external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00
Code Issues Releases Wiki Activity

Rename ffmpeg-4.2-fit to ffmpeg-4-fit

Browse source
This commit is contained in:
winlin 2021-03-02 17:48:40 +08:00
parent b19074721c
commit 27712fdda7
720 changed files with 14 additions and 14 deletions
Download patch file Download diff file Expand all files Collapse all files

23
trunk/3rdparty/ffmpeg-4-fit/libswresample/Makefile vendored Normal file
View file

@ -0,0 +1,23 @@
NAME = swresample
DESC = FFmpeg audio resampling library
FFLIBS = avutil
HEADERS = swresample.h \
version.h \
OBJS = audioconvert.o \
dither.o \
options.o \
rematrix.o \
resample.o \
resample_dsp.o \
swresample.o \
swresample_frame.o \
OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o
OBJS-$(CONFIG_SHARED) += log2_tab.o
# Windows resource file
SLIBOBJS-$(HAVE_GNU_WINDRES) += swresampleres.o
TESTPROGS = swresample

7
trunk/3rdparty/ffmpeg-4-fit/libswresample/aarch64/Makefile vendored Normal file
View file

@ -0,0 +1,7 @@
OBJS += aarch64/audio_convert_init.o \
aarch64/resample_init.o
OBJS-$(CONFIG_NEON_CLOBBER_TEST) += aarch64/neontest.o
NEON-OBJS += aarch64/audio_convert_neon.o \
aarch64/resample.o

67
trunk/3rdparty/ffmpeg-4-fit/libswresample/aarch64/audio_convert_init.c vendored Normal file
View file

@ -0,0 +1,67 @@
/*
* This file is part of libswresample.
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/cpu.h"
#include "libavutil/aarch64/cpu.h"
#include "libavutil/samplefmt.h"
#include "libswresample/swresample_internal.h"
#include "libswresample/audioconvert.h"
void swri_oldapi_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len);
void swri_oldapi_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src, int len, int channels);
void swri_oldapi_conv_fltp_to_s16_nch_neon(int16_t *dst, float *const *src, int len, int channels);
static void conv_flt_to_s16_neon(uint8_t **dst, const uint8_t **src, int len){
swri_oldapi_conv_flt_to_s16_neon((int16_t*)*dst, (const float*)*src, len);
}
static void conv_fltp_to_s16_2ch_neon(uint8_t **dst, const uint8_t **src, int len){
swri_oldapi_conv_fltp_to_s16_2ch_neon((int16_t*)*dst, (float *const*)src, len, 2);
}
static void conv_fltp_to_s16_nch_neon(uint8_t **dst, const uint8_t **src, int len){
int channels;
for(channels=3; channels<SWR_CH_MAX && src[channels]; channels++)
;
swri_oldapi_conv_fltp_to_s16_nch_neon((int16_t*)*dst, (float *const*)src, len, channels);
}
av_cold void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels)
{
int cpu_flags = av_get_cpu_flags();
ac->simd_f= NULL;
if (have_neon(cpu_flags)) {
if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = conv_flt_to_s16_neon;
if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP && channels == 2)
ac->simd_f = conv_fltp_to_s16_2ch_neon;
if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP && channels > 2)
ac->simd_f = conv_fltp_to_s16_nch_neon;
if(ac->simd_f)
ac->in_simd_align_mask = ac->out_simd_align_mask = 15;
}
}

120
trunk/3rdparty/ffmpeg-4-fit/libswresample/aarch64/resample_init.c vendored Normal file
View file

@ -0,0 +1,120 @@
/*
* Audio resampling
*
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "libavutil/cpu.h"
#include "libavutil/avassert.h"
#include "libavutil/aarch64/cpu.h"
#include "libswresample/resample.h"
#define DECLARE_RESAMPLE_COMMON_TEMPLATE(TYPE, DELEM, FELEM, FELEM2, OUT) \
\
void ff_resample_common_apply_filter_x4_##TYPE##_neon(FELEM2 *acc, const DELEM *src, \
const FELEM *filter, int length); \
\
void ff_resample_common_apply_filter_x8_##TYPE##_neon(FELEM2 *acc, const DELEM *src, \
const FELEM *filter, int length); \
\
static int ff_resample_common_##TYPE##_neon(ResampleContext *c, void *dest, const void *source, \
int n, int update_ctx) \
{ \
DELEM *dst = dest; \
const DELEM *src = source; \
int dst_index; \
int index = c->index; \
int frac = c->frac; \
int sample_index = 0; \
int x4_aligned_filter_length = c->filter_length & ~3; \
int x8_aligned_filter_length = c->filter_length & ~7; \
\
while (index >= c->phase_count) { \
sample_index++; \
index -= c->phase_count; \
} \
\
for (dst_index = 0; dst_index < n; dst_index++) { \
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index; \
\
FELEM2 val = 0; \
int i = 0; \
if (x8_aligned_filter_length >= 8) { \
ff_resample_common_apply_filter_x8_##TYPE##_neon(&val, &src[sample_index], \
filter, x8_aligned_filter_length); \
i += x8_aligned_filter_length; \
\
} else if (x4_aligned_filter_length >= 4) { \
ff_resample_common_apply_filter_x4_##TYPE##_neon(&val, &src[sample_index], \
filter, x4_aligned_filter_length); \
i += x4_aligned_filter_length; \
} \
for (; i < c->filter_length; i++) { \
val += src[sample_index + i] * (FELEM2)filter[i]; \
} \
OUT(dst[dst_index], val); \
\
frac += c->dst_incr_mod; \
index += c->dst_incr_div; \
if (frac >= c->src_incr) { \
frac -= c->src_incr; \
index++; \
} \
\
while (index >= c->phase_count) { \
sample_index++; \
index -= c->phase_count; \
} \
} \
\
if (update_ctx) { \
c->frac = frac; \
c->index = index; \
} \
\
return sample_index; \
} \
#define OUT(d, v) d = v
DECLARE_RESAMPLE_COMMON_TEMPLATE(float, float, float, float, OUT)
#undef OUT
#define OUT(d, v) (v) = ((v) + (1<<(14)))>>15; (d) = av_clip_int16(v)
DECLARE_RESAMPLE_COMMON_TEMPLATE(s16, int16_t, int16_t, int32_t, OUT)
#undef OUT
av_cold void swri_resample_dsp_aarch64_init(ResampleContext *c)
{
int cpu_flags = av_get_cpu_flags();
if (!have_neon(cpu_flags))
return;
switch(c->format) {
case AV_SAMPLE_FMT_FLTP:
c->dsp.resample_common = ff_resample_common_float_neon;
break;
case AV_SAMPLE_FMT_S16P:
c->dsp.resample_common = ff_resample_common_s16_neon;
break;
}
}

8
trunk/3rdparty/ffmpeg-4-fit/libswresample/arm/Makefile vendored Normal file
View file

@ -0,0 +1,8 @@
OBJS += arm/audio_convert_init.o \
arm/resample_init.o
OBJS-$(CONFIG_NEON_CLOBBER_TEST) += arm/neontest.o
NEON-OBJS += arm/audio_convert_neon.o \
arm/resample.o

67
trunk/3rdparty/ffmpeg-4-fit/libswresample/arm/audio_convert_init.c vendored Normal file
View file

@ -0,0 +1,67 @@
/*
* This file is part of libswresample.
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/cpu.h"
#include "libavutil/arm/cpu.h"
#include "libavutil/samplefmt.h"
#include "libswresample/swresample_internal.h"
#include "libswresample/audioconvert.h"
void swri_oldapi_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len);
void swri_oldapi_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src, int len, int channels);
void swri_oldapi_conv_fltp_to_s16_nch_neon(int16_t *dst, float *const *src, int len, int channels);
static void conv_flt_to_s16_neon(uint8_t **dst, const uint8_t **src, int len){
swri_oldapi_conv_flt_to_s16_neon((int16_t*)*dst, (const float*)*src, len);
}
static void conv_fltp_to_s16_2ch_neon(uint8_t **dst, const uint8_t **src, int len){
swri_oldapi_conv_fltp_to_s16_2ch_neon((int16_t*)*dst, (float *const*)src, len, 2);
}
static void conv_fltp_to_s16_nch_neon(uint8_t **dst, const uint8_t **src, int len){
int channels;
for(channels=3; channels<SWR_CH_MAX && src[channels]; channels++)
;
swri_oldapi_conv_fltp_to_s16_nch_neon((int16_t*)*dst, (float *const*)src, len, channels);
}
av_cold void swri_audio_convert_init_arm(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels)
{
int cpu_flags = av_get_cpu_flags();
ac->simd_f= NULL;
if (have_neon(cpu_flags)) {
if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = conv_flt_to_s16_neon;
if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP && channels == 2)
ac->simd_f = conv_fltp_to_s16_2ch_neon;
if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP && channels > 2)
ac->simd_f = conv_fltp_to_s16_nch_neon;
if(ac->simd_f)
ac->in_simd_align_mask = ac->out_simd_align_mask = 15;
}
}

120
trunk/3rdparty/ffmpeg-4-fit/libswresample/arm/resample_init.c vendored Normal file
View file

@ -0,0 +1,120 @@
/*
* Audio resampling
*
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "libavutil/cpu.h"
#include "libavutil/avassert.h"
#include "libavutil/arm/cpu.h"
#include "libswresample/resample.h"
#define DECLARE_RESAMPLE_COMMON_TEMPLATE(TYPE, DELEM, FELEM, FELEM2, OUT) \
\
void ff_resample_common_apply_filter_x4_##TYPE##_neon(FELEM2 *acc, const DELEM *src, \
const FELEM *filter, int length); \
\
void ff_resample_common_apply_filter_x8_##TYPE##_neon(FELEM2 *acc, const DELEM *src, \
const FELEM *filter, int length); \
\
static int ff_resample_common_##TYPE##_neon(ResampleContext *c, void *dest, const void *source, \
int n, int update_ctx) \
{ \
DELEM *dst = dest; \
const DELEM *src = source; \
int dst_index; \
int index = c->index; \
int frac = c->frac; \
int sample_index = 0; \
int x4_aligned_filter_length = c->filter_length & ~3; \
int x8_aligned_filter_length = c->filter_length & ~7; \
\
while (index >= c->phase_count) { \
sample_index++; \
index -= c->phase_count; \
} \
\
for (dst_index = 0; dst_index < n; dst_index++) { \
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index; \
\
FELEM2 val = 0; \
int i = 0; \
if (x8_aligned_filter_length >= 8) { \
ff_resample_common_apply_filter_x8_##TYPE##_neon(&val, &src[sample_index], \
filter, x8_aligned_filter_length); \
i += x8_aligned_filter_length; \
\
} else if (x4_aligned_filter_length >= 4) { \
ff_resample_common_apply_filter_x4_##TYPE##_neon(&val, &src[sample_index], \
filter, x4_aligned_filter_length); \
i += x4_aligned_filter_length; \
} \
for (; i < c->filter_length; i++) { \
val += src[sample_index + i] * (FELEM2)filter[i]; \
} \
OUT(dst[dst_index], val); \
\
frac += c->dst_incr_mod; \
index += c->dst_incr_div; \
if (frac >= c->src_incr) { \
frac -= c->src_incr; \
index++; \
} \
\
while (index >= c->phase_count) { \
sample_index++; \
index -= c->phase_count; \
} \
} \
\
if (update_ctx) { \
c->frac = frac; \
c->index = index; \
} \
\
return sample_index; \
} \
#define OUT(d, v) d = v
DECLARE_RESAMPLE_COMMON_TEMPLATE(float, float, float, float, OUT)
#undef OUT
#define OUT(d, v) (v) = ((v) + (1<<(14)))>>15; (d) = av_clip_int16(v)
DECLARE_RESAMPLE_COMMON_TEMPLATE(s16, int16_t, int16_t, int32_t, OUT)
#undef OUT
av_cold void swri_resample_dsp_arm_init(ResampleContext *c)
{
int cpu_flags = av_get_cpu_flags();
if (!have_neon(cpu_flags))
return;
switch(c->format) {
case AV_SAMPLE_FMT_FLTP:
c->dsp.resample_common = ff_resample_common_float_neon;
break;
case AV_SAMPLE_FMT_S16P:
c->dsp.resample_common = ff_resample_common_s16_neon;
break;
}
}

247
trunk/3rdparty/ffmpeg-4-fit/libswresample/audioconvert.c vendored Normal file
View file

@ -0,0 +1,247 @@
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avstring.h"
#include "libavutil/avassert.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "audioconvert.h"
#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
//FIXME rounding ?
#define CONV_FUNC(ofmt, otype, ifmt, expr)\
static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)\
{\
uint8_t *end2 = end - 3*os;\
while(po < end2){\
*(otype*)po = expr; pi += is; po += os;\
*(otype*)po = expr; pi += is; po += os;\
*(otype*)po = expr; pi += is; po += os;\
*(otype*)po = expr; pi += is; po += os;\
}\
while(po < end){\
*(otype*)po = expr; pi += is; po += os;\
}\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<8)
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<24)
CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8 , (uint64_t)((*(const uint8_t*)pi - 0x80U))<<56)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16, (uint64_t)(*(const int16_t*)pi)<<48)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32, (uint64_t)(*(const int32_t*)pi)<<32)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S64, (*(const int64_t*)pi>>56) + 0x80)
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi>>48)
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi>>32)
CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S64, *(const int64_t*)pi*(1.0f/ (INT64_C(1)<<63)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S64, *(const int64_t*)pi*(1.0 / (INT64_C(1)<<63)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float*)pi * (INT64_C(1)<<63)))
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double*)pi * (INT64_C(1)<<63)))
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
#define FMT_PAIR_FUNC(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME(out, in)
static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = {
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_S16),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_S32),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_FLT),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_DBL),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_S64),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),
};
static void cpy1(uint8_t **dst, const uint8_t **src, int len){
memcpy(*dst, *src, len);
}
static void cpy2(uint8_t **dst, const uint8_t **src, int len){
memcpy(*dst, *src, 2*len);
}
static void cpy4(uint8_t **dst, const uint8_t **src, int len){
memcpy(*dst, *src, 4*len);
}
static void cpy8(uint8_t **dst, const uint8_t **src, int len){
memcpy(*dst, *src, 8*len);
}
AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, const int *ch_map,
int flags)
{
AudioConvert *ctx;
conv_func_type *f = fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt) + AV_SAMPLE_FMT_NB*av_get_packed_sample_fmt(in_fmt)];
if (!f)
return NULL;
ctx = av_mallocz(sizeof(*ctx));
if (!ctx)
return NULL;
if(channels == 1){
in_fmt = av_get_planar_sample_fmt( in_fmt);
out_fmt = av_get_planar_sample_fmt(out_fmt);
}
ctx->channels = channels;
ctx->conv_f = f;
ctx->ch_map = ch_map;
if (in_fmt == AV_SAMPLE_FMT_U8 || in_fmt == AV_SAMPLE_FMT_U8P)
memset(ctx->silence, 0x80, sizeof(ctx->silence));
if(out_fmt == in_fmt && !ch_map) {
switch(av_get_bytes_per_sample(in_fmt)){
case 1:ctx->simd_f = cpy1; break;
case 2:ctx->simd_f = cpy2; break;
case 4:ctx->simd_f = cpy4; break;
case 8:ctx->simd_f = cpy8; break;
}
}
if(HAVE_X86ASM && HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);
if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);
if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);
return ctx;
}
void swri_audio_convert_free(AudioConvert **ctx)
{
av_freep(ctx);
}
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
{
int ch;
int off=0;
const int os= (out->planar ? 1 :out->ch_count) *out->bps;
unsigned misaligned = 0;
av_assert0(ctx->channels == out->ch_count);
if (ctx->in_simd_align_mask) {
int planes = in->planar ? in->ch_count : 1;
unsigned m = 0;
for (ch = 0; ch < planes; ch++)
m |= (intptr_t)in->ch[ch];
misaligned |= m & ctx->in_simd_align_mask;
}
if (ctx->out_simd_align_mask) {
int planes = out->planar ? out->ch_count : 1;
unsigned m = 0;
for (ch = 0; ch < planes; ch++)
m |= (intptr_t)out->ch[ch];
misaligned |= m & ctx->out_simd_align_mask;
}
//FIXME optimize common cases
if(ctx->simd_f && !ctx->ch_map && !misaligned){
off = len&~15;
av_assert1(off>=0);
av_assert1(off<=len);
av_assert2(ctx->channels == SWR_CH_MAX || !in->ch[ctx->channels]);
if(off>0){
if(out->planar == in->planar){
int planes = out->planar ? out->ch_count : 1;
for(ch=0; ch<planes; ch++){
ctx->simd_f(out->ch+ch, (const uint8_t **)in->ch+ch, off * (out->planar ? 1 :out->ch_count));
}
}else{
ctx->simd_f(out->ch, (const uint8_t **)in->ch, off);
}
}
if(off == len)
return 0;
}
for(ch=0; ch<ctx->channels; ch++){
const int ich= ctx->ch_map ? ctx->ch_map[ch] : ch;
const int is= ich < 0 ? 0 : (in->planar ? 1 : in->ch_count) * in->bps;
const uint8_t *pi= ich < 0 ? ctx->silence : in->ch[ich];
uint8_t *po= out->ch[ch];
uint8_t *end= po + os*len;
if(!po)
continue;
ctx->conv_f(po+off*os, pi+off*is, is, os, end);
}
return 0;
}

78
trunk/3rdparty/ffmpeg-4-fit/libswresample/audioconvert.h vendored Normal file
View file

@ -0,0 +1,78 @@
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_AUDIOCONVERT_H
#define SWRESAMPLE_AUDIOCONVERT_H
/**
* @file
* Audio format conversion routines
*/
#include "swresample_internal.h"
#include "libavutil/cpu.h"
typedef void (conv_func_type)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end);
typedef void (simd_func_type)(uint8_t **dst, const uint8_t **src, int len);
typedef struct AudioConvert {
int channels;
int in_simd_align_mask;
int out_simd_align_mask;
conv_func_type *conv_f;
simd_func_type *simd_f;
const int *ch_map;
uint8_t silence[8]; ///< silence input sample
}AudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param in_fmt Input sample format
* @param channels Number of channels
* @param flags See AV_CPU_FLAG_xx
* @param ch_map list of the channels id to pick from the source stream, NULL
* if all channels must be selected
* @return NULL on error
*/
AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, const int *ch_map,
int flags);
/**
* Free audio sample format converter context.
* and set the pointer to NULL
*/
void swri_audio_convert_free(AudioConvert **ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] in array of input buffers for each channel
* @param len length of audio frame size (measured in samples)
*/
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len);
#endif /* SWRESAMPLE_AUDIOCONVERT_H */

