src/audio/multiband_drc/multiband_drc.c +++ b/src/audio/multiband_drc/multiband_drc.c @@ -388,8 +388,8 @@ static int multiband_drc_get_config(struct processing_module *mod, return comp_data_blob_get_cmd(cd->model_handler, cdata, fragment_size); } -static void multiband_drc_set_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static void multiband_drc_set_alignment(struct audio_stream *source, + struct audio_stream *sink) { /* Currently no optimizations those would use wider loads and stores */ audio_stream_init_alignment_constants(1, 1, source); @@ -403,8 +403,8 @@ static int multiband_drc_process(struct processing_module *mod, { struct multiband_drc_comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; - struct audio_stream __sparse_cache *source = input_buffers[0].data; - struct audio_stream __sparse_cache *sink = output_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; + struct audio_stream *sink = output_buffers[0].data; int frames = input_buffers[0].size; int ret; @@ -435,7 +435,7 @@ static int multiband_drc_params(struct processing_module *mod) struct sof_ipc_stream_params comp_params; struct comp_dev *dev = mod->dev; struct comp_buffer *sinkb; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; enum sof_ipc_frame valid_fmt, frame_fmt; int i, ret; @@ -466,13 +466,13 @@ static int multiband_drc_params(struct processing_module *mod) #endif /* CONFIG_IPC_MAJOR_4 */ static int multiband_drc_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) { struct multiband_drc_comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; int channels; int rate; int ret = 0; diff --git a/src/audio/multiband_drc/multiband_drc_generic.c b/src/audio/multiband_drc/multiband_drc_generic.c index e4d3014c9270..fff21e2653bf 100644 --- a/src/audio/multiband_drc/multiband_drc_generic.c +++ b/src/audio/multiband_drc/multiband_drc_generic.c @@ -11,8 +11,8 @@ #include static void multiband_drc_default_pass(const struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { audio_stream_copy(source, 0, sink, 0, audio_stream_get_channels(source) * frames); @@ -203,8 +203,8 @@ static void multiband_drc_process_deemp(struct multiband_drc_state *state, */ #if CONFIG_FORMAT_S16LE static void multiband_drc_s16_default(const struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { struct multiband_drc_comp_data *cd = module_get_private_data(mod); @@ -270,8 +270,8 @@ static void multiband_drc_s16_default(const struct processing_module *mod, #if CONFIG_FORMAT_S24LE static void multiband_drc_s24_default(const struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { struct multiband_drc_comp_data *cd = module_get_private_data(mod); @@ -337,8 +337,8 @@ static void multiband_drc_s24_default(const struct processing_module *mod, #if CONFIG_FORMAT_S32LE static void multiband_drc_s32_default(const struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames) { struct multiband_drc_comp_data *cd = module_get_private_data(mod); diff --git a/src/audio/mux/mux.c b/src/audio/mux/mux.c index 3223f515438c..ca35d73b6131 100644 --- a/src/audio/mux/mux.c +++ b/src/audio/mux/mux.c @@ -274,7 +274,7 @@ static void set_mux_params(struct processing_module *mod) struct comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; struct comp_buffer *sink, *source; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; struct list_item *source_list; int j; const uint32_t byte_align = 1; @@ -375,10 +375,10 @@ static struct mux_look_up *get_lookup_table(struct comp_dev *dev, struct comp_da } static void mux_prepare_active_look_up(struct comp_data *cd, - struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources) + struct audio_stream *sink, + const struct audio_stream **sources) { - const struct audio_stream __sparse_cache *source; + const struct audio_stream *source; int elem; int active_elem = 0; @@ -400,8 +400,8 @@ static void mux_prepare_active_look_up(struct comp_data *cd, } static void demux_prepare_active_look_up(struct comp_data *cd, - struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, + struct audio_stream *sink, + const struct audio_stream *source, struct mux_look_up *look_up) { int elem; @@ -429,8 +429,8 @@ static int demux_process(struct processing_module *mod, struct comp_dev *dev = mod->dev; struct list_item *clist; struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; - struct audio_stream __sparse_cache *sinks_stream[MUX_MAX_STREAMS] = { NULL }; + struct comp_buffer *sink_c; + struct audio_stream *sinks_stream[MUX_MAX_STREAMS] = { NULL }; struct mux_look_up *look_ups[MUX_MAX_STREAMS] = { NULL }; int frames; int sink_bytes; @@ -487,9 +487,9 @@ static int mux_process(struct processing_module *mod, struct comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; struct comp_buffer *source; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; struct list_item *clist; - const struct audio_stream __sparse_cache *sources_stream[MUX_MAX_STREAMS] = { NULL }; + const struct audio_stream *sources_stream[MUX_MAX_STREAMS] = { NULL }; int frames = 0; int sink_bytes; int source_bytes; @@ -560,7 +560,7 @@ static int mux_reset(struct processing_module *mod) list_for_item(blist, &dev->bsource_list) { struct comp_buffer *source = container_of(blist, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); + struct comp_buffer *source_c = buffer_acquire(source); int state = source_c->source->state; buffer_release(source_c); @@ -578,16 +578,16 @@ static int mux_reset(struct processing_module *mod) } static int mux_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) { struct comp_dev *dev = mod->dev; struct comp_data *cd = module_get_private_data(mod); struct list_item *blist; struct comp_buffer *source; struct comp_buffer *sink; - struct comp_buffer __sparse_cache *source_c; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *source_c; + struct comp_buffer *sink_c; struct sof_mux_config *config; size_t blob_size; int state; diff --git a/src/audio/mux/mux_generic.c b/src/audio/mux/mux_generic.c index dd58cedbbd6b..a00a36cc084b 100644 --- a/src/audio/mux/mux_generic.c +++ b/src/audio/mux/mux_generic.c @@ -19,11 +19,11 @@ LOG_MODULE_DECLARE(muxdemux, CONFIG_SOF_LOG_LEVEL); -static void mux_check_for_wrap(struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, +static void mux_check_for_wrap(struct audio_stream *sink, + const struct audio_stream **sources, struct mux_look_up *lookup) { - const struct audio_stream __sparse_cache *source; + const struct audio_stream *source; uint32_t elem; /* check sources and destinations for wrap */ @@ -36,8 +36,8 @@ static void mux_check_for_wrap(struct audio_stream __sparse_cache *sink, } } -static void demux_check_for_wrap(struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, +static void demux_check_for_wrap(struct audio_stream *sink, + const struct audio_stream *source, struct mux_look_up *lookup) { uint32_t elem; @@ -53,7 +53,7 @@ static void demux_check_for_wrap(struct audio_stream __sparse_cache *sink, #if CONFIG_FORMAT_S16LE -static uint32_t demux_calc_frames_without_wrap_s16(struct audio_stream __sparse_cache *sink, +static uint32_t demux_calc_frames_without_wrap_s16(struct audio_stream *sink, const struct audio_stream __sparse_cache *source, struct mux_look_up *lookup) @@ -79,12 +79,12 @@ static uint32_t demux_calc_frames_without_wrap_s16(struct audio_stream __sparse_ return min_frames; } -static uint32_t mux_calc_frames_without_wrap_s16(struct audio_stream __sparse_cache *sink, +static uint32_t mux_calc_frames_without_wrap_s16(struct audio_stream *sink, const struct audio_stream __sparse_cache **sources, struct mux_look_up *lookup) { - const struct audio_stream __sparse_cache *source; + const struct audio_stream *source; uint32_t frames; uint32_t min_frames; uint32_t elem; @@ -111,11 +111,11 @@ static uint32_t mux_calc_frames_without_wrap_s16(struct audio_stream __sparse_ca return min_frames; } -static void mux_init_look_up_pointers_s16(struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, +static void mux_init_look_up_pointers_s16(struct audio_stream *sink, + const struct audio_stream **sources, struct mux_look_up *lookup) { - const struct audio_stream __sparse_cache *source; + const struct audio_stream *source; uint32_t elem; /* init pointers */ @@ -132,8 +132,8 @@ static void mux_init_look_up_pointers_s16(struct audio_stream __sparse_cache *si } } -static void demux_init_look_up_pointers_s16(struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, +static void demux_init_look_up_pointers_s16(struct audio_stream *sink, + const struct audio_stream *source, struct mux_look_up *lookup) { uint32_t elem; @@ -162,8 +162,8 @@ static void demux_init_look_up_pointers_s16(struct audio_stream __sparse_cache * * @param[in] frames Number of frames to process. * @param[in] lookup mux look up table. */ -static void demux_s16le(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames, +static void demux_s16le(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames, struct mux_look_up *lookup) { uint32_t i; @@ -215,8 +215,8 @@ static void demux_s16le(struct comp_dev *dev, struct audio_stream __sparse_cache * @param[in] frames Number of frames to process. * @param[in] lookup mux look up table. */ -static void mux_s16le(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t frames, +static void mux_s16le(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t frames, struct mux_look_up *lookup) { uint32_t i; @@ -259,12 +259,12 @@ static void mux_s16le(struct comp_dev *dev, struct audio_stream __sparse_cache * #if CONFIG_FORMAT_S24LE || CONFIG_FORMAT_S32LE -static uint32_t mux_calc_frames_without_wrap_s32(struct audio_stream __sparse_cache *sink, +static uint32_t mux_calc_frames_without_wrap_s32(struct audio_stream *sink, const struct audio_stream __sparse_cache **sources, struct mux_look_up *lookup) { - const struct audio_stream __sparse_cache *source; + const struct audio_stream *source; uint32_t frames; uint32_t min_frames; uint32_t elem; @@ -289,7 +289,7 @@ static uint32_t mux_calc_frames_without_wrap_s32(struct audio_stream __sparse_ca return min_frames; } -static uint32_t demux_calc_frames_without_wrap_s32(struct audio_stream __sparse_cache *sink, +static uint32_t demux_calc_frames_without_wrap_s32(struct audio_stream *sink, const struct audio_stream __sparse_cache *source, struct mux_look_up *lookup) @@ -313,11 +313,11 @@ static uint32_t demux_calc_frames_without_wrap_s32(struct audio_stream __sparse_ return min_frames; } -static void mux_init_look_up_pointers_s32(struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, +static void mux_init_look_up_pointers_s32(struct audio_stream *sink, + const struct audio_stream **sources, struct mux_look_up *lookup) { - const struct audio_stream __sparse_cache *source; + const struct audio_stream *source; uint32_t elem; /* init pointers */ @@ -334,8 +334,8 @@ static void mux_init_look_up_pointers_s32(struct audio_stream __sparse_cache *si } } -static void demux_init_look_up_pointers_s32(struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, +static void demux_init_look_up_pointers_s32(struct audio_stream *sink, + const struct audio_stream *source, struct mux_look_up *lookup) { uint32_t elem; @@ -364,8 +364,8 @@ static void demux_init_look_up_pointers_s32(struct audio_stream __sparse_cache * * @param[in] frames Number of frames to process. * @param[in] lookup mux look up table. */ -static void demux_s32le(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames, +static void demux_s32le(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames, struct mux_look_up *lookup) { uint32_t i; @@ -417,8 +417,8 @@ static void demux_s32le(struct comp_dev *dev, struct audio_stream __sparse_cache * @param[in] frames Number of frames to process. * @param[in] lookup mux look up table. */ -static void mux_s32le(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t frames, +static void mux_s32le(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t frames, struct mux_look_up *lookup) { uint32_t i; @@ -534,7 +534,7 @@ mux_func mux_get_processing_function(struct processing_module *mod) source_list); for (i = 0; i < ARRAY_SIZE(mux_func_map); i++) { - struct comp_buffer __sparse_cache *sink_c = buffer_acquire(sinkb); + struct comp_buffer *sink_c = buffer_acquire(sinkb); enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&sink_c->stream); @@ -558,7 +558,7 @@ demux_func demux_get_processing_function(struct processing_module *mod) sink_list); for (i = 0; i < ARRAY_SIZE(mux_func_map); i++) { - struct comp_buffer __sparse_cache *source_c = buffer_acquire(sourceb); + struct comp_buffer *source_c = buffer_acquire(sourceb); enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&source_c->stream); buffer_release(source_c); diff --git a/src/audio/pcm_converter/pcm_converter.c b/src/audio/pcm_converter/pcm_converter.c index 6c3753b8d8e7..5335ee295b51 100644 --- a/src/audio/pcm_converter/pcm_converter.c +++ b/src/audio/pcm_converter/pcm_converter.c @@ -15,8 +15,8 @@ #include #include -int pcm_convert_as_linear(const struct audio_stream __sparse_cache *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, uint32_t ooffset, +int pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples, pcm_converter_lin_func converter) { const int s_size_in = audio_stream_sample_bytes(source); diff --git a/src/audio/pcm_converter/pcm_converter_generic.c b/src/audio/pcm_converter/pcm_converter_generic.c index db3bbe9b88d4..fd8e5a5e5002 100644 --- a/src/audio/pcm_converter/pcm_converter_generic.c +++ b/src/audio/pcm_converter/pcm_converter_generic.c @@ -34,8 +34,8 @@ #define BYTES_TO_S32_SAMPLES 2 #if CONFIG_PCM_CONVERTER_FORMAT_U8 && CONFIG_PCM_CONVERTER_FORMAT_S32LE -static int pcm_convert_u8_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_u8_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { uint8_t *src = audio_stream_get_rptr(source); @@ -63,8 +63,8 @@ static int pcm_convert_u8_to_s32(const struct audio_stream __sparse_cache *sourc return samples; } -static int pcm_convert_s32_to_u8(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_u8(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -95,8 +95,8 @@ static int pcm_convert_s32_to_u8(const struct audio_stream __sparse_cache *sourc #if CONFIG_PCM_CONVERTER_FORMAT_S16LE && CONFIG_PCM_CONVERTER_FORMAT_S24LE -static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_to_s24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int16_t *src = audio_stream_get_rptr(source); @@ -124,8 +124,8 @@ static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *sour return samples; } -static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_to_s16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -157,8 +157,8 @@ static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *sour #if CONFIG_PCM_CONVERTER_FORMAT_S16LE && CONFIG_PCM_CONVERTER_FORMAT_S32LE -static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int16_t *src = audio_stream_get_rptr(source); @@ -186,8 +186,8 @@ static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *sour return samples; } -static int pcm_convert_s32_to_s16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_s16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -219,8 +219,8 @@ static int pcm_convert_s32_to_s16(const struct audio_stream __sparse_cache *sour #if CONFIG_PCM_CONVERTER_FORMAT_S24LE && CONFIG_PCM_CONVERTER_FORMAT_S32LE -static int pcm_convert_s24_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -248,8 +248,8 @@ static int pcm_convert_s24_to_s32(const struct audio_stream __sparse_cache *sour return samples; } -static int pcm_convert_s32_to_s24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_s24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -277,8 +277,8 @@ static int pcm_convert_s32_to_s24(const struct audio_stream __sparse_cache *sour return samples; } -static int pcm_convert_s32_to_s24_be(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_s24_be(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -442,16 +442,16 @@ static void pcm_convert_f_to_s16_lin(const void *psrc, void *pdst, dst[i] = sat_int16(_pcm_convert_f_to_i(src[i], 15)); } -static int pcm_convert_s16_to_f(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_to_f(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, pcm_convert_s16_to_f_lin); } -static int pcm_convert_f_to_s16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_f_to_s16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -486,16 +486,16 @@ static void pcm_convert_f_to_s24_lin(const void *psrc, void *pdst, dst[i] = sat_int24(_pcm_convert_f_to_i(src[i], 23)); } -static int pcm_convert_s24_to_f(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_to_f(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, pcm_convert_s24_to_f_lin); } -static int pcm_convert_f_to_s24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_f_to_s24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -530,16 +530,16 @@ static void pcm_convert_f_to_s32_lin(const void *psrc, void *pdst, dst[i] = _pcm_convert_f_to_i(src[i], 31); } -static int pcm_convert_s32_to_f(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_f(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, pcm_convert_s32_to_f_lin); } -static int pcm_convert_f_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_f_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -599,8 +599,8 @@ const struct pcm_func_map pcm_func_map[] = { const size_t pcm_func_count = ARRAY_SIZE(pcm_func_map); #if CONFIG_PCM_CONVERTER_FORMAT_S16_C16_AND_S16_C32 -static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int16_t *src = audio_stream_get_rptr(source); @@ -628,8 +628,8 @@ static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -658,8 +658,8 @@ static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream __sparse_cac } #endif #if CONFIG_PCM_CONVERTER_FORMAT_S16_C32_AND_S32_C32 -static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -687,8 +687,8 @@ static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -717,8 +717,8 @@ static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream __sparse_cac } #endif #if CONFIG_PCM_CONVERTER_FORMAT_S16_C32_AND_S24_C32 -static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -746,8 +746,8 @@ static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -777,8 +777,8 @@ static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream __sparse_cac #endif #if CONFIG_PCM_CONVERTER_FORMAT_S24_C24_AND_S24_C32 -static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { uint8_t *src = audio_stream_get_rptr(source); @@ -807,8 +807,8 @@ static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -841,8 +841,8 @@ static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream __sparse_cac } /* 2x24bit samples are packed into 3x16bit samples for hda link dma */ -static int pcm_convert_s24_c32_to_s24_c24_link_gtw(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_c32_to_s24_c24_link_gtw(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); diff --git a/src/audio/pcm_converter/pcm_converter_hifi3.c b/src/audio/pcm_converter/pcm_converter_hifi3.c index ac5c14b379bc..da7e2ff1c7d1 100644 --- a/src/audio/pcm_converter/pcm_converter_hifi3.c +++ b/src/audio/pcm_converter/pcm_converter_hifi3.c @@ -36,8 +36,8 @@ * \param[in,out] sink Destination buffer. * \param[in] samples Number of samples to process. */ -static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_to_s24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { ae_int16x4 sample = AE_ZERO16(); @@ -110,8 +110,8 @@ static ae_int32x2 pcm_shift_s24_to_s16(ae_int32x2 sample) * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_to_s16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { ae_int16x4 sample = AE_ZERO16(); @@ -181,8 +181,8 @@ static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *sour * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int16_t *src = audio_stream_get_rptr(source); @@ -237,8 +237,8 @@ static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *sour * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s32_to_s16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_s16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -303,8 +303,8 @@ static int pcm_convert_s32_to_s16(const struct audio_stream __sparse_cache *sour * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s24_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -368,8 +368,8 @@ static ae_int32x2 pcm_shift_s32_to_s24(ae_int32x2 sample) * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s32_to_s24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_s24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -413,8 +413,8 @@ static int pcm_convert_s32_to_s24(const struct audio_stream __sparse_cache *sour return samples; } -static int pcm_convert_s32_to_s24_be(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_s24_be(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -541,8 +541,8 @@ static void pcm_convert_f_to_s16_lin(const void *psrc, void *pdst, * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s16_to_f(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s16_to_f(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -556,8 +556,8 @@ static int pcm_convert_s16_to_f(const struct audio_stream __sparse_cache *source * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_f_to_s16(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_f_to_s16(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -642,8 +642,8 @@ static void pcm_convert_f_to_s24_lin(const void *psrc, void *pdst, * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s24_to_f(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s24_to_f(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -657,8 +657,8 @@ static int pcm_convert_s24_to_f(const struct audio_stream __sparse_cache *source * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_f_to_s24(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_f_to_s24(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -739,8 +739,8 @@ static void pcm_convert_f_to_s32_lin(const void *psrc, void *pdst, * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_s32_to_f(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_s32_to_f(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -754,8 +754,8 @@ static int pcm_convert_s32_to_f(const struct audio_stream __sparse_cache *source * \param[in] samples Number of samples to process. * \return error code or number of processed samples. */ -static int pcm_convert_f_to_s32(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +static int pcm_convert_f_to_s32(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples, @@ -810,9 +810,9 @@ const struct pcm_func_map pcm_func_map[] = { const size_t pcm_func_count = ARRAY_SIZE(pcm_func_map); #if CONFIG_PCM_CONVERTER_FORMAT_S16_C16_AND_S16_C32 -static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int16_t *src = audio_stream_get_rptr(source); @@ -860,9 +860,9 @@ static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -919,9 +919,9 @@ static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream __sparse_cac #endif #if CONFIG_PCM_CONVERTER_FORMAT_S16_C32_AND_S32_C32 -static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -962,9 +962,9 @@ static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -1007,9 +1007,9 @@ static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream __sparse_cac #endif #if CONFIG_PCM_CONVERTER_FORMAT_S16_C32_AND_S24_C32 -static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -1062,9 +1062,9 @@ static ae_int32x2 pcm_shift_s24_c32_to_s16(ae_int32x2 sample) return sample; } -static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); @@ -1109,9 +1109,9 @@ static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream __sparse_cac #endif #if CONFIG_PCM_CONVERTER_FORMAT_S24_C24_AND_S24_C32 -static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { uint8_t *src = audio_stream_get_rptr(source); @@ -1156,9 +1156,9 @@ static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream __sparse_cac return samples; } -static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream __sparse_cache *source, +static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples) { int32_t *src = audio_stream_get_rptr(source); diff --git a/src/audio/pipeline/pipeline-graph.c b/src/audio/pipeline/pipeline-graph.c index 160e14e15441..46b9f9b510f8 100644 --- a/src/audio/pipeline/pipeline-graph.c +++ b/src/audio/pipeline/pipeline-graph.c @@ -168,7 +168,7 @@ struct pipeline *pipeline_new(uint32_t pipeline_id, uint32_t priority, uint32_t static void buffer_set_comp(struct comp_buffer *buffer, struct comp_dev *comp, int dir) { - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(buffer); + struct comp_buffer *buffer_c = buffer_acquire(buffer); if (dir == PPL_CONN_DIR_COMP_TO_BUFFER) buffer_c->source = comp; @@ -410,7 +410,7 @@ int pipeline_for_each_comp(struct comp_dev *current, /* run this operation further */ list_for_item(clist, buffer_list) { struct comp_buffer *buffer = buffer_from_list(clist, dir); - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct comp_dev *buffer_comp; int err = 0; diff --git a/src/audio/pipeline/pipeline-params.c b/src/audio/pipeline/pipeline-params.c index 43974e290fb4..a0a53749320c 100644 --- a/src/audio/pipeline/pipeline-params.c +++ b/src/audio/pipeline/pipeline-params.c @@ -30,7 +30,7 @@ static int pipeline_comp_params_neg(struct comp_dev *current, int dir) { struct pipeline_data *ppl_data = ctx->comp_data; - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(calling_buf); + struct comp_buffer *buf_c = buffer_acquire(calling_buf); int err = 0; pipe_dbg(current->pipeline, "pipeline_comp_params_neg(), current->comp.id = %u, dir = %u", @@ -131,7 +131,7 @@ static int pipeline_comp_params(struct comp_dev *current, } /* save params changes made by component */ -static void pipeline_update_buffer_pcm_params(struct comp_buffer __sparse_cache *buffer, +static void pipeline_update_buffer_pcm_params(struct comp_buffer *buffer, void *data) { struct sof_ipc_stream_params *params = data; @@ -188,7 +188,7 @@ static int pipeline_comp_hw_params_buf(struct comp_dev *current, return ret; /* set buffer parameters */ if (calling_buf) { - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(calling_buf); + struct comp_buffer *buf_c = buffer_acquire(calling_buf); ret = buffer_set_params(buf_c, &ppl_data->params->params, BUFFER_UPDATE_IF_UNSET); diff --git a/src/audio/rtnr/rtnr.c b/src/audio/rtnr/rtnr.c index 8c3ce8496c14..270559b0728f 100644 --- a/src/audio/rtnr/rtnr.c +++ b/src/audio/rtnr/rtnr.c @@ -325,7 +325,7 @@ static int rtnr_params(struct comp_dev *dev, struct sof_ipc_stream_params *param int ret; struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *sinkb, *sourceb; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; bool channels_valid; comp_info(dev, "rtnr_params()"); @@ -727,7 +727,7 @@ static int rtnr_trigger(struct comp_dev *dev, int cmd) } static void rtnr_copy_from_sof_stream(struct audio_stream_rtnr *dst, - struct audio_stream __sparse_cache *src) + struct audio_stream *src) { dst->size = audio_stream_get_size(src); @@ -739,7 +739,7 @@ static void rtnr_copy_from_sof_stream(struct audio_stream_rtnr *dst, dst->end_addr = audio_stream_get_end_addr(src); } -static void rtnr_copy_to_sof_stream(struct audio_stream __sparse_cache *dst, +static void rtnr_copy_to_sof_stream(struct audio_stream *dst, struct audio_stream_rtnr *src) { audio_stream_set_size(dst, src->size); @@ -851,7 +851,7 @@ static int rtnr_prepare(struct comp_dev *dev) { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *sinkb; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; int ret; comp_dbg(dev, "rtnr_prepare()"); diff --git a/src/audio/selector/selector.c b/src/audio/selector/selector.c index 8a39db81b2ef..8748f07ea1a3 100644 --- a/src/audio/selector/selector.c +++ b/src/audio/selector/selector.c @@ -60,7 +60,7 @@ static int selector_verify_params(struct comp_dev *dev, { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *buffer, *sinkb; - struct comp_buffer __sparse_cache *buffer_c, *sink_c; + struct comp_buffer *buffer_c, *sink_c; uint32_t in_channels; uint32_t out_channels; @@ -358,7 +358,7 @@ static int selector_cmd(struct comp_dev *dev, int cmd, void *data, static int selector_trigger(struct comp_dev *dev, int cmd) { struct comp_buffer *sourceb; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; enum sof_comp_type type; int ret; @@ -391,7 +391,7 @@ static int selector_copy(struct comp_dev *dev) { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *sink, *source; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; uint32_t frames; uint32_t source_bytes; uint32_t sink_bytes; @@ -444,7 +444,7 @@ static int selector_prepare(struct comp_dev *dev) { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *sinkb, *sourceb; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; size_t sink_size; int ret; @@ -673,7 +673,7 @@ static void set_selector_params(struct processing_module *mod, { struct comp_dev *dev = mod->dev; struct comp_data *cd = module_get_private_data(mod); - struct comp_buffer __sparse_cache *source; + struct comp_buffer *source; const struct sof_selector_ipc4_config *sel_cfg = &cd->sel_ipc4_cfg; const struct ipc4_audio_format *out_fmt = NULL; struct comp_buffer *src_buf; @@ -700,7 +700,7 @@ static void set_selector_params(struct processing_module *mod, list_for_item(sink_list, &dev->bsink_list) { struct comp_buffer *sink_buf = container_of(sink_list, struct comp_buffer, source_list); - struct comp_buffer __sparse_cache *sink = buffer_acquire(sink_buf); + struct comp_buffer *sink = buffer_acquire(sink_buf); ipc4_update_buffer_format(sink, out_fmt); audio_stream_set_channels(&sink->stream, params->channels); @@ -729,7 +729,7 @@ static int selector_verify_params(struct processing_module *mod, struct comp_dev *dev = mod->dev; struct comp_data *cd = module_get_private_data(mod); struct comp_buffer *buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; uint32_t in_channels = cd->config.in_channels_count; uint32_t out_channels = cd->config.out_channels_count; @@ -862,14 +862,14 @@ static int selector_process(struct processing_module *mod, * \return Error code. */ static int selector_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) { struct comp_data *cd = module_get_private_data(mod); struct module_data *md = &mod->priv; struct comp_dev *dev = mod->dev; struct comp_buffer *sinkb, *sourceb; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; size_t sink_size; int ret; diff --git a/src/audio/selector/selector_generic.c b/src/audio/selector/selector_generic.c index ca9e72f656d5..923277ec3f46 100644 --- a/src/audio/selector/selector_generic.c +++ b/src/audio/selector/selector_generic.c @@ -32,8 +32,8 @@ LOG_MODULE_DECLARE(selector, CONFIG_SOF_LOG_LEVEL); * \param[in,out] source Source buffer. * \param[in] frames Number of frames to process. */ -static void sel_s16le_1ch(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames) +static void sel_s16le_1ch(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames) { struct comp_data *cd = comp_get_drvdata(dev); int16_t *src = audio_stream_get_rptr(source); @@ -72,8 +72,8 @@ static void sel_s16le_1ch(struct comp_dev *dev, struct audio_stream __sparse_cac * \param[in,out] source Source buffer. * \param[in] frames Number of frames to process. */ -static void sel_s16le_nch(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames) +static void sel_s16le_nch(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames) { int8_t *src = audio_stream_get_rptr(source); int8_t *dst = audio_stream_get_wptr(sink); @@ -104,8 +104,8 @@ static void sel_s16le_nch(struct comp_dev *dev, struct audio_stream __sparse_cac * \param[in,out] source Source buffer. * \param[in] frames Number of frames to process. */ -static void sel_s32le_1ch(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames) +static void sel_s32le_1ch(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames) { struct comp_data *cd = comp_get_drvdata(dev); int32_t *src = audio_stream_get_rptr(source); @@ -144,8 +144,8 @@ static void sel_s32le_1ch(struct comp_dev *dev, struct audio_stream __sparse_cac * \param[in,out] source Source buffer. * \param[in] frames Number of frames to process. */ -static void sel_s32le_nch(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames) +static void sel_s32le_nch(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames) { int8_t *src = audio_stream_get_rptr(source); int8_t *dst = audio_stream_get_wptr(sink); @@ -206,8 +206,8 @@ static void sel_s16le(struct processing_module *mod, struct input_stream_buffer struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int16_t *src = audio_stream_get_rptr(source); int16_t *dest = audio_stream_get_wptr(sink); int nmax; @@ -277,8 +277,8 @@ static void sel_s32le(struct processing_module *mod, struct input_stream_buffer struct output_stream_buffer *bsink, uint32_t frames) { struct comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *src = audio_stream_get_rptr(source); int32_t *dest = audio_stream_get_wptr(sink); int nmax; diff --git a/src/audio/sink_api_helper.c b/src/audio/sink_api_helper.c index 4655fdbde347..519912e49967 100644 --- a/src/audio/sink_api_helper.c +++ b/src/audio/sink_api_helper.c @@ -7,19 +7,19 @@ #include #include -void sink_init(struct sof_sink __sparse_cache *sink, const struct sink_ops *ops, +void sink_init(struct sof_sink *sink, const struct sink_ops *ops, struct sof_audio_stream_params *audio_stream_params) { sink->ops = ops; sink->audio_stream_params = audio_stream_params; } -size_t sink_get_free_size(struct sof_sink __sparse_cache *sink) +size_t sink_get_free_size(struct sof_sink *sink) { return sink->ops->get_free_size(sink); } -int sink_get_buffer(struct sof_sink __sparse_cache *sink, size_t req_size, +int sink_get_buffer(struct sof_sink *sink, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size) { int ret; @@ -35,7 +35,7 @@ int sink_get_buffer(struct sof_sink __sparse_cache *sink, size_t req_size, return ret; } -int sink_commit_buffer(struct sof_sink __sparse_cache *sink, size_t commit_size) +int sink_commit_buffer(struct sof_sink *sink, size_t commit_size) { int ret; @@ -56,47 +56,47 @@ int sink_commit_buffer(struct sof_sink __sparse_cache *sink, size_t commit_size) return ret; } -size_t sink_get_num_of_processed_bytes(struct sof_sink __sparse_cache *sink) +size_t sink_get_num_of_processed_bytes(struct sof_sink *sink) { return sink->num_of_bytes_processed; } -void sink_reset_num_of_processed_bytes(struct sof_sink __sparse_cache *sink) +void sink_reset_num_of_processed_bytes(struct sof_sink *sink) { sink->num_of_bytes_processed = 0; } -enum sof_ipc_frame sink_get_frm_fmt(struct sof_sink __sparse_cache *sink) +enum sof_ipc_frame sink_get_frm_fmt(struct sof_sink *sink) { return sink->audio_stream_params->frame_fmt; } -enum sof_ipc_frame sink_get_valid_fmt(struct sof_sink __sparse_cache *sink) +enum sof_ipc_frame sink_get_valid_fmt(struct sof_sink *sink) { return sink->audio_stream_params->valid_sample_fmt; } -uint32_t sink_get_rate(struct sof_sink __sparse_cache *sink) +uint32_t sink_get_rate(struct sof_sink *sink) { return sink->audio_stream_params->rate; } -uint32_t sink_get_channels(struct sof_sink __sparse_cache *sink) +uint32_t sink_get_channels(struct sof_sink *sink) { return sink->audio_stream_params->channels; } -uint32_t sink_get_buffer_fmt(struct sof_sink __sparse_cache *sink) +uint32_t sink_get_buffer_fmt(struct sof_sink *sink) { return sink->audio_stream_params->buffer_fmt; } -bool sink_get_overrun(struct sof_sink __sparse_cache *sink) +bool sink_get_overrun(struct sof_sink *sink) { return sink->audio_stream_params->overrun_permitted; } -int sink_set_frm_fmt(struct sof_sink __sparse_cache *sink, enum sof_ipc_frame frame_fmt) +int sink_set_frm_fmt(struct sof_sink *sink, enum sof_ipc_frame frame_fmt) { sink->audio_stream_params->frame_fmt = frame_fmt; @@ -106,7 +106,7 @@ int sink_set_frm_fmt(struct sof_sink __sparse_cache *sink, enum sof_ipc_frame fr return 0; } -int sink_set_valid_fmt(struct sof_sink __sparse_cache *sink, +int sink_set_valid_fmt(struct sof_sink *sink, enum sof_ipc_frame valid_sample_fmt) { sink->audio_stream_params->valid_sample_fmt = valid_sample_fmt; @@ -115,7 +115,7 @@ int sink_set_valid_fmt(struct sof_sink __sparse_cache *sink, return 0; } -int sink_set_rate(struct sof_sink __sparse_cache *sink, unsigned int rate) +int sink_set_rate(struct sof_sink *sink, unsigned int rate) { sink->audio_stream_params->rate = rate; if (sink->ops->on_audio_format_set) @@ -123,7 +123,7 @@ int sink_set_rate(struct sof_sink __sparse_cache *sink, unsigned int rate) return 0; } -int sink_set_channels(struct sof_sink __sparse_cache *sink, unsigned int channels) +int sink_set_channels(struct sof_sink *sink, unsigned int channels) { sink->audio_stream_params->channels = channels; if (sink->ops->on_audio_format_set) @@ -131,7 +131,7 @@ int sink_set_channels(struct sof_sink __sparse_cache *sink, unsigned int channel return 0; } -int sink_set_buffer_fmt(struct sof_sink __sparse_cache *sink, uint32_t buffer_fmt) +int sink_set_buffer_fmt(struct sof_sink *sink, uint32_t buffer_fmt) { sink->audio_stream_params->buffer_fmt = buffer_fmt; if (sink->ops->on_audio_format_set) @@ -139,7 +139,7 @@ int sink_set_buffer_fmt(struct sof_sink __sparse_cache *sink, uint32_t buffer_fm return 0; } -int sink_set_overrun(struct sof_sink __sparse_cache *sink, bool overrun_permitted) +int sink_set_overrun(struct sof_sink *sink, bool overrun_permitted) { sink->audio_stream_params->overrun_permitted = overrun_permitted; if (sink->ops->on_audio_format_set) @@ -147,19 +147,19 @@ int sink_set_overrun(struct sof_sink __sparse_cache *sink, bool overrun_permitte return 0; } -size_t sink_get_frame_bytes(struct sof_sink __sparse_cache *sink) +size_t sink_get_frame_bytes(struct sof_sink *sink) { return get_frame_bytes(sink_get_frm_fmt(sink), sink_get_channels(sink)); } -size_t sink_get_free_frames(struct sof_sink __sparse_cache *sink) +size_t sink_get_free_frames(struct sof_sink *sink) { return sink_get_free_size(sink) / sink_get_frame_bytes(sink); } -int sink_set_params(struct sof_sink __sparse_cache *sink, +int sink_set_params(struct sof_sink *sink, struct sof_ipc_stream_params *params, bool force_update) { if (sink->ops->audio_set_ipc_params) @@ -167,7 +167,7 @@ int sink_set_params(struct sof_sink __sparse_cache *sink, return 0; } -int sink_set_alignment_constants(struct sof_sink __sparse_cache *sink, +int sink_set_alignment_constants(struct sof_sink *sink, const uint32_t byte_align, const uint32_t frame_align_req) { @@ -176,12 +176,12 @@ int sink_set_alignment_constants(struct sof_sink __sparse_cache *sink, return 0; } -void sink_set_obs(struct sof_sink __sparse_cache *sink, size_t obs) +void sink_set_obs(struct sof_sink *sink, size_t obs) { sink->obs = obs; } -size_t sink_get_obs(struct sof_sink __sparse_cache *sink) +size_t sink_get_obs(struct sof_sink *sink) { return sink->obs; } diff --git a/src/audio/sink_source_utils.c b/src/audio/sink_source_utils.c index 671c1752d544..5a0d17608be4 100644 --- a/src/audio/sink_source_utils.c +++ b/src/audio/sink_source_utils.c @@ -12,8 +12,8 @@ #include #include -int source_to_sink_copy(struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink, bool free, size_t size) +int source_to_sink_copy(struct sof_source *source, + struct sof_sink *sink, bool free, size_t size) { uint8_t *src_ptr; uint8_t *src_begin; diff --git a/src/audio/smart_amp/smart_amp.c b/src/audio/smart_amp/smart_amp.c index d0108aa346c7..351db6cbf9c6 100644 --- a/src/audio/smart_amp/smart_amp.c +++ b/src/audio/smart_amp/smart_amp.c @@ -26,8 +26,8 @@ DECLARE_TR_CTX(maxim_dsm_comp_tr, SOF_UUID(maxim_dsm_comp_uuid), #define SOF_SMART_AMP_MODEL 1 typedef int(*smart_amp_proc)(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, uint32_t frames, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames, int8_t *chan_map, bool is_feedback); struct smart_amp_data { @@ -518,7 +518,7 @@ static int smart_amp_trigger(struct comp_dev *dev, int cmd) case COMP_TRIGGER_START: case COMP_TRIGGER_RELEASE: if (sad->feedback_buf) { - struct comp_buffer __sparse_cache *buf = buffer_acquire(sad->feedback_buf); + struct comp_buffer *buf = buffer_acquire(sad->feedback_buf); buffer_zero(buf); buffer_release(buf); } @@ -534,8 +534,8 @@ static int smart_amp_trigger(struct comp_dev *dev, int cmd) } static int smart_amp_process(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames, int8_t *chan_map, bool is_feedback) { @@ -558,7 +558,7 @@ 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); + struct comp_buffer *source_buf = buffer_acquire(sad->source_buf); enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&source_buf->stream); buffer_release(source_buf); @@ -577,8 +577,8 @@ static smart_amp_proc get_smart_amp_process(struct comp_dev *dev) static int smart_amp_copy(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); - struct comp_buffer __sparse_cache *sink_buf = buffer_acquire(sad->sink_buf); + struct comp_buffer *source_buf = buffer_acquire(sad->source_buf); + struct comp_buffer *sink_buf = buffer_acquire(sad->sink_buf); uint32_t avail_passthrough_frames; uint32_t avail_feedback_frames; uint32_t avail_frames; @@ -595,7 +595,7 @@ static int smart_amp_copy(struct comp_dev *dev) avail_frames = avail_passthrough_frames; if (sad->feedback_buf) { - struct comp_buffer __sparse_cache *feedback_buf = buffer_acquire(sad->feedback_buf); + struct comp_buffer *feedback_buf = buffer_acquire(sad->feedback_buf); if (comp_get_state(dev, feedback_buf->source) == dev->state) { /* feedback */ @@ -661,7 +661,7 @@ static int smart_amp_reset(struct comp_dev *dev) static int smart_amp_prepare(struct comp_dev *dev) { struct smart_amp_data *sad = comp_get_drvdata(dev); - struct comp_buffer __sparse_cache *source_c, *buf_c; + struct comp_buffer *source_c, *buf_c; struct list_item *blist; int ret; int bitwidth; diff --git a/src/audio/smart_amp/smart_amp_generic.c b/src/audio/smart_amp/smart_amp_generic.c index 3ad4c1ce28de..35dffeb0e449 100644 --- a/src/audio/smart_amp/smart_amp_generic.c +++ b/src/audio/smart_amp/smart_amp_generic.c @@ -24,9 +24,9 @@ static void smart_amp_fb_generic(int32_t x) #if CONFIG_FORMAT_S16LE static void smart_amp_s16_ff_default(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames) { int16_t *x; @@ -52,9 +52,9 @@ static void smart_amp_s16_ff_default(const struct comp_dev *dev, #if CONFIG_FORMAT_S24LE static void smart_amp_s24_ff_default(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames) { int32_t *x; @@ -80,9 +80,9 @@ static void smart_amp_s24_ff_default(const struct comp_dev *dev, #if CONFIG_FORMAT_S32LE static void smart_amp_s32_ff_default(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames) { int32_t *x; @@ -106,9 +106,9 @@ static void smart_amp_s32_ff_default(const struct comp_dev *dev, #if CONFIG_FORMAT_S16LE static void smart_amp_s16_fb_default(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames) { int16_t *x; @@ -130,9 +130,9 @@ static void smart_amp_s16_fb_default(const struct comp_dev *dev, #if CONFIG_FORMAT_S24LE static void smart_amp_s24_fb_default(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames) { int32_t *x; @@ -154,9 +154,9 @@ static void smart_amp_s24_fb_default(const struct comp_dev *dev, #if CONFIG_FORMAT_S32LE static void smart_amp_s32_fb_default(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames) { int32_t *x; diff --git a/src/audio/smart_amp/smart_amp_maxim_dsm.c b/src/audio/smart_amp/smart_amp_maxim_dsm.c index ed6b5a218511..165d159678dc 100644 --- a/src/audio/smart_amp/smart_amp_maxim_dsm.c +++ b/src/audio/smart_amp/smart_amp_maxim_dsm.c @@ -579,7 +579,7 @@ int smart_amp_check_audio_fmt(int sample_rate, int ch_num) } static int smart_amp_get_buffer(int32_t *buf, uint32_t frames, - const struct audio_stream __sparse_cache *stream, + const struct audio_stream *stream, int8_t *chan_map, uint32_t num_ch) { int idx, ch; @@ -629,7 +629,7 @@ static int smart_amp_get_buffer(int32_t *buf, uint32_t frames, } static int smart_amp_put_buffer(int32_t *buf, uint32_t frames, - const struct audio_stream __sparse_cache *stream, + const struct audio_stream *stream, int8_t *chan_map, uint32_t num_ch_in, uint32_t num_ch_out) { @@ -681,8 +681,8 @@ static int smart_amp_put_buffer(int32_t *buf, uint32_t frames, } int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, int8_t *chan_map, + const struct audio_stream *source, + const struct audio_stream *sink, int8_t *chan_map, struct smart_amp_mod_struct_t *hspk, uint32_t num_ch_in, uint32_t num_ch_out) { @@ -741,8 +741,8 @@ int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames, } int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, int8_t *chan_map, + const struct audio_stream *source, + const struct audio_stream *sink, int8_t *chan_map, struct smart_amp_mod_struct_t *hspk, uint32_t num_ch) { diff --git a/src/audio/source_api_helper.c b/src/audio/source_api_helper.c index 91985aeed969..1483fb6603c9 100644 --- a/src/audio/source_api_helper.c +++ b/src/audio/source_api_helper.c @@ -7,7 +7,7 @@ #include #include -void source_init(struct sof_source __sparse_cache *source, const struct source_ops *ops, +void source_init(struct sof_source *source, const struct source_ops *ops, struct sof_audio_stream_params *audio_stream_params) { source->ops = ops; @@ -15,12 +15,12 @@ void source_init(struct sof_source __sparse_cache *source, const struct source_o source->audio_stream_params = audio_stream_params; } -size_t source_get_data_available(struct sof_source __sparse_cache *source) +size_t source_get_data_available(struct sof_source *source) { return source->ops->get_data_available(source); } -int source_get_data(struct sof_source __sparse_cache *source, size_t req_size, +int source_get_data(struct sof_source *source, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size) { int ret; @@ -35,7 +35,7 @@ int source_get_data(struct sof_source __sparse_cache *source, size_t req_size, return ret; } -int source_release_data(struct sof_source __sparse_cache *source, size_t free_size) +int source_release_data(struct sof_source *source, size_t free_size) { int ret; @@ -56,47 +56,47 @@ int source_release_data(struct sof_source __sparse_cache *source, size_t free_si return ret; } -size_t source_get_num_of_processed_bytes(struct sof_source __sparse_cache *source) +size_t source_get_num_of_processed_bytes(struct sof_source *source) { return source->num_of_bytes_processed; } -void source_reset_num_of_processed_bytes(struct sof_source __sparse_cache *source) +void source_reset_num_of_processed_bytes(struct sof_source *source) { source->num_of_bytes_processed = 0; } -enum sof_ipc_frame source_get_frm_fmt(struct sof_source __sparse_cache *source) +enum sof_ipc_frame source_get_frm_fmt(struct sof_source *source) { return source->audio_stream_params->frame_fmt; } -enum sof_ipc_frame source_get_valid_fmt(struct sof_source __sparse_cache *source) +enum sof_ipc_frame source_get_valid_fmt(struct sof_source *source) { return source->audio_stream_params->valid_sample_fmt; } -unsigned int source_get_rate(struct sof_source __sparse_cache *source) +unsigned int source_get_rate(struct sof_source *source) { return source->audio_stream_params->rate; } -unsigned int source_get_channels(struct sof_source __sparse_cache *source) +unsigned int source_get_channels(struct sof_source *source) { return source->audio_stream_params->channels; } -uint32_t source_get_buffer_fmt(struct sof_source __sparse_cache *source) +uint32_t source_get_buffer_fmt(struct sof_source *source) { return source->audio_stream_params->buffer_fmt; } -bool source_get_underrun(struct sof_source __sparse_cache *source) +bool source_get_underrun(struct sof_source *source) { return source->audio_stream_params->underrun_permitted; } -int source_set_valid_fmt(struct sof_source __sparse_cache *source, +int source_set_valid_fmt(struct sof_source *source, enum sof_ipc_frame valid_sample_fmt) { source->audio_stream_params->valid_sample_fmt = valid_sample_fmt; @@ -105,7 +105,7 @@ int source_set_valid_fmt(struct sof_source __sparse_cache *source, return 0; } -int source_set_rate(struct sof_source __sparse_cache *source, unsigned int rate) +int source_set_rate(struct sof_source *source, unsigned int rate) { source->audio_stream_params->rate = rate; if (source->ops->on_audio_format_set) @@ -113,7 +113,7 @@ int source_set_rate(struct sof_source __sparse_cache *source, unsigned int rate) return 0; } -int source_set_channels(struct sof_source __sparse_cache *source, unsigned int channels) +int source_set_channels(struct sof_source *source, unsigned int channels) { source->audio_stream_params->channels = channels; if (source->ops->on_audio_format_set) @@ -121,7 +121,7 @@ int source_set_channels(struct sof_source __sparse_cache *source, unsigned int c return 0; } -int source_set_buffer_fmt(struct sof_source __sparse_cache *source, uint32_t buffer_fmt) +int source_set_buffer_fmt(struct sof_source *source, uint32_t buffer_fmt) { source->audio_stream_params->buffer_fmt = buffer_fmt; if (source->ops->on_audio_format_set) @@ -129,7 +129,7 @@ int source_set_buffer_fmt(struct sof_source __sparse_cache *source, uint32_t buf return 0; } -int source_set_underrun(struct sof_source __sparse_cache *source, bool underrun_permitted) +int source_set_underrun(struct sof_source *source, bool underrun_permitted) { source->audio_stream_params->underrun_permitted = underrun_permitted; if (source->ops->on_audio_format_set) @@ -137,19 +137,19 @@ int source_set_underrun(struct sof_source __sparse_cache *source, bool underrun_ return 0; } -size_t source_get_frame_bytes(struct sof_source __sparse_cache *source) +size_t source_get_frame_bytes(struct sof_source *source) { return get_frame_bytes(source_get_frm_fmt(source), source_get_channels(source)); } -size_t source_get_data_frames_available(struct sof_source __sparse_cache *source) +size_t source_get_data_frames_available(struct sof_source *source) { return source_get_data_available(source) / source_get_frame_bytes(source); } -int source_set_params(struct sof_source __sparse_cache *source, +int source_set_params(struct sof_source *source, struct sof_ipc_stream_params *params, bool force_update) { if (source->ops->audio_set_ipc_params) @@ -157,7 +157,7 @@ int source_set_params(struct sof_source __sparse_cache *source, return 0; } -int source_set_alignment_constants(struct sof_source __sparse_cache *source, +int source_set_alignment_constants(struct sof_source *source, const uint32_t byte_align, const uint32_t frame_align_req) { @@ -166,12 +166,12 @@ int source_set_alignment_constants(struct sof_source __sparse_cache *source, return 0; } -void source_set_ibs(struct sof_source __sparse_cache *source, size_t ibs) +void source_set_ibs(struct sof_source *source, size_t ibs) { source->ibs = ibs; } -size_t source_get_ibs(struct sof_source __sparse_cache *source) +size_t source_get_ibs(struct sof_source *source) { return source->ibs; } diff --git a/src/audio/src/src.c b/src/audio/src/src.c index 82501728c59c..65b8ae965abd 100644 --- a/src/audio/src/src.c +++ b/src/audio/src/src.c @@ -101,8 +101,8 @@ struct comp_data { int sink_frames; int sample_container_bytes; int channels_count; - int (*src_func)(struct comp_data *cd, struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink); + int (*src_func)(struct comp_data *cd, struct sof_source *source, + struct sof_sink *sink); void (*polyphase_func)(struct src_stage_prm *s); }; @@ -335,15 +335,15 @@ int src_polyphase_init(struct polyphase_src *src, struct src_param *p, } /* Fallback function */ -static int src_fallback(struct comp_data *cd, struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) +static int src_fallback(struct comp_data *cd, struct sof_source *source, + struct sof_sink *sink) { return 0; } /* Normal 2 stage SRC */ static int src_2s(struct comp_data *cd, - struct sof_source __sparse_cache *source, struct sof_sink __sparse_cache *sink) + struct sof_source *source, struct sof_sink *sink) { struct src_stage_prm s1; struct src_stage_prm s2; @@ -443,8 +443,8 @@ static int src_2s(struct comp_data *cd, } /* 1 stage SRC for simple conversions */ -static int src_1s(struct comp_data *cd, struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) +static int src_1s(struct comp_data *cd, struct sof_source *source, + struct sof_sink *sink) { struct src_stage_prm s1; int ret; @@ -482,8 +482,8 @@ static int src_1s(struct comp_data *cd, struct sof_source __sparse_cache *source } /* A fast copy function for same in and out rate */ -static int src_copy_sxx(struct comp_data *cd, struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) +static int src_copy_sxx(struct comp_data *cd, struct sof_source *source, + struct sof_sink *sink) { int frames = cd->param.blk_in; @@ -531,7 +531,7 @@ static int src_stream_pcm_sink_rate_check(struct ipc4_config_src cfg, * set up param then verify param. BTW for IPC3 path, the param is sent by * host driver. */ -static int src_set_params(struct processing_module *mod, struct sof_sink __sparse_cache *sink) +static int src_set_params(struct processing_module *mod, struct sof_sink *sink) { struct sof_ipc_stream_params src_params; struct sof_ipc_stream_params *params = mod->stream_params; @@ -561,7 +561,7 @@ static int src_set_params(struct processing_module *mod, struct sof_sink __spars return ret; } -static void src_set_sink_params(struct comp_dev *dev, struct sof_sink __sparse_cache *sink) +static void src_set_sink_params(struct comp_dev *dev, struct sof_sink *sink) { struct processing_module *mod = comp_get_drvdata(dev); struct comp_data *cd = module_get_private_data(mod); @@ -610,12 +610,12 @@ static int src_stream_pcm_source_rate_check(struct ipc_config_src cfg, return 0; } -static int src_set_params(struct processing_module *mod, struct sof_sink __sparse_cache *sink) +static int src_set_params(struct processing_module *mod, struct sof_sink *sink) { return 0; } -static void src_set_sink_params(struct comp_dev *dev, struct sof_sink __sparse_cache *sink) +static void src_set_sink_params(struct comp_dev *dev, struct sof_sink *sink) { /* empty */ } @@ -624,8 +624,8 @@ static void src_set_sink_params(struct comp_dev *dev, struct sof_sink __sparse_c #error "No or invalid IPC MAJOR version selected." #endif /* CONFIG_IPC_MAJOR_4 */ -static void src_set_alignment(struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) +static void src_set_alignment(struct sof_source *source, + struct sof_sink *sink) { const uint32_t byte_align = 1; const uint32_t frame_align_req = 1; @@ -672,8 +672,8 @@ static int src_verify_params(struct processing_module *mod) } static bool src_get_copy_limits(struct comp_data *cd, - struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) + struct sof_source *source, + struct sof_sink *sink) { struct src_param *sp; struct src_stage *s1; @@ -716,8 +716,8 @@ static bool src_get_copy_limits(struct comp_data *cd, } static int src_params_general(struct processing_module *mod, - struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) + struct sof_source *source, + struct sof_sink *sink) { struct comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; @@ -826,8 +826,8 @@ static int src_params_general(struct processing_module *mod, } static int src_prepare_general(struct processing_module *mod, - struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink) + struct sof_source *source, + struct sof_sink *sink) { struct comp_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; @@ -983,8 +983,8 @@ static int src_init(struct processing_module *mod) } static int src_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; @@ -1002,8 +1002,8 @@ static int src_prepare(struct processing_module *mod, static bool src_is_ready_to_process(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) { struct comp_data *cd = module_get_private_data(mod); @@ -1011,8 +1011,8 @@ static bool src_is_ready_to_process(struct processing_module *mod, } static int src_process(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) { struct comp_data *cd = module_get_private_data(mod); diff --git a/src/audio/tdfb/tdfb.c b/src/audio/tdfb/tdfb.c index d1b04b7faf42..976119264e02 100644 --- a/src/audio/tdfb/tdfb.c +++ b/src/audio/tdfb/tdfb.c @@ -660,8 +660,8 @@ static int tdfb_process(struct processing_module *mod, { struct comp_dev *dev = mod->dev; struct tdfb_comp_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = input_buffers[0].data; - struct audio_stream __sparse_cache *sink = output_buffers[0].data; + struct audio_stream *source = input_buffers[0].data; + struct audio_stream *sink = output_buffers[0].data; int frame_count = input_buffers[0].size; int ret; @@ -715,8 +715,8 @@ static int tdfb_process(struct processing_module *mod, return 0; } -static void tdfb_set_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static void tdfb_set_alignment(struct audio_stream *source, + struct audio_stream *sink) { const uint32_t byte_align = 1; const uint32_t frame_align_req = 2; /* Process multiples of 2 frames */ @@ -726,12 +726,12 @@ static void tdfb_set_alignment(struct audio_stream __sparse_cache *source, } static int tdfb_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) { struct tdfb_comp_data *cd = module_get_private_data(mod); struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; struct comp_dev *dev = mod->dev; enum sof_ipc_frame frame_fmt; int source_channels; diff --git a/src/audio/tdfb/tdfb_generic.c b/src/audio/tdfb/tdfb_generic.c index 8306bb6a8991..22f74702ac06 100644 --- a/src/audio/tdfb/tdfb_generic.c +++ b/src/audio/tdfb/tdfb_generic.c @@ -58,8 +58,8 @@ static inline void tdfb_core(struct tdfb_comp_data *cd, int in_nch, int out_nch) void tdfb_fir_s16(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int16_t *x = audio_stream_get_rptr(source); int16_t *y = audio_stream_get_wptr(sink); int fmax; @@ -104,8 +104,8 @@ void tdfb_fir_s16(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource void tdfb_fir_s24(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x = audio_stream_get_rptr(source); int32_t *y = audio_stream_get_wptr(sink); int fmax; @@ -150,8 +150,8 @@ void tdfb_fir_s24(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource void tdfb_fir_s32(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x = audio_stream_get_rptr(source); int32_t *y = audio_stream_get_wptr(sink); int fmax; diff --git a/src/audio/tdfb/tdfb_hifi3.c b/src/audio/tdfb/tdfb_hifi3.c index 97b5229d02dd..421784db9d1d 100644 --- a/src/audio/tdfb/tdfb_hifi3.c +++ b/src/audio/tdfb/tdfb_hifi3.c @@ -19,8 +19,8 @@ void tdfb_fir_s16(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct sof_tdfb_config *cfg = cd->config; struct fir_state_32x16 *f; ae_int16x4 d; @@ -108,8 +108,8 @@ void tdfb_fir_s16(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource void tdfb_fir_s24(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct sof_tdfb_config *cfg = cd->config; struct fir_state_32x16 *f; ae_int32x2 d; @@ -197,8 +197,8 @@ void tdfb_fir_s24(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource void tdfb_fir_s32(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; struct sof_tdfb_config *cfg = cd->config; struct fir_state_32x16 *f; ae_int32x2 d; diff --git a/src/audio/tdfb/tdfb_hifiep.c b/src/audio/tdfb/tdfb_hifiep.c index eba8d7c7807e..1f5b71dbe612 100644 --- a/src/audio/tdfb/tdfb_hifiep.c +++ b/src/audio/tdfb/tdfb_hifiep.c @@ -58,8 +58,8 @@ static inline void tdfb_core(struct tdfb_comp_data *cd, int in_nch, int out_nch) void tdfb_fir_s16(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int16_t *x = audio_stream_get_rptr(source); int16_t *y = audio_stream_get_wptr(sink); int fmax; @@ -104,8 +104,8 @@ void tdfb_fir_s16(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource void tdfb_fir_s24(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x = audio_stream_get_rptr(source); int32_t *y = audio_stream_get_wptr(sink); int fmax; @@ -150,8 +150,8 @@ void tdfb_fir_s24(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource void tdfb_fir_s32(struct tdfb_comp_data *cd, struct input_stream_buffer *bsource, struct output_stream_buffer *bsink, int frames) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x = audio_stream_get_rptr(source); int32_t *y = audio_stream_get_wptr(sink); int fmax; diff --git a/src/audio/tone.c b/src/audio/tone.c index 65c934246909..0f744cb75d4d 100644 --- a/src/audio/tone.c +++ b/src/audio/tone.c @@ -95,7 +95,7 @@ struct comp_data { uint32_t frame_bytes; uint32_t rate; struct tone_state sg[PLATFORM_MAX_CHANNELS]; - void (*tone_func)(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, + void (*tone_func)(struct comp_dev *dev, struct audio_stream *sink, uint32_t frames); }; @@ -113,7 +113,7 @@ static inline void tone_circ_inc_wrap(int32_t **ptr, int32_t *end, size_t size) *ptr = (int32_t *)((size_t)*ptr - size); } -static void tone_s32_default(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, +static void tone_s32_default(struct comp_dev *dev, struct audio_stream *sink, uint32_t frames) { struct comp_data *cd = comp_get_drvdata(dev); @@ -429,7 +429,7 @@ static int tone_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; + struct comp_buffer *source_c, *sink_c; sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list); @@ -634,7 +634,7 @@ static int tone_trigger(struct comp_dev *dev, int cmd) static int tone_copy(struct comp_dev *dev) { struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; struct comp_data *cd = comp_get_drvdata(dev); uint32_t free; int ret = 0; diff --git a/src/audio/up_down_mixer/up_down_mixer.c b/src/audio/up_down_mixer/up_down_mixer.c index e1887865ac70..7736b681dd24 100644 --- a/src/audio/up_down_mixer/up_down_mixer.c +++ b/src/audio/up_down_mixer/up_down_mixer.c @@ -400,8 +400,8 @@ static int up_down_mixer_init(struct processing_module *mod) /* just stubs for now. Remove these after making these ops optional in the module adapter */ static int up_down_mixer_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) { struct up_down_mixer_data *cd = module_get_private_data(mod); struct comp_dev *dev = mod->dev; diff --git a/src/audio/volume/volume.c b/src/audio/volume/volume.c index a4128136a5e5..797b201b03b5 100644 --- a/src/audio/volume/volume.c +++ b/src/audio/volume/volume.c @@ -57,7 +57,7 @@ LOG_MODULE_REGISTER(volume, CONFIG_SOF_LOG_LEVEL); * \param[in] frames Number of frames. * \param[in,out] prev_sum Previous sum of channel samples. */ -static uint32_t vol_zc_get_s16(const struct audio_stream __sparse_cache *source, +static uint32_t vol_zc_get_s16(const struct audio_stream *source, uint32_t frames, int64_t *prev_sum) { uint32_t curr_frames = frames; @@ -105,7 +105,7 @@ static uint32_t vol_zc_get_s16(const struct audio_stream __sparse_cache *source, * \param[in] frames Number of frames. * \param[in,out] prev_sum Previous sum of channel samples. */ -static uint32_t vol_zc_get_s24(const struct audio_stream __sparse_cache *source, +static uint32_t vol_zc_get_s24(const struct audio_stream *source, uint32_t frames, int64_t *prev_sum) { int64_t sum; @@ -153,7 +153,7 @@ static uint32_t vol_zc_get_s24(const struct audio_stream __sparse_cache *source, * \param[in] frames Number of frames. * \param[in,out] prev_sum Previous sum of channel samples. */ -static uint32_t vol_zc_get_s32(const struct audio_stream __sparse_cache *source, +static uint32_t vol_zc_get_s32(const struct audio_stream *source, uint32_t frames, int64_t *prev_sum) { int64_t sum; @@ -575,7 +575,7 @@ static int volume_process(struct processing_module *mod, * \param[in,out] dev Volume base component device. */ static vol_zc_func vol_get_zc_function(struct comp_dev *dev, - struct comp_buffer __sparse_cache *sinkb) + struct comp_buffer *sinkb) { int i; @@ -593,8 +593,8 @@ static vol_zc_func vol_get_zc_function(struct comp_dev *dev, * \param[in,out] source Structure pointer of source. * \param[in,out] sink Structure pointer of sink. */ -static void volume_set_alignment(struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink) +static void volume_set_alignment(struct audio_stream *source, + struct audio_stream *sink) { #if XCHAL_HAVE_HIFI3 || XCHAL_HAVE_HIFI4 @@ -631,14 +631,14 @@ static void volume_set_alignment(struct audio_stream __sparse_cache *source, * to also do some type of conversion here. */ static int volume_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) { struct vol_data *cd = module_get_private_data(mod); struct module_data *md = &mod->priv; struct comp_dev *dev = mod->dev; struct comp_buffer *sourceb, *sinkb; - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; uint32_t sink_period_bytes; int ret; int i; diff --git a/src/audio/volume/volume.h b/src/audio/volume/volume.h index 856c27568d84..140ec3e7af2f 100644 --- a/src/audio/volume/volume.h +++ b/src/audio/volume/volume.h @@ -134,7 +134,7 @@ typedef void (*vol_scale_func)(struct processing_module *mod, struct input_strea /** * \brief volume interface for function getting nearest zero crossing frame */ -typedef uint32_t (*vol_zc_func)(const struct audio_stream __sparse_cache *source, +typedef uint32_t (*vol_zc_func)(const struct audio_stream *source, uint32_t frames, int64_t *prev_sum); /** @@ -205,7 +205,7 @@ struct comp_zc_func_map { * \param[in] cd Volume data structure. */ static inline vol_scale_func vol_get_processing_function(struct comp_dev *dev, - struct comp_buffer __sparse_cache *sinkb, + struct comp_buffer *sinkb, struct vol_data *cd) { int i; diff --git a/src/audio/volume/volume_generic.c b/src/audio/volume/volume_generic.c index 966f62b7da34..30498bc992eb 100644 --- a/src/audio/volume/volume_generic.c +++ b/src/audio/volume/volume_generic.c @@ -61,8 +61,8 @@ static void vol_s24_to_s24(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t vol; int32_t *x, *x0; int32_t *y, *y0; @@ -110,8 +110,8 @@ static void vol_passthrough_s24_to_s24(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x; int32_t *y; int nmax, n; @@ -153,8 +153,8 @@ static void vol_s32_to_s32(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t vol; int32_t *x, *x0; int32_t *y, *y0; @@ -205,8 +205,8 @@ static void vol_passthrough_s32_to_s32(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x; int32_t *y; int nmax, n; @@ -247,8 +247,8 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t vol; int16_t *x, *x0; int16_t *y, *y0; @@ -296,8 +296,8 @@ static void vol_passthrough_s16_to_s16(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int16_t *x; int16_t *y; int nmax, n; diff --git a/src/audio/volume/volume_generic_with_peakvol.c b/src/audio/volume/volume_generic_with_peakvol.c index fb0cff0bee3a..0ef720d78665 100644 --- a/src/audio/volume/volume_generic_with_peakvol.c +++ b/src/audio/volume/volume_generic_with_peakvol.c @@ -57,8 +57,8 @@ static void vol_s24_to_s24(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t vol; int32_t *x, *x0; int32_t *y, *y0; @@ -112,8 +112,8 @@ static void vol_passthrough_s24_to_s24(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x, *x0; int32_t *y, *y0; int nmax, n, i, j; @@ -166,8 +166,8 @@ static void vol_s32_to_s32(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t vol; int32_t *x, *x0; int32_t *y, *y0; @@ -224,8 +224,8 @@ static void vol_passthrough_s32_to_s32(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x, *x0; int32_t *y, *y0; int nmax, n, i, j; @@ -280,8 +280,8 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t vol; int16_t *x, *x0; int16_t *y, *y0; @@ -336,8 +336,8 @@ static void vol_passthrough_s16_to_s16(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *x, *x0; int32_t *y, *y0; int nmax, n, i, j; diff --git a/src/audio/volume/volume_hifi3.c b/src/audio/volume/volume_hifi3.c index 05746ab40f67..e32b8551c775 100644 --- a/src/audio/volume/volume_hifi3.c +++ b/src/audio/volume/volume_hifi3.c @@ -61,8 +61,8 @@ static void vol_s24_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -147,8 +147,8 @@ static void vol_passthrough_s24_to_s24_s32(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, m; ae_valign inu = AE_ZALIGN64(); @@ -198,8 +198,8 @@ static void vol_s32_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -289,8 +289,8 @@ static void vol_passthrough_s32_to_s24_s32(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, m; ae_valign inu = AE_ZALIGN64(); @@ -339,8 +339,8 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 volume0 = AE_ZERO32(); ae_f32x2 volume1 = AE_ZERO32(); ae_f32x2 out_sample0 = AE_ZERO32(); @@ -436,8 +436,8 @@ static void vol_passthrough_s16_to_s16(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f16x4 in_sample = AE_ZERO16(); int i, n, m; ae_valign inu = AE_ZALIGN64(); diff --git a/src/audio/volume/volume_hifi3_with_peakvol.c b/src/audio/volume/volume_hifi3_with_peakvol.c index 5d0ff31b0385..9bda00a27bc7 100644 --- a/src/audio/volume/volume_hifi3_with_peakvol.c +++ b/src/audio/volume/volume_hifi3_with_peakvol.c @@ -41,8 +41,8 @@ static void vol_s24_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -116,8 +116,8 @@ static void vol_passthrough_s24_to_s24_s32(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int channel, n, i, m; ae_f32 *in0 = (ae_f32 *)audio_stream_wrap(source, (char *)audio_stream_get_rptr(source) @@ -176,8 +176,8 @@ static void vol_s32_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -252,8 +252,8 @@ static void vol_passthrough_s32_to_s24_s32(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, channel, m; const int channels_count = audio_stream_get_channels(sink); @@ -312,8 +312,8 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 volume = AE_ZERO32(); ae_f32x2 out_sample0 = AE_ZERO32(); ae_f16x4 in_sample = AE_ZERO16(); @@ -394,8 +394,8 @@ static void vol_passthrough_s16_to_s16(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f16x4 in_sample = AE_ZERO16(); int i, n, channel, m; ae_f16 *in; diff --git a/src/audio/volume/volume_hifi4.c b/src/audio/volume/volume_hifi4.c index 880aef23f139..d0d114dcfc5d 100644 --- a/src/audio/volume/volume_hifi4.c +++ b/src/audio/volume/volume_hifi4.c @@ -61,8 +61,8 @@ static void vol_s24_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -147,8 +147,8 @@ static void vol_passthrough_s24_to_s24_s32(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, m; ae_valign inu = AE_ZALIGN64(); @@ -198,8 +198,8 @@ static void vol_s32_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -289,8 +289,8 @@ static void vol_passthrough_s32_to_s24_s32(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, m; ae_valign inu = AE_ZALIGN64(); @@ -338,8 +338,8 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 volume0 = AE_ZERO32(); ae_f32x2 volume1 = AE_ZERO32(); ae_f32x2 out_sample0 = AE_ZERO32(); @@ -435,8 +435,8 @@ static void vol_passthrough_s16_to_s16(struct processing_module *mod, struct output_stream_buffer *bsink, uint32_t frames, uint32_t attenuation) { - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f16x4 in_sample = AE_ZERO16(); int i, n, m; ae_valign inu = AE_ZALIGN64(); diff --git a/src/audio/volume/volume_hifi4_with_peakvol.c b/src/audio/volume/volume_hifi4_with_peakvol.c index df1eafff2788..cbb08e297b87 100644 --- a/src/audio/volume/volume_hifi4_with_peakvol.c +++ b/src/audio/volume/volume_hifi4_with_peakvol.c @@ -54,8 +54,8 @@ static void vol_s24_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -150,8 +150,8 @@ static void vol_passthrough_s24_to_s24_s32(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, m; @@ -217,8 +217,8 @@ static void vol_s32_to_s24_s32(struct processing_module *mod, struct input_strea uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); ae_f32x2 out_sample = AE_ZERO32(); ae_f32x2 volume = AE_ZERO32(); @@ -321,8 +321,8 @@ static void vol_passthrough_s32_to_s24_s32(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 in_sample = AE_ZERO32(); int i, n, m; ae_valign inu = AE_ZALIGN64(); @@ -385,8 +385,8 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f32x2 volume0 = AE_ZERO32(); ae_f32x2 volume1 = AE_ZERO32(); ae_f32x2 out_sample0 = AE_ZERO32(); @@ -501,8 +501,8 @@ static void vol_passthrough_s16_to_s16(struct processing_module *mod, uint32_t attenuation) { struct vol_data *cd = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; ae_f16x4 in_sample = AE_ZERO16(); int i, n, m; ae_valign inu = AE_ZALIGN64(); diff --git a/src/audio/volume/volume_ipc3.c b/src/audio/volume/volume_ipc3.c index c1378acbdd45..3881c4ae7f8b 100644 --- a/src/audio/volume/volume_ipc3.c +++ b/src/audio/volume/volume_ipc3.c @@ -37,7 +37,7 @@ LOG_MODULE_DECLARE(volume, CONFIG_SOF_LOG_LEVEL); void set_volume_process(struct vol_data *cd, struct comp_dev *dev, bool source_or_sink) { struct comp_buffer *bufferb; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; if (source_or_sink) bufferb = list_first_item(&dev->bsource_list, diff --git a/src/audio/volume/volume_ipc4.c b/src/audio/volume/volume_ipc4.c index 0ec4c4c275f0..23cd6c7281eb 100644 --- a/src/audio/volume/volume_ipc4.c +++ b/src/audio/volume/volume_ipc4.c @@ -366,7 +366,7 @@ int volume_get_config(struct processing_module *mod, static int volume_params(struct processing_module *mod) { struct