audio: remove S8, U16, U24, U32 formats

They are useless. Not only are they actually rarely in use; but
libavcodec doesn't even output them, as libavcodec has no such sample
formats for decoded audio.

Even if it should happen that we actually still need them (e.g. if doing
direct hardware output), there are better solutions. Swapping the sign
is a fast and lossless operation and can be done inplace, so AO actually
needing it could do this directly.

If you wonder why we keep U8 instead of S8: because libavcodec does it.

4 files changed, 1 insertions, 116 deletions

diff --git a/audio/filter/af.c b/audio/filter/af.c
index 67083c493b..c0535f957c 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -56,7 +56,6 @@ extern const struct af_info af_info_scaletempo;
 extern const struct af_info af_info_bs2b;
 extern const struct af_info af_info_lavfi;
 extern const struct af_info af_info_convert24;
-extern const struct af_info af_info_convertsignendian;
 extern const struct af_info af_info_rubberband;
 
 static const struct af_info *const filter_list[] = {
@@ -94,7 +93,6 @@ static const struct af_info *const filter_list[] = {
 #endif
     // Must come last, because they're fallback format conversion filter
     &af_info_convert24,
-    &af_info_convertsignendian,
     NULL
 };
 
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index 10d6f4e3fc..beb4dc9b96 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -57,12 +57,8 @@ static int filter_##name(struct af_instance *af, struct mp_audio *data) \
 #define FILTERS \
     FILTER(FLOAT,   f)      \
     FILTER(S32,     s32)    \
-    FILTER(U32,     u32)    \
     FILTER(S24,     s24)    \
-    FILTER(U24,     u24)    \
     FILTER(S16,     s16)    \
-    FILTER(U16,     u16)    \
-    FILTER(S8,      s8)     \
     FILTER(U8,      u8)
 
 #define FILTER DEF_FILTER
diff --git a/audio/filter/af_convert24.c b/audio/filter/af_convert24.c
index ab04c931b6..a81e84e2af 100644
--- a/audio/filter/af_convert24.c
+++ b/audio/filter/af_convert24.c
@@ -24,9 +24,7 @@
 
 static bool test_conversion(int src_format, int dst_format)
 {
-    return (src_format == AF_FORMAT_U24 && dst_format == AF_FORMAT_U32) ||
-           (src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
-           (src_format == AF_FORMAT_U32 && dst_format == AF_FORMAT_U24) ||
+    return (src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
            (src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24);
 }
 
diff --git a/audio/filter/af_convertsignendian.c b/audio/filter/af_convertsignendian.c
deleted file mode 100644
index abbd260a0f..0000000000
--- a/audio/filter/af_convertsignendian.c
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdlib.h>
-#include <assert.h>
-
-#include "af.h"
-#include "audio/format.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;
-    if (((src_format & ~AF_FORMAT_SIGN_MASK) ==
-         (dst_format & ~AF_FORMAT_SIGN_MASK)) &&
-        ((src_format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_I))
-        return true;
-    return false;
-}
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
-    switch (cmd) {
-    case AF_CONTROL_REINIT: {
-        struct mp_audio *in = arg;
-        struct mp_audio orig_in = *in;
-        struct mp_audio *out = af->data;
-
-        if (!test_conversion(in->format, out->format))
-            return AF_DETACH;
-
-        out->rate = in->rate;
-        mp_audio_set_channels(out, &in->channels);
-
-        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
-    }
-    case AF_CONTROL_SET_FORMAT: {
-        mp_audio_set_format(af->data, *(int*)arg);
-        return AF_OK;
-    }
-    }
-    return AF_UNKNOWN;
-}
-
-static void si2us(void *data, int len, int bps)
-{
-    ptrdiff_t i = -(len * bps);
-    uint8_t *p = &((uint8_t *)data)[len * bps];
-    if (BYTE_ORDER == LITTLE_ENDIAN && bps > 1)
-        p += bps - 1;
-    if (len <= 0)
-        return;
-    do {
-        p[i] ^= 0x80;
-    } while (i += bps);
-}
-
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    if (!data)
-        return 0;
-    if (af_make_writeable(af, data) < 0) {
-        talloc_free(data);
-        return -1;
-    }
-
-    int infmt = data->format;
-    int outfmt = af->data->format;
-    size_t len = data->samples * data->nch;
-
-    if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK))
-        si2us(data->planes[0], len, data->bps);
-
-    mp_audio_set_format(data, outfmt);
-    af_add_output_frame(af, data);
-    return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
-    af->control = control;
-    af->filter_frame = filter;
-    return AF_OK;
-}
-
-const struct af_info af_info_convertsignendian = {
-    .info = "Convert between sample format sign",
-    .name = "convertsign",
-    .open = af_open,
-    .test_conversion = test_conversion,
-};