148
trunk/3rdparty/ffmpeg-4-fit/libswresample/dither.c vendored Normal file
View file

@ -0,0 +1,148 @@
/*
* Copyright (C) 2012-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/avassert.h"
#include "swresample_internal.h"
#include "noise_shaping_data.c"
int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt) {
double scale = s->dither.noise_scale;
#define TMP_EXTRA 2
double *tmp = av_malloc_array(len + TMP_EXTRA, sizeof(double));
int i;
if (!tmp)
return AVERROR(ENOMEM);
for(i=0; i<len + TMP_EXTRA; i++){
double v;
seed = seed* 1664525 + 1013904223;
switch(s->dither.method){
case SWR_DITHER_RECTANGULAR: v= ((double)seed) / UINT_MAX - 0.5; break;
default:
av_assert0(s->dither.method < SWR_DITHER_NB);
v = ((double)seed) / UINT_MAX;
seed = seed*1664525 + 1013904223;
v-= ((double)seed) / UINT_MAX;
break;
}
tmp[i] = v;
}
for(i=0; i<len; i++){
double v;
switch(s->dither.method){
default:
av_assert0(s->dither.method < SWR_DITHER_NB);
v = tmp[i];
break;
case SWR_DITHER_TRIANGULAR_HIGHPASS :
v = (- tmp[i] + 2*tmp[i+1] - tmp[i+2]) / sqrt(6);
break;
}
v*= scale;
switch(noise_fmt){
case AV_SAMPLE_FMT_S16P: ((int16_t*)dst)[i] = v; break;
case AV_SAMPLE_FMT_S32P: ((int32_t*)dst)[i] = v; break;
case AV_SAMPLE_FMT_FLTP: ((float *)dst)[i] = v; break;
case AV_SAMPLE_FMT_DBLP: ((double *)dst)[i] = v; break;
default: av_assert0(0);
}
}
av_free(tmp);
return 0;
}
av_cold int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt)
{
int i;
double scale = 0;
if (s->dither.method > SWR_DITHER_TRIANGULAR_HIGHPASS && s->dither.method <= SWR_DITHER_NS)
return AVERROR(EINVAL);
out_fmt = av_get_packed_sample_fmt(out_fmt);
in_fmt = av_get_packed_sample_fmt( in_fmt);
if(in_fmt == AV_SAMPLE_FMT_FLT || in_fmt == AV_SAMPLE_FMT_DBL){
if(out_fmt == AV_SAMPLE_FMT_S32) scale = 1.0/(1LL<<31);
if(out_fmt == AV_SAMPLE_FMT_S16) scale = 1.0/(1LL<<15);
if(out_fmt == AV_SAMPLE_FMT_U8 ) scale = 1.0/(1LL<< 7);
}
if(in_fmt == AV_SAMPLE_FMT_S32 && out_fmt == AV_SAMPLE_FMT_S32 && (s->dither.output_sample_bits&31)) scale = 1;
if(in_fmt == AV_SAMPLE_FMT_S32 && out_fmt == AV_SAMPLE_FMT_S16) scale = 1<<16;
if(in_fmt == AV_SAMPLE_FMT_S32 && out_fmt == AV_SAMPLE_FMT_U8 ) scale = 1<<24;
if(in_fmt == AV_SAMPLE_FMT_S16 && out_fmt == AV_SAMPLE_FMT_U8 ) scale = 1<<8;
scale *= s->dither.scale;
if (out_fmt == AV_SAMPLE_FMT_S32 && s->dither.output_sample_bits)
scale *= 1<<(32-s->dither.output_sample_bits);
if (scale == 0) {
s->dither.method = 0;
return 0;
}
s->dither.ns_pos = 0;
s->dither.noise_scale= scale;
s->dither.ns_scale = scale;
s->dither.ns_scale_1 = scale ? 1/scale : 0;
memset(s->dither.ns_errors, 0, sizeof(s->dither.ns_errors));
for (i=0; filters[i].coefs; i++) {
const filter_t *f = &filters[i];
if (llabs(s->out_sample_rate - f->rate)*20 <= f->rate && f->name == s->dither.method) {
int j;
s->dither.ns_taps = f->len;
for (j=0; j<f->len; j++)
s->dither.ns_coeffs[j] = f->coefs[j];
s->dither.ns_scale_1 *= 1 - exp(f->gain_cB * M_LN10 * 0.005) * 2 / (1<<(8*av_get_bytes_per_sample(out_fmt)));
break;
}
}
if (!filters[i].coefs && s->dither.method > SWR_DITHER_NS) {
av_log(s, AV_LOG_WARNING, "Requested noise shaping dither not available at this sampling rate, using triangular hp dither\n");
s->dither.method = SWR_DITHER_TRIANGULAR_HIGHPASS;
}
return 0;
}
#define TEMPLATE_DITHER_S16
#include "dither_template.c"
#undef TEMPLATE_DITHER_S16
#define TEMPLATE_DITHER_S32
#include "dither_template.c"
#undef TEMPLATE_DITHER_S32
#define TEMPLATE_DITHER_FLT
#include "dither_template.c"
#undef TEMPLATE_DITHER_FLT
#define TEMPLATE_DITHER_DBL
#include "dither_template.c"
#undef TEMPLATE_DITHER_DBL

84
trunk/3rdparty/ffmpeg-4-fit/libswresample/dither_template.c vendored Normal file
View file

@ -0,0 +1,84 @@
/*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#if defined(TEMPLATE_DITHER_DBL)
# define RENAME(N) N ## _double
# define DELEM double
# define CLIP(v) while(0)
#elif defined(TEMPLATE_DITHER_FLT)
# define RENAME(N) N ## _float
# define DELEM float
# define CLIP(v) while(0)
#elif defined(TEMPLATE_DITHER_S32)
# define RENAME(N) N ## _int32
# define DELEM int32_t
# define CLIP(v) v = FFMAX(FFMIN(v, INT32_MAX), INT32_MIN)
#elif defined(TEMPLATE_DITHER_S16)
# define RENAME(N) N ## _int16
# define DELEM int16_t
# define CLIP(v) v = FFMAX(FFMIN(v, INT16_MAX), INT16_MIN)
#else
ERROR
#endif
void RENAME(swri_noise_shaping)(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count){
int pos = s->dither.ns_pos;
int i, j, ch;
int taps = s->dither.ns_taps;
float S = s->dither.ns_scale;
float S_1 = s->dither.ns_scale_1;
av_assert2((taps&3) != 2);
av_assert2((taps&3) != 3 || s->dither.ns_coeffs[taps] == 0);
for (ch=0; ch<srcs->ch_count; ch++) {
const float *noise = ((const float *)noises->ch[ch]) + s->dither.noise_pos;
const DELEM *src = (const DELEM*)srcs->ch[ch];
DELEM *dst = (DELEM*)dsts->ch[ch];
float *ns_errors = s->dither.ns_errors[ch];
const float *ns_coeffs = s->dither.ns_coeffs;
pos = s->dither.ns_pos;
for (i=0; i<count; i++) {
double d1, d = src[i]*S_1;
for(j=0; j<taps-2; j+=4) {
d -= ns_coeffs[j ] * ns_errors[pos + j ]
+ns_coeffs[j + 1] * ns_errors[pos + j + 1]
+ns_coeffs[j + 2] * ns_errors[pos + j + 2]
+ns_coeffs[j + 3] * ns_errors[pos + j + 3];
}
if(j < taps)
d -= ns_coeffs[j] * ns_errors[pos + j];
pos = pos ? pos - 1 : taps - 1;
d1 = rint(d + noise[i]);
ns_errors[pos + taps] = ns_errors[pos] = d1 - d;
d1 *= S;
CLIP(d1);
dst[i] = d1;
}
}
s->dither.ns_pos = pos;
}
#undef RENAME
#undef DELEM
#undef CLIP

224
trunk/3rdparty/ffmpeg-4-fit/libswresample/noise_shaping_data.c vendored Normal file
View file

@ -0,0 +1,224 @@
/* Effect: dither/noise-shape Copyright (c) 2008-9 robs@users.sourceforge.net
*
* This library is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or (at
* your option) any later version.
*
* This library 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 Lesser
* General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this library; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
typedef struct {
int rate;
enum {fir, iir} type;
size_t len;
int gain_cB; /* Chosen so clips are few if any, but not guaranteed none. */
double const * coefs;
enum SwrDitherType name;
} filter_t;
static double const lip44[] = {2.033, -2.165, 1.959, -1.590, .6149};
static double const fwe44[] = {
2.412, -3.370, 3.937, -4.174, 3.353, -2.205, 1.281, -.569, .0847};
static double const mew44[] = {
1.662, -1.263, .4827, -.2913, .1268, -.1124, .03252, -.01265, -.03524};
static double const iew44[] = {
2.847, -4.685, 6.214, -7.184, 6.639, -5.032, 3.263, -1.632, .4191};
static double const ges44[] = {
2.2061, -.4706, -.2534, -.6214, 1.0587, .0676, -.6054, -.2738};
static double const ges48[] = {
2.2374, -.7339, -.1251, -.6033, .903, .0116, -.5853, -.2571};
static double const shi48[] = {
2.8720729351043701172, -5.0413231849670410156, 6.2442994117736816406,
-5.8483986854553222656, 3.7067542076110839844, -1.0495119094848632812,
-1.1830236911773681641, 2.1126792430877685547, -1.9094531536102294922,
0.99913084506988525391, -0.17090806365013122559, -0.32615602016448974609,
0.39127644896507263184, -0.26876461505889892578, 0.097676105797290802002,
-0.023473845794796943665,
};
static double const shi44[] = {
2.6773197650909423828, -4.8308925628662109375, 6.570110321044921875,
-7.4572014808654785156, 6.7263274192810058594, -4.8481650352478027344,
2.0412089824676513672, 0.7006359100341796875, -2.9537565708160400391,
4.0800385475158691406, -4.1845216751098632812, 3.3311812877655029297,
-2.1179926395416259766, 0.879302978515625, -0.031759146600961685181,
-0.42382788658142089844, 0.47882103919982910156, -0.35490813851356506348,
0.17496839165687561035, -0.060908168554306030273,
};
static double const shi38[] = {
1.6335992813110351562, -2.2615492343902587891, 2.4077029228210449219,
-2.6341717243194580078, 2.1440362930297851562, -1.8153258562088012695,
1.0816224813461303711, -0.70302653312683105469, 0.15991993248462677002,
0.041549518704414367676, -0.29416576027870178223, 0.2518316805362701416,
-0.27766478061676025391, 0.15785403549671173096, -0.10165894031524658203,
0.016833892092108726501,
};
static double const shi32[] =
{ /* dmaker 32000: bestmax=4.99659 (inverted) */
0.82118552923202515,
-1.0063692331314087,
0.62341964244842529,
-1.0447187423706055,
0.64532512426376343,
-0.87615132331848145,
0.52219754457473755,
-0.67434263229370117,
0.44954317808151245,
-0.52557498216629028,
0.34567299485206604,
-0.39618203043937683,
0.26791760325431824,
-0.28936097025871277,
0.1883765310049057,
-0.19097308814525604,
0.10431359708309174,
-0.10633844882249832,
0.046832218766212463,
-0.039653312414884567,
};
static double const shi22[] =
{ /* dmaker 22050: bestmax=5.77762 (inverted) */
0.056581053882837296,
-0.56956905126571655,
-0.40727734565734863,
-0.33870288729667664,
-0.29810553789138794,
-0.19039161503314972,
-0.16510021686553955,
-0.13468159735202789,
-0.096633769571781158,
-0.081049129366874695,
-0.064953058958053589,
-0.054459091275930405,
-0.043378707021474838,
-0.03660014271736145,
-0.026256965473294258,
-0.018786206841468811,
-0.013387725688517094,
-0.0090983230620622635,
-0.0026585909072309732,
-0.00042083300650119781,
};
static double const shi16[] =
{ /* dmaker 16000: bestmax=5.97128 (inverted) */
-0.37251132726669312,
-0.81423574686050415,
-0.55010956525802612,
-0.47405767440795898,
-0.32624706625938416,
-0.3161766529083252,
-0.2286367267370224,
-0.22916607558727264,
-0.19565616548061371,
-0.18160104751586914,
-0.15423151850700378,
-0.14104481041431427,
-0.11844276636838913,
-0.097583092749118805,
-0.076493598520755768,
-0.068106919527053833,
-0.041881654411554337,
-0.036922425031661987,
-0.019364040344953537,
-0.014994367957115173,
};
static double const shi11[] =
{ /* dmaker 11025: bestmax=5.9406 (inverted) */
-0.9264228343963623,
-0.98695987462997437,
-0.631156325340271,
-0.51966935396194458,
-0.39738872647285461,
-0.35679301619529724,
-0.29720726609230042,
-0.26310476660728455,
-0.21719355881214142,
-0.18561814725399017,
-0.15404847264289856,
-0.12687471508979797,
-0.10339745879173279,
-0.083688631653785706,
-0.05875682458281517,
-0.046893671154975891,
-0.027950936928391457,
-0.020740609616041183,
-0.009366452693939209,
-0.0060260160826146603,
};
static double const shi08[] =
{ /* dmaker 8000: bestmax=5.56234 (inverted) */
-1.202863335609436,
-0.94103097915649414,
-0.67878556251525879,
-0.57650017738342285,
-0.50004476308822632,
-0.44349345564842224,
-0.37833768129348755,
-0.34028723835945129,
-0.29413089156150818,
-0.24994957447052002,
-0.21715600788593292,
-0.18792112171649933,
-0.15268312394618988,
-0.12135542929172516,
-0.099610626697540283,
-0.075273610651493073,
-0.048787496984004974,
-0.042586319148540497,
-0.028991291299462318,
-0.011869125068187714,
};
static double const shl48[] = {
2.3925774097442626953, -3.4350297451019287109, 3.1853709220886230469,
-1.8117271661758422852, -0.20124770700931549072, 1.4759907722473144531,
-1.7210904359817504883, 0.97746700048446655273, -0.13790138065814971924,
-0.38185903429985046387, 0.27421241998672485352, 0.066584214568138122559,
-0.35223302245140075684, 0.37672343850135803223, -0.23964276909828186035,
0.068674825131893157959,
};
static double const shl44[] = {
2.0833916664123535156, -3.0418450832366943359, 3.2047898769378662109,
-2.7571926116943359375, 1.4978630542755126953, -0.3427594602108001709,
-0.71733748912811279297, 1.0737057924270629883, -1.0225815773010253906,
0.56649994850158691406, -0.20968692004680633545, -0.065378531813621520996,
0.10322438180446624756, -0.067442022264003753662, -0.00495197344571352005,
0,
};
static double const shh44[] = {
3.0259189605712890625, -6.0268716812133789062, 9.195003509521484375,
-11.824929237365722656, 12.767142295837402344, -11.917946815490722656,
9.1739168167114257812, -5.3712320327758789062, 1.1393624544143676758,
2.4484779834747314453, -4.9719839096069335938, 6.0392003059387207031,
-5.9359521865844726562, 4.903278350830078125, -3.5527443885803222656,
2.1909697055816650391, -1.1672389507293701172, 0.4903914332389831543,
-0.16519790887832641602, 0.023217858746647834778,
};
static const filter_t filters[] = {
{44100, fir, 5, 210, lip44, SWR_DITHER_NS_LIPSHITZ},
{46000, fir, 9, 276, fwe44, SWR_DITHER_NS_F_WEIGHTED},
{46000, fir, 9, 160, mew44, SWR_DITHER_NS_MODIFIED_E_WEIGHTED},
{46000, fir, 9, 321, iew44, SWR_DITHER_NS_IMPROVED_E_WEIGHTED},
// {48000, iir, 4, 220, ges48, SWR_DITHER_NS_GESEMANN},
// {44100, iir, 4, 230, ges44, SWR_DITHER_NS_GESEMANN},
{48000, fir, 16, 301, shi48, SWR_DITHER_NS_SHIBATA},
{44100, fir, 20, 333, shi44, SWR_DITHER_NS_SHIBATA},
{37800, fir, 16, 240, shi38, SWR_DITHER_NS_SHIBATA},
{32000, fir, 20, 240/*TBD*/, shi32, SWR_DITHER_NS_SHIBATA},
{22050, fir, 20, 240/*TBD*/, shi22, SWR_DITHER_NS_SHIBATA},
{16000, fir, 20, 240/*TBD*/, shi16, SWR_DITHER_NS_SHIBATA},
{11025, fir, 20, 240/*TBD*/, shi11, SWR_DITHER_NS_SHIBATA},
{ 8000, fir, 20, 240/*TBD*/, shi08, SWR_DITHER_NS_SHIBATA},
{48000, fir, 16, 250, shl48, SWR_DITHER_NS_LOW_SHIBATA},
{44100, fir, 15, 250, shl44, SWR_DITHER_NS_LOW_SHIBATA},
{44100, fir, 20, 383, shh44, SWR_DITHER_NS_HIGH_SHIBATA},
{ 0, fir, 0, 0, NULL, SWR_DITHER_NONE},
};

156
trunk/3rdparty/ffmpeg-4-fit/libswresample/options.c vendored Normal file
View file