sof_ipc_stream_params *params = mod->stream_params; - struct comp_buffer __sparse_cache *sink_c, *source_c; + struct comp_buffer *sink_c, *source_c; struct comp_buffer *sinkb, *sourceb; struct comp_dev *dev = mod->dev; diff --git a/src/include/ipc4/base-config.h b/src/include/ipc4/base-config.h index 26eeb07bf355..2e54478efd3a 100644 --- a/src/include/ipc4/base-config.h +++ b/src/include/ipc4/base-config.h @@ -242,7 +242,7 @@ struct sof_ipc_stream_params; void ipc4_base_module_cfg_to_stream_params(const struct ipc4_base_module_cfg *base_cfg, struct sof_ipc_stream_params *params); struct comp_buffer; -void ipc4_update_buffer_format(struct comp_buffer __sparse_cache *buf_c, +void ipc4_update_buffer_format(struct comp_buffer *buf_c, const struct ipc4_audio_format *fmt); #endif diff --git a/src/include/ipc4/copier.h b/src/include/ipc4/copier.h index 285b6ca3f4d4..23953d6aca97 100644 --- a/src/include/ipc4/copier.h +++ b/src/include/ipc4/copier.h @@ -272,7 +272,7 @@ struct copier_data { }; int apply_attenuation(struct comp_dev *dev, struct copier_data *cd, - struct comp_buffer __sparse_cache *sink, int frame); + struct comp_buffer *sink, int frame); pcm_converter_func get_converter_func(const struct ipc4_audio_format *in_fmt, const struct ipc4_audio_format *out_fmt, diff --git a/src/include/ipc4/mixin_mixout.h b/src/include/ipc4/mixin_mixout.h index f44ef97e0b37..49184e2aa51b 100644 --- a/src/include/ipc4/mixin_mixout.h +++ b/src/include/ipc4/mixin_mixout.h @@ -105,15 +105,15 @@ struct ipc4_mixer_mode_config { /** * \brief normal mode mixin_mixout processing function interface */ -typedef void (*normal_mix_func)(struct audio_stream __sparse_cache *sink, int32_t start_frame, +typedef void (*normal_mix_func)(struct audio_stream *sink, int32_t start_frame, int32_t mixed_frames, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, int32_t frame_count, uint16_t gain); /** * \brief mixin_mixout mute processing function interface */ -typedef void (*mute_func) (struct audio_stream __sparse_cache *stream, int32_t channel_index, +typedef void (*mute_func) (struct audio_stream *stream, int32_t channel_index, int32_t start_frame, int32_t mixed_frames, int32_t frame_count); /** diff --git a/src/include/sof/audio/aria/aria.h b/src/include/sof/audio/aria/aria.h index 64e34e09c86b..6718f84a72da 100644 --- a/src/include/sof/audio/aria/aria.h +++ b/src/include/sof/audio/aria/aria.h @@ -48,14 +48,14 @@ * \brief aria get data function interface */ typedef void (*aria_get_data_func)(struct processing_module *mod, - struct audio_stream __sparse_cache *sink, int frames); + struct audio_stream *sink, int frames); struct aria_data; /** * \brief Aria gain processing function */ 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); aria_get_data_func aria_algo_get_data_func(struct processing_module *mod); diff --git a/src/include/sof/audio/audio_stream.h b/src/include/sof/audio/audio_stream.h index ecfc63432d86..1bbc23b1e85e 100644 --- a/src/include/sof/audio/audio_stream.h +++ b/src/include/sof/audio/audio_stream.h @@ -104,148 +104,148 @@ struct audio_stream { struct sof_audio_stream_params runtime_stream_params; }; -static inline void *audio_stream_get_rptr(const struct audio_stream __sparse_cache *buf) +static inline void *audio_stream_get_rptr(const struct audio_stream *buf) { return buf->r_ptr; } -static inline void *audio_stream_get_wptr(const struct audio_stream __sparse_cache *buf) +static inline void *audio_stream_get_wptr(const struct audio_stream *buf) { return buf->w_ptr; } -static inline void *audio_stream_get_end_addr(const struct audio_stream __sparse_cache *buf) +static inline void *audio_stream_get_end_addr(const struct audio_stream *buf) { return buf->end_addr; } -static inline void *audio_stream_get_addr(const struct audio_stream __sparse_cache *buf) +static inline void *audio_stream_get_addr(const struct audio_stream *buf) { return buf->addr; } -static inline uint32_t audio_stream_get_size(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_get_size(const struct audio_stream *buf) { return buf->size; } -static inline uint32_t audio_stream_get_avail(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_get_avail(const struct audio_stream *buf) { return buf->avail; } -static inline uint32_t audio_stream_get_free(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_get_free(const struct audio_stream *buf) { return buf->free; } static inline enum sof_ipc_frame audio_stream_get_frm_fmt( - const struct audio_stream __sparse_cache *buf) + const struct audio_stream *buf) { return buf->runtime_stream_params.frame_fmt; } static inline enum sof_ipc_frame audio_stream_get_valid_fmt( - const struct audio_stream __sparse_cache *buf) + const struct audio_stream *buf) { return buf->runtime_stream_params.valid_sample_fmt; } -static inline uint32_t audio_stream_get_rate(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_get_rate(const struct audio_stream *buf) { return buf->runtime_stream_params.rate; } -static inline uint32_t audio_stream_get_channels(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_get_channels(const struct audio_stream *buf) { return buf->runtime_stream_params.channels; } -static inline bool audio_stream_get_underrun(const struct audio_stream __sparse_cache *buf) +static inline bool audio_stream_get_underrun(const struct audio_stream *buf) { return buf->runtime_stream_params.underrun_permitted; } -static inline uint32_t audio_stream_get_buffer_fmt(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_get_buffer_fmt(const struct audio_stream *buf) { return buf->runtime_stream_params.buffer_fmt; } -static inline bool audio_stream_get_overrun(const struct audio_stream __sparse_cache *buf) +static inline bool audio_stream_get_overrun(const struct audio_stream *buf) { return buf->runtime_stream_params.overrun_permitted; } -static inline void audio_stream_set_rptr(struct audio_stream __sparse_cache *buf, void *val) +static inline void audio_stream_set_rptr(struct audio_stream *buf, void *val) { buf->r_ptr = val; } -static inline void audio_stream_set_wptr(struct audio_stream __sparse_cache *buf, void *val) +static inline void audio_stream_set_wptr(struct audio_stream *buf, void *val) { buf->w_ptr = val; } -static inline void audio_stream_set_end_addr(struct audio_stream __sparse_cache *buf, void *val) +static inline void audio_stream_set_end_addr(struct audio_stream *buf, void *val) { buf->end_addr = val; } -static inline void audio_stream_set_addr(struct audio_stream __sparse_cache *buf, void *val) +static inline void audio_stream_set_addr(struct audio_stream *buf, void *val) { buf->addr = val; } -static inline void audio_stream_set_size(struct audio_stream __sparse_cache *buf, uint32_t val) +static inline void audio_stream_set_size(struct audio_stream *buf, uint32_t val) { buf->size = val; } -static inline void audio_stream_set_avail(struct audio_stream __sparse_cache *buf, uint32_t val) +static inline void audio_stream_set_avail(struct audio_stream *buf, uint32_t val) { buf->avail = val; } -static inline void audio_stream_set_free(struct audio_stream __sparse_cache *buf, uint32_t val) +static inline void audio_stream_set_free(struct audio_stream *buf, uint32_t val) { buf->free = val; } -static inline void audio_stream_set_frm_fmt(struct audio_stream __sparse_cache *buf, +static inline void audio_stream_set_frm_fmt(struct audio_stream *buf, enum sof_ipc_frame val) { buf->runtime_stream_params.frame_fmt = val; } -static inline void audio_stream_set_valid_fmt(struct audio_stream __sparse_cache *buf, +static inline void audio_stream_set_valid_fmt(struct audio_stream *buf, enum sof_ipc_frame val) { buf->runtime_stream_params.valid_sample_fmt = val; } -static inline void audio_stream_set_rate(struct audio_stream __sparse_cache *buf, uint32_t val) +static inline void audio_stream_set_rate(struct audio_stream *buf, uint32_t val) { buf->runtime_stream_params.rate = val; } -static inline void audio_stream_set_channels(struct audio_stream __sparse_cache *buf, uint16_t val) +static inline void audio_stream_set_channels(struct audio_stream *buf, uint16_t val) { buf->runtime_stream_params.channels = val; } -static inline void audio_stream_set_underrun(struct audio_stream __sparse_cache *buf, +static inline void audio_stream_set_underrun(struct audio_stream *buf, bool underrun_permitted) { buf->runtime_stream_params.underrun_permitted = underrun_permitted; } -static inline void audio_stream_set_overrun(struct audio_stream __sparse_cache *buf, +static inline void audio_stream_set_overrun(struct audio_stream *buf, bool overrun_permitted) { buf->runtime_stream_params.overrun_permitted = overrun_permitted; } -static inline void audio_stream_set_buffer_fmt(struct audio_stream __sparse_cache *buf, +static inline void audio_stream_set_buffer_fmt(struct audio_stream *buf, uint32_t buffer_fmt) { buf->runtime_stream_params.buffer_fmt = buffer_fmt; @@ -356,7 +356,7 @@ static inline void audio_stream_set_buffer_fmt(struct audio_stream __sparse_cach * @param params Parameters (frame format, rate, number of channels). * @return 0 if succeeded, error code otherwise. */ -static inline int audio_stream_set_params(struct audio_stream __sparse_cache *buffer, +static inline int audio_stream_set_params(struct audio_stream *buffer, struct sof_ipc_stream_params *params) { if (!params) @@ -374,7 +374,7 @@ static inline int audio_stream_set_params(struct audio_stream __sparse_cache *bu * @param buf Component buffer. * @return Period size in bytes. */ -static inline uint32_t audio_stream_frame_bytes(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_frame_bytes(const struct audio_stream *buf) { return get_frame_bytes(buf->runtime_stream_params.frame_fmt, buf->runtime_stream_params.channels); @@ -385,7 +385,7 @@ static inline uint32_t audio_stream_frame_bytes(const struct audio_stream __spar * @param buf Component buffer. * @return Size of sample in bytes. */ -static inline uint32_t audio_stream_sample_bytes(const struct audio_stream __sparse_cache *buf) +static inline uint32_t audio_stream_sample_bytes(const struct audio_stream *buf) { return get_sample_bytes(buf->runtime_stream_params.frame_fmt); } @@ -423,7 +423,7 @@ static inline uint32_t audio_stream_frame_align_get(const uint32_t byte_align, */ static inline void audio_stream_init_alignment_constants(const uint32_t byte_align, const uint32_t frame_align_req, - struct audio_stream __sparse_cache *stream) + struct audio_stream *stream) { uint32_t process_size; uint32_t frame_size = audio_stream_frame_bytes(stream); @@ -441,7 +441,7 @@ static inline void audio_stream_init_alignment_constants(const uint32_t byte_ali * @param frames Number of processing frames. * @return Period size in bytes. */ -static inline uint32_t audio_stream_period_bytes(const struct audio_stream __sparse_cache *buf, +static inline uint32_t audio_stream_period_bytes(const struct audio_stream *buf, uint32_t frames) { return frames * audio_stream_frame_bytes(buf); @@ -454,7 +454,7 @@ static inline uint32_t audio_stream_period_bytes(const struct audio_stream __spa * @param ptr Pointer * @return Pointer, adjusted if necessary. */ -static inline void *audio_stream_wrap(const struct audio_stream __sparse_cache *buffer, +static inline void *audio_stream_wrap(const struct audio_stream *buffer, void *ptr) { if (ptr >= buffer->end_addr) @@ -492,7 +492,7 @@ static inline void *cir_buf_wrap(void *ptr, void *buf_addr, void *buf_end) * @param ptr Pointer * @return Pointer, adjusted if necessary. */ -static inline void *audio_stream_rewind_wrap(const struct audio_stream __sparse_cache *buffer, +static inline void *audio_stream_rewind_wrap(const struct audio_stream *buffer, void *ptr) { if (ptr < buffer->addr) @@ -509,7 +509,7 @@ static inline void *audio_stream_rewind_wrap(const struct audio_stream __sparse_ * @return amount of data available for processing in bytes */ static inline uint32_t -audio_stream_get_avail_bytes(const struct audio_stream __sparse_cache *stream) +audio_stream_get_avail_bytes(const struct audio_stream *stream) { /* * In case of underrun-permitted stream, report buffer full instead of @@ -529,7 +529,7 @@ audio_stream_get_avail_bytes(const struct audio_stream __sparse_cache *stream) * @return amount of data available for processing in samples */ static inline uint32_t -audio_stream_get_avail_samples(const struct audio_stream __sparse_cache *stream) +audio_stream_get_avail_samples(const struct audio_stream *stream) { return audio_stream_get_avail_bytes(stream) / audio_stream_sample_bytes(stream); @@ -541,7 +541,7 @@ audio_stream_get_avail_samples(const struct audio_stream __sparse_cache *stream) * @return amount of data available for processing in frames */ static inline uint32_t -audio_stream_get_avail_frames(const struct audio_stream __sparse_cache *stream) +audio_stream_get_avail_frames(const struct audio_stream *stream) { return audio_stream_get_avail_bytes(stream) / audio_stream_frame_bytes(stream); @@ -553,7 +553,7 @@ audio_stream_get_avail_frames(const struct audio_stream __sparse_cache *stream) * @return amount of space free in bytes */ static inline uint32_t -audio_stream_get_free_bytes(const struct audio_stream __sparse_cache *stream) +audio_stream_get_free_bytes(const struct audio_stream *stream) { /* * In case of overrun-permitted stream, report buffer empty instead of @@ -573,7 +573,7 @@ audio_stream_get_free_bytes(const struct audio_stream __sparse_cache *stream) * @return amount of space free in samples */ static inline uint32_t -audio_stream_get_free_samples(const struct audio_stream __sparse_cache *stream) +audio_stream_get_free_samples(const struct audio_stream *stream) { return audio_stream_get_free_bytes(stream) / audio_stream_sample_bytes(stream); @@ -585,7 +585,7 @@ audio_stream_get_free_samples(const struct audio_stream __sparse_cache *stream) * @return amount of space free in frames */ static inline uint32_t -audio_stream_get_free_frames(const struct audio_stream __sparse_cache *stream) +audio_stream_get_free_frames(const struct audio_stream *stream) { return audio_stream_get_free_bytes(stream) / audio_stream_frame_bytes(stream); @@ -601,8 +601,8 @@ audio_stream_get_free_frames(const struct audio_stream __sparse_cache *stream) * @return 1 if there is not enough free space in sink. * @return -1 if there is not enough data in source. */ -static inline int audio_stream_can_copy_bytes(const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, +static inline int audio_stream_can_copy_bytes(const struct audio_stream *source, + const struct audio_stream *sink, uint32_t bytes) { /* check for underrun */ @@ -626,8 +626,8 @@ static inline int audio_stream_can_copy_bytes(const struct audio_stream __sparse * @return Number of bytes. */ static inline uint32_t -audio_stream_get_copy_bytes(const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink) +audio_stream_get_copy_bytes(const struct audio_stream *source, + const struct audio_stream *sink) { uint32_t avail = audio_stream_get_avail_bytes(source); uint32_t free = audio_stream_get_free_bytes(sink); @@ -647,8 +647,8 @@ audio_stream_get_copy_bytes(const struct audio_stream __sparse_cache *source, * @return Number of frames. */ static inline uint32_t -audio_stream_avail_frames(const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink) +audio_stream_avail_frames(const struct audio_stream *source, + const struct audio_stream *sink) { uint32_t src_frames = audio_stream_get_avail_frames(source); uint32_t sink_frames = audio_stream_get_free_frames(sink); @@ -665,8 +665,8 @@ audio_stream_avail_frames(const struct audio_stream __sparse_cache *source, * @return Number of frames. */ static inline uint32_t -audio_stream_avail_frames_aligned(const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink) +audio_stream_avail_frames_aligned(const struct audio_stream *source, + const struct audio_stream *sink) { uint32_t src_frames = (audio_stream_get_avail_bytes(source) >> source->runtime_stream_params.align_shift_idx) * @@ -683,7 +683,7 @@ audio_stream_avail_frames_aligned(const struct audio_stream __sparse_cache *sour * @param buffer Buffer to update. * @param bytes Number of written bytes. */ -static inline void audio_stream_produce(struct audio_stream __sparse_cache *buffer, +static inline void audio_stream_produce(struct audio_stream *buffer, uint32_t bytes) { buffer->w_ptr = audio_stream_wrap(buffer, @@ -711,7 +711,7 @@ static inline void audio_stream_produce(struct audio_stream __sparse_cache *buff * @param buffer Buffer to update. * @param bytes Number of read bytes. */ -static inline void audio_stream_consume(struct audio_stream __sparse_cache *buffer, +static inline void audio_stream_consume(struct audio_stream *buffer, uint32_t bytes) { buffer->r_ptr = audio_stream_wrap(buffer, @@ -734,7 +734,7 @@ static inline void audio_stream_consume(struct audio_stream __sparse_cache *buff * Resets the buffer. * @param buffer Buffer to reset. */ -static inline void audio_stream_reset(struct audio_stream __sparse_cache *buffer) +static inline void audio_stream_reset(struct audio_stream *buffer) { /* reset read and write pointer to buffer bas */ buffer->w_ptr = buffer->addr; @@ -753,7 +753,7 @@ static inline void audio_stream_reset(struct audio_stream __sparse_cache *buffer * @param buff_addr Address of the memory block to assign. * @param size Size of the memory block in bytes. */ -void audio_stream_init(struct audio_stream __sparse_cache *audio_stream, +void audio_stream_init(struct audio_stream *audio_stream, void *buff_addr, uint32_t size); /** @@ -762,7 +762,7 @@ void audio_stream_init(struct audio_stream __sparse_cache *audio_stream, * @param buffer Buffer. * @param bytes Size of the fragment to invalidate. */ -static inline void audio_stream_invalidate(struct audio_stream __sparse_cache *buffer, +static inline void audio_stream_invalidate(struct audio_stream *buffer, uint32_t bytes) { uint32_t head_size = bytes; @@ -786,7 +786,7 @@ static inline void audio_stream_invalidate(struct audio_stream __sparse_cache *b * @param buffer Buffer. * @param bytes Size of the fragment to write back. */ -static inline void audio_stream_writeback(struct audio_stream __sparse_cache *buffer, +static inline void audio_stream_writeback(struct audio_stream *buffer, uint32_t