mirror of https://github.com/thesofproject/sof.git
pcm_converter: add __sparse_cache annotations
PCM converter functions operate on locked and cached buffers, add __sparse_cache annotations to them. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
This commit is contained in:
Guennadi Liakhovetski
Liam Girdwood
parent
f8bbd24831
commit
95d6499ad6
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@ -10,12 +10,13 @@
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* \authors Karol Trzcinski <karolx.trzcinski@linux.intel.com>
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*/
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#include <sof/compiler_attributes.h>
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#include <sof/audio/audio_stream.h>
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#include <sof/audio/pcm_converter.h>
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#include <sof/debug/panic.h>
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int pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
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struct audio_stream *sink, uint32_t ooffset,
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int pcm_convert_as_linear(const struct audio_stream __sparse_cache *source, uint32_t ioffset,
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struct audio_stream __sparse_cache *sink, uint32_t ooffset,
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uint32_t samples, pcm_converter_lin_func converter)
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{
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const int s_size_in = audio_stream_sample_bytes(source);
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@ -20,6 +20,7 @@
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#include <sof/audio/format.h>
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#include <sof/bit.h>
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#include <sof/common.h>
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#include <sof/compiler_attributes.h>
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#include <ipc/stream.h>
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#include <stddef.h>
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@ -30,8 +31,8 @@
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#if CONFIG_PCM_CONVERTER_FORMAT_S16LE && CONFIG_PCM_CONVERTER_FORMAT_S24LE
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static int pcm_convert_s16_to_s24(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int16_t *src = source->r_ptr;
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@ -59,8 +60,8 @@ static int pcm_convert_s16_to_s24(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s24_to_s16(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -92,8 +93,8 @@ static int pcm_convert_s24_to_s16(const struct audio_stream *source,
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#if CONFIG_PCM_CONVERTER_FORMAT_S16LE && CONFIG_PCM_CONVERTER_FORMAT_S32LE
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static int pcm_convert_s16_to_s32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int16_t *src = source->r_ptr;
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@ -121,8 +122,8 @@ static int pcm_convert_s16_to_s32(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s32_to_s16(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s32_to_s16(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -154,8 +155,8 @@ static int pcm_convert_s32_to_s16(const struct audio_stream *source,
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#if CONFIG_PCM_CONVERTER_FORMAT_S24LE && CONFIG_PCM_CONVERTER_FORMAT_S32LE
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static int pcm_convert_s24_to_s32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_to_s32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -183,8 +184,8 @@ static int pcm_convert_s24_to_s32(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s32_to_s24(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s32_to_s24(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -212,8 +213,8 @@ static int pcm_convert_s32_to_s24(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s32_to_s24_be(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s32_to_s24_be(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -377,16 +378,16 @@ static void pcm_convert_f_to_s16_lin(const void *psrc, void *pdst,
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dst[i] = sat_int16(_pcm_convert_f_to_i(src[i], 15));
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}
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static int pcm_convert_s16_to_f(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_to_f(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
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pcm_convert_s16_to_f_lin);
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}
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static int pcm_convert_f_to_s16(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_f_to_s16(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
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@ -421,16 +422,16 @@ static void pcm_convert_f_to_s24_lin(const void *psrc, void *pdst,
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dst[i] = sat_int24(_pcm_convert_f_to_i(src[i], 23));
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}
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static int pcm_convert_s24_to_f(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_to_f(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
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pcm_convert_s24_to_f_lin);
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}
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static int pcm_convert_f_to_s24(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_f_to_s24(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
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@ -465,16 +466,16 @@ static void pcm_convert_f_to_s32_lin(const void *psrc, void *pdst,
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dst[i] = _pcm_convert_f_to_i(src[i], 31);
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}
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static int pcm_convert_s32_to_f(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s32_to_f(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
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pcm_convert_s32_to_f_lin);
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}
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static int pcm_convert_f_to_s32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_f_to_s32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
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@ -527,8 +528,8 @@ const struct pcm_func_map pcm_func_map[] = {
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const size_t pcm_func_count = ARRAY_SIZE(pcm_func_map);
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#if CONFIG_PCM_CONVERTER_FORMAT_S16_C16_AND_S16_C32
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static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int16_t *src = source->r_ptr;
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@ -556,8 +557,8 @@ static int pcm_convert_s16_c16_to_s16_c32(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -586,8 +587,8 @@ static int pcm_convert_s16_c32_to_s16_c16(const struct audio_stream *source,
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}
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#endif
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#if CONFIG_PCM_CONVERTER_FORMAT_S16_C32_AND_S32_C32
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static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -615,8 +616,8 @@ static int pcm_convert_s16_c32_to_s32_c32(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -645,8 +646,8 @@ static int pcm_convert_s32_c32_to_s16_c32(const struct audio_stream *source,
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}
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#endif
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#if CONFIG_PCM_CONVERTER_FORMAT_S16_C32_AND_S24_C32
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static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -674,8 +675,8 @@ static int pcm_convert_s16_c32_to_s24_c32(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -705,8 +706,8 @@ static int pcm_convert_s24_c32_to_s16_c32(const struct audio_stream *source,
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#endif
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#if CONFIG_PCM_CONVERTER_FORMAT_S24_C24_AND_S24_C32
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static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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uint8_t *src = source->r_ptr;
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@ -735,8 +736,8 @@ static int pcm_convert_s24_c24_to_s24_c32(const struct audio_stream *source,
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return samples;
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}
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static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -769,8 +770,8 @@ static int pcm_convert_s24_c32_to_s24_c24(const struct audio_stream *source,
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}
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/* 2x24bit samples are packed into 3x16bit samples for hda link dma */
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static int pcm_convert_s24_c32_to_s24_c24_link_gtw(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_c32_to_s24_c24_link_gtw(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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int32_t *src = source->r_ptr;
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@ -17,6 +17,7 @@
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#include <sof/audio/buffer.h>
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#include <sof/common.h>
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#include <sof/compiler_attributes.h>
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#include <ipc/stream.h>
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#include <xtensa/tie/xt_hifi3.h>
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@ -45,8 +46,8 @@ static void pcm_converter_setup_circular(const struct audio_stream *source)
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* \param[in,out] sink Destination buffer.