@ -0,0 +1,156 @@
/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "swresample_internal.h"
#include <float.h>
#define C30DB M_SQRT2
#define C15DB 1.189207115
#define C__0DB 1.0
#define C_15DB 0.840896415
#define C_30DB M_SQRT1_2
#define C_45DB 0.594603558
#define C_60DB 0.5
#define OFFSET(x) offsetof(SwrContext,x)
#define PARAM AV_OPT_FLAG_AUDIO_PARAM
static const AVOption options[]={
{"ich" , "set input channel count" , OFFSET(user_in_ch_count ), AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"in_channel_count" , "set input channel count" , OFFSET(user_in_ch_count ), AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"och" , "set output channel count" , OFFSET(user_out_ch_count ), AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"out_channel_count" , "set output channel count" , OFFSET(user_out_ch_count ), AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"uch" , "set used channel count" , OFFSET(user_used_ch_count), AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"used_channel_count" , "set used channel count" , OFFSET(user_used_ch_count), AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"isr" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"in_sample_rate" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"osr" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"out_sample_rate" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"isf" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},
{"in_sample_fmt" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},
{"osf" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},
{"out_sample_fmt" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},
{"tsf" , "set internal sample format" , OFFSET(user_int_sample_fmt), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},
{"internal_sample_fmt" , "set internal sample format" , OFFSET(user_int_sample_fmt), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},
{"icl" , "set input channel layout" , OFFSET(user_in_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, INT64_MIN, INT64_MAX , PARAM, "channel_layout"},
{"in_channel_layout" , "set input channel layout" , OFFSET(user_in_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, INT64_MIN, INT64_MAX , PARAM, "channel_layout"},
{"ocl" , "set output channel layout" , OFFSET(user_out_ch_layout), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, INT64_MIN, INT64_MAX , PARAM, "channel_layout"},
{"out_channel_layout" , "set output channel layout" , OFFSET(user_out_ch_layout), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, INT64_MIN, INT64_MAX , PARAM, "channel_layout"},
{"clev" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"center_mix_level" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"slev" , "set surround mix level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"surround_mix_level" , "set surround mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"lfe_mix_level" , "set LFE mix level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
{"rmvol" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
{"rematrix_volume" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
{"rematrix_maxval" , "set rematrix maxval" , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0 }, 0 , 1000 , PARAM},
{"flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
{"swr_flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
{"res" , "force resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
{"dither_scale" , "set dither scale" , OFFSET(dither.scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
{"dither_method" , "set dither method" , OFFSET(user_dither_method),AV_OPT_TYPE_INT, {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
{"rectangular" , "select rectangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
{"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
{"triangular_hp" , "select triangular dither with high pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"lipshitz" , "select Lipshitz noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"shibata" , "select Shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"low_shibata" , "select low Shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"high_shibata" , "select high Shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"f_weighted" , "select f-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"modified_e_weighted" , "select modified-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"improved_e_weighted" , "select improved-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM },
{"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 24 , PARAM },
{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_BOOL , {.i64=1 }, 0 , 1 , PARAM },
{"exact_rational" , "enable exact rational" , OFFSET(exact_rational) , AV_OPT_TYPE_BOOL , {.i64=1 }, 0 , 1 , PARAM },
{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
/* duplicate option in order to work with avconv */
{"resample_cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},
{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},
{"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"},
{"precision" , "set soxr resampling precision (in bits)"
, OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
{"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
, OFFSET(cheby) , AV_OPT_TYPE_BOOL , {.i64=0 }, 0 , 1 , PARAM },
{"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
, OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
{"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
, OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
{"comp_duration" , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
, OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
{"max_soft_comp" , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
, OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
{"async" , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
, OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
{"first_pts" , "Assume the first pts should be this value (in samples)."
, OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE }, INT64_MIN,INT64_MAX, PARAM },
{ "matrix_encoding" , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
{ "none", "select none", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "filter_type" , "select swr filter type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
{ "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "blackman_nuttall", "select Blackman Nuttall windowed sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser" , "select Kaiser windowed sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser_beta" , "set swr Kaiser window beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_DOUBLE , {.dbl=9 }, 2 , 16 , PARAM },
{ "output_sample_bits" , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT , {.i64=0 }, 0 , 64 , PARAM },
{0}
};
static const char* context_to_name(void* ptr) {
return "SWR";
}
static const AVClass av_class = {
.class_name = "SWResampler",
.item_name = context_to_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.log_level_offset_offset = OFFSET(log_level_offset),
.parent_log_context_offset = OFFSET(log_ctx),
.category = AV_CLASS_CATEGORY_SWRESAMPLER,
};
const AVClass *swr_get_class(void)
{
return &av_class;
}
av_cold struct SwrContext *swr_alloc(void){
SwrContext *s= av_mallocz(sizeof(SwrContext));
if(s){
s->av_class= &av_class;
av_opt_set_defaults(s);
}
return s;
}

576
trunk/3rdparty/ffmpeg-4-fit/libswresample/rematrix.c vendored Normal file
View file

@ -0,0 +1,576 @@
/*
* Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "swresample_internal.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#define TEMPLATE_REMATRIX_FLT
#include "rematrix_template.c"
#undef TEMPLATE_REMATRIX_FLT
#define TEMPLATE_REMATRIX_DBL
#include "rematrix_template.c"
#undef TEMPLATE_REMATRIX_DBL
#define TEMPLATE_REMATRIX_S16
#include "rematrix_template.c"
#define TEMPLATE_CLIP
#include "rematrix_template.c"
#undef TEMPLATE_CLIP
#undef TEMPLATE_REMATRIX_S16
#define TEMPLATE_REMATRIX_S32
#include "rematrix_template.c"
#undef TEMPLATE_REMATRIX_S32
#define FRONT_LEFT 0
#define FRONT_RIGHT 1
#define FRONT_CENTER 2
#define LOW_FREQUENCY 3
#define BACK_LEFT 4
#define BACK_RIGHT 5
#define FRONT_LEFT_OF_CENTER 6
#define FRONT_RIGHT_OF_CENTER 7
#define BACK_CENTER 8
#define SIDE_LEFT 9
#define SIDE_RIGHT 10
#define TOP_CENTER 11
#define TOP_FRONT_LEFT 12
#define TOP_FRONT_CENTER 13
#define TOP_FRONT_RIGHT 14
#define TOP_BACK_LEFT 15
#define TOP_BACK_CENTER 16
#define TOP_BACK_RIGHT 17
#define NUM_NAMED_CHANNELS 18
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride)
{
int nb_in, nb_out, in, out;
if (!s || s->in_convert) // s needs to be allocated but not initialized
return AVERROR(EINVAL);
memset(s->matrix, 0, sizeof(s->matrix));
memset(s->matrix_flt, 0, sizeof(s->matrix_flt));
nb_in = (s->user_in_ch_count > 0) ? s->user_in_ch_count :
av_get_channel_layout_nb_channels(s->user_in_ch_layout);
nb_out = (s->user_out_ch_count > 0) ? s->user_out_ch_count :
av_get_channel_layout_nb_channels(s->user_out_ch_layout);
for (out = 0; out < nb_out; out++) {
for (in = 0; in < nb_in; in++)
s->matrix_flt[out][in] = s->matrix[out][in] = matrix[in];
matrix += stride;
}
s->rematrix_custom = 1;
return 0;
}
static int even(int64_t layout){
if(!layout) return 1;
if(layout&(layout-1)) return 1;
return 0;
}
static int clean_layout(void *s, int64_t layout){
if(layout && layout != AV_CH_FRONT_CENTER && !(layout&(layout-1))) {
char buf[128];
av_get_channel_layout_string(buf, sizeof(buf), -1, layout);
av_log(s, AV_LOG_VERBOSE, "Treating %s as mono\n", buf);
return AV_CH_FRONT_CENTER;
}
return layout;
}
static int sane_layout(int64_t layout){
if(!(layout & AV_CH_LAYOUT_SURROUND)) // at least 1 front speaker
return 0;
if(!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT))) // no asymetric front
return 0;
if(!even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT))) // no asymetric side
return 0;
if(!even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT)))
return 0;
if(!even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER)))
return 0;
if(av_get_channel_layout_nb_channels(layout) >= SWR_CH_MAX)
return 0;
return 1;
}
av_cold int swr_build_matrix(uint64_t in_ch_layout_param, uint64_t out_ch_layout_param,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double maxval,
double rematrix_volume, double *matrix_param,
int stride, enum AVMatrixEncoding matrix_encoding, void *log_context)
{
int i, j, out_i;
double matrix[NUM_NAMED_CHANNELS][NUM_NAMED_CHANNELS]={{0}};
int64_t unaccounted, in_ch_layout, out_ch_layout;
double maxcoef=0;
char buf[128];
in_ch_layout = clean_layout(log_context, in_ch_layout_param);
out_ch_layout = clean_layout(log_context, out_ch_layout_param);
if( out_ch_layout == AV_CH_LAYOUT_STEREO_DOWNMIX
&& (in_ch_layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == 0
)
out_ch_layout = AV_CH_LAYOUT_STEREO;
if( in_ch_layout == AV_CH_LAYOUT_STEREO_DOWNMIX
&& (out_ch_layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == 0
)
in_ch_layout = AV_CH_LAYOUT_STEREO;
if(!sane_layout(in_ch_layout)){
av_get_channel_layout_string(buf, sizeof(buf), -1, in_ch_layout_param);
av_log(log_context, AV_LOG_ERROR, "Input channel layout '%s' is not supported\n", buf);
return AVERROR(EINVAL);
}
if(!sane_layout(out_ch_layout)){
av_get_channel_layout_string(buf, sizeof(buf), -1, out_ch_layout_param);
av_log(log_context, AV_LOG_ERROR, "Output channel layout '%s' is not supported\n", buf);
return AVERROR(EINVAL);
}
for(i=0; i<FF_ARRAY_ELEMS(matrix); i++){
if(in_ch_layout & out_ch_layout & (1ULL<<i))
matrix[i][i]= 1.0;
}
unaccounted= in_ch_layout & ~out_ch_layout;
//FIXME implement dolby surround
//FIXME implement full ac3
if(unaccounted & AV_CH_FRONT_CENTER){
if((out_ch_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO){
if(in_ch_layout & AV_CH_LAYOUT_STEREO) {
matrix[ FRONT_LEFT][FRONT_CENTER]+= center_mix_level;
matrix[FRONT_RIGHT][FRONT_CENTER]+= center_mix_level;
} else {
matrix[ FRONT_LEFT][FRONT_CENTER]+= M_SQRT1_2;
matrix[FRONT_RIGHT][FRONT_CENTER]+= M_SQRT1_2;
}
}else
av_assert0(0);
}
if(unaccounted & AV_CH_LAYOUT_STEREO){
if(out_ch_layout & AV_CH_FRONT_CENTER){
matrix[FRONT_CENTER][ FRONT_LEFT]+= M_SQRT1_2;
matrix[FRONT_CENTER][FRONT_RIGHT]+= M_SQRT1_2;
if(in_ch_layout & AV_CH_FRONT_CENTER)
matrix[FRONT_CENTER][ FRONT_CENTER] = center_mix_level*sqrt(2);
}else
av_assert0(0);
}
if(unaccounted & AV_CH_BACK_CENTER){
if(out_ch_layout & AV_CH_BACK_LEFT){
matrix[ BACK_LEFT][BACK_CENTER]+= M_SQRT1_2;
matrix[BACK_RIGHT][BACK_CENTER]+= M_SQRT1_2;
}else if(out_ch_layout & AV_CH_SIDE_LEFT){
matrix[ SIDE_LEFT][BACK_CENTER]+= M_SQRT1_2;
matrix[SIDE_RIGHT][BACK_CENTER]+= M_SQRT1_2;
}else if(out_ch_layout & AV_CH_FRONT_LEFT){
if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY ||
matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
if (unaccounted & (AV_CH_BACK_LEFT | AV_CH_SIDE_LEFT)) {
matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
} else {
matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level;
}
} else {
matrix[ FRONT_LEFT][BACK_CENTER]+= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_CENTER]+= surround_mix_level * M_SQRT1_2;
}
}else if(out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][BACK_CENTER]+= surround_mix_level * M_SQRT1_2;
}else
av_assert0(0);
}
if(unaccounted & AV_CH_BACK_LEFT){
if(out_ch_layout & AV_CH_BACK_CENTER){
matrix[BACK_CENTER][ BACK_LEFT]+= M_SQRT1_2;
matrix[BACK_CENTER][BACK_RIGHT]+= M_SQRT1_2;
}else if(out_ch_layout & AV_CH_SIDE_LEFT){
if(in_ch_layout & AV_CH_SIDE_LEFT){
matrix[ SIDE_LEFT][ BACK_LEFT]+= M_SQRT1_2;
matrix[SIDE_RIGHT][BACK_RIGHT]+= M_SQRT1_2;
}else{
matrix[ SIDE_LEFT][ BACK_LEFT]+= 1.0;
matrix[SIDE_RIGHT][BACK_RIGHT]+= 1.0;
}
}else if(out_ch_layout & AV_CH_FRONT_LEFT){
if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) {
matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * M_SQRT1_2;
} else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * SQRT3_2;
matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * SQRT3_2;
} else {
matrix[ FRONT_LEFT][ BACK_LEFT] += surround_mix_level;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level;
}
}else if(out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][BACK_LEFT ]+= surround_mix_level*M_SQRT1_2;
matrix[ FRONT_CENTER][BACK_RIGHT]+= surround_mix_level*M_SQRT1_2;
}else
av_assert0(0);
}
if(unaccounted & AV_CH_SIDE_LEFT){
if(out_ch_layout & AV_CH_BACK_LEFT){
/* if back channels do not exist in the input, just copy side
channels to back channels, otherwise mix side into back */
if (in_ch_layout & AV_CH_BACK_LEFT) {
matrix[BACK_LEFT ][SIDE_LEFT ] += M_SQRT1_2;
matrix[BACK_RIGHT][SIDE_RIGHT] += M_SQRT1_2;
} else {
matrix[BACK_LEFT ][SIDE_LEFT ] += 1.0;
matrix[BACK_RIGHT][SIDE_RIGHT] += 1.0;
}
}else if(out_ch_layout & AV_CH_BACK_CENTER){
matrix[BACK_CENTER][ SIDE_LEFT]+= M_SQRT1_2;
matrix[BACK_CENTER][SIDE_RIGHT]+= M_SQRT1_2;
}else if(out_ch_layout & AV_CH_FRONT_LEFT){
if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) {
matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2;
} else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * SQRT3_2;
matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * SQRT3_2;
} else {
matrix[ FRONT_LEFT][ SIDE_LEFT] += surround_mix_level;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level;
}
}else if(out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][SIDE_LEFT ]+= surround_mix_level * M_SQRT1_2;
matrix[ FRONT_CENTER][SIDE_RIGHT]+= surround_mix_level * M_SQRT1_2;
}else
av_assert0(0);
}
if(unaccounted & AV_CH_FRONT_LEFT_OF_CENTER){
if(out_ch_layout & AV_CH_FRONT_LEFT){
matrix[ FRONT_LEFT][ FRONT_LEFT_OF_CENTER]+= 1.0;
matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER]+= 1.0;
}else if(out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][ FRONT_LEFT_OF_CENTER]+= M_SQRT1_2;
matrix[ FRONT_CENTER][FRONT_RIGHT_OF_CENTER]+= M_SQRT1_2;
}else
av_assert0(0);
}
/* mix LFE into front left/right or center */
if (unaccounted & AV_CH_LOW_FREQUENCY) {
if (out_ch_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][LOW_FREQUENCY] += lfe_mix_level;
} else if (out_ch_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2;
} else
av_assert0(0);
}
for(out_i=i=0; i<64; i++){
double sum=0;
int in_i=0;
if((out_ch_layout & (1ULL<<i)) == 0)
continue;
for(j=0; j<64; j++){
if((in_ch_layout & (1ULL<<j)) == 0)
continue;
if (i < FF_ARRAY_ELEMS(matrix) && j < FF_ARRAY_ELEMS(matrix[0]))
matrix_param[stride*out_i + in_i] = matrix[i][j];
else
matrix_param[stride*out_i + in_i] = i == j && (in_ch_layout & out_ch_layout & (1ULL<<i));
sum += fabs(matrix_param[stride*out_i + in_i]);
in_i++;
}
maxcoef= FFMAX(maxcoef, sum);
out_i++;
}
if(rematrix_volume < 0)
maxcoef = -rematrix_volume;
if(maxcoef > maxval || rematrix_volume < 0){
maxcoef /= maxval;
for(i=0; i<SWR_CH_MAX; i++)
for(j=0; j<SWR_CH_MAX; j++){
matrix_param[stride*i + j] /= maxcoef;
}
}
if(rematrix_volume > 0){
for(i=0; i<SWR_CH_MAX; i++)
for(j=0; j<SWR_CH_MAX; j++){
matrix_param[stride*i + j] *= rematrix_volume;
}
}
av_log(log_context, AV_LOG_DEBUG, "Matrix coefficients:\n");
for(i=0; i<av_get_channel_layout_nb_channels(out_ch_layout); i++){
const char *c =
av_get_channel_name(av_channel_layout_extract_channel(out_ch_layout, i));
av_log(log_context, AV_LOG_DEBUG, "%s: ", c ? c : "?");
for(j=0; j<av_get_channel_layout_nb_channels(in_ch_layout); j++){
c = av_get_channel_name(av_channel_layout_extract_channel(in_ch_layout, j));
av_log(log_context, AV_LOG_DEBUG, "%s:%f ", c ? c : "?", matrix_param[stride*i + j]);
}
av_log(log_context, AV_LOG_DEBUG, "\n");
}
return 0;
}
av_cold static int auto_matrix(SwrContext *s)
{
double maxval;
int ret;
if (s->rematrix_maxval > 0) {
maxval = s->rematrix_maxval;
} else if ( av_get_packed_sample_fmt(s->out_sample_fmt) < AV_SAMPLE_FMT_FLT
|| av_get_packed_sample_fmt(s->int_sample_fmt) < AV_SAMPLE_FMT_FLT) {
maxval = 1.0;
} else
maxval = INT_MAX;
memset(s->matrix, 0, sizeof(s->matrix));
ret = swr_build_matrix(s->in_ch_layout, s->out_ch_layout,
s->clev, s->slev, s->lfe_mix_level,
maxval, s->rematrix_volume, (double*)s->matrix,
s->matrix[1] - s->matrix[0], s->matrix_encoding, s);
if (ret >= 0 && s->int_sample_fmt == AV_SAMPLE_FMT_FLTP) {
int i, j;
for (i = 0; i < FF_ARRAY_ELEMS(s->matrix[0]); i++)
for (j = 0; j < FF_ARRAY_ELEMS(s->matrix[0]); j++)
s->matrix_flt[i][j] = s->matrix[i][j];
}
return ret;
}
av_cold int swri_rematrix_init(SwrContext *s){
int i, j;
int nb_in = s->used_ch_count;
int nb_out = s->out.ch_count;
s->mix_any_f = NULL;
if (!s->rematrix_custom) {
int r = auto_matrix(s);
if (r)
return r;
}
if (s->midbuf.fmt == AV_SAMPLE_FMT_S16P){
int maxsum = 0;
s->native_matrix = av_calloc(nb_in * nb_out, sizeof(int));
s->native_one = av_mallocz(sizeof(int));
if (!s->native_matrix || !s->native_one)
return AVERROR(ENOMEM);
for (i = 0; i < nb_out; i++) {
double rem = 0;
int sum = 0;
for (j = 0; j < nb_in; j++) {
double target = s->matrix[i][j] * 32768 + rem;
((int*)s->native_matrix)[i * nb_in + j] = lrintf(target);
rem += target - ((int*)s->native_matrix)[i * nb_in + j];
sum += FFABS(((int*)s->native_matrix)[i * nb_in + j]);
}
maxsum = FFMAX(maxsum, sum);
}
*((int*)s->native_one) = 32768;
if (maxsum <= 32768) {
s->mix_1_1_f = (mix_1_1_func_type*)copy_s16;
s->mix_2_1_f = (mix_2_1_func_type*)sum2_s16;
s->mix_any_f = (mix_any_func_type*)get_mix_any_func_s16(s);
} else {
s->mix_1_1_f = (mix_1_1_func_type*)copy_clip_s16;
s->mix_2_1_f = (mix_2_1_func_type*)sum2_clip_s16;
s->mix_any_f = (mix_any_func_type*)get_mix_any_func_clip_s16(s);
}
}else if(s->midbuf.fmt == AV_SAMPLE_FMT_FLTP){
s->native_matrix = av_calloc(nb_in * nb_out, sizeof(float));
s->native_one = av_mallocz(sizeof(float));
if (!s->native_matrix || !s->native_one)
return AVERROR(ENOMEM);
for (i = 0; i < nb_out; i++)
for (j = 0; j < nb_in; j++)
((float*)s->native_matrix)[i * nb_in + j] = s->matrix[i][j];
*((float*)s->native_one) = 1.0;
s->mix_1_1_f = (mix_1_1_func_type*)copy_float;
s->mix_2_1_f = (mix_2_1_func_type*)sum2_float;
s->mix_any_f = (mix_any_func_type*)get_mix_any_func_float(s);
}else if(s->midbuf.fmt == AV_SAMPLE_FMT_DBLP){
s->native_matrix = av_calloc(nb_in * nb_out, sizeof(double));
s->native_one = av_mallocz(sizeof(double));
if (!s->native_matrix || !s->native_one)
return AVERROR(ENOMEM);
for (i = 0; i < nb_out; i++)
for (j = 0; j < nb_in; j++)
((double*)s->native_matrix)[i * nb_in + j] = s->matrix[i][j];
*((double*)s->native_one) = 1.0;
s->mix_1_1_f = (mix_1_1_func_type*)copy_double;
s->mix_2_1_f = (mix_2_1_func_type*)sum2_double;
s->mix_any_f = (mix_any_func_type*)get_mix_any_func_double(s);
}else if(s->midbuf.fmt == AV_SAMPLE_FMT_S32P){
s->native_one = av_mallocz(sizeof(int));
if (!s->native_one)
return AVERROR(ENOMEM);
s->native_matrix = av_calloc(nb_in * nb_out, sizeof(int));
if (!s->native_matrix) {
av_freep(&s->native_one);
return AVERROR(ENOMEM);
}
for (i = 0; i < nb_out; i++) {
double rem = 0;
for (j = 0; j < nb_in; j++) {
double target = s->matrix[i][j] * 32768 + rem;
((int*)s->native_matrix)[i * nb_in + j] = lrintf(target);
rem += target - ((int*)s->native_matrix)[i * nb_in + j];
}
}
*((int*)s->native_one) = 32768;
s->mix_1_1_f = (mix_1_1_func_type*)copy_s32;
s->mix_2_1_f = (mix_2_1_func_type*)sum2_s32;
s->mix_any_f = (mix_any_func_type*)get_mix_any_func_s32(s);
}else
av_assert0(0);
//FIXME quantize for integeres
for (i = 0; i < SWR_CH_MAX; i++) {
int ch_in=0;
for (j = 0; j < SWR_CH_MAX; j++) {
s->matrix32[i][j]= lrintf(s->matrix[i][j] * 32768);
if(s->matrix[i][j])
s->matrix_ch[i][++ch_in]= j;
}
s->matrix_ch[i][0]= ch_in;
}
if(HAVE_X86ASM && HAVE_MMX)
return swri_rematrix_init_x86(s);
return 0;
}
av_cold void swri_rematrix_free(SwrContext *s){
av_freep(&s->native_matrix);
av_freep(&s->native_one);
av_freep(&s->native_simd_matrix);
av_freep(&s->native_simd_one);
}
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy){
int out_i, in_i, i, j;
int len1 = 0;
int off = 0;
if(s->mix_any_f) {
s->mix_any_f(out->ch, (const uint8_t **)in->ch, s->native_matrix, len);
return 0;
}
if(s->mix_2_1_simd || s->mix_1_1_simd){
len1= len&~15;
off = len1 * out->bps;
}
av_assert0(!s->out_ch_layout || out->ch_count == av_get_channel_layout_nb_channels(s->out_ch_layout));
av_assert0(!s-> in_ch_layout || in ->ch_count == av_get_channel_layout_nb_channels(s-> in_ch_layout));
for(out_i=0; out_i<out->ch_count; out_i++){
switch(s->matrix_ch[out_i][0]){
case 0:
if(mustcopy)
memset(out->ch[out_i], 0, len * av_get_bytes_per_sample(s->int_sample_fmt));
break;
case 1:
in_i= s->matrix_ch[out_i][1];
if(s->matrix[out_i][in_i]!=1.0){
if(s->mix_1_1_simd && len1)
s->mix_1_1_simd(out->ch[out_i] , in->ch[in_i] , s->native_simd_matrix, in->ch_count*out_i + in_i, len1);
if(len != len1)
s->mix_1_1_f (out->ch[out_i]+off, in->ch[in_i]+off, s->native_matrix, in->ch_count*out_i + in_i, len-len1);
}else if(mustcopy){
memcpy(out->ch[out_i], in->ch[in_i], len*out->bps);
}else{
out->ch[out_i]= in->ch[in_i];
}
break;
case 2: {
int in_i1 = s->matrix_ch[out_i][1];
int in_i2 = s->matrix_ch[out_i][2];
if(s->mix_2_1_simd && len1)
s->mix_2_1_simd(out->ch[out_i] , in->ch[in_i1] , in->ch[in_i2] , s->native_simd_matrix, in->ch_count*out_i + in_i1, in->ch_count*out_i + in_i2, len1);
else
s->mix_2_1_f (out->ch[out_i] , in->ch[in_i1] , in->ch[in_i2] , s->native_matrix, in->ch_count*out_i + in_i1, in->ch_count*out_i + in_i2, len1);
if(len != len1)
s->mix_2_1_f (out->ch[out_i]+off, in->ch[in_i1]+off, in->ch[in_i2]+off, s->native_matrix, in->ch_count*out_i + in_i1, in->ch_count*out_i + in_i2, len-len1);
break;}
default:
if(s->int_sample_fmt == AV_SAMPLE_FMT_FLTP){
for(i=0; i<len; i++){
float v=0;
for(j=0; j<s->matrix_ch[out_i][0]; j++){
in_i= s->matrix_ch[out_i][1+j];
v+= ((float*)in->ch[in_i])[i] * s->matrix_flt[out_i][in_i];
}
((float*)out->ch[out_i])[i]= v;
}
}else if(s->int_sample_fmt == AV_SAMPLE_FMT_DBLP){
for(i=0; i<len; i++){
double v=0;
for(j=0; j<s->matrix_ch[out_i][0]; j++){
in_i= s->matrix_ch[out_i][1+j];
v+= ((double*)in->ch[in_i])[i] * s->matrix[out_i][in_i];
}
((double*)out->ch[out_i])[i]= v;
}
}else{
for(i=0; i<len; i++){
int v=0;
for(j=0; j<s->matrix_ch[out_i][0]; j++){
in_i= s->matrix_ch[out_i][1+j];
v+= ((int16_t*)in->ch[in_i])[i] * s->matrix32[out_i][in_i];
}
((int16_t*)out->ch[out_i])[i]= (v + 16384)>>15;
}
}
}
}
return 0;
}