bytes) { uint32_t head_size = bytes; @@ -811,7 +811,7 @@ static inline void audio_stream_writeback(struct audio_stream __sparse_cache *bu * @return Number of data samples to buffer wrap. */ static inline int -audio_stream_bytes_without_wrap(const struct audio_stream __sparse_cache *source, +audio_stream_bytes_without_wrap(const struct audio_stream *source, const void *ptr) { assert((intptr_t)source->end_addr >= (intptr_t)ptr); @@ -827,7 +827,7 @@ audio_stream_bytes_without_wrap(const struct audio_stream __sparse_cache *source * need to add size of sample to returned bytes count. */ static inline int -audio_stream_rewind_bytes_without_wrap(const struct audio_stream __sparse_cache *source, +audio_stream_rewind_bytes_without_wrap(const struct audio_stream *source, const void *ptr) { assert((intptr_t)ptr >= (intptr_t)source->addr); @@ -843,7 +843,7 @@ audio_stream_rewind_bytes_without_wrap(const struct audio_stream __sparse_cache * @return Previous position of the write pointer. */ static inline uint32_t -*audio_stream_rewind_wptr_by_bytes(const struct audio_stream __sparse_cache *source, +*audio_stream_rewind_wptr_by_bytes(const struct audio_stream *source, const uint32_t bytes) { void *wptr = audio_stream_get_wptr(source); @@ -866,7 +866,7 @@ static inline uint32_t * @return Number of data s16 samples until circular wrap need at end */ static inline int -audio_stream_samples_without_wrap_s16(const struct audio_stream __sparse_cache *source, +audio_stream_samples_without_wrap_s16(const struct audio_stream *source, const void *ptr) { int to_end = (int16_t *)source->end_addr - (int16_t *)ptr; @@ -883,7 +883,7 @@ audio_stream_samples_without_wrap_s16(const struct audio_stream __sparse_cache * * @return Number of data s24 samples until circular wrap need at end */ static inline int -audio_stream_samples_without_wrap_s24(const struct audio_stream __sparse_cache *source, +audio_stream_samples_without_wrap_s24(const struct audio_stream *source, const void *ptr) { int to_end = (int32_t *)source->end_addr - (int32_t *)ptr; @@ -900,7 +900,7 @@ audio_stream_samples_without_wrap_s24(const struct audio_stream __sparse_cache * * @return Number of data s32 samples until circular wrap need at end */ static inline int -audio_stream_samples_without_wrap_s32(const struct audio_stream __sparse_cache *source, +audio_stream_samples_without_wrap_s32(const struct audio_stream *source, const void *ptr) { int to_end = (int32_t *)source->end_addr - (int32_t *)ptr; @@ -947,7 +947,7 @@ static inline int cir_buf_samples_without_wrap_s32(void *ptr, void *buf_end) * @return Number of data frames to buffer wrap. */ static inline uint32_t -audio_stream_frames_without_wrap(const struct audio_stream __sparse_cache *source, +audio_stream_frames_without_wrap(const struct audio_stream *source, const void *ptr) { uint32_t bytes = audio_stream_bytes_without_wrap(source, ptr); @@ -965,8 +965,8 @@ audio_stream_frames_without_wrap(const struct audio_stream __sparse_cache *sourc * @param samples Number of samples to copy. * @return number of processed samples. */ -int audio_stream_copy(const struct audio_stream __sparse_cache *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, uint32_t ooffset, uint32_t samples); +int audio_stream_copy(const struct audio_stream *source, uint32_t ioffset, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples); /** * Copies data from one circular buffer to another circular buffer. @@ -990,7 +990,7 @@ void cir_buf_copy(void *src, void *src_addr, void *src_end, void *dst, * @param samples Number of samples to copy. */ void audio_stream_copy_from_linear(const void *linear_source, int ioffset, - struct audio_stream __sparse_cache *sink, int ooffset, + struct audio_stream *sink, int ooffset, unsigned int samples); /** @@ -1001,7 +1001,7 @@ void audio_stream_copy_from_linear(const void *linear_source, int ioffset, * @param ooffset Offset (in samples) in sink buffer to start writing to. * @param samples Number of samples to copy. */ -void audio_stream_copy_to_linear(const struct audio_stream __sparse_cache *source, int ioffset, +void audio_stream_copy_to_linear(const struct audio_stream *source, int ioffset, void *linear_sink, int ooffset, unsigned int samples); /** @@ -1011,7 +1011,7 @@ void audio_stream_copy_to_linear(const struct audio_stream __sparse_cache *sourc * @return 0 if there is enough free space in buffer. * @return 1 if there is not enough free space in buffer. */ -static inline int audio_stream_set_zero(struct audio_stream __sparse_cache *buffer, +static inline int audio_stream_set_zero(struct audio_stream *buffer, uint32_t bytes) { uint32_t head_size = bytes; @@ -1073,14 +1073,14 @@ static inline void audio_stream_fmt_conversion(enum ipc4_bit_depth depth, /** get a handler to source API * NOTE! to use the handlers the buffer must be acquired by buffer_acquire */ -static inline struct sof_source __sparse_cache * -audio_stream_get_source(struct audio_stream __sparse_cache *audio_stream) +static inline struct sof_source * +audio_stream_get_source(struct audio_stream *audio_stream) { return &audio_stream->source_api; } -static inline struct sof_sink __sparse_cache * -audio_stream_get_sink(struct audio_stream __sparse_cache *audio_stream) +static inline struct sof_sink * +audio_stream_get_sink(struct audio_stream *audio_stream) { return &audio_stream->sink_api; } diff --git a/src/include/sof/audio/buffer.h b/src/include/sof/audio/buffer.h index 51a505b209cf..6391e9962900 100644 --- a/src/include/sof/audio/buffer.h +++ b/src/include/sof/audio/buffer.h @@ -163,7 +163,7 @@ struct comp_buffer { /* Only to be used for synchronous same-core notifications! */ struct buffer_cb_transact { - struct comp_buffer __sparse_cache *buffer; + struct comp_buffer *buffer; uint32_t transaction_amount; void *transaction_begin_address; }; @@ -192,30 +192,30 @@ struct buffer_cb_free { struct comp_buffer *buffer_alloc(uint32_t size, uint32_t caps, uint32_t flags, uint32_t align, bool is_shared); struct comp_buffer *buffer_new(const struct sof_ipc_buffer *desc, bool is_shared); -int buffer_set_size(struct comp_buffer __sparse_cache *buffer, uint32_t size, uint32_t alignment); +int buffer_set_size(struct comp_buffer *buffer, uint32_t size, uint32_t alignment); void buffer_free(struct comp_buffer *buffer); -void buffer_zero(struct comp_buffer __sparse_cache *buffer); +void buffer_zero(struct comp_buffer *buffer); /* called by a component after producing data into this buffer */ -void comp_update_buffer_produce(struct comp_buffer __sparse_cache *buffer, uint32_t bytes); +void comp_update_buffer_produce(struct comp_buffer *buffer, uint32_t bytes); /* called by a component after consuming data from this buffer */ -void comp_update_buffer_consume(struct comp_buffer __sparse_cache *buffer, uint32_t bytes); +void comp_update_buffer_consume(struct comp_buffer *buffer, uint32_t bytes); -int buffer_set_params(struct comp_buffer __sparse_cache *buffer, +int buffer_set_params(struct comp_buffer *buffer, struct sof_ipc_stream_params *params, bool force_update); -bool buffer_params_match(struct comp_buffer __sparse_cache *buffer, +bool buffer_params_match(struct comp_buffer *buffer, struct sof_ipc_stream_params *params, uint32_t flag); -static inline void buffer_stream_invalidate(struct comp_buffer __sparse_cache *buffer, +static inline void buffer_stream_invalidate(struct comp_buffer *buffer, uint32_t bytes) { if (buffer->is_shared) audio_stream_invalidate(&buffer->stream, bytes); } -static inline void buffer_stream_writeback(struct comp_buffer __sparse_cache *buffer, +static inline void buffer_stream_writeback(struct comp_buffer *buffer, uint32_t bytes) { if (buffer->is_shared) @@ -223,13 +223,13 @@ static inline void buffer_stream_writeback(struct comp_buffer __sparse_cache *bu } /* stubs for acquire/release for compilation, to be removed at last step */ -__must_check static inline struct comp_buffer __sparse_cache *buffer_acquire( +__must_check static inline struct comp_buffer *buffer_acquire( struct comp_buffer *buffer) { - return (struct comp_buffer __sparse_cache *)buffer; + return (struct comp_buffer *)buffer; } -static inline void buffer_release(struct comp_buffer __sparse_cache *buffer) +static inline void buffer_release(struct comp_buffer *buffer) { (void)buffer; } @@ -252,14 +252,14 @@ void buffer_detach(struct comp_buffer *buffer, struct list_item *head, int dir); static inline struct comp_dev *buffer_get_comp(struct comp_buffer *buffer, int dir) { - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(buffer); + struct comp_buffer *buffer_c = buffer_acquire(buffer); struct comp_dev *comp = dir == PPL_DIR_DOWNSTREAM ? buffer_c->sink : buffer_c->source; buffer_release(buffer_c); return comp; } -static inline void buffer_reset_pos(struct comp_buffer __sparse_cache *buffer, void *data) +static inline void buffer_reset_pos(struct comp_buffer *buffer, void *data) { /* reset rw pointers and avail/free bytes counters */ audio_stream_reset(&buffer->stream); @@ -269,7 +269,7 @@ static inline void buffer_reset_pos(struct comp_buffer __sparse_cache *buffer, v } /* Run-time buffer re-configuration calls this too, so it must use cached access */ -static inline void buffer_init(struct comp_buffer __sparse_cache *buffer, +static inline void buffer_init(struct comp_buffer *buffer, uint32_t size, uint32_t caps) { buffer->caps = caps; @@ -278,7 +278,7 @@ static inline void buffer_init(struct comp_buffer __sparse_cache *buffer, audio_stream_init(&buffer->stream, buffer->stream.addr, size); } -static inline void buffer_reset_params(struct comp_buffer __sparse_cache *buffer, void *data) +static inline void buffer_reset_params(struct comp_buffer *buffer, void *data) { buffer->hw_params_configured = false; } diff --git a/src/include/sof/audio/component.h b/src/include/sof/audio/component.h index f860cdcb3d90..b4a61d765e21 100644 --- a/src/include/sof/audio/component.h +++ b/src/include/sof/audio/component.h @@ -817,7 +817,7 @@ static inline void component_set_nearest_period_frames(struct comp_dev *current, * @param copy_bytes Requested size of data to be available. */ static inline void comp_underrun(struct comp_dev *dev, - struct comp_buffer __sparse_cache *source, + struct comp_buffer *source, uint32_t copy_bytes) { LOG_MODULE_DECLARE(component, CONFIG_SOF_LOG_LEVEL); @@ -839,7 +839,7 @@ static inline void comp_underrun(struct comp_dev *dev, * @param sink Sink buffer. * @param copy_bytes Requested size of free space to be available. */ -static inline void comp_overrun(struct comp_dev *dev, struct comp_buffer __sparse_cache *sink, +static inline void comp_overrun(struct comp_dev *dev, struct comp_buffer *sink, uint32_t copy_bytes) { LOG_MODULE_DECLARE(component, CONFIG_SOF_LOG_LEVEL); @@ -864,8 +864,8 @@ static inline void comp_overrun(struct comp_dev *dev, struct comp_buffer __spars * @param[in] sink Sink buffer. * @param[out] cl Current copy limits. */ -void comp_get_copy_limits(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +void comp_get_copy_limits(struct comp_buffer *source, + struct comp_buffer *sink, struct comp_copy_limits *cl); /** @@ -877,8 +877,8 @@ void comp_get_copy_limits(struct comp_buffer __sparse_cache *source, * @param[in] sink Buffer of sink. * @param[out] cl Current copy limits. */ -void comp_get_copy_limits_frame_aligned(const struct comp_buffer __sparse_cache *source, - const struct comp_buffer __sparse_cache *sink, +void comp_get_copy_limits_frame_aligned(const struct comp_buffer *source, + const struct comp_buffer *sink, struct comp_copy_limits *cl); /** @@ -894,7 +894,7 @@ void comp_get_copy_limits_with_lock(struct comp_buffer *source, struct comp_buffer *sink, struct comp_copy_limits *cl) { - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; source_c = buffer_acquire(source); sink_c = buffer_acquire(sink); @@ -919,7 +919,7 @@ void comp_get_copy_limits_with_lock_frame_aligned(struct comp_buffer *source, struct comp_buffer *sink, struct comp_copy_limits *cl) { - struct comp_buffer __sparse_cache *source_c, *sink_c; + struct comp_buffer *source_c, *sink_c; source_c = buffer_acquire(source); sink_c = buffer_acquire(sink); diff --git a/src/include/sof/audio/dcblock/dcblock.h b/src/include/sof/audio/dcblock/dcblock.h index 35019364b33f..2d0d46317e5c 100644 --- a/src/include/sof/audio/dcblock/dcblock.h +++ b/src/include/sof/audio/dcblock/dcblock.h @@ -43,8 +43,8 @@ struct dcblock_state { struct comp_data; typedef void (*dcblock_func)(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames); /* DC Blocking Filter component private data */ diff --git a/src/include/sof/audio/drc/drc.h b/src/include/sof/audio/drc/drc.h index ee480e1d8692..e7571096313e 100644 --- a/src/include/sof/audio/drc/drc.h +++ b/src/include/sof/audio/drc/drc.h @@ -60,8 +60,8 @@ struct drc_state { }; typedef void (*drc_func)(struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames); /* DRC component private data */ @@ -83,8 +83,8 @@ extern const struct drc_proc_fnmap drc_proc_fnmap[]; extern const size_t drc_proc_fncount; void drc_default_pass(struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, uint32_t frames); + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames); /** * \brief Returns DRC processing function. */ diff --git a/src/include/sof/audio/igo_nr/igo_nr_comp.h b/src/include/sof/audio/igo_nr/igo_nr_comp.h index 50159adb1a08..64c663a99944 100644 --- a/src/include/sof/audio/igo_nr/igo_nr_comp.h +++ b/src/include/sof/audio/igo_nr/igo_nr_comp.h @@ -41,8 +41,8 @@ struct comp_data { int32_t sink_frames_max; /* Max # of frames to process at sink */ int32_t frames; /* IO buffer length */ void (*igo_nr_func)(struct comp_data *cd, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, int32_t frames); }; diff --git a/src/include/sof/audio/mixer.h b/src/include/sof/audio/mixer.h index 2dd5e3c80885..118aaf1c914f 100644 --- a/src/include/sof/audio/mixer.h +++ b/src/include/sof/audio/mixer.h @@ -34,16 +34,16 @@ void sys_comp_module_mixer_interface_init(void); /* mixer component private data */ struct mixer_data { - void (*mix_func)(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t count, + void (*mix_func)(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t count, uint32_t frames); }; /** * \brief mixer processing function interface */ -typedef void (*mixer_func)(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t num_sources, +typedef void (*mixer_func)(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t num_sources, uint32_t frames); /** \brief Volume processing functions map. */ @@ -64,7 +64,7 @@ extern const size_t mixer_func_count; * \param[in] sinkb Sink buffer to match against */ static inline mixer_func mixer_get_processing_function(struct comp_dev *dev, - struct comp_buffer __sparse_cache *sinkb) + struct comp_buffer *sinkb) { int i; diff --git a/src/include/sof/audio/module_adapter/module/generic.h b/src/include/sof/audio/module_adapter/module/generic.h index 28f5c0765d0c..038f774df3eb 100644 --- a/src/include/sof/audio/module_adapter/module/generic.h +++ b/src/include/sof/audio/module_adapter/module/generic.h @@ -245,14 +245,14 @@ void *module_allocate_memory(struct processing_module *mod, uint32_t size, uint3 int module_free_memory(struct processing_module *mod, void *ptr); void module_free_all_memory(struct processing_module *mod); int module_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); static inline bool module_is_ready_to_process(struct processing_module *mod, - struct sof_source __sparse_cache **sources, + struct sof_source **sources, int num_of_sources, - struct sof_sink __sparse_cache **sinks, + struct sof_sink **sinks, int num_of_sinks) { struct module_data *md = &mod->priv; @@ -272,8 +272,8 @@ bool module_is_ready_to_process(struct processing_module *mod, } int module_process_sink_src(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 module_process_legacy(struct processing_module *mod, struct input_stream_buffer *input_buffers, int num_input_buffers, struct output_stream_buffer *output_buffers, @@ -319,8 +319,8 @@ static inline void module_update_buffer_position(struct input_stream_buffer *inp struct output_stream_buffer *output_buffers, uint32_t frames) { - struct audio_stream __sparse_cache *source = input_buffers->data; - struct audio_stream __sparse_cache *sink = output_buffers->data; + struct audio_stream *source = input_buffers->data; + struct audio_stream *sink = output_buffers->data; input_buffers->consumed += audio_stream_frame_bytes(source) * frames; output_buffers->size += audio_stream_frame_bytes(sink) * frames; diff --git a/src/include/sof/audio/module_adapter/module/module_interface.h b/src/include/sof/audio/module_adapter/module/module_interface.h index e4b7ddcfb7d4..5a4e27c354f7 100644 --- a/src/include/sof/audio/module_adapter/module/module_interface.h +++ b/src/include/sof/audio/module_adapter/module/module_interface.h @@ -158,8 +158,8 @@ struct module_interface { * component preparation in .prepare() */ int (*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); /** * (optional) return true if the module is ready to process @@ -177,8 +177,8 @@ struct module_interface { * the module */ bool (*is_ready_to_process)(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); /** * Module specific processing procedure @@ -200,8 +200,8 @@ struct module_interface { * - sinks are handlers to sink API struct sink*[] */ int (*process)(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); /** * process_audio_stream (depreciated) diff --git a/src/include/sof/audio/multiband_drc/multiband_drc.h b/src/include/sof/audio/multiband_drc/multiband_drc.h index a5058bd158dc..2c331215a8ee 100644 --- a/src/include/sof/audio/multiband_drc/multiband_drc.h +++ b/src/include/sof/audio/multiband_drc/multiband_drc.h @@ -28,8 +28,8 @@ struct multiband_drc_state { }; typedef void (*multiband_drc_func)(const struct processing_module *mod, - const struct audio_stream __sparse_cache *source, - struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + struct audio_stream *sink, uint32_t frames); /* Multiband DRC component private data */ diff --git a/src/include/sof/audio/mux.h b/src/include/sof/audio/mux.h index 2c42c2506edd..e8eb9bc9de54 100644 --- a/src/include/sof/audio/mux.h +++ b/src/include/sof/audio/mux.h @@ -65,11 +65,11 @@ struct mux_stream_data { uint8_t reserved2[3]; // padding to ensure proper alignment of following instances } __attribute__((packed, aligned(4))); -typedef void(*demux_func)(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames, +typedef void(*demux_func)(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames, struct mux_look_up *look_up); -typedef void(*mux_func)(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache **sources, uint32_t frames, +typedef void(*mux_func)(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream **sources, uint32_t frames, struct mux_look_up *look_up); /** diff --git a/src/include/sof/audio/pcm_converter.h b/src/include/sof/audio/pcm_converter.h index 09d20f5568b6..a06dc79f632a 100644 --- a/src/include/sof/audio/pcm_converter.h +++ b/src/include/sof/audio/pcm_converter.h @@ -45,8 +45,8 @@ struct audio_stream; * \param samples number of samples to convert * \return error code or number of processed samples. */ -typedef int (*pcm_converter_func)(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +typedef int (*pcm_converter_func)(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t samples); /** @@ -156,8 +156,8 @@ pcm_get_conversion_vc_function(enum sof_ipc_frame in_bits, * \param converter core conversion function working on linear memory regions * \return error code or number of processed samples */ -int pcm_convert_as_linear(const struct audio_stream __sparse_cache *source, uint32_t ioffset, - struct audio_stream __sparse_cache *sink, uint32_t ooffset, +int pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset, + struct audio_stream *sink, uint32_t ooffset, uint32_t samples, pcm_converter_lin_func converter); #endif /* __SOF_AUDIO_PCM_CONVERTER_H__ */ diff --git a/src/include/sof/audio/pipeline.h b/src/include/sof/audio/pipeline.h index d80e4380aa82..8dc5ea16d51d 100644 --- a/src/include/sof/audio/pipeline.h +++ b/src/include/sof/audio/pipeline.h @@ -100,7 +100,7 @@ struct pipeline_walk_context { int (*comp_func)(struct comp_dev *cd, struct comp_buffer *buffer, struct pipeline_walk_context *ctx, int dir); void *comp_data; - void (*buff_func)(struct comp_buffer __sparse_cache *buffer, void *data); + void (*buff_func)(struct comp_buffer *buffer, void *data); void *buff_data; struct comp_buffer *incoming; /**< pipelines to be scheduled after trigger walk */ diff --git a/src/include/sof/audio/selector.h b/src/include/sof/audio/selector.h index e07576430322..cd4aa926c7f4 100644 --- a/src/include/sof/audio/selector.h +++ b/src/include/sof/audio/selector.h @@ -100,8 +100,8 @@ struct sof_selector_avs_ipc4_config { }; #else -typedef void (*sel_func)(struct comp_dev *dev, struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *source, uint32_t frames); +typedef void (*sel_func)(struct comp_dev *dev, struct audio_stream *sink, + const struct audio_stream *source, uint32_t frames); #endif /** \brief Selector component private data. */ diff --git a/src/include/sof/audio/sink_api.h b/src/include/sof/audio/sink_api.h index 07172f332555..83aeae514317 100644 --- a/src/include/sof/audio/sink_api.h +++ b/src/include/sof/audio/sink_api.h @@ -54,13 +54,13 @@ struct sof_ipc_stream_params; * Retrieves size of free space available in sink (in bytes) * return number of free bytes in buffer available to immediate filling */ -size_t sink_get_free_size(struct sof_sink __sparse_cache *sink); +size_t sink_get_free_size(struct sof_sink *sink); /** * Retrieves size of free space available in sink (in frames) * return number of free bytes in buffer available to immediate filling */ -size_t sink_get_free_frames(struct sof_sink __sparse_cache *sink); +size_t sink_get_free_frames(struct sof_sink *sink); /** * Get a circular buffer to operate on (to write). @@ -81,7 +81,7 @@ size_t sink_get_free_frames(struct sof_sink __sparse_cache *sink); * @retval -ENODATA if req_size is bigger than free space * */ -int sink_get_buffer(struct sof_sink __sparse_cache *sink, size_t req_size, +int sink_get_buffer(struct sof_sink *sink, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size); /** @@ -96,7 +96,7 @@ int sink_get_buffer(struct sof_sink __sparse_cache *sink, size_t req_size, * commit_buffer with commit_size==MAXINT * @return proper error code (0 on success) */ -int sink_commit_buffer(struct sof_sink __sparse_cache *sink, size_t commit_size); +int sink_commit_buffer(struct sof_sink *sink, size_t commit_size); /** * Get total number of bytes processed by the sink (meaning - committed by sink_commit_buffer()) @@ -104,33 +104,33 @@ int sink_commit_buffer(struct sof_sink __sparse_cache *sink, size_t commit_size) * @param sink a handler to sink * @return total number of processed data */ -size_t sink_get_num_of_processed_bytes(struct sof_sink __sparse_cache *sink); +size_t sink_get_num_of_processed_bytes(struct sof_sink *sink); /** * sets counter of total number of bytes processed to zero */ -void sink_reset_num_of_processed_bytes(struct sof_sink __sparse_cache *sink); +void sink_reset_num_of_processed_bytes(struct sof_sink *sink); /** get size of a single audio frame (in bytes) */ -size_t sink_get_frame_bytes(struct sof_sink __sparse_cache *sink); +size_t sink_get_frame_bytes(struct sof_sink *sink); /** set of functions for retrieve audio parameters */ -enum sof_ipc_frame sink_get_frm_fmt(struct sof_sink __sparse_cache *sink); -enum sof_ipc_frame sink_get_valid_fmt(struct sof_sink __sparse_cache *sink); -uint32_t sink_get_rate(struct sof_sink __sparse_cache *sink); -uint32_t sink_get_channels(struct sof_sink __sparse_cache *sink); -uint32_t sink_get_buffer_fmt(struct sof_sink __sparse_cache *sink); -bool sink_get_overrun(struct sof_sink __sparse_cache *sink); +enum sof_ipc_frame sink_get_frm_fmt(struct sof_sink *sink); +enum sof_ipc_frame sink_get_valid_fmt(struct sof_sink *sink); +uint32_t sink_get_rate(struct sof_sink *sink); +uint32_t sink_get_channels(struct sof_sink *sink); +uint32_t sink_get_buffer_fmt(struct sof_sink *sink); +bool sink_get_overrun(struct sof_sink *sink); /** set of functions for setting audio parameters */ -int sink_set_frm_fmt(struct sof_sink __sparse_cache *sink, enum sof_ipc_frame frame_fmt); -int sink_set_valid_fmt(struct sof_sink __sparse_cache *sink, enum sof_ipc_frame valid_sample_fmt); -int sink_set_rate(struct sof_sink __sparse_cache *sink, unsigned int rate); -int sink_set_channels(struct sof_sink __sparse_cache *sink, unsigned int channels); -int sink_set_overrun(struct sof_sink __sparse_cache *sink, bool overrun_permitted); -int sink_set_buffer_fmt(struct sof_sink __sparse_cache *sink, uint32_t buffer_fmt); -void sink_set_obs(struct sof_sink __sparse_cache *sink, size_t obs); -size_t sink_get_obs(struct sof_sink __sparse_cache *sink); +int sink_set_frm_fmt(struct sof_sink *sink, enum sof_ipc_frame frame_fmt); +int sink_set_valid_fmt(struct sof_sink *sink, enum sof_ipc_frame valid_sample_fmt); +int sink_set_rate(struct sof_sink *sink, unsigned int rate); +int sink_set_channels(struct sof_sink *sink, unsigned int channels); +int sink_set_overrun(struct sof_sink *sink, bool overrun_permitted); +int sink_set_buffer_fmt(struct sof_sink *sink, uint32_t buffer_fmt); +void sink_set_obs(struct sof_sink *sink, size_t obs); +size_t sink_get_obs(struct sof_sink *sink); /** * initial set of audio parameters, provided in sof_ipc_stream_params @@ -140,7 +140,7 @@ size_t sink_get_obs(struct sof_sink __sparse_cache *sink); * @param force_update tells the implementation that the params should override actual settings * @return 0 if success */ -int sink_set_params(struct sof_sink __sparse_cache *sink, +int sink_set_params(struct sof_sink *sink, struct sof_ipc_stream_params *params, bool force_update); /** @@ -158,7 +158,7 @@ int sink_set_params(struct sof_sink __sparse_cache *sink, * * @return 0 if success */ -int sink_set_alignment_constants(struct sof_sink __sparse_cache *sink, +int sink_set_alignment_constants(struct sof_sink *sink, const uint32_t byte_align, const uint32_t frame_align_req); diff --git a/src/include/sof/audio/sink_api_implementation.h b/src/include/sof/audio/sink_api_implementation.h index 15db32152039..71c694d8a774 100644 --- a/src/include/sof/audio/sink_api_implementation.h +++ b/src/include/sof/audio/sink_api_implementation.h @@ -27,18 +27,18 @@ struct sink_ops { /** * see comment of sink_get_free_size() */ - size_t (*get_free_size)(struct sof_sink __sparse_cache *sink); + size_t (*get_free_size)(struct sof_sink *sink); /** * see comment of sink_get_buffer() */ - int (*get_buffer)(struct sof_sink __sparse_cache *sink, size_t req_size, + int (*get_buffer)(struct sof_sink *sink, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size); /** * see comment of sink_commit_buffer() */ - int (*commit_buffer)(struct sof_sink __sparse_cache *sink, size_t commit_size); + int (*commit_buffer)(struct sof_sink *sink, size_t commit_size); /** * OPTIONAL: Notification to the sink implementation about changes in audio format @@ -49,20 +49,20 @@ struct sink_ops { * * @retval 0 if success, negative if new parameters are not supported */ - int (*on_audio_format_set)(struct sof_sink __sparse_cache *sink); + int (*on_audio_format_set)(struct sof_sink *sink); /** * OPTIONAL * see sink_set_params comments */ - int (*audio_set_ipc_params)(struct sof_sink __sparse_cache *sink, + int (*audio_set_ipc_params)(struct sof_sink *sink, struct sof_ipc_stream_params *params, bool force_update); /** * OPTIONAL * see comment for sink_set_alignment_constants */ - int (*set_alignment_constants)(struct sof_sink __sparse_cache *sink, + int (*set_alignment_constants)(struct sof_sink *sink, const uint32_t byte_align, const uint32_t frame_align_req); }; @@ -86,7 +86,7 @@ struct sof_sink { * the implementation must ensure coherent access to the parameteres * in case of i.e. cross core shared queue, it must be located in non-cached memory */ -void sink_init(struct sof_sink __sparse_cache *sink, const struct sink_ops *ops, +void sink_init(struct sof_sink *sink, const struct sink_ops *ops, struct sof_audio_stream_params *audio_stream_params); #endif /* __SOF_SINK_API_IMPLEMENTATION_H__ */ diff --git a/src/include/sof/audio/sink_source_utils.h b/src/include/sof/audio/sink_source_utils.h index 8f550b1d2276..ca8752f07294 100644 --- a/src/include/sof/audio/sink_source_utils.h +++ b/src/include/sof/audio/sink_source_utils.h @@ -19,7 +19,7 @@ * if false, data will remain in the source * @param size number of bytes to be copied */ -int source_to_sink_copy(struct sof_source __sparse_cache *source, - struct sof_sink __sparse_cache *sink, bool free, size_t size); +int source_to_sink_copy(struct sof_source *source, + struct sof_sink *sink, bool free, size_t size); #endif /* SINK_SOURCE_UTILS_H */ diff --git a/src/include/sof/audio/smart_amp/smart_amp.h b/src/include/sof/audio/smart_amp/smart_amp.h index 6438d7f082bc..d19f2e00e627 100644 --- a/src/include/sof/audio/smart_amp/smart_amp.h +++ b/src/include/sof/audio/smart_amp/smart_amp.h @@ -136,9 +136,9 @@ struct smart_amp_mod_struct_t { }; typedef void (*smart_amp_func)(const struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, - const struct audio_stream __sparse_cache *feedback, + const struct audio_stream *source, + const struct audio_stream *sink, + const struct audio_stream *feedback, uint32_t frames); struct smart_amp_func_map { uint16_t frame_fmt; @@ -151,14 +151,14 @@ int smart_amp_init(struct smart_amp_mod_struct_t *hspk, struct comp_dev *dev); int smart_amp_flush(struct smart_amp_mod_struct_t *hspk, struct comp_dev *dev); /* Feed forward processing function */ int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, int8_t *chan_map, + const struct audio_stream *source, + const struct audio_stream *sink, int8_t *chan_map, struct smart_amp_mod_struct_t *hspk, uint32_t num_ch_in, uint32_t num_ch_out); /* Feedback processing function */ int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, int8_t *chan_map, + const struct audio_stream *source, + const struct audio_stream *sink, int8_t *chan_map, struct smart_amp_mod_struct_t *hspk, uint32_t num_ch); /* memory usage calculation for the component */ diff --git a/src/include/sof/audio/source_api.h b/src/include/sof/audio/source_api.h index bfbcf530a81b..785b15167b17 100644 --- a/src/include/sof/audio/source_api.h +++ b/src/include/sof/audio/source_api.h @@ -54,13 +54,13 @@ struct sof_ipc_stream_params; * Retrieves size of available data (in bytes) * return number of bytes that are available for immediate use */ -size_t source_get_data_available(struct sof_source __sparse_cache *source); +size_t source_get_data_available(struct sof_source *source); /** * Retrieves size of available data (in frames) * return number of bytes that are available for immediate use */ -size_t source_get_data_frames_available(struct sof_source __sparse_cache *source); +size_t source_get_data_frames_available(struct sof_source *source); /** * Retrieves a fragment of circular data to be used by the caller (to read) @@ -93,7 +93,7 @@ size_t source_get_data_frames_available(struct sof_source __sparse_cache *source * * @retval -ENODATA if req_size is bigger than available data */ -int source_get_data(struct sof_source __sparse_cache *source, size_t req_size, +int source_get_data(struct sof_source *source, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size); /** @@ -110,38 +110,38 @@ int source_get_data(struct sof_source __sparse_cache *source, size_t req_size, * * @return proper error code (0 on success) */ -int source_release_data(struct sof_source __sparse_cache *source, size_t free_size); +int source_release_data(struct sof_source *source, size_t free_size); /** * Get total number of bytes processed by the source (meaning - freed by source_release_data()) */ -size_t source_get_num_of_processed_bytes(struct sof_source __sparse_cache *source); +size_t source_get_num_of_processed_bytes(struct sof_source *source); /** * sets counter of total number of bytes processed to zero */ -void source_reset_num_of_processed_bytes(struct sof_source __sparse_cache *source); +void source_reset_num_of_processed_bytes(struct sof_source *source); /** get size of a single audio frame (in bytes) */ -size_t source_get_frame_bytes(struct sof_source __sparse_cache *source); +size_t source_get_frame_bytes(struct sof_source *source); /** set of functions for retrieve audio parameters */ -enum sof_ipc_frame source_get_frm_fmt(struct sof_source __sparse_cache *source); -enum sof_ipc_frame source_get_valid_fmt(struct sof_source __sparse_cache *source); -unsigned int source_get_rate(struct sof_source __sparse_cache *source); -unsigned int source_get_channels(struct sof_source __sparse_cache *source); -uint32_t source_get_buffer_fmt(struct sof_source __sparse_cache *source); -bool source_get_underrun(struct sof_source __sparse_cache *source); +enum sof_ipc_frame source_get_frm_fmt(struct sof_source *source); +enum sof_ipc_frame source_get_valid_fmt(struct sof_source *source); +unsigned int source_get_rate(struct sof_source *source); +unsigned int source_get_channels(struct sof_source *source); +uint32_t source_get_buffer_fmt(struct sof_source *source); +bool source_get_underrun(struct sof_source *source); /** set of functions for setting audio parameters */ -int source_set_valid_fmt(struct sof_source __sparse_cache *source, +int source_set_valid_fmt(struct sof_source *source, enum sof_ipc_frame valid_sample_fmt); -int source_set_rate(struct sof_source __sparse_cache *source, unsigned int rate); -int source_set_channels(struct sof_source __sparse_cache *source, unsigned int channels); -int source_set_underrun(struct sof_source __sparse_cache *source, bool underrun_permitted); -int source_set_buffer_fmt(struct sof_source __sparse_cache *source, uint32_t buffer_fmt); -void source_set_ibs(struct sof_source __sparse_cache *source, size_t ibs); -size_t source_get_ibs(struct sof_source __sparse_cache *source); +int source_set_rate(struct sof_source *source, unsigned int rate); +int source_set_channels(struct sof_source *source, unsigned int channels); +int source_set_underrun(struct sof_source *source, bool underrun_permitted); +int source_set_buffer_fmt(struct sof_source *source, uint32_t buffer_fmt); +void source_set_ibs(struct sof_source *source, size_t ibs); +size_t source_get_ibs(struct sof_source *source); /** * initial set of audio parameters, provided in sof_ipc_stream_params @@ -151,7 +151,7 @@ size_t source_get_ibs(struct sof_source __sparse_cache *source); * @param force_update tells the implementation that the params should override actual settings * @return 0 if success */ -int source_set_params(struct sof_source __sparse_cache *source, +int source_set_params(struct sof_source *source, struct sof_ipc_stream_params *params, bool force_update); /** @@ -169,7 +169,7 @@ int source_set_params(struct sof_source __sparse_cache *source, * * @return 0 if success */ -int source_set_alignment_constants(struct sof_source __sparse_cache *source, +int source_set_alignment_constants(struct sof_source *source, const uint32_t byte_align, const uint32_t frame_align_req); diff --git a/src/include/sof/audio/source_api_implementation.h b/src/include/sof/audio/source_api_implementation.h index c75330cde019..4977c81e86d2 100644 --- a/src/include/sof/audio/source_api_implementation.h +++ b/src/include/sof/audio/source_api_implementation.h @@ -27,18 +27,18 @@ struct source_ops { /** * see comment of source_get_data_available() */ - size_t (*get_data_available)(struct sof_source __sparse_cache *source); + size_t (*get_data_available)(struct sof_source *source); /** * see comment of source_get_data_available() */ - int (*get_data)(struct sof_source __sparse_cache *source, size_t req_size, + int (*get_data)(struct sof_source *source, size_t req_size, void **data_ptr, void **buffer_start, size_t *buffer_size); /** * see comment of source_release_data() */ - int (*release_data)(struct sof_source __sparse_cache *source, size_t free_size); + int (*release_data)(struct sof_source *source, size_t free_size); /** * OPTIONAL: Notification to the source implementation about changes in audio format @@ -49,20 +49,20 @@ struct source_ops { * * @retval 0 if success, negative if new parameteres are not supported */ - int (*on_audio_format_set)(struct sof_source __sparse_cache *source); + int (*on_audio_format_set)(struct sof_source *source); /** * OPTIONAL * see source_set_params comments */ - int (*audio_set_ipc_params)(struct sof_source __sparse_cache *source, + int (*audio_set_ipc_params)(struct sof_source *source, struct sof_ipc_stream_params *params, bool force_update); /** * OPTIONAL * see comment for source_set_alignment_constants */ - int (*set_alignment_constants)(struct sof_source __sparse_cache *source, + int (*set_alignment_constants)(struct sof_source *source, const uint32_t byte_align, const uint32_t frame_align_req); }; @@ -86,7 +86,7 @@ struct sof_source { * the implementation must ensure coherent access to the parameteres * in case of i.e. cross core shared queue, it must be located in non-cached memory */ -void source_init(struct sof_source __sparse_cache *source, const struct source_ops *ops, +void source_init(struct sof_source *source, const struct source_ops *ops, struct sof_audio_stream_params *audio_stream_params); #endif /* __SOF_SOURCE_API_IMPLEMENTATION_H__ */ diff --git a/src/include/sof/math/fir_hifi3.h b/src/include/sof/math/fir_hifi3.h index a3d4b19ce0ce..aa1102ef2a1f 100644 --- a/src/include/sof/math/fir_hifi3.h +++ b/src/include/sof/math/fir_hifi3.h @@ -48,7 +48,7 @@ static inline void fir_core_setup_circular(struct fir_state_32x16 *fir) } /* Setup circular for component buffer */ -static inline void fir_comp_setup_circular(const struct audio_stream __sparse_cache *buffer) +static inline void fir_comp_setup_circular(const struct audio_stream *buffer) { AE_SETCBEGIN0(audio_stream_get_addr(buffer)); AE_SETCEND0(audio_stream_get_end_addr(buffer)); diff --git a/src/ipc/ipc-helper.c b/src/ipc/ipc-helper.c index ed6d742dc235..331822c3795a 100644 --- a/src/ipc/ipc-helper.c +++ b/src/ipc/ipc-helper.c @@ -80,7 +80,7 @@ int32_t ipc_comp_pipe_id(const struct ipc_comp_dev *icd) */ static void comp_update_params(uint32_t flag, struct sof_ipc_stream_params *params, - struct comp_buffer __sparse_cache *buffer) + struct comp_buffer *buffer) { if (flag & BUFF_PARAMS_FRAME_FMT) params->frame_fmt = audio_stream_get_frm_fmt(&buffer->stream); @@ -104,7 +104,7 @@ int comp_verify_params(struct comp_dev *dev, uint32_t flag, struct list_item *clist; struct comp_buffer *sinkb; struct comp_buffer *buf; - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; int dir = dev->direction; if (!params) { @@ -291,7 +291,7 @@ int ipc_comp_free(struct ipc *ipc, uint32_t comp_id) irq_local_disable(flags); list_for_item_safe(clist, tmp, &icd->cd->bsource_list) { struct comp_buffer *buffer = container_of(clist, struct comp_buffer, sink_list); - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(buffer); + struct comp_buffer *buffer_c = buffer_acquire(buffer); buffer_c->sink = NULL; buffer_release(buffer_c); @@ -304,7 +304,7 @@ int ipc_comp_free(struct ipc *ipc, uint32_t comp_id) list_for_item_safe(clist, tmp, &icd->cd->bsink_list) { struct comp_buffer *buffer = container_of(clist, struct comp_buffer, source_list); - struct comp_buffer __sparse_cache *buffer_c = buffer_acquire(buffer); + struct comp_buffer *buffer_c = buffer_acquire(buffer); buffer_c->source = NULL; buffer_release(buffer_c); diff --git a/src/ipc/ipc3/dai.c b/src/ipc/ipc3/dai.c index decee57299b9..ccc41524e002 100644 --- a/src/ipc/ipc3/dai.c +++ b/src/ipc/ipc3/dai.c @@ -103,7 +103,7 @@ int ipc_dai_data_config(struct dai_data *dd, struct comp_dev *dev) { struct ipc_config_dai *dai = &dd->ipc_config; struct sof_ipc_dai_config *config = ipc_from_dai_config(dd->dai_spec_config); - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; if (!config) { comp_err(dev, "dai_data_config(): no config set for dai %d type %d", diff --git a/src/ipc/ipc3/helper.c b/src/ipc/ipc3/helper.c index 2bd55ec22717..9a9bbf0ac21d 100644 --- a/src/ipc/ipc3/helper.c +++ b/src/ipc/ipc3/helper.c @@ -487,7 +487,7 @@ int ipc_buffer_free(struct ipc *ipc, uint32_t buffer_id) unsigned int core; bool sink_active = false; bool source_active = false; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; /* check whether buffer exists */ ibd = ipc_get_buffer_by_id(ipc, buffer_id); diff --git a/src/ipc/ipc4/helper.c b/src/ipc/ipc4/helper.c index 1d2838c1ed57..206773d23c0c 100644 --- a/src/ipc/ipc4/helper.c +++ b/src/ipc/ipc4/helper.c @@ -241,7 +241,7 @@ static int ipc_pipeline_module_free(uint32_t pipeline_id) /* free sink buffer allocated by current component in bind function */ list_for_item_safe(list, _list, &icd->cd->bsink_list) { - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct comp_dev *sink; buffer = container_of(list, struct comp_buffer, source_list); @@ -257,7 +257,7 @@ static int ipc_pipeline_module_free(uint32_t pipeline_id) /* free source buffer allocated by current component in bind function */ list_for_item_safe(list, _list, &icd->cd->bsource_list) { - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct comp_dev *source; buffer = container_of(list, struct comp_buffer, sink_list); @@ -337,7 +337,7 @@ int ipc_comp_connect(struct ipc *ipc, ipc_pipe_comp_connect *_connect) { struct ipc4_module_bind_unbind *bu; struct comp_buffer *buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct comp_dev *source; struct comp_dev *sink; struct ipc4_base_module_cfg source_src_cfg; @@ -490,7 +490,7 @@ int ipc_comp_disconnect(struct ipc *ipc, ipc_pipe_comp_connect *_connect) buffer_id = IPC4_COMP_ID(bu->extension.r.src_queue, bu->extension.r.dst_queue); list_for_item(sink_list, &src->bsink_list) { struct comp_buffer *buf = container_of(sink_list, struct comp_buffer, source_list); - struct comp_buffer __sparse_cache *buf_c = buffer_acquire(buf); + struct comp_buffer *buf_c = buffer_acquire(buf); bool found = buf_c->id == buffer_id; buffer_release(buf_c); @@ -841,7 +841,7 @@ void ipc4_base_module_cfg_to_stream_params(const struct ipc4_base_module_cfg *ba params->chmap[i] = (base_cfg->audio_fmt.ch_map >> i * 4) & 0xf; } -void ipc4_update_buffer_format(struct comp_buffer __sparse_cache *buf_c, +void ipc4_update_buffer_format(struct comp_buffer *buf_c, const struct ipc4_audio_format *fmt) { enum sof_ipc_frame valid_fmt, frame_fmt; diff --git a/src/lib/dma.c b/src/lib/dma.c index 6115cfba1e85..f124a040c4c3 100644 --- a/src/lib/dma.c +++ b/src/lib/dma.c @@ -321,11 +321,11 @@ void dma_sg_free(struct dma_sg_elem_array *elem_array) dma_sg_init(elem_array); } -int dma_buffer_copy_from(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +int dma_buffer_copy_from(struct comp_buffer *source, + struct comp_buffer *sink, dma_process_func process, uint32_t source_bytes) { - struct audio_stream __sparse_cache *istream = &source->stream; + struct audio_stream *istream = &source->stream; uint32_t samples = source_bytes / audio_stream_sample_bytes(istream); uint32_t sink_bytes = audio_stream_sample_bytes(&sink->stream) * @@ -350,11 +350,11 @@ int dma_buffer_copy_from(struct comp_buffer __sparse_cache *source, return ret; } -int dma_buffer_copy_to(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +int dma_buffer_copy_to(struct comp_buffer *source, + struct comp_buffer *sink, dma_process_func process, uint32_t sink_bytes) { - struct audio_stream __sparse_cache *ostream = &sink->stream; + struct audio_stream *ostream = &sink->stream; uint32_t samples = sink_bytes / audio_stream_sample_bytes(ostream); uint32_t source_bytes = audio_stream_sample_bytes(&source->stream) * @@ -379,11 +379,11 @@ int dma_buffer_copy_to(struct comp_buffer __sparse_cache *source, return ret; } -int dma_buffer_copy_from_no_consume(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +int dma_buffer_copy_from_no_consume(struct comp_buffer *source, + struct comp_buffer *sink, dma_process_func process, uint32_t source_bytes) { - struct audio_stream __sparse_cache *istream = &source->stream; + struct audio_stream *istream = &source->stream; uint32_t samples = source_bytes / audio_stream_sample_bytes(istream); uint32_t sink_bytes = audio_stream_sample_bytes(&sink->stream) * diff --git a/src/probe/probe.c b/src/probe/probe.c index 91344e08c56a..fe1b8ae9efa1 100644 --- a/src/probe/probe.c +++ b/src/probe/probe.c @@ -879,7 +879,7 @@ static void probe_cb_produce(void *arg, enum notify_id type, void *data) { struct probe_pdata *_probe = probe_get(); struct buffer_cb_transact *cb_data = data; - struct comp_buffer __sparse_cache *buffer = cb_data->buffer; + struct comp_buffer *buffer = cb_data->buffer; struct probe_dma_ext *dma; uint32_t buffer_id; uint32_t head, tail; @@ -1066,7 +1066,7 @@ static bool probe_purpose_needs_ext_dma(uint32_t purpose) static struct comp_buffer *ipc4_get_buffer(struct ipc_comp_dev *dev, probe_point_id_t probe_point) { struct comp_buffer *buf; - struct comp_buffer __sparse_cache *buf_c; + struct comp_buffer *buf_c; struct list_item *sink_list, *source_list; unsigned int queue_id; diff --git a/src/samples/audio/detect_test.c b/src/samples/audio/detect_test.c index 55058f6ca047..632ad4a6a00d 100644 --- a/src/samples/audio/detect_test.c +++ b/src/samples/audio/detect_test.c @@ -101,7 +101,7 @@ struct comp_data { struct ipc_msg *msg; /**< host notification */ void (*detect_func)(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, uint32_t frames); + const struct audio_stream *source, uint32_t frames); struct sof_ipc_comp_event event; #if CONFIG_AMS @@ -209,7 +209,7 @@ void detect_test_notify(const struct comp_dev *dev) } static void default_detect_test(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, uint32_t frames) { struct comp_data *cd = comp_get_drvdata(dev); @@ -797,7 +797,7 @@ static int test_keyword_params(struct comp_dev *dev, { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *sourceb; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; unsigned int channels, rate; enum sof_ipc_frame frame_fmt; int err; @@ -881,7 +881,7 @@ static int test_keyword_copy(struct comp_dev *dev) { struct comp_data *cd = comp_get_drvdata(dev); struct comp_buffer *source; - struct comp_buffer __sparse_cache *source_c; + struct comp_buffer *source_c; uint32_t frames; comp_dbg(dev, "test_keyword_copy()"); diff --git a/src/samples/audio/kwd_nn_detect_test.c b/src/samples/audio/kwd_nn_detect_test.c index df0a8c80af98..ef71a441580b 100644 --- a/src/samples/audio/kwd_nn_detect_test.c +++ b/src/samples/audio/kwd_nn_detect_test.c @@ -35,7 +35,7 @@ static int kwd_nn_detect_postprocess(uint8_t confidences[KWD_NN_CONFIDENCES_SIZE } void kwd_nn_detect_test(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, + const struct audio_stream *source, uint32_t frames) { void *src; diff --git a/src/samples/audio/smart_amp_test_ipc3.c b/src/samples/audio/smart_amp_test_ipc3.c index b2056c6cb7d0..dc58cd0727d9 100644 --- a/src/samples/audio/smart_amp_test_ipc3.c +++ b/src/samples/audio/smart_amp_test_ipc3.c @@ -28,8 +28,8 @@ DECLARE_TR_CTX(smart_amp_comp_tr, SOF_UUID(smart_amp_comp_uuid), LOG_LEVEL_INFO); typedef int(*smart_amp_proc)(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, uint32_t frames, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames, int8_t *chan_map); struct smart_amp_data { @@ -316,7 +316,7 @@ static int smart_amp_trigger(struct comp_dev *dev, int cmd) case COMP_TRIGGER_START: case COMP_TRIGGER_RELEASE: if (sad->feedback_buf) { - struct comp_buffer __sparse_cache *buf = buffer_acquire(sad->feedback_buf); + struct comp_buffer *buf = buffer_acquire(sad->feedback_buf); buffer_zero(buf); buffer_release(buf); } @@ -332,8 +332,8 @@ static int smart_amp_trigger(struct comp_dev *dev, int cmd) } static int smart_amp_process_s16(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames, int8_t *chan_map) { struct smart_amp_data *sad = comp_get_drvdata(dev); @@ -364,8 +364,8 @@ static int smart_amp_process_s16(struct comp_dev *dev, } static int smart_amp_process_s32(struct comp_dev *dev, - const struct audio_stream __sparse_cache *source, - const struct audio_stream __sparse_cache *sink, + const struct audio_stream *source, + const struct audio_stream *sink, uint32_t frames, int8_t *chan_map) { struct smart_amp_data *sad = comp_get_drvdata(dev); @@ -397,7 +397,7 @@ 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) + struct comp_buffer *buf) { switch (audio_stream_get_frm_fmt(&buf->stream)) { case SOF_IPC_FRAME_S16_LE: @@ -414,8 +414,8 @@ static smart_amp_proc get_smart_amp_process(struct comp_dev *dev, static int smart_amp_copy(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); - struct comp_buffer __sparse_cache *sink_buf = buffer_acquire(sad->sink_buf); + struct comp_buffer *source_buf = buffer_acquire(sad->source_buf); + struct comp_buffer *sink_buf = buffer_acquire(sad->sink_buf); uint32_t avail_passthrough_frames; uint32_t avail_feedback_frames; uint32_t avail_frames = 0; @@ -431,7 +431,7 @@ static int smart_amp_copy(struct comp_dev *dev) &sink_buf->stream); if (sad->feedback_buf) { - struct comp_buffer __sparse_cache *buf = buffer_acquire(sad->feedback_buf); + struct comp_buffer *buf = buffer_acquire(sad->feedback_buf); if (buf->source && comp_get_state(dev, buf->source) == dev->state) { /* feedback */ @@ -496,7 +496,7 @@ static int smart_amp_prepare(struct comp_dev *dev) { struct smart_amp_data *sad = comp_get_drvdata(dev); struct comp_buffer *source_buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct list_item *blist; int ret; @@ -535,7 +535,7 @@ static int smart_amp_prepare(struct comp_dev *dev) sad->in_channels = audio_stream_get_channels(&buffer_c->stream); if (sad->feedback_buf) { - struct comp_buffer __sparse_cache *buf = buffer_acquire(sad->feedback_buf); + struct comp_buffer *buf = buffer_acquire(sad->feedback_buf); audio_stream_set_channels(&buf->stream, sad->config.feedback_channels); audio_stream_set_rate(&buf->stream, audio_stream_get_rate(&buffer_c->stream)); diff --git a/src/samples/audio/smart_amp_test_ipc4.c b/src/samples/audio/smart_amp_test_ipc4.c index a256de9c3d28..802e54a39169 100644 --- a/src/samples/audio/smart_amp_test_ipc4.c +++ b/src/samples/audio/smart_amp_test_ipc4.c @@ -92,7 +92,7 @@ static void smart_amp_set_params(struct processing_module *mod) struct comp_dev *dev = mod->dev; struct smart_amp_data *sad = module_get_private_data(mod); struct comp_buffer *sink; - struct comp_buffer __sparse_cache *sink_c; + struct comp_buffer *sink_c; ipc4_base_module_cfg_to_stream_params(&mod->priv.cfg.base_cfg, params); @@ -196,8 +196,8 @@ static int smart_amp_process_s16(struct processing_module *mod, uint32_t frames, int8_t *chan_map) { struct smart_amp_data *sad = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int16_t *src; int16_t *dest; uint32_t in_frag = 0; @@ -229,8 +229,8 @@ static int smart_amp_process_s32(struct processing_module *mod, uint32_t frames, int8_t *chan_map) { struct smart_amp_data *sad = module_get_private_data(mod); - struct audio_stream __sparse_cache *source = bsource->data; - struct audio_stream __sparse_cache *sink = bsink->data; + struct audio_stream *source = bsource->data; + struct audio_stream *sink = bsink->data; int32_t *src; int32_t *dest; uint32_t in_frag = 0; @@ -258,7 +258,7 @@ static int smart_amp_process_s32(struct processing_module *mod, } static smart_amp_proc get_smart_amp_process(struct comp_dev *dev, - struct comp_buffer __sparse_cache *buf) + struct comp_buffer *buf) { switch (audio_stream_get_frm_fmt(&buf->stream)) { case SOF_IPC_FRAME_S16_LE: @@ -278,8 +278,8 @@ static int smart_amp_process(struct processing_module *mod, { struct smart_amp_data *sad = module_get_private_data(mod); struct comp_dev *dev = mod->dev; - struct comp_buffer __sparse_cache *fb_buf_c; - struct comp_buffer __sparse_cache *buf; + struct comp_buffer *fb_buf_c; + struct comp_buffer *buf; struct module_source_info __sparse_cache *mod_source_info; struct input_stream_buffer *fb_input = NULL; /* if there is only one input stream, it should be the source input */ @@ -294,7 +294,7 @@ static int smart_amp_process(struct processing_module *mod, if (num_input_buffers == SMART_AMP_NUM_IN_PINS) for (i = 0; i < num_input_buffers; i++) { buf = attr_container_of(input_buffers[i].data, - struct comp_buffer __sparse_cache, + struct comp_buffer, stream, __sparse_cache); if (IPC4_SINK_QUEUE_ID(buf->id) == SOF_SMART_AMP_FEEDBACK_QUEUE_ID) { @@ -345,14 +345,14 @@ static int smart_amp_reset(struct processing_module *mod) } static int smart_amp_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) { struct smart_amp_data *sad = module_get_private_data(mod); struct comp_dev *dev = mod->dev; struct comp_buffer *source_buffer; struct comp_buffer *sink_buffer; - struct comp_buffer __sparse_cache *buffer_c; + struct comp_buffer *buffer_c; struct list_item *blist; int ret; diff --git a/xtos/include/sof/lib/dma.h b/xtos/include/sof/lib/dma.h index 85080e70c18b..82f4992cbe88 100644 --- a/xtos/include/sof/lib/dma.h +++ b/xtos/include/sof/lib/dma.h @@ -254,8 +254,8 @@ struct dma_info { }; struct audio_stream; -typedef int (*dma_process_func)(const struct audio_stream __sparse_cache *source, - uint32_t ioffset, struct audio_stream __sparse_cache *sink, +typedef int (*dma_process_func)(const struct audio_stream *source, + uint32_t ioffset, struct audio_stream *sink, uint32_t ooffset, uint32_t frames); /** @@ -530,8 +530,8 @@ typedef void (*dma_process)(const struct audio_stream *, struct audio_stream *, uint32_t); /* copies data from DMA buffer using provided processing function */ -int dma_buffer_copy_from(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +int dma_buffer_copy_from(struct comp_buffer *source, + struct comp_buffer *sink, dma_process_func process, uint32_t source_bytes); /* @@ -539,13 +539,13 @@ int dma_buffer_copy_from(struct comp_buffer __sparse_cache *source, * conversion function. DMA buffer consume should be performed after the data has been copied * to all sinks. */ -int dma_buffer_copy_from_no_consume(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +int dma_buffer_copy_from_no_consume(struct comp_buffer *source, + struct comp_buffer *sink, dma_process_func process, uint32_t source_bytes); /* copies data to DMA buffer using provided processing function */ -int dma_buffer_copy_to(struct comp_buffer __sparse_cache *source, - struct comp_buffer __sparse_cache *sink, +int dma_buffer_copy_to(struct comp_buffer *source, + struct comp_buffer *sink, dma_process_func process, uint32_t sink_bytes); /* generic DMA DSP <-> Host copier */