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-rw-r--r--audio/filter/af_bs2b.c69
-rw-r--r--audio/filter/af_convert24.c1
-rw-r--r--audio/filter/af_convertsignendian.c50
3 files changed, 30 insertions, 90 deletions
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index 0a2bd8e552..29d646a9a9 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -48,20 +48,12 @@ static int filter_##name(struct af_instance *af, struct mp_audio *data, int f) \
}
FILTER(f, float)
-FILTER(fbe, float)
-FILTER(fle, float)
-FILTER(s32be, int32_t)
-FILTER(u32be, uint32_t)
-FILTER(s32le, int32_t)
-FILTER(u32le, uint32_t)
-FILTER(s24be, bs2b_int24_t)
-FILTER(u24be, bs2b_uint24_t)
-FILTER(s24le, bs2b_int24_t)
-FILTER(u24le, bs2b_uint24_t)
-FILTER(s16be, int16_t)
-FILTER(u16be, uint16_t)
-FILTER(s16le, int16_t)
-FILTER(u16le, uint16_t)
+FILTER(s32, int32_t)
+FILTER(u32, uint32_t)
+FILTER(s24, bs2b_int24_t)
+FILTER(u24, bs2b_uint24_t)
+FILTER(s16, int16_t)
+FILTER(u16, uint16_t)
FILTER(s8, int8_t)
FILTER(u8, uint8_t)
@@ -85,47 +77,26 @@ static int control(struct af_instance *af, int cmd, void *arg)
/* check for formats supported by libbs2b
and assign corresponding handlers */
switch (format) {
- case AF_FORMAT_FLOAT_BE:
- af->filter = filter_fbe;
- break;
- case AF_FORMAT_FLOAT_LE:
- af->filter = filter_fle;
- break;
- case AF_FORMAT_S32_BE:
- af->filter = filter_s32be;
- break;
- case AF_FORMAT_U32_BE:
- af->filter = filter_u32be;
- break;
- case AF_FORMAT_S32_LE:
- af->filter = filter_s32le;
- break;
- case AF_FORMAT_U32_LE:
- af->filter = filter_u32le;
- break;
- case AF_FORMAT_S24_BE:
- af->filter = filter_s24be;
- break;
- case AF_FORMAT_U24_BE:
- af->filter = filter_u24be;
+ case AF_FORMAT_FLOAT:
+ af->filter = filter_f;
break;
- case AF_FORMAT_S24_LE:
- af->filter = filter_s24le;
+ case AF_FORMAT_S32:
+ af->filter = filter_s32;
break;
- case AF_FORMAT_U24_LE:
- af->filter = filter_u24le;
+ case AF_FORMAT_U32:
+ af->filter = filter_u32;
break;
- case AF_FORMAT_S16_BE:
- af->filter = filter_s16be;
+ case AF_FORMAT_S24:
+ af->filter = filter_s24;
break;
- case AF_FORMAT_U16_BE:
- af->filter = filter_u16be;
+ case AF_FORMAT_U24:
+ af->filter = filter_u24;
break;
- case AF_FORMAT_S16_LE:
- af->filter = filter_s16le;
+ case AF_FORMAT_S16:
+ af->filter = filter_s16;
break;
- case AF_FORMAT_U16_LE:
- af->filter = filter_u16le;
+ case AF_FORMAT_U16:
+ af->filter = filter_u16;
break;
case AF_FORMAT_S8:
af->filter = filter_s8;
diff --git a/audio/filter/af_convert24.c b/audio/filter/af_convert24.c
index 38799bd8fc..e59317fe1b 100644
--- a/audio/filter/af_convert24.c
+++ b/audio/filter/af_convert24.c
@@ -20,6 +20,7 @@
#include "audio/format.h"
#include "af.h"
+#include "osdep/endian.h"
static bool test_conversion(int src_format, int dst_format)
{
diff --git a/audio/filter/af_convertsignendian.c b/audio/filter/af_convertsignendian.c
index a0b47b38a3..7e7e436352 100644
--- a/audio/filter/af_convertsignendian.c
+++ b/audio/filter/af_convertsignendian.c
@@ -20,21 +20,16 @@
#include "af.h"
#include "audio/format.h"
-#include "osdep/mpbswap.h"
+#include "osdep/endian.h"
static bool test_conversion(int src_format, int dst_format)
{
if ((src_format & AF_FORMAT_PLANAR) ||
(dst_format & AF_FORMAT_PLANAR))
return false;
- int src_noend = src_format & ~AF_FORMAT_END_MASK;
- int dst_noend = dst_format & ~AF_FORMAT_END_MASK;
- // We can swap endian for all formats, but sign only for integer formats.
- if (src_noend == dst_noend)
- return true;
- if (((src_noend & ~AF_FORMAT_SIGN_MASK) ==
- (dst_noend & ~AF_FORMAT_SIGN_MASK)) &&
- ((src_noend & AF_FORMAT_POINT_MASK) == AF_FORMAT_I))
+ if (((src_format & ~AF_FORMAT_SIGN_MASK) ==
+ (dst_format & ~AF_FORMAT_SIGN_MASK)) &&
+ ((src_format & AF_FORMAT_POINT_MASK) == AF_FORMAT_I))
return true;
return false;
}
@@ -63,34 +58,11 @@ static int control(struct af_instance *af, int cmd, void *arg)
return AF_UNKNOWN;
}
-static void endian(void *data, int len, int bps)
-{
- switch (bps) {
- case 2:
- for (int i = 0; i < len; i++) {
- ((uint16_t*)data)[i] = bswap_16(((uint16_t *)data)[i]);
- }
- break;
- case 3:
- for(int i = 0; i < len; i++) {
- uint8_t s = ((uint8_t *)data)[3 * i];
- ((uint8_t *)data)[3 * i] = ((uint8_t *)data)[3 * i + 2];
- ((uint8_t *)data)[3 * i + 2] = s;
- }
- break;
- case 4:
- for(int i = 0; i < len; i++) {
- ((uint32_t*)data)[i] = bswap_32(((uint32_t *)data)[i]);
- }
- break;
- }
-}
-
-static void si2us(void *data, int len, int bps, bool le)
+static void si2us(void *data, int len, int bps)
{
ptrdiff_t i = -(len * bps);
uint8_t *p = &((uint8_t *)data)[len * bps];
- if (le && bps > 1)
+ if (BYTE_ORDER == LITTLE_ENDIAN && bps > 1)
p += bps - 1;
if (len <= 0)
return;
@@ -105,12 +77,8 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags)
int outfmt = af->data->format;
size_t len = data->samples * data->nch;
- if ((infmt & AF_FORMAT_END_MASK) != (outfmt & AF_FORMAT_END_MASK))
- endian(data->planes[0], len, data->bps);
-
if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK))
- si2us(data->planes[0], len, data->bps,
- (outfmt & AF_FORMAT_END_MASK) == AF_FORMAT_LE);
+ si2us(data->planes[0], len, data->bps);
mp_audio_set_format(data, outfmt);
return 0;
@@ -124,8 +92,8 @@ static int af_open(struct af_instance *af)
}
const struct af_info af_info_convertsignendian = {
- .info = "Convert between sample format sign/endian",
- .name = "convertsignendian",
+ .info = "Convert between sample format sign",
+ .name = "convertsign",
.open = af_open,
.test_conversion = test_conversion,
};
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 17:55:47 +0000