111
trunk/3rdparty/ffmpeg-4-fit/libswresample/rematrix_template.c vendored Normal file
View file

@ -0,0 +1,111 @@
/*
* Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#if defined(TEMPLATE_REMATRIX_FLT)
# define R(x) x
# define SAMPLE float
# define COEFF float
# define INTER float
# define RENAME(x) x ## _float
#elif defined(TEMPLATE_REMATRIX_DBL)
# define R(x) x
# define SAMPLE double
# define COEFF double
# define INTER double
# define RENAME(x) x ## _double
#elif defined(TEMPLATE_REMATRIX_S16)
# define SAMPLE int16_t
# define COEFF int
# define INTER int
# ifdef TEMPLATE_CLIP
# define R(x) av_clip_int16(((x) + 16384)>>15)
# define RENAME(x) x ## _clip_s16
# else
# define R(x) (((x) + 16384)>>15)
# define RENAME(x) x ## _s16
# endif
#elif defined(TEMPLATE_REMATRIX_S32)
# define R(x) (((x) + 16384)>>15)
# define SAMPLE int32_t
# define COEFF int
# define INTER int64_t
# define RENAME(x) x ## _s32
#endif
typedef void (RENAME(mix_any_func_type))(SAMPLE **out, const SAMPLE **in1, COEFF *coeffp, integer len);
static void RENAME(sum2)(SAMPLE *out, const SAMPLE *in1, const SAMPLE *in2, COEFF *coeffp, integer index1, integer index2, integer len){
int i;
INTER coeff1 = coeffp[index1];
INTER coeff2 = coeffp[index2];
for(i=0; i<len; i++)
out[i] = R(coeff1*in1[i] + coeff2*in2[i]);
}
static void RENAME(copy)(SAMPLE *out, const SAMPLE *in, COEFF *coeffp, integer index, integer len){
int i;
INTER coeff = coeffp[index];
for(i=0; i<len; i++)
out[i] = R(coeff*in[i]);
}
static void RENAME(mix6to2)(SAMPLE **out, const SAMPLE **in, COEFF *coeffp, integer len){
int i;
for(i=0; i<len; i++) {
INTER t = in[2][i]*(INTER)coeffp[0*6+2] + in[3][i]*(INTER)coeffp[0*6+3];
out[0][i] = R(t + in[0][i]*(INTER)coeffp[0*6+0] + in[4][i]*(INTER)coeffp[0*6+4]);
out[1][i] = R(t + in[1][i]*(INTER)coeffp[1*6+1] + in[5][i]*(INTER)coeffp[1*6+5]);
}
}
static void RENAME(mix8to2)(SAMPLE **out, const SAMPLE **in, COEFF *coeffp, integer len){
int i;
for(i=0; i<len; i++) {
INTER t = in[2][i]*(INTER)coeffp[0*8+2] + in[3][i]*(INTER)coeffp[0*8+3];
out[0][i] = R(t + in[0][i]*(INTER)coeffp[0*8+0] + in[4][i]*(INTER)coeffp[0*8+4] + in[6][i]*(INTER)coeffp[0*8+6]);
out[1][i] = R(t + in[1][i]*(INTER)coeffp[1*8+1] + in[5][i]*(INTER)coeffp[1*8+5] + in[7][i]*(INTER)coeffp[1*8+7]);
}
}
static RENAME(mix_any_func_type) *RENAME(get_mix_any_func)(SwrContext *s){
if( s->out_ch_layout == AV_CH_LAYOUT_STEREO && (s->in_ch_layout == AV_CH_LAYOUT_5POINT1 || s->in_ch_layout == AV_CH_LAYOUT_5POINT1_BACK)
&& s->matrix[0][2] == s->matrix[1][2] && s->matrix[0][3] == s->matrix[1][3]
&& !s->matrix[0][1] && !s->matrix[0][5] && !s->matrix[1][0] && !s->matrix[1][4]
)
return RENAME(mix6to2);
if( s->out_ch_layout == AV_CH_LAYOUT_STEREO && s->in_ch_layout == AV_CH_LAYOUT_7POINT1
&& s->matrix[0][2] == s->matrix[1][2] && s->matrix[0][3] == s->matrix[1][3]
&& !s->matrix[0][1] && !s->matrix[0][5] && !s->matrix[1][0] && !s->matrix[1][4]
&& !s->matrix[0][7] && !s->matrix[1][6]
)
return RENAME(mix8to2);
return NULL;
}
#undef R
#undef SAMPLE
#undef COEFF
#undef INTER
#undef RENAME

622
trunk/3rdparty/ffmpeg-4-fit/libswresample/resample.c vendored Normal file
View file

@ -0,0 +1,622 @@
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
* bessel function: Copyright (c) 2006 Xiaogang Zhang
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avassert.h"
#include "resample.h"
static inline double eval_poly(const double *coeff, int size, double x) {
double sum = coeff[size-1];
int i;
for (i = size-2; i >= 0; --i) {
sum *= x;
sum += coeff[i];
}
return sum;
}
/**
* 0th order modified bessel function of the first kind.
* Algorithm taken from the Boost project, source:
* https://searchcode.com/codesearch/view/14918379/
* Use, modification and distribution are subject to the
* Boost Software License, Version 1.0 (see notice below).
* Boost Software License - Version 1.0 - August 17th, 2003
Permission is hereby granted, free of charge, to any person or organization
obtaining a copy of the software and accompanying documentation covered by
this license (the "Software") to use, reproduce, display, distribute,
execute, and transmit the Software, and to prepare derivative works of the
Software, and to permit third-parties to whom the Software is furnished to
do so, all subject to the following:
The copyright notices in the Software and this entire statement, including
the above license grant, this restriction and the following disclaimer,
must be included in all copies of the Software, in whole or in part, and
all derivative works of the Software, unless such copies or derivative
works are solely in the form of machine-executable object code generated by
a source language processor.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
DEALINGS IN THE SOFTWARE.
*/
static double bessel(double x) {
// Modified Bessel function of the first kind of order zero
// minimax rational approximations on intervals, see
// Blair and Edwards, Chalk River Report AECL-4928, 1974
static const double p1[] = {
-2.2335582639474375249e+15,
-5.5050369673018427753e+14,
-3.2940087627407749166e+13,
-8.4925101247114157499e+11,
-1.1912746104985237192e+10,
-1.0313066708737980747e+08,
-5.9545626019847898221e+05,
-2.4125195876041896775e+03,
-7.0935347449210549190e+00,
-1.5453977791786851041e-02,
-2.5172644670688975051e-05,
-3.0517226450451067446e-08,
-2.6843448573468483278e-11,
-1.5982226675653184646e-14,
-5.2487866627945699800e-18,
};
static const double q1[] = {
-2.2335582639474375245e+15,
7.8858692566751002988e+12,
-1.2207067397808979846e+10,
1.0377081058062166144e+07,
-4.8527560179962773045e+03,
1.0,
};
static const double p2[] = {
-2.2210262233306573296e-04,
1.3067392038106924055e-02,
-4.4700805721174453923e-01,
5.5674518371240761397e+00,
-2.3517945679239481621e+01,
3.1611322818701131207e+01,
-9.6090021968656180000e+00,
};
static const double q2[] = {
-5.5194330231005480228e-04,
3.2547697594819615062e-02,
-1.1151759188741312645e+00,
1.3982595353892851542e+01,
-6.0228002066743340583e+01,
8.5539563258012929600e+01,
-3.1446690275135491500e+01,
1.0,
};
double y, r, factor;
if (x == 0)
return 1.0;
x = fabs(x);
if (x <= 15) {
y = x * x;
return eval_poly(p1, FF_ARRAY_ELEMS(p1), y) / eval_poly(q1, FF_ARRAY_ELEMS(q1), y);
}
else {
y = 1 / x - 1.0 / 15;
r = eval_poly(p2, FF_ARRAY_ELEMS(p2), y) / eval_poly(q2, FF_ARRAY_ELEMS(q2), y);
factor = exp(x) / sqrt(x);
return factor * r;
}
}
/**
* builds a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param filter_type filter type
* @param kaiser_beta kaiser window beta
* @return 0 on success, negative on error
*/
static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
int filter_type, double kaiser_beta){
int ph, i;
int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1;
double x, y, w, t, s;
double *tab = av_malloc_array(tap_count+1, sizeof(*tab));
double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut));
const int center= (tap_count-1)/2;
double norm = 0;
int ret = AVERROR(ENOMEM);
if (!tab || !sin_lut)
goto fail;
av_assert0(tap_count == 1 || tap_count % 2 == 0);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
if (factor == 1.0) {
for (ph = 0; ph < ph_nb; ph++)
sin_lut[ph] = sin(M_PI * ph / phase_count) * (center & 1 ? 1 : -1);
}
for(ph = 0; ph < ph_nb; ph++) {
s = sin_lut[ph];
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else if (factor == 1.0)
y = s / x;
else
y = sin(x) / x;
switch(filter_type){
case SWR_FILTER_TYPE_CUBIC:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0*x / (factor*tap_count);
t = -cos(w);
y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t);
break;
case SWR_FILTER_TYPE_KAISER:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
break;
default:
av_assert0(0);
}
tab[i] = y;
s = -s;
if (!ph)
norm += y;
}
/* normalize so that an uniform color remains the same */
switch(c->format){
case AV_SAMPLE_FMT_S16P:
for(i=0;i<tap_count;i++)
((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm));
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];
break;
case AV_SAMPLE_FMT_S32P:
for(i=0;i<tap_count;i++)
((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];
break;
case AV_SAMPLE_FMT_FLTP:
for(i=0;i<tap_count;i++)
((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];
break;
case AV_SAMPLE_FMT_DBLP:
for(i=0;i<tap_count;i++)
((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];
break;
}
}
#if 0
{
#define LEN 1024
int j,k;
double sine[LEN + tap_count];
double filtered[LEN];
double maxff=-2, minff=2, maxsf=-2, minsf=2;
for(i=0; i<LEN; i++){
double ss=0, sf=0, ff=0;
for(j=0; j<LEN+tap_count; j++)
sine[j]= cos(i*j*M_PI/LEN);
for(j=0; j<LEN; j++){
double sum=0;
ph=0;
for(k=0; k<tap_count; k++)
sum += filter[ph * tap_count + k] * sine[k+j];
filtered[j]= sum / (1<<FILTER_SHIFT);
ss+= sine[j + center] * sine[j + center];
ff+= filtered[j] * filtered[j];
sf+= sine[j + center] * filtered[j];
}
ss= sqrt(2*ss/LEN);
ff= sqrt(2*ff/LEN);
sf= 2*sf/LEN;
maxff= FFMAX(maxff, ff);
minff= FFMIN(minff, ff);
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
minff=minsf= 2;
maxff=maxsf= -2;
}
}
}
#endif
ret = 0;
fail:
av_free(tab);
av_free(sin_lut);
return ret;
}
static void resample_free(ResampleContext **cc){
ResampleContext *c = *cc;
if(!c)
return;
av_freep(&c->filter_bank);
av_freep(cc);
}
static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta,
double precision, int cheby, int exact_rational)
{
double cutoff = cutoff0? cutoff0 : 0.97;
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
int phase_count_compensation = phase_count;
int filter_length = FFMAX((int)ceil(filter_size/factor), 1);
if (filter_length > 1)
filter_length = FFALIGN(filter_length, 2);
if (exact_rational) {
int phase_count_exact, phase_count_exact_den;
av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX);
if (phase_count_exact <= phase_count) {
phase_count_compensation = phase_count_exact * (phase_count / phase_count_exact);
phase_count = phase_count_exact;
}
}
if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor
|| c->filter_length != filter_length || c->format != format
|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
resample_free(&c);
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
c->format= format;
c->felem_size= av_get_bytes_per_sample(c->format);
switch(c->format){
case AV_SAMPLE_FMT_S16P:
c->filter_shift = 15;
break;
case AV_SAMPLE_FMT_S32P:
c->filter_shift = 30;
break;
case AV_SAMPLE_FMT_FLTP:
case AV_SAMPLE_FMT_DBLP:
c->filter_shift = 0;
break;
default:
av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
av_assert0(0);
}
if (filter_size/factor > INT32_MAX/256) {
av_log(NULL, AV_LOG_ERROR, "Filter length too large\n");
goto error;
}
c->phase_count = phase_count;
c->linear = linear;
c->factor = factor;
c->filter_length = filter_length;
c->filter_alloc = FFALIGN(c->filter_length, 8);
c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size);
c->filter_type = filter_type;
c->kaiser_beta = kaiser_beta;
c->phase_count_compensation = phase_count_compensation;
if (!c->filter_bank)
goto error;
if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta))
goto error;
memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
}
c->compensation_distance= 0;
if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
goto error;
while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
c->dst_incr *= 2;
c->src_incr *= 2;
}
c->ideal_dst_incr = c->dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
c->index= -phase_count*((c->filter_length-1)/2);
c->frac= 0;
swri_resample_dsp_init(c);
return c;
error:
av_freep(&c->filter_bank);
av_free(c);
return NULL;
}
static int rebuild_filter_bank_with_compensation(ResampleContext *c)
{
uint8_t *new_filter_bank;
int new_src_incr, new_dst_incr;
int phase_count = c->phase_count_compensation;
int ret;
if (phase_count == c->phase_count)
return 0;
av_assert0(!c->frac && !c->dst_incr_mod);
new_filter_bank = av_calloc(c->filter_alloc, (phase_count + 1) * c->felem_size);
if (!new_filter_bank)
return AVERROR(ENOMEM);
ret = build_filter(c, new_filter_bank, c->factor, c->filter_length, c->filter_alloc,
phase_count, 1 << c->filter_shift, c->filter_type, c->kaiser_beta);
if (ret < 0) {
av_freep(&new_filter_bank);
return ret;
}
memcpy(new_filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, new_filter_bank, (c->filter_alloc-1)*c->felem_size);
memcpy(new_filter_bank + (c->filter_alloc*phase_count )*c->felem_size, new_filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
if (!av_reduce(&new_src_incr, &new_dst_incr, c->src_incr,
c->dst_incr * (int64_t)(phase_count/c->phase_count), INT32_MAX/2))
{
av_freep(&new_filter_bank);
return AVERROR(EINVAL);
}
c->src_incr = new_src_incr;
c->dst_incr = new_dst_incr;
while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
c->dst_incr *= 2;
c->src_incr *= 2;
}
c->ideal_dst_incr = c->dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
c->index *= phase_count / c->phase_count;
c->phase_count = phase_count;
av_freep(&c->filter_bank);
c->filter_bank = new_filter_bank;
return 0;
}
static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
int ret;
if (compensation_distance && sample_delta) {
ret = rebuild_filter_bank_with_compensation(c);
if (ret < 0)
return ret;
}
c->compensation_distance= compensation_distance;
if (compensation_distance)
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
else
c->dst_incr = c->ideal_dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
return 0;
}
static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i;
int av_unused mm_flags = av_get_cpu_flags();
int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 &&
(mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2;
int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;
if (c->compensation_distance)
dst_size = FFMIN(dst_size, c->compensation_distance);
src_size = FFMIN(src_size, max_src_size);
*consumed = 0;
if (c->filter_length == 1 && c->phase_count == 1) {
int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*c->index;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
int new_size = (src_size * (int64_t)c->src_incr - c->frac + c->dst_incr - 1) / c->dst_incr;
dst_size = FFMAX(FFMIN(dst_size, new_size), 0);
if (dst_size > 0) {
for (i = 0; i < dst->ch_count; i++) {
c->dsp.resample_one(dst->ch[i], src->ch[i], dst_size, index2, incr);
if (i+1 == dst->ch_count) {
c->index += dst_size * c->dst_incr_div;
c->index += (c->frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;
av_assert2(c->index >= 0);
*consumed = c->index;
c->frac = (c->frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;
c->index = 0;
}
}
}
} else {
int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;
int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
int (*resample_func)(struct ResampleContext *c, void *dst,
const void *src, int n, int update_ctx);
dst_size = FFMAX(FFMIN(dst_size, delta_n), 0);
if (dst_size > 0) {
/* resample_linear and resample_common should have same behavior
* when frac and dst_incr_mod are zero */
resample_func = (c->linear && (c->frac || c->dst_incr_mod)) ?
c->dsp.resample_linear : c->dsp.resample_common;
for (i = 0; i < dst->ch_count; i++)
*consumed = resample_func(c, dst->ch[i], src->ch[i], dst_size, i+1 == dst->ch_count);
}
}
if(need_emms)
emms_c();
if (c->compensation_distance) {
c->compensation_distance -= dst_size;
if (!c->compensation_distance) {
c->dst_incr = c->ideal_dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
}
}
return dst_size;
}
static int64_t get_delay(struct SwrContext *s, int64_t base){
ResampleContext *c = s->resample;
int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
num *= c->phase_count;
num -= c->index;
num *= c->src_incr;
num -= c->frac;
return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr * c->phase_count);
}
static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
ResampleContext *c = s->resample;
// The + 2 are added to allow implementations to be slightly inaccurate, they should not be needed currently.
// They also make it easier to proof that changes and optimizations do not
// break the upper bound.
int64_t num = s->in_buffer_count + 2LL + in_samples;
num *= c->phase_count;
num -= c->index;
num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) * c->phase_count, AV_ROUND_UP) + 2;
if (c->compensation_distance) {
if (num > INT_MAX)
return AVERROR(EINVAL);
num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);
}
return num;
}
static int resample_flush(struct SwrContext *s) {
ResampleContext *c = s->resample;
AudioData *a= &s->in_buffer;
int i, j, ret;
int reflection = (FFMIN(s->in_buffer_count, c->filter_length) + 1) / 2;
if((ret = swri_realloc_audio(a, s->in_buffer_index + s->in_buffer_count + reflection)) < 0)
return ret;
av_assert0(a->planar);
for(i=0; i<a->ch_count; i++){
for(j=0; j<reflection; j++){
memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
}
}
s->in_buffer_count += reflection;
return 0;
}
// in fact the whole handle multiple ridiculously small buffers might need more thinking...
static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,
int in_count, int *out_idx, int *out_sz)
{
int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;
if (c->index >= 0)
return 0;
if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)
return res;
// copy
for (n = *out_sz; n < num; n++) {
for (ch = 0; ch < src->ch_count; ch++) {
memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);
}
}
// if not enough data is in, return and wait for more
if (num < c->filter_length + 1) {
*out_sz = num;
*out_idx = c->filter_length;
return INT_MAX;
}
// else invert
for (n = 1; n <= c->filter_length; n++) {
for (ch = 0; ch < src->ch_count; ch++) {
memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),
dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
c->felem_size);
}
}
res = num - *out_sz;
*out_idx = c->filter_length;
while (c->index < 0) {
--*out_idx;
c->index += c->phase_count;
}
*out_sz = FFMAX(*out_sz + c->filter_length,
1 + c->filter_length * 2) - *out_idx;
return FFMAX(res, 0);
}
struct Resampler const swri_resampler={
resample_init,
resample_free,
multiple_resample,
resample_flush,
set_compensation,
get_delay,
invert_initial_buffer,
get_out_samples,
};

68
trunk/3rdparty/ffmpeg-4-fit/libswresample/resample.h vendored Normal file
View file

@ -0,0 +1,68 @@
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_RESAMPLE_H
#define SWRESAMPLE_RESAMPLE_H
#include "libavutil/log.h"
#include "libavutil/samplefmt.h"
#include "swresample_internal.h"
typedef struct ResampleContext {
const AVClass *av_class;
uint8_t *filter_bank;
int filter_length;
int filter_alloc;
int ideal_dst_incr;
int dst_incr;
int dst_incr_div;
int dst_incr_mod;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_count;
int linear;
enum SwrFilterType filter_type;
double kaiser_beta;
double factor;
enum AVSampleFormat format;
int felem_size;
int filter_shift;
int phase_count_compensation; /* desired phase_count when compensation is enabled */
struct {
void (*resample_one)(void *dst, const void *src,
int n, int64_t index, int64_t incr);
int (*resample_common)(struct ResampleContext *c, void *dst,
const void *src, int n, int update_ctx);
int (*resample_linear)(struct ResampleContext *c, void *dst,
const void *src, int n, int update_ctx);
} dsp;
} ResampleContext;
void swri_resample_dsp_init(ResampleContext *c);
void swri_resample_dsp_x86_init(ResampleContext *c);
void swri_resample_dsp_arm_init(ResampleContext *c);
void swri_resample_dsp_aarch64_init(ResampleContext *c);
#endif /* SWRESAMPLE_RESAMPLE_H */

74
trunk/3rdparty/ffmpeg-4-fit/libswresample/resample_dsp.c vendored Normal file
View file

@ -0,0 +1,74 @@
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "resample.h"
#define TEMPLATE_RESAMPLE_S16
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S16
#define TEMPLATE_RESAMPLE_S32
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S32
#define TEMPLATE_RESAMPLE_FLT
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT
#define TEMPLATE_RESAMPLE_DBL
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_DBL
void swri_resample_dsp_init(ResampleContext *c)
{
switch(c->format){
case AV_SAMPLE_FMT_S16P:
c->dsp.resample_one = resample_one_int16;
c->dsp.resample_common = resample_common_int16;
c->dsp.resample_linear = resample_linear_int16;
break;
case AV_SAMPLE_FMT_S32P:
c->dsp.resample_one = resample_one_int32;
c->dsp.resample_common = resample_common_int32;
c->dsp.resample_linear = resample_linear_int32;
break;
case AV_SAMPLE_FMT_FLTP:
c->dsp.resample_one = resample_one_float;
c->dsp.resample_common = resample_common_float;
c->dsp.resample_linear = resample_linear_float;
break;
case AV_SAMPLE_FMT_DBLP:
c->dsp.resample_one = resample_one_double;
c->dsp.resample_common = resample_common_double;
c->dsp.resample_linear = resample_linear_double;
break;
}
if (ARCH_X86) swri_resample_dsp_x86_init(c);
else if (ARCH_ARM) swri_resample_dsp_arm_init(c);
else if (ARCH_AARCH64) swri_resample_dsp_aarch64_init(c);
}

212
trunk/3rdparty/ffmpeg-4-fit/libswresample/resample_template.c vendored Normal file
View file

@ -0,0 +1,212 @@
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#if defined(TEMPLATE_RESAMPLE_DBL)
# define RENAME(N) N ## _double
# define FILTER_SHIFT 0
# define DELEM double
# define FELEM double
# define FELEM2 double
# define FOFFSET 0
# define OUT(d, v) d = v
#elif defined(TEMPLATE_RESAMPLE_FLT)
# define RENAME(N) N ## _float
# define FILTER_SHIFT 0
# define DELEM float
# define FELEM float
# define FELEM2 float
# define FOFFSET 0
# define OUT(d, v) d = v
#elif defined(TEMPLATE_RESAMPLE_S32)
# define RENAME(N) N ## _int32
# define FILTER_SHIFT 30
# define DELEM int32_t
# define FELEM int32_t
# define FELEM2 int64_t
# define FELEM_MAX INT32_MAX
# define FELEM_MIN INT32_MIN
# define FOFFSET (1<<(FILTER_SHIFT-1))
# define OUT(d, v) (d) = av_clipl_int32((v)>>FILTER_SHIFT)
#elif defined(TEMPLATE_RESAMPLE_S16)
# define RENAME(N) N ## _int16
# define FILTER_SHIFT 15
# define DELEM int16_t
# define FELEM int16_t
# define FELEM2 int32_t
# define FELEML int64_t
# define FELEM_MAX INT16_MAX
# define FELEM_MIN INT16_MIN
# define FOFFSET (1<<(FILTER_SHIFT-1))
# define OUT(d, v) (d) = av_clip_int16((v)>>FILTER_SHIFT)
#endif
static void RENAME(resample_one)(void *dest, const void *source,
int dst_size, int64_t index2, int64_t incr)
{
DELEM *dst = dest;
const DELEM *src = source;
int dst_index;
for (dst_index = 0; dst_index < dst_size; dst_index++) {
dst[dst_index] = src[index2 >> 32];
index2 += incr;
}
}
static int RENAME(resample_common)(ResampleContext *c,
void *dest, const void *source,
int n, int update_ctx)
{
DELEM *dst = dest;
const DELEM *src = source;
int dst_index;
int index= c->index;
int frac= c->frac;
int sample_index = 0;
while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
for (dst_index = 0; dst_index < n; dst_index++) {
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index;
FELEM2 val = FOFFSET;
FELEM2 val2= 0;
int i;
for (i = 0; i + 1 < c->filter_length; i+=2) {
val += src[sample_index + i ] * (FELEM2)filter[i ];
val2 += src[sample_index + i + 1] * (FELEM2)filter[i + 1];
}
if (i < c->filter_length)
val += src[sample_index + i ] * (FELEM2)filter[i ];
#ifdef FELEML
OUT(dst[dst_index], val + (FELEML)val2);
#else
OUT(dst[dst_index], val + val2);
#endif
frac += c->dst_incr_mod;
index += c->dst_incr_div;
if (frac >= c->src_incr) {
frac -= c->src_incr;
index++;
}
while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
}
if(update_ctx){
c->frac= frac;
c->index= index;
}
return sample_index;
}
static int RENAME(resample_linear)(ResampleContext *c,
void *dest, const void *source,
int n, int update_ctx)
{
DELEM *dst = dest;
const DELEM *src = source;
int dst_index;
int index= c->index;
int frac= c->frac;
int sample_index = 0;
#if FILTER_SHIFT == 0
double inv_src_incr = 1.0 / c->src_incr;
#endif
while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
for (dst_index = 0; dst_index < n; dst_index++) {
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index;
FELEM2 val = FOFFSET, v2 = FOFFSET;
int i;
for (i = 0; i < c->filter_length; i++) {
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_alloc];
}
#ifdef FELEML
val += (v2 - val) * (FELEML) frac / c->src_incr;
#else
# if FILTER_SHIFT == 0
val += (v2 - val) * inv_src_incr * frac;
# else
val += (v2 - val) / c->src_incr * frac;
# endif
#endif
OUT(dst[dst_index], val);
frac += c->dst_incr_mod;
index += c->dst_incr_div;
if (frac >= c->src_incr) {
frac -= c->src_incr;
index++;
}
while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
}
if(update_ctx){
c->frac= frac;
c->index= index;
}
return sample_index;
}
#undef RENAME
#undef FILTER_SHIFT
#undef DELEM
#undef FELEM
#undef FELEM2
#undef FELEML
#undef FELEM_MAX
#undef FELEM_MIN
#undef OUT
#undef FOFFSET

949
trunk/3rdparty/ffmpeg-4-fit/libswresample/swresample.c vendored Normal file
View file

@ -0,0 +1,949 @@
/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "swresample_internal.h"
#include "audioconvert.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/internal.h"
#include <float.h>
#define ALIGN 32
#include "libavutil/ffversion.h"
const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
unsigned swresample_version(void)
{
av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
return LIBSWRESAMPLE_VERSION_INT;
}
const char *swresample_configuration(void)
{
return FFMPEG_CONFIGURATION;
}
const char *swresample_license(void)
{
#define LICENSE_PREFIX "libswresample license: "
return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
}
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
if(!s || s->in_convert) // s needs to be allocated but not initialized
return AVERROR(EINVAL);
s->channel_map = channel_map;
return 0;
}
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx){
if(!s) s= swr_alloc();
if(!s) return NULL;
s->log_level_offset= log_offset;
s->log_ctx= log_ctx;
if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
goto fail;
if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
goto fail;
if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
goto fail;
if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
goto fail;
if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
goto fail;
if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
goto fail;
if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> user_in_ch_layout), 0) < 0)
goto fail;
if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->user_out_ch_layout), 0) < 0)
goto fail;
av_opt_set_int(s, "uch", 0, 0);
return s;
fail:
av_log(s, AV_LOG_ERROR, "Failed to set option\n");
swr_free(&s);
return NULL;
}
static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
a->fmt = fmt;
a->bps = av_get_bytes_per_sample(fmt);
a->planar= av_sample_fmt_is_planar(fmt);
if (a->ch_count == 1)
a->planar = 1;
}
static void free_temp(AudioData *a){
av_free(a->data);
memset(a, 0, sizeof(*a));
}
static void clear_context(SwrContext *s){
s->in_buffer_index= 0;
s->in_buffer_count= 0;
s->resample_in_constraint= 0;
memset(s->in.ch, 0, sizeof(s->in.ch));
memset(s->out.ch, 0, sizeof(s->out.ch));
free_temp(&s->postin);
free_temp(&s->midbuf);
free_temp(&s->preout);
free_temp(&s->in_buffer);
free_temp(&s->silence);
free_temp(&s->drop_temp);
free_temp(&s->dither.noise);
free_temp(&s->dither.temp);
swri_audio_convert_free(&s-> in_convert);
swri_audio_convert_free(&s->out_convert);
swri_audio_convert_free(&s->full_convert);
swri_rematrix_free(s);
s->delayed_samples_fixup = 0;
s->flushed = 0;
}
av_cold void swr_free(SwrContext **ss){
SwrContext *s= *ss;
if(s){
clear_context(s);
if (s->resampler)
s->resampler->free(&s->resample);
}
av_freep(ss);
}
av_cold void swr_close(SwrContext *s){
clear_context(s);
}
av_cold int swr_init(struct SwrContext *s){
int ret;
char l1[1024], l2[1024];
clear_context(s);
if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
return AVERROR(EINVAL);
}
if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
return AVERROR(EINVAL);
}
if(s-> in_sample_rate <= 0){
av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate);
return AVERROR(EINVAL);
}
if(s->out_sample_rate <= 0){
av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate);
return AVERROR(EINVAL);
}
s->out.ch_count = s-> user_out_ch_count;
s-> in.ch_count = s-> user_in_ch_count;
s->used_ch_count = s->user_used_ch_count;
s-> in_ch_layout = s-> user_in_ch_layout;
s->out_ch_layout = s->user_out_ch_layout;
s->int_sample_fmt= s->user_int_sample_fmt;
s->dither.method = s->user_dither_method;
if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
s->in_ch_layout = 0;
}
if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
s->out_ch_layout = 0;
}
switch(s->engine){
#if CONFIG_LIBSOXR
case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
#endif
case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
default:
av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
return AVERROR(EINVAL);
}
if(!s->used_ch_count)
s->used_ch_count= s->in.ch_count;
if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
s-> in_ch_layout= 0;
}
if(!s-> in_ch_layout)
s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
if(!s->out_ch_layout)
s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
s->rematrix_custom;
if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
&& av_get_bytes_per_sample(s->out_sample_fmt) <= 2){
s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
}else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
&& !s->rematrix
&& s->out_sample_rate==s->in_sample_rate
&& !(s->flags & SWR_FLAG_RESAMPLE)){
s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
}else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
&& av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
&& !s->rematrix
&& s->out_sample_rate == s->in_sample_rate
&& !(s->flags & SWR_FLAG_RESAMPLE)
&& s->engine != SWR_ENGINE_SOXR){
s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
}else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
}else{
s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
}
}
av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
&&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
&&s->int_sample_fmt != AV_SAMPLE_FMT_S64P
&&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
&&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, s16p/s32p/s64p/fltp/dblp are supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
return AVERROR(EINVAL);
}
set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
set_audiodata_fmt(&s->out, s->out_sample_fmt);
if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
if (!s->async && s->min_compensation >= FLT_MAX/2)
s->async = 1;
s->firstpts =
s->outpts = s->firstpts_in_samples * s->out_sample_rate;
} else
s->firstpts = AV_NOPTS_VALUE;
if (s->async) {
if (s->min_compensation >= FLT_MAX/2)
s->min_compensation = 0.001;
if (s->async > 1.0001) {
s->max_soft_compensation = s->async / (double) s->in_sample_rate;
}
}
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby, s->exact_rational);
if (!s->resample) {
av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
return AVERROR(ENOMEM);
}
}else
s->resampler->free(&s->resample);
if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
&& s->int_sample_fmt != AV_SAMPLE_FMT_S32P
&& s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
&& s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
&& s->resample){
av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16p/s32p/fltp/dblp\n");
ret = AVERROR(EINVAL);
goto fail;
}
#define RSC 1 //FIXME finetune
if(!s-> in.ch_count)
s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
if(!s->used_ch_count)
s->used_ch_count= s->in.ch_count;
if(!s->out.ch_count)
s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
if(!s-> in.ch_count){
av_assert0(!s->in_ch_layout);
av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
ret = AVERROR(EINVAL);
goto fail;
}
av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) {
av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
ret = AVERROR(EINVAL);
goto fail;
}
if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) {
av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
ret = AVERROR(EINVAL);
goto fail;
}
if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
"but there is not enough information to do it\n", l1, l2);
ret = AVERROR(EINVAL);
goto fail;
}
av_assert0(s->used_ch_count);
av_assert0(s->out.ch_count);
s->resample_first= RSC*s->out.ch_count/s->used_ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
s->in_buffer= s->in;
s->silence = s->in;
s->drop_temp= s->out;
if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
goto fail;
if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
return 0;
}
s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
s->int_sample_fmt, s->out.ch_count, NULL, 0);
if (!s->in_convert || !s->out_convert) {
ret = AVERROR(ENOMEM);
goto fail;
}
s->postin= s->in;
s->preout= s->out;
s->midbuf= s->in;
if(s->channel_map){
s->postin.ch_count=
s->midbuf.ch_count= s->used_ch_count;
if(s->resample)
s->in_buffer.ch_count= s->used_ch_count;
}
if(!s->resample_first){
s->midbuf.ch_count= s->out.ch_count;
if(s->resample)
s->in_buffer.ch_count = s->out.ch_count;
}
set_audiodata_fmt(&s->postin, s->int_sample_fmt);
set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
set_audiodata_fmt(&s->preout, s->int_sample_fmt);
if(s->resample){
set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
}
av_assert0(!s->preout.count);
s->dither.noise = s->preout;
s->dither.temp = s->preout;
if (s->dither.method > SWR_DITHER_NS) {
s->dither.noise.bps = 4;
s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP;
s->dither.noise_scale = 1;
}
if(s->rematrix || s->dither.method) {
ret = swri_rematrix_init(s);
if (ret < 0)
goto fail;
}
return 0;
fail:
swr_close(s);
return ret;
}
int swri_realloc_audio(AudioData *a, int count){
int i, countb;
AudioData old;
if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
return AVERROR(EINVAL);
if(a->count >= count)
return 0;
count*=2;
countb= FFALIGN(count*a->bps, ALIGN);
old= *a;
av_assert0(a->bps);
av_assert0(a->ch_count);
a->data= av_mallocz_array(countb, a->ch_count);
if(!a->data)
return AVERROR(ENOMEM);
for(i=0; i<a->ch_count; i++){
a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
}
if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
av_freep(&old.data);
a->count= count;
return 1;
}
static void copy(AudioData *out, AudioData *in,
int count){
av_assert0(out->planar == in->planar);
av_assert0(out->bps == in->bps);
av_assert0(out->ch_count == in->ch_count);
if(out->planar){
int ch;
for(ch=0; ch<out->ch_count; ch++)
memcpy(out->ch[ch], in->ch[ch], count*out->bps);
}else
memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
}
static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
int i;
if(!in_arg){
memset(out->ch, 0, sizeof(out->ch));
}else if(out->planar){
for(i=0; i<out->ch_count; i++)
out->ch[i]= in_arg[i];
}else{
for(i=0; i<out->ch_count; i++)
out->ch[i]= in_arg[0] + i*out->bps;
}
}
static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
int i;
if(out->planar){
for(i=0; i<out->ch_count; i++)
in_arg[i]= out->ch[i];
}else{
in_arg[0]= out->ch[0];
}
}
/**
*
* out may be equal in.
*/
static void buf_set(AudioData *out, AudioData *in, int count){
int ch;
if(in->planar){
for(ch=0; ch<out->ch_count; ch++)
out->ch[ch]= in->ch[ch] + count*out->bps;
}else{
for(ch=out->ch_count-1; ch>=0; ch--)
out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
}
}
/**
*
* @return number of samples output per channel
*/
static int resample(SwrContext *s, AudioData *out_param, int out_count,
const AudioData * in_param, int in_count){
AudioData in, out, tmp;
int ret_sum=0;
int border=0;
int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
av_assert1(s->in_buffer.ch_count == in_param->ch_count);
av_assert1(s->in_buffer.planar == in_param->planar);
av_assert1(s->in_buffer.fmt == in_param->fmt);
tmp=out=*out_param;
in = *in_param;
border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
&in, in_count, &s->in_buffer_index, &s->in_buffer_count);
if (border == INT_MAX) {
return 0;
} else if (border < 0) {
return border;
} else if (border) {
buf_set(&in, &in, border);
in_count -= border;
s->resample_in_constraint = 0;
}
do{
int ret, size, consumed;
if(!s->resample_in_constraint && s->in_buffer_count){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
out_count -= ret;
ret_sum += ret;
buf_set(&out, &out, ret);
s->in_buffer_count -= consumed;
s->in_buffer_index += consumed;
if(!in_count)
break;
if(s->in_buffer_count <= border){
buf_set(&in, &in, -s->in_buffer_count);
in_count += s->in_buffer_count;
s->in_buffer_count=0;
s->in_buffer_index=0;
border = 0;
}
}
if((s->flushed || in_count > padless) && !s->in_buffer_count){
s->in_buffer_index=0;
ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
out_count -= ret;
ret_sum += ret;
buf_set(&out, &out, ret);
in_count -= consumed;
buf_set(&in, &in, consumed);
}
//TODO is this check sane considering the advanced copy avoidance below
size= s->in_buffer_index + s->in_buffer_count + in_count;
if( size > s->in_buffer.count
&& s->in_buffer_count + in_count <= s->in_buffer_index){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
copy(&s->in_buffer, &tmp, s->in_buffer_count);
s->in_buffer_index=0;
}else
if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
return ret;
if(in_count){
int count= in_count;
if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
copy(&tmp, &in, /*in_*/count);
s->in_buffer_count += count;
in_count -= count;
border += count;
buf_set(&in, &in, count);
s->resample_in_constraint= 0;
if(s->in_buffer_count != count || in_count)
continue;
if (padless) {
padless = 0;
continue;
}
}
break;
}while(1);
s->resample_in_constraint= !!out_count;
return ret_sum;
}
static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
AudioData *in , int in_count){
AudioData *postin, *midbuf, *preout;
int ret/*, in_max*/;
AudioData preout_tmp, midbuf_tmp;
if(s->full_convert){
av_assert0(!s->resample);
swri_audio_convert(s->full_convert, out, in, in_count);
return out_count;
}
// in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
// in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
if((ret=swri_realloc_audio(&s->postin, in_count))<0)
return ret;
if(s->resample_first){
av_assert0(s->midbuf.ch_count == s->used_ch_count);
if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
return ret;
}else{
av_assert0(s->midbuf.ch_count == s->out.ch_count);
if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
return ret;
}
if((ret=swri_realloc_audio(&s->preout, out_count))<0)
return ret;
postin= &s->postin;
midbuf_tmp= s->midbuf;
midbuf= &midbuf_tmp;
preout_tmp= s->preout;
preout= &preout_tmp;
if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
postin= in;
if(s->resample_first ? !s->resample : !s->rematrix)
midbuf= postin;
if(s->resample_first ? !s->rematrix : !s->resample)
preout= midbuf;
if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
&& !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
if(preout==in){
out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
copy(out, in, out_count);
return out_count;
}
else if(preout==postin) preout= midbuf= postin= out;
else if(preout==midbuf) preout= midbuf= out;
else preout= out;
}
if(in != postin){
swri_audio_convert(s->in_convert, postin, in, in_count);
}
if(s->resample_first){
if(postin != midbuf)
out_count= resample(s, midbuf, out_count, postin, in_count);
if(midbuf != preout)
swri_rematrix(s, preout, midbuf, out_count, preout==out);
}else{
if(postin != midbuf)
swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
if(midbuf != preout)
out_count= resample(s, preout, out_count, midbuf, in_count);
}
if(preout != out && out_count){
AudioData *conv_src = preout;
if(s->dither.method){
int ch;
int dither_count= FFMAX(out_count, 1<<16);
if (preout == in) {
conv_src = &s->dither.temp;
if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
return ret;
}
if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
return ret;
if(ret)
for(ch=0; ch<s->dither.noise.ch_count; ch++)
if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
return ret;
av_assert0(s->dither.noise.ch_count == preout->ch_count);
if(s->dither.noise_pos + out_count > s->dither.noise.count)
s->dither.noise_pos = 0;
if (s->dither.method < SWR_DITHER_NS){
if (s->mix_2_1_simd) {
int len1= out_count&~15;
int off = len1 * preout->bps;
if(len1)
for(ch=0; ch<preout->ch_count; ch++)
s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);
if(out_count != len1)
for(ch=0; ch<preout->ch_count; ch++)
s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off, s->native_one, 0, 0, out_count - len1);
} else {
for(ch=0; ch<preout->ch_count; ch++)
s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
}
} else {
switch(s->int_sample_fmt) {
case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
}
}
s->dither.noise_pos += out_count;
}
//FIXME packed doesn't need more than 1 chan here!
swri_audio_convert(s->out_convert, out, conv_src, out_count);
}
return out_count;
}
int swr_is_initialized(struct SwrContext *s) {
return !!s->in_buffer.ch_count;
}
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
const uint8_t *in_arg [SWR_CH_MAX], int in_count){
AudioData * in= &s->in;
AudioData *out= &s->out;
int av_unused max_output;
if (!swr_is_initialized(s)) {
av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
return AVERROR(EINVAL);
}
#if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
max_output = swr_get_out_samples(s, in_count);
#endif
while(s->drop_output > 0){
int ret;
uint8_t *tmp_arg[SWR_CH_MAX];
#define MAX_DROP_STEP 16384
if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
return ret;
reversefill_audiodata(&s->drop_temp, tmp_arg);
s->drop_output *= -1; //FIXME find a less hackish solution
ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
s->drop_output *= -1;
in_count = 0;
if(ret>0) {
s->drop_output -= ret;
if (!s->drop_output && !out_arg)
return 0;
continue;
}
av_assert0(s->drop_output);
return 0;
}
if(!in_arg){
if(s->resample){
if (!s->flushed)
s->resampler->flush(s);
s->resample_in_constraint = 0;
s->flushed = 1;
}else if(!s->in_buffer_count){
return 0;
}
}else
fill_audiodata(in , (void*)in_arg);
fill_audiodata(out, out_arg);
if(s->resample){
int ret = swr_convert_internal(s, out, out_count, in, in_count);
if(ret>0 && !s->drop_output)
s->outpts += ret * (int64_t)s->in_sample_rate;
av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
return ret;
}else{
AudioData tmp= *in;
int ret2=0;
int ret, size;
size = FFMIN(out_count, s->in_buffer_count);
if(size){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
ret= swr_convert_internal(s, out, size, &tmp, size);
if(ret<0)
return ret;
ret2= ret;
s->in_buffer_count -= ret;
s->in_buffer_index += ret;
buf_set(out, out, ret);
out_count -= ret;
if(!s->in_buffer_count)
s->in_buffer_index = 0;
}
if(in_count){
size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
if(in_count > out_count) { //FIXME move after swr_convert_internal
if( size > s->in_buffer.count
&& s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
copy(&s->in_buffer, &tmp, s->in_buffer_count);
s->in_buffer_index=0;
}else
if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
return ret;
}
if(out_count){
size = FFMIN(in_count, out_count);
ret= swr_convert_internal(s, out, size, in, size);
if(ret<0)
return ret;
buf_set(in, in, ret);
in_count -= ret;
ret2 += ret;
}
if(in_count){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
copy(&tmp, in, in_count);
s->in_buffer_count += in_count;
}
}
if(ret2>0 && !s->drop_output)
s->outpts += ret2 * (int64_t)s->in_sample_rate;
av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
return ret2;
}
}
int swr_drop_output(struct SwrContext *s, int count){
const uint8_t *tmp_arg[SWR_CH_MAX];
s->drop_output += count;
if(s->drop_output <= 0)
return 0;
av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
}
int swr_inject_silence(struct SwrContext *s, int count){
int ret, i;
uint8_t *tmp_arg[SWR_CH_MAX];
if(count <= 0)
return 0;
#define MAX_SILENCE_STEP 16384
while (count > MAX_SILENCE_STEP) {
if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
return ret;
count -= MAX_SILENCE_STEP;
}
if((ret=swri_realloc_audio(&s->silence, count))<0)
return ret;
if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
} else
memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
reversefill_audiodata(&s->silence, tmp_arg);
av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
return ret;
}
int64_t swr_get_delay(struct SwrContext *s, int64_t base){
if (s->resampler && s->resample){
return s->resampler->get_delay(s, base);
}else{
return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
}
}
int swr_get_out_samples(struct SwrContext *s, int in_samples)
{
int64_t out_samples;
if (in_samples < 0)
return AVERROR(EINVAL);
if (s->resampler && s->resample) {
if (!s->resampler->get_out_samples)
return AVERROR(ENOSYS);
out_samples = s->resampler->get_out_samples(s, in_samples);
} else {
out_samples = s->in_buffer_count + in_samples;
av_assert0(s->out_sample_rate == s->in_sample_rate);
}
if (out_samples > INT_MAX)
return AVERROR(EINVAL);
return out_samples;
}
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
int ret;
if (!s || compensation_distance < 0)
return AVERROR(EINVAL);
if (!compensation_distance && sample_delta)
return AVERROR(EINVAL);
if (!s->resample) {
s->flags |= SWR_FLAG_RESAMPLE;
ret = swr_init(s);
if (ret < 0)
return ret;
}
if (!s->resampler->set_compensation){
return AVERROR(EINVAL);
}else{
return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
}
}
int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
if(pts == INT64_MIN)
return s->outpts;
if (s->firstpts == AV_NOPTS_VALUE)
s->outpts = s->firstpts = pts;
if(s->min_compensation >= FLT_MAX) {
return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
} else {
int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
if(fabs(fdelta) > s->min_compensation) {
if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
int ret;
if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
if(ret<0){
av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
}
} else if(s->soft_compensation_duration && s->max_soft_compensation) {
int duration = s->out_sample_rate * s->soft_compensation_duration;
double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
swr_set_compensation(s, comp, duration);
}
}
return s->outpts;
}
}

