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Revert "pcm_converter: Convert linear memory regions" #3159

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Merged
lgirdwood merged 1 commit into from
Jul 10, 2020
Merged

Revert "pcm_converter: Convert linear memory regions" #3159

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144 changes: 66 additions & 78 deletions src/audio/pcm_converter/pcm_converter_generic.c
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Original file line number Diff line number Diff line change
Expand Up @@ -26,126 +26,114 @@

#if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE

static void pcm_convert_s16_to_s24_lin(const void *psrc, void *pdst,
uint32_t samples)
{
const int16_t *src = psrc;
int32_t *dst = pdst;
uint32_t i;

for (i = 0; i < samples; i++)
dst[i] = src[i] << 8;
}

static void pcm_convert_s24_to_s16_lin(const void *psrc, void *pdst,
uint32_t samples)
{
const int32_t *src = psrc;
int16_t *dst = pdst;
uint32_t i;

for (i = 0; i < samples; i++)
dst[i] = sat_int16(Q_SHIFT_RND(sign_extend_s24(src[i]), 23, 15));
}

static void pcm_convert_s16_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s16_to_s24_lin);
uint32_t buff_frag = 0;
int16_t *src;
int32_t *dst;
uint32_t i;

for (i = 0; i < samples; i++) {
src = audio_stream_read_frag_s16(source, buff_frag + ioffset);
dst = audio_stream_write_frag_s32(sink, buff_frag + ooffset);
*dst = *src << 8;
buff_frag++;
}
}

static void pcm_convert_s24_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s24_to_s16_lin);
}

#endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE */

#if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE

static void pcm_convert_s16_to_s32_lin(const void *psrc, void *pdst,
uint32_t samples)
{
const int32_t *src = psrc;
int32_t *dst = pdst;
uint32_t buff_frag = 0;
int32_t *src;
int16_t *dst;
uint32_t i;

for (i = 0; i < samples; i++)
dst[i] = src[i] << 16;
for (i = 0; i < samples; i++) {
src = audio_stream_read_frag_s32(source, buff_frag + ioffset);
dst = audio_stream_write_frag_s16(sink, buff_frag + ooffset);
*dst = sat_int16(Q_SHIFT_RND(sign_extend_s24(*src), 23, 15));
buff_frag++;
}
}

static void pcm_convert_s32_to_s16_lin(const void *psrc, void *pdst,
uint32_t samples)
{
const int32_t *src = psrc;
int32_t *dst = pdst;
uint32_t i;
#endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE */

for (i = 0; i < samples; i++)
dst[i] = sat_int16(Q_SHIFT_RND(src[i], 31, 15));
}
#if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE

static void pcm_convert_s16_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s16_to_s32_lin);
uint32_t buff_frag = 0;
int16_t *src;
int32_t *dst;
uint32_t i;

for (i = 0; i < samples; i++) {
src = audio_stream_read_frag_s16(source, buff_frag + ioffset);
dst = audio_stream_write_frag_s32(sink, buff_frag + ooffset);
*dst = *src << 16;
buff_frag++;
}
}

static void pcm_convert_s32_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s32_to_s16_lin);
}

#endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE */

#if CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE

static void pcm_convert_s24_to_s32_lin(const void *psrc, void *pdst,
uint32_t samples)
{
const int32_t *src = psrc;
int32_t *dst = pdst;
uint32_t buff_frag = 0;
int32_t *src;
int16_t *dst;
uint32_t i;

for (i = 0; i < samples; i++)
dst[i] = src[i] << 8;
for (i = 0; i < samples; i++) {
src = audio_stream_read_frag_s32(source, buff_frag + ioffset);
dst = audio_stream_write_frag_s16(sink, buff_frag + ooffset);
*dst = sat_int16(Q_SHIFT_RND(*src, 31, 15));
buff_frag++;
}
}

static void pcm_convert_s32_to_s24_lin(const void *psrc, void *pdst,
uint32_t samples)
{
const int32_t *src = psrc;
int32_t *dst = pdst;
uint32_t i;
#endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE */

for (i = 0; i < samples; i++)
dst[i] = sat_int24(Q_SHIFT_RND(src[i], 31, 23));
}
#if CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE

static void pcm_convert_s24_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s24_to_s32_lin);
uint32_t buff_frag = 0;
int32_t *src;
int32_t *dst;
uint32_t i;

for (i = 0; i < samples; i++) {
src = audio_stream_read_frag_s32(source, buff_frag + ioffset);
dst = audio_stream_write_frag_s32(sink, buff_frag + ooffset);
*dst = *src << 8;
buff_frag++;
}
}

static void pcm_convert_s32_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s32_to_s24_lin);
uint32_t buff_frag = 0;
int32_t *src;
int32_t *dst;
uint32_t i;

for (i = 0; i < samples; i++) {
src = audio_stream_read_frag_s32(source, buff_frag + ioffset);
dst = audio_stream_write_frag_s32(sink, buff_frag + ooffset);
*dst = sat_int24(Q_SHIFT_RND(*src, 31, 23));
buff_frag++;
}
}

#endif /* CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE */
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