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* \param[in] samples Number of samples to process.
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*/
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static int pcm_convert_s16_to_s24(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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ae_int16 *in = audio_stream_read_frag(source, ioffset, sizeof(int16_t));
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@ -150,8 +151,8 @@ static ae_int32x2 pcm_shift_s24_to_s16(ae_int32x2 sample)
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* \param[in] samples Number of samples to process.
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* \return error code or number of processed samples.
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*/
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static int pcm_convert_s24_to_s16(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
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@ -262,8 +263,8 @@ static int pcm_convert_s24_to_s16(const struct audio_stream *source,
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* \param[in] samples Number of samples to process.
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* \return error code or number of processed samples.
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*/
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static int pcm_convert_s16_to_s32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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ae_int16 *in = audio_stream_read_frag(source, ioffset,
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@ -349,8 +350,8 @@ static int pcm_convert_s16_to_s32(const struct audio_stream *source,
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* \param[in] samples Number of samples to process.
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* \return error code or number of processed samples.
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*/
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static int pcm_convert_s32_to_s16(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s32_to_s16(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
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@ -457,8 +458,8 @@ static int pcm_convert_s32_to_s16(const struct audio_stream *source,
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* \param[in] samples Number of samples to process.
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* \return error code or number of processed samples.
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*/
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static int pcm_convert_s24_to_s32(const struct audio_stream *source,
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uint32_t ioffset, struct audio_stream *sink,
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static int pcm_convert_s24_to_s32(const struct audio_stream __sparse_cache *source,
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uint32_t ioffset, struct audio_stream __sparse_cache *sink,
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uint32_t ooffset, uint32_t samples)
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{
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ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
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@ -550,8 +551,8 @@ static ae_int32x2 pcm_shift_s32_to_s24(ae_int32x2 sample)
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|||
* \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 *source,
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||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_s32_to_s24(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
|
||||
|
|
@ -704,8 +705,8 @@ 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 *source,
|
||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_s16_to_f(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
|
||||
|
|
@ -719,8 +720,8 @@ static int pcm_convert_s16_to_f(const struct audio_stream *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 *source,
|
||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_f_to_s16(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
|
||||
|
|
@ -805,8 +806,8 @@ 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 *source,
|
||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_s24_to_f(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
|
||||
|
|
@ -820,8 +821,8 @@ static int pcm_convert_s24_to_f(const struct audio_stream *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 *source,
|
||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_f_to_s24(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
|
||||
|
|
@ -902,8 +903,8 @@ 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 *source,
|
||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_s32_to_f(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
|
||||
|
|
@ -917,8 +918,8 @@ static int pcm_convert_s32_to_f(const struct audio_stream *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 *source,
|
||||
uint32_t ioffset, struct audio_stream *sink,
|
||||
static int pcm_convert_f_to_s32(const struct audio_stream __sparse_cache *source,
|
||||
uint32_t ioffset, struct audio_stream __sparse_cache *sink,
|
||||
uint32_t ooffset, uint32_t samples)
|
||||
{
|
||||
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
|
||||
|
|
|
|||
|
|
@ -167,8 +167,8 @@ 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 *source, uint32_t ioffset,
|
||||
struct audio_stream *sink, uint32_t ooffset,
|
||||
int pcm_convert_as_linear(const struct audio_stream __sparse_cache *source, uint32_t ioffset,
|
||||
struct audio_stream __sparse_cache *sink, uint32_t ooffset,
|
||||
uint32_t samples, pcm_converter_lin_func converter);
|
||||
|
||||
#endif /* __SOF_AUDIO_PCM_CONVERTER_H__ */
|
||||
|
|
|
|||
Loading…
Reference in New Issue