579
trunk/3rdparty/ffmpeg-4-fit/libswresample/swresample.h vendored Normal file
View file

@ -0,0 +1,579 @@
/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_SWRESAMPLE_H
#define SWRESAMPLE_SWRESAMPLE_H
/**
* @file
* @ingroup lswr
* libswresample public header
*/
/**
* @defgroup lswr libswresample
* @{
*
* Audio resampling, sample format conversion and mixing library.
*
* Interaction with lswr is done through SwrContext, which is
* allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
* must be set with the @ref avoptions API.
*
* The first thing you will need to do in order to use lswr is to allocate
* SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
* are using the former, you must set options through the @ref avoptions API.
* The latter function provides the same feature, but it allows you to set some
* common options in the same statement.
*
* For example the following code will setup conversion from planar float sample
* format to interleaved signed 16-bit integer, downsampling from 48kHz to
* 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
* matrix). This is using the swr_alloc() function.
* @code
* SwrContext *swr = swr_alloc();
* av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
* av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
* av_opt_set_int(swr, "in_sample_rate", 48000, 0);
* av_opt_set_int(swr, "out_sample_rate", 44100, 0);
* av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
* av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
* @endcode
*
* The same job can be done using swr_alloc_set_opts() as well:
* @code
* SwrContext *swr = swr_alloc_set_opts(NULL, // we're allocating a new context
* AV_CH_LAYOUT_STEREO, // out_ch_layout
* AV_SAMPLE_FMT_S16, // out_sample_fmt
* 44100, // out_sample_rate
* AV_CH_LAYOUT_5POINT1, // in_ch_layout
* AV_SAMPLE_FMT_FLTP, // in_sample_fmt
* 48000, // in_sample_rate
* 0, // log_offset
* NULL); // log_ctx
* @endcode
*
* Once all values have been set, it must be initialized with swr_init(). If
* you need to change the conversion parameters, you can change the parameters
* using @ref AVOptions, as described above in the first example; or by using
* swr_alloc_set_opts(), but with the first argument the allocated context.
* You must then call swr_init() again.
*
* The conversion itself is done by repeatedly calling swr_convert().
* Note that the samples may get buffered in swr if you provide insufficient
* output space or if sample rate conversion is done, which requires "future"
* samples. Samples that do not require future input can be retrieved at any
* time by using swr_convert() (in_count can be set to 0).
* At the end of conversion the resampling buffer can be flushed by calling
* swr_convert() with NULL in and 0 in_count.
*
* The samples used in the conversion process can be managed with the libavutil
* @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
* function used in the following example.
*
* The delay between input and output, can at any time be found by using
* swr_get_delay().
*
* The following code demonstrates the conversion loop assuming the parameters
* from above and caller-defined functions get_input() and handle_output():
* @code
* uint8_t **input;
* int in_samples;
*
* while (get_input(&input, &in_samples)) {
* uint8_t *output;
* int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
* in_samples, 44100, 48000, AV_ROUND_UP);
* av_samples_alloc(&output, NULL, 2, out_samples,
* AV_SAMPLE_FMT_S16, 0);
* out_samples = swr_convert(swr, &output, out_samples,
* input, in_samples);
* handle_output(output, out_samples);
* av_freep(&output);
* }
* @endcode
*
* When the conversion is finished, the conversion
* context and everything associated with it must be freed with swr_free().
* A swr_close() function is also available, but it exists mainly for
* compatibility with libavresample, and is not required to be called.
*
* There will be no memory leak if the data is not completely flushed before
* swr_free().
*/
#include <stdint.h>
#include "libavutil/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/samplefmt.h"
#include "libswresample/version.h"
/**
* @name Option constants
* These constants are used for the @ref avoptions interface for lswr.
* @{
*
*/
#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?
/** Dithering algorithms */
enum SwrDitherType {
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR,
SWR_DITHER_TRIANGULAR,
SWR_DITHER_TRIANGULAR_HIGHPASS,
SWR_DITHER_NS = 64, ///< not part of API/ABI
SWR_DITHER_NS_LIPSHITZ,
SWR_DITHER_NS_F_WEIGHTED,
SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
SWR_DITHER_NS_SHIBATA,
SWR_DITHER_NS_LOW_SHIBATA,
SWR_DITHER_NS_HIGH_SHIBATA,
SWR_DITHER_NB, ///< not part of API/ABI
};
/** Resampling Engines */
enum SwrEngine {
SWR_ENGINE_SWR, /**< SW Resampler */
SWR_ENGINE_SOXR, /**< SoX Resampler */
SWR_ENGINE_NB, ///< not part of API/ABI
};
/** Resampling Filter Types */
enum SwrFilterType {
SWR_FILTER_TYPE_CUBIC, /**< Cubic */
SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall windowed sinc */
SWR_FILTER_TYPE_KAISER, /**< Kaiser windowed sinc */
};
/**
* @}
*/
/**
* The libswresample context. Unlike libavcodec and libavformat, this structure
* is opaque. This means that if you would like to set options, you must use
* the @ref avoptions API and cannot directly set values to members of the
* structure.
*/
typedef struct SwrContext SwrContext;
/**
* Get the AVClass for SwrContext. It can be used in combination with
* AV_OPT_SEARCH_FAKE_OBJ for examining options.
*
* @see av_opt_find().
* @return the AVClass of SwrContext
*/
const AVClass *swr_get_class(void);
/**
* @name SwrContext constructor functions
* @{
*/
/**
* Allocate SwrContext.
*
* If you use this function you will need to set the parameters (manually or
* with swr_alloc_set_opts()) before calling swr_init().
*
* @see swr_alloc_set_opts(), swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc(void);
/**
* Initialize context after user parameters have been set.
* @note The context must be configured using the AVOption API.
*
* @see av_opt_set_int()
* @see av_opt_set_dict()
*
* @param[in,out] s Swr context to initialize
* @return AVERROR error code in case of failure.
*/
int swr_init(struct SwrContext *s);
/**
* Check whether an swr context has been initialized or not.
*
* @param[in] s Swr context to check
* @see swr_init()
* @return positive if it has been initialized, 0 if not initialized
*/
int swr_is_initialized(struct SwrContext *s);
/**
* Allocate SwrContext if needed and set/reset common parameters.
*
* This function does not require s to be allocated with swr_alloc(). On the
* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
* on the allocated context.
*
* @param s existing Swr context if available, or NULL if not
* @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
* @param out_sample_rate output sample rate (frequency in Hz)
* @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
* @param in_sample_rate input sample rate (frequency in Hz)
* @param log_offset logging level offset
* @param log_ctx parent logging context, can be NULL
*
* @see swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx);
/**
* @}
*
* @name SwrContext destructor functions
* @{
*/
/**
* Free the given SwrContext and set the pointer to NULL.
*
* @param[in] s a pointer to a pointer to Swr context
*/
void swr_free(struct SwrContext **s);
/**
* Closes the context so that swr_is_initialized() returns 0.
*
* The context can be brought back to life by running swr_init(),
* swr_init() can also be used without swr_close().
* This function is mainly provided for simplifying the usecase
* where one tries to support libavresample and libswresample.
*
* @param[in,out] s Swr context to be closed
*/
void swr_close(struct SwrContext *s);
/**
* @}
*
* @name Core conversion functions
* @{
*/
/** Convert audio.
*
* in and in_count can be set to 0 to flush the last few samples out at the
* end.
*
* If more input is provided than output space, then the input will be buffered.
* You can avoid this buffering by using swr_get_out_samples() to retrieve an
* upper bound on the required number of output samples for the given number of
* input samples. Conversion will run directly without copying whenever possible.
*
* @param s allocated Swr context, with parameters set
* @param out output buffers, only the first one need be set in case of packed audio
* @param out_count amount of space available for output in samples per channel
* @param in input buffers, only the first one need to be set in case of packed audio
* @param in_count number of input samples available in one channel
*
* @return number of samples output per channel, negative value on error
*/
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
const uint8_t **in , int in_count);
/**
* Convert the next timestamp from input to output
* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
*
* @note There are 2 slightly differently behaving modes.
* @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
* in this case timestamps will be passed through with delays compensated
* @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
* in this case the output timestamps will match output sample numbers.
* See ffmpeg-resampler(1) for the two modes of compensation.
*
* @param s[in] initialized Swr context
* @param pts[in] timestamp for the next input sample, INT64_MIN if unknown
* @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
* function used internally for timestamp compensation.
* @return the output timestamp for the next output sample
*/
int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
/**
* @}
*
* @name Low-level option setting functions
* These functons provide a means to set low-level options that is not possible
* with the AVOption API.
* @{
*/
/**
* Activate resampling compensation ("soft" compensation). This function is
* internally called when needed in swr_next_pts().
*
* @param[in,out] s allocated Swr context. If it is not initialized,
* or SWR_FLAG_RESAMPLE is not set, swr_init() is
* called with the flag set.
* @param[in] sample_delta delta in PTS per sample
* @param[in] compensation_distance number of samples to compensate for
* @return >= 0 on success, AVERROR error codes if:
* @li @c s is NULL,
* @li @c compensation_distance is less than 0,
* @li @c compensation_distance is 0 but sample_delta is not,
* @li compensation unsupported by resampler, or
* @li swr_init() fails when called.
*/
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
/**
* Set a customized input channel mapping.
*
* @param[in,out] s allocated Swr context, not yet initialized
* @param[in] channel_map customized input channel mapping (array of channel
* indexes, -1 for a muted channel)
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
/**
* Generate a channel mixing matrix.
*
* This function is the one used internally by libswresample for building the
* default mixing matrix. It is made public just as a utility function for
* building custom matrices.
*
* @param in_layout input channel layout
* @param out_layout output channel layout
* @param center_mix_level mix level for the center channel
* @param surround_mix_level mix level for the surround channel(s)
* @param lfe_mix_level mix level for the low-frequency effects channel
* @param rematrix_maxval if 1.0, coefficients will be normalized to prevent
* overflow. if INT_MAX, coefficients will not be
* normalized.
* @param[out] matrix mixing coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o.
* @param stride distance between adjacent input channels in the
* matrix array
* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
* @param log_ctx parent logging context, can be NULL
* @return 0 on success, negative AVERROR code on failure
*/
int swr_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double rematrix_maxval,
double rematrix_volume, double *matrix,
int stride, enum AVMatrixEncoding matrix_encoding,
void *log_ctx);
/**
* Set a customized remix matrix.
*
* @param s allocated Swr context, not yet initialized
* @param matrix remix coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o
* @param stride offset between lines of the matrix
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
/**
* @}
*
* @name Sample handling functions
* @{
*/
/**
* Drops the specified number of output samples.
*
* This function, along with swr_inject_silence(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_drop_output(struct SwrContext *s, int count);
/**
* Injects the specified number of silence samples.
*
* This function, along with swr_drop_output(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_inject_silence(struct SwrContext *s, int count);
/**
* Gets the delay the next input sample will experience relative to the next output sample.
*
* Swresample can buffer data if more input has been provided than available
* output space, also converting between sample rates needs a delay.
* This function returns the sum of all such delays.
* The exact delay is not necessarily an integer value in either input or
* output sample rate. Especially when downsampling by a large value, the
* output sample rate may be a poor choice to represent the delay, similarly
* for upsampling and the input sample rate.
*
* @param s swr context
* @param base timebase in which the returned delay will be:
* @li if it's set to 1 the returned delay is in seconds
* @li if it's set to 1000 the returned delay is in milliseconds
* @li if it's set to the input sample rate then the returned
* delay is in input samples
* @li if it's set to the output sample rate then the returned
* delay is in output samples
* @li if it's the least common multiple of in_sample_rate and
* out_sample_rate then an exact rounding-free delay will be
* returned
* @returns the delay in 1 / @c base units.
*/
int64_t swr_get_delay(struct SwrContext *s, int64_t base);
/**
* Find an upper bound on the number of samples that the next swr_convert
* call will output, if called with in_samples of input samples. This
* depends on the internal state, and anything changing the internal state
* (like further swr_convert() calls) will may change the number of samples
* swr_get_out_samples() returns for the same number of input samples.
*
* @param in_samples number of input samples.
* @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
* or swr_set_compensation() invalidates this limit
* @note it is recommended to pass the correct available buffer size
* to all functions like swr_convert() even if swr_get_out_samples()
* indicates that less would be used.
* @returns an upper bound on the number of samples that the next swr_convert
* will output or a negative value to indicate an error
*/
int swr_get_out_samples(struct SwrContext *s, int in_samples);
/**
* @}
*
* @name Configuration accessors
* @{
*/
/**
* Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
*
* This is useful to check if the build-time libswresample has the same version
* as the run-time one.
*
* @returns the unsigned int-typed version
*/
unsigned swresample_version(void);
/**
* Return the swr build-time configuration.
*
* @returns the build-time @c ./configure flags
*/
const char *swresample_configuration(void);
/**
* Return the swr license.
*
* @returns the license of libswresample, determined at build-time
*/
const char *swresample_license(void);
/**
* @}
*
* @name AVFrame based API
* @{
*/
/**
* Convert the samples in the input AVFrame and write them to the output AVFrame.
*
* Input and output AVFrames must have channel_layout, sample_rate and format set.
*
* If the output AVFrame does not have the data pointers allocated the nb_samples
* field will be set using av_frame_get_buffer()
* is called to allocate the frame.
*
* The output AVFrame can be NULL or have fewer allocated samples than required.
* In this case, any remaining samples not written to the output will be added
* to an internal FIFO buffer, to be returned at the next call to this function
* or to swr_convert().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. swr_get_delay() tells the number of
* remaining samples. To get this data as output, call this function or
* swr_convert() with NULL input.
*
* If the SwrContext configuration does not match the output and
* input AVFrame settings the conversion does not take place and depending on
* which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
* or the result of a bitwise-OR of them is returned.
*
* @see swr_delay()
* @see swr_convert()
* @see swr_get_delay()
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure or nonmatching
* configuration.
*/
int swr_convert_frame(SwrContext *swr,
AVFrame *output, const AVFrame *input);
/**
* Configure or reconfigure the SwrContext using the information
* provided by the AVFrames.
*
* The original resampling context is reset even on failure.
* The function calls swr_close() internally if the context is open.
*
* @see swr_close();
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure.
*/
int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);
/**
* @}
* @}
*/
#endif /* SWRESAMPLE_SWRESAMPLE_H */

159
trunk/3rdparty/ffmpeg-4-fit/libswresample/swresample_frame.c vendored Normal file
View file

@ -0,0 +1,159 @@
/*
* Copyright (c) 2014 Luca Barbato <lu_zero@gentoo.org>
* Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "swresample_internal.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
int swr_config_frame(SwrContext *s, const AVFrame *out, const AVFrame *in)
{
swr_close(s);
if (in) {
if (av_opt_set_int(s, "icl", in->channel_layout, 0) < 0)
goto fail;
if (av_opt_set_int(s, "isf", in->format, 0) < 0)
goto fail;
if (av_opt_set_int(s, "isr", in->sample_rate, 0) < 0)
goto fail;
}
if (out) {
if (av_opt_set_int(s, "ocl", out->channel_layout, 0) < 0)
goto fail;
if (av_opt_set_int(s, "osf", out->format, 0) < 0)
goto fail;
if (av_opt_set_int(s, "osr", out->sample_rate, 0) < 0)
goto fail;
}
return 0;
fail:
av_log(s, AV_LOG_ERROR, "Failed to set option\n");
return AVERROR(EINVAL);
}
static int config_changed(SwrContext *s,
const AVFrame *out, const AVFrame *in)
{
int ret = 0;
if (in) {
if (s->in_ch_layout != in->channel_layout ||
s->in_sample_rate != in->sample_rate ||
s->in_sample_fmt != in->format) {
ret |= AVERROR_INPUT_CHANGED;
}
}
if (out) {
if (s->out_ch_layout != out->channel_layout ||
s->out_sample_rate != out->sample_rate ||
s->out_sample_fmt != out->format) {
ret |= AVERROR_OUTPUT_CHANGED;
}
}
return ret;
}
static inline int convert_frame(SwrContext *s,
AVFrame *out, const AVFrame *in)
{
int ret;
uint8_t **out_data = NULL;
const uint8_t **in_data = NULL;
int out_nb_samples = 0, in_nb_samples = 0;
if (out) {
out_data = out->extended_data;
out_nb_samples = out->nb_samples;
}
if (in) {
in_data = (const uint8_t **)in->extended_data;
in_nb_samples = in->nb_samples;
}
ret = swr_convert(s, out_data, out_nb_samples, in_data, in_nb_samples);
if (ret < 0) {
if (out)
out->nb_samples = 0;
return ret;
}
if (out)
out->nb_samples = ret;
return 0;
}
static inline int available_samples(AVFrame *out)
{
int bytes_per_sample = av_get_bytes_per_sample(out->format);
int samples = out->linesize[0] / bytes_per_sample;
if (av_sample_fmt_is_planar(out->format)) {
return samples;
} else {
int channels = av_get_channel_layout_nb_channels(out->channel_layout);
return samples / channels;
}
}
int swr_convert_frame(SwrContext *s,
AVFrame *out, const AVFrame *in)
{
int ret, setup = 0;
if (!swr_is_initialized(s)) {
if ((ret = swr_config_frame(s, out, in)) < 0)
return ret;
if ((ret = swr_init(s)) < 0)
return ret;
setup = 1;
} else {
// return as is or reconfigure for input changes?
if ((ret = config_changed(s, out, in)))
return ret;
}
if (out) {
if (!out->linesize[0]) {
out->nb_samples = swr_get_delay(s, s->out_sample_rate) + 3;
if (in) {
out->nb_samples += in->nb_samples*(int64_t)s->out_sample_rate / s->in_sample_rate;
}
if ((ret = av_frame_get_buffer(out, 0)) < 0) {
if (setup)
swr_close(s);
return ret;
}
} else {
if (!out->nb_samples)
out->nb_samples = available_samples(out);
}
}
return convert_frame(s, out, in);
}

