diff options
Diffstat (limited to 'audio/filter')
| -rw-r--r-- | audio/filter/af.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_bs2b.c | 4 | ||||
| -rw-r--r-- | audio/filter/af_convert24.c | 4 | ||||
| -rw-r--r-- | audio/filter/af_convertsignendian.c | 107 |
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, -}; |