222
trunk/3rdparty/ffmpeg-4-fit/libswresample/swresample_internal.h vendored Normal file
View file

@ -0,0 +1,222 @@
/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_SWRESAMPLE_INTERNAL_H
#define SWRESAMPLE_SWRESAMPLE_INTERNAL_H
#include "swresample.h"
#include "libavutil/channel_layout.h"
#include "config.h"
#define SWR_CH_MAX 64
#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
#define NS_TAPS 20
#if ARCH_X86_64
typedef int64_t integer;
#else
typedef int integer;
#endif
typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
typedef struct AudioData{
uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel
uint8_t *data; ///< samples buffer
int ch_count; ///< number of channels
int bps; ///< bytes per sample
int count; ///< number of samples
int planar; ///< 1 if planar audio, 0 otherwise
enum AVSampleFormat fmt; ///< sample format
} AudioData;
struct DitherContext {
int method;
int noise_pos;
float scale;
float noise_scale; ///< Noise scale
int ns_taps; ///< Noise shaping dither taps
float ns_scale; ///< Noise shaping dither scale
float ns_scale_1; ///< Noise shaping dither scale^-1
int ns_pos; ///< Noise shaping dither position
float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients
float ns_errors[SWR_CH_MAX][2*NS_TAPS];
AudioData noise; ///< noise used for dithering
AudioData temp; ///< temporary storage when writing into the input buffer isn't possible
int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly
};
typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational);
typedef void (* resample_free_func)(struct ResampleContext **c);
typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
typedef int (* resample_flush_func)(struct SwrContext *c);
typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
struct Resampler {
resample_init_func init;
resample_free_func free;
multiple_resample_func multiple_resample;
resample_flush_func flush;
set_compensation_func set_compensation;
get_delay_func get_delay;
invert_initial_buffer_func invert_initial_buffer;
get_out_samples_func get_out_samples;
};
extern struct Resampler const swri_resampler;
extern struct Resampler const swri_soxr_resampler;
struct SwrContext {
const AVClass *av_class; ///< AVClass used for AVOption and av_log()
int log_level_offset; ///< logging level offset
void *log_ctx; ///< parent logging context
enum AVSampleFormat in_sample_fmt; ///< input sample format
enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
enum AVSampleFormat out_sample_fmt; ///< output sample format
int64_t in_ch_layout; ///< input channel layout
int64_t out_ch_layout; ///< output channel layout
int in_sample_rate; ///< input sample rate
int out_sample_rate; ///< output sample rate
int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
float slev; ///< surround mixing level
float clev; ///< center mixing level
float lfe_mix_level; ///< LFE mixing level
float rematrix_volume; ///< rematrixing volume coefficient
float rematrix_maxval; ///< maximum value for rematrixing output
int matrix_encoding; /**< matrixed stereo encoding */
const int *channel_map; ///< channel index (or -1 if muted channel) map
int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
int engine;
int user_in_ch_count; ///< User set input channel count
int user_out_ch_count; ///< User set output channel count
int user_used_ch_count; ///< User set used channel count
int64_t user_in_ch_layout; ///< User set input channel layout
int64_t user_out_ch_layout; ///< User set output channel layout
enum AVSampleFormat user_int_sample_fmt; ///< User set internal sample format
int user_dither_method; ///< User set dither method
struct DitherContext dither;
int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
int exact_rational; /**< if 1 then enable non power of 2 phase_count */
double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
int filter_type; /**< swr resampling filter type */
double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
double precision; /**< soxr resampling precision (in bits) */
int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
float min_compensation; ///< swr minimum below which no compensation will happen
float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
float soft_compensation_duration; ///< swr duration over which soft compensation is applied
float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
int64_t firstpts_in_samples; ///< swr first pts in samples
int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
int rematrix_custom; ///< flag to indicate that a custom matrix has been defined
AudioData in; ///< input audio data
AudioData postin; ///< post-input audio data: used for rematrix/resample
AudioData midbuf; ///< intermediate audio data (postin/preout)
AudioData preout; ///< pre-output audio data: used for rematrix/resample
AudioData out; ///< converted output audio data
AudioData in_buffer; ///< cached audio data (convert and resample purpose)
AudioData silence; ///< temporary with silence
AudioData drop_temp; ///< temporary used to discard output
int in_buffer_index; ///< cached buffer position
int in_buffer_count; ///< cached buffer length
int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
int flushed; ///< 1 if data is to be flushed and no further input is expected
int64_t outpts; ///< output PTS
int64_t firstpts; ///< first PTS
int drop_output; ///< number of output samples to drop
double delayed_samples_fixup; ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
struct AudioConvert *in_convert; ///< input conversion context
struct AudioConvert *out_convert; ///< output conversion context
struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
struct ResampleContext *resample; ///< resampling context
struct Resampler const *resampler; ///< resampler virtual function table
double matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
float matrix_flt[SWR_CH_MAX][SWR_CH_MAX]; ///< single precision floating point rematrixing coefficients
uint8_t *native_matrix;
uint8_t *native_one;
uint8_t *native_simd_one;
uint8_t *native_simd_matrix;
int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients
uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
mix_1_1_func_type *mix_1_1_f;
mix_1_1_func_type *mix_1_1_simd;
mix_2_1_func_type *mix_2_1_f;
mix_2_1_func_type *mix_2_1_simd;
mix_any_func_type *mix_any_f;
/* TODO: callbacks for ASM optimizations */
};
av_warn_unused_result
int swri_realloc_audio(AudioData *a, int count);
void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count);
av_warn_unused_result
int swri_rematrix_init(SwrContext *s);
void swri_rematrix_free(SwrContext *s);
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
int swri_rematrix_init_x86(struct SwrContext *s);
av_warn_unused_result
int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
av_warn_unused_result
int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
void swri_audio_convert_init_aarch64(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
void swri_audio_convert_init_arm(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
void swri_audio_convert_init_x86(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
#endif

55
trunk/3rdparty/ffmpeg-4-fit/libswresample/swresampleres.rc vendored Normal file
View file

@ -0,0 +1,55 @@
/*
* Windows resource file for libswresample
*
* Copyright (C) 2012 James Almer
* Copyright (C) 2013 Tiancheng "Timothy" Gu
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <windows.h>
#include "libswresample/version.h"
#include "libavutil/ffversion.h"
#include "config.h"
1 VERSIONINFO
FILEVERSION LIBSWRESAMPLE_VERSION_MAJOR, LIBSWRESAMPLE_VERSION_MINOR, LIBSWRESAMPLE_VERSION_MICRO, 0
PRODUCTVERSION LIBSWRESAMPLE_VERSION_MAJOR, LIBSWRESAMPLE_VERSION_MINOR, LIBSWRESAMPLE_VERSION_MICRO, 0
FILEFLAGSMASK VS_FFI_FILEFLAGSMASK
FILEOS VOS_NT_WINDOWS32
FILETYPE VFT_DLL
{
BLOCK "StringFileInfo"
{
BLOCK "040904B0"
{
VALUE "CompanyName", "FFmpeg Project"
VALUE "FileDescription", "FFmpeg audio resampling library"
VALUE "FileVersion", AV_STRINGIFY(LIBSWRESAMPLE_VERSION)
VALUE "InternalName", "libswresample"
VALUE "LegalCopyright", "Copyright (C) 2000-" AV_STRINGIFY(CONFIG_THIS_YEAR) " FFmpeg Project"
VALUE "OriginalFilename", "swresample" BUILDSUF "-" AV_STRINGIFY(LIBSWRESAMPLE_VERSION_MAJOR) SLIBSUF
VALUE "ProductName", "FFmpeg"
VALUE "ProductVersion", FFMPEG_VERSION
}
}
BLOCK "VarFileInfo"
{
VALUE "Translation", 0x0409, 0x04B0
}
}

45
trunk/3rdparty/ffmpeg-4-fit/libswresample/version.h vendored Normal file
View file

@ -0,0 +1,45 @@
/*
* Version macros.
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_VERSION_H
#define SWRESAMPLE_VERSION_H
/**
* @file
* Libswresample version macros
*/
#include "libavutil/avutil.h"
#define LIBSWRESAMPLE_VERSION_MAJOR 3
#define LIBSWRESAMPLE_VERSION_MINOR 5
#define LIBSWRESAMPLE_VERSION_MICRO 100
#define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \
LIBSWRESAMPLE_VERSION_MICRO)
#define LIBSWRESAMPLE_VERSION AV_VERSION(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \
LIBSWRESAMPLE_VERSION_MICRO)
#define LIBSWRESAMPLE_BUILD LIBSWRESAMPLE_VERSION_INT
#define LIBSWRESAMPLE_IDENT "SwR" AV_STRINGIFY(LIBSWRESAMPLE_VERSION)
#endif /* SWRESAMPLE_VERSION_H */

9
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/Makefile vendored Normal file
View file

@ -0,0 +1,9 @@
X86ASM-OBJS += x86/audio_convert.o\
x86/rematrix.o\
x86/resample.o\
OBJS += x86/audio_convert_init.o\
x86/rematrix_init.o\
x86/resample_init.o\
OBJS-$(CONFIG_XMM_CLOBBER_TEST) += x86/w64xmmtest.o

739
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/audio_convert.asm vendored Normal file
View file

@ -0,0 +1,739 @@
;******************************************************************************
;* Copyright (c) 2012 Michael Niedermayer
;*
;* This file is part of FFmpeg.
;*
;* FFmpeg is free software; you can redistribute it and/or
;* modify it under the terms of the GNU Lesser General Public
;* License as published by the Free Software Foundation; either
;* version 2.1 of the License, or (at your option) any later version.
;*
;* FFmpeg 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
;* Lesser General Public License for more details.
;*
;* You should have received a copy of the GNU Lesser General Public
;* License along with FFmpeg; if not, write to the Free Software
;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
;******************************************************************************
%include "libavutil/x86/x86util.asm"
SECTION_RODATA 32
flt2pm31: times 8 dd 4.6566129e-10
flt2p31 : times 8 dd 2147483648.0
flt2p15 : times 8 dd 32768.0
word_unpack_shuf : db 0, 1, 4, 5, 8, 9,12,13, 2, 3, 6, 7,10,11,14,15
SECTION .text
;to, from, a/u, log2_outsize, log_intsize, const
%macro PACK_2CH 5-7
cglobal pack_2ch_%2_to_%1_%3, 3, 4, 6, dst, src, len, src2
mov src2q , [srcq+gprsize]
mov srcq , [srcq]
mov dstq , [dstq]
%ifidn %3, a
test dstq, mmsize-1
jne pack_2ch_%2_to_%1_u_int %+ SUFFIX
test srcq, mmsize-1
jne pack_2ch_%2_to_%1_u_int %+ SUFFIX
test src2q, mmsize-1
jne pack_2ch_%2_to_%1_u_int %+ SUFFIX
%else
pack_2ch_%2_to_%1_u_int %+ SUFFIX:
%endif
lea srcq , [srcq + (1<<%5)*lenq]
lea src2q, [src2q + (1<<%5)*lenq]
lea dstq , [dstq + (2<<%4)*lenq]
neg lenq
%7 m0,m1,m2,m3,m4,m5
.next:
%if %4 >= %5
mov%3 m0, [ srcq +(1<<%5)*lenq]
mova m1, m0
mov%3 m2, [ src2q+(1<<%5)*lenq]
%if %5 == 1
punpcklwd m0, m2
punpckhwd m1, m2
%else
punpckldq m0, m2
punpckhdq m1, m2
%endif
%6 m0,m1,m2,m3,m4,m5
%else
mov%3 m0, [ srcq +(1<<%5)*lenq]
mov%3 m1, [mmsize + srcq +(1<<%5)*lenq]
mov%3 m2, [ src2q+(1<<%5)*lenq]
mov%3 m3, [mmsize + src2q+(1<<%5)*lenq]
%6 m0,m1,m2,m3,m4,m5
mova m2, m0
punpcklwd m0, m1
punpckhwd m2, m1
SWAP 1,2
%endif
mov%3 [ dstq+(2<<%4)*lenq], m0
mov%3 [ mmsize + dstq+(2<<%4)*lenq], m1
%if %4 > %5
mov%3 [2*mmsize + dstq+(2<<%4)*lenq], m2
mov%3 [3*mmsize + dstq+(2<<%4)*lenq], m3
add lenq, 4*mmsize/(2<<%4)
%else
add lenq, 2*mmsize/(2<<%4)
%endif
jl .next
REP_RET
%endmacro
%macro UNPACK_2CH 5-7
cglobal unpack_2ch_%2_to_%1_%3, 3, 4, 7, dst, src, len, dst2
mov dst2q , [dstq+gprsize]
mov srcq , [srcq]
mov dstq , [dstq]
%ifidn %3, a
test dstq, mmsize-1
jne unpack_2ch_%2_to_%1_u_int %+ SUFFIX
test srcq, mmsize-1
jne unpack_2ch_%2_to_%1_u_int %+ SUFFIX
test dst2q, mmsize-1
jne unpack_2ch_%2_to_%1_u_int %+ SUFFIX
%else
unpack_2ch_%2_to_%1_u_int %+ SUFFIX:
%endif
lea srcq , [srcq + (2<<%5)*lenq]
lea dstq , [dstq + (1<<%4)*lenq]
lea dst2q, [dst2q + (1<<%4)*lenq]
neg lenq
%7 m0,m1,m2,m3,m4,m5
mova m6, [word_unpack_shuf]
.next:
mov%3 m0, [ srcq +(2<<%5)*lenq]
mov%3 m2, [ mmsize + srcq +(2<<%5)*lenq]
%if %5 == 1
%ifidn SUFFIX, _ssse3
pshufb m0, m6
mova m1, m0
pshufb m2, m6
punpcklqdq m0,m2
punpckhqdq m1,m2
%else
mova m1, m0
punpcklwd m0,m2
punpckhwd m1,m2
mova m2, m0
punpcklwd m0,m1
punpckhwd m2,m1
mova m1, m0
punpcklwd m0,m2
punpckhwd m1,m2
%endif
%else
mova m1, m0
shufps m0, m2, 10001000b
shufps m1, m2, 11011101b
%endif
%if %4 < %5
mov%3 m2, [2*mmsize + srcq +(2<<%5)*lenq]
mova m3, m2
mov%3 m4, [3*mmsize + srcq +(2<<%5)*lenq]
shufps m2, m4, 10001000b
shufps m3, m4, 11011101b
SWAP 1,2
%endif
%6 m0,m1,m2,m3,m4,m5
mov%3 [ dstq+(1<<%4)*lenq], m0
%if %4 > %5
mov%3 [ dst2q+(1<<%4)*lenq], m2
mov%3 [ mmsize + dstq+(1<<%4)*lenq], m1
mov%3 [ mmsize + dst2q+(1<<%4)*lenq], m3
add lenq, 2*mmsize/(1<<%4)
%else
mov%3 [ dst2q+(1<<%4)*lenq], m1
add lenq, mmsize/(1<<%4)
%endif
jl .next
REP_RET
%endmacro
%macro CONV 5-7
cglobal %2_to_%1_%3, 3, 3, 6, dst, src, len
mov srcq , [srcq]
mov dstq , [dstq]
%ifidn %3, a
test dstq, mmsize-1
jne %2_to_%1_u_int %+ SUFFIX
test srcq, mmsize-1
jne %2_to_%1_u_int %+ SUFFIX
%else
%2_to_%1_u_int %+ SUFFIX:
%endif
lea srcq , [srcq + (1<<%5)*lenq]
lea dstq , [dstq + (1<<%4)*lenq]
neg lenq
%7 m0,m1,m2,m3,m4,m5
.next:
mov%3 m0, [ srcq +(1<<%5)*lenq]
mov%3 m1, [ mmsize + srcq +(1<<%5)*lenq]
%if %4 < %5
mov%3 m2, [2*mmsize + srcq +(1<<%5)*lenq]
mov%3 m3, [3*mmsize + srcq +(1<<%5)*lenq]
%endif
%6 m0,m1,m2,m3,m4,m5
mov%3 [ dstq+(1<<%4)*lenq], m0
mov%3 [ mmsize + dstq+(1<<%4)*lenq], m1
%if %4 > %5
mov%3 [2*mmsize + dstq+(1<<%4)*lenq], m2
mov%3 [3*mmsize + dstq+(1<<%4)*lenq], m3
add lenq, 4*mmsize/(1<<%4)
%else
add lenq, 2*mmsize/(1<<%4)
%endif
jl .next
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
%macro PACK_6CH 8
cglobal pack_6ch_%2_to_%1_%3, 2, 8, %6, dst, src, src1, src2, src3, src4, src5, len
%if ARCH_X86_64
mov lend, r2d
%else
%define lend dword r2m
%endif
mov src1q, [srcq+1*gprsize]
mov src2q, [srcq+2*gprsize]
mov src3q, [srcq+3*gprsize]
mov src4q, [srcq+4*gprsize]
mov src5q, [srcq+5*gprsize]
mov srcq, [srcq]
mov dstq, [dstq]
%ifidn %3, a
test dstq, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
test srcq, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
test src1q, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
test src2q, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
test src3q, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
test src4q, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
test src5q, mmsize-1
jne pack_6ch_%2_to_%1_u_int %+ SUFFIX
%else
pack_6ch_%2_to_%1_u_int %+ SUFFIX:
%endif
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
sub src4q, srcq
sub src5q, srcq
%8 x,x,x,x,m7,x
.loop:
mov%3 m0, [srcq ]
mov%3 m1, [srcq+src1q]
mov%3 m2, [srcq+src2q]
mov%3 m3, [srcq+src3q]
mov%3 m4, [srcq+src4q]
mov%3 m5, [srcq+src5q]
%if cpuflag(sse)
SBUTTERFLYPS 0, 1, 6
SBUTTERFLYPS 2, 3, 6
SBUTTERFLYPS 4, 5, 6
%if cpuflag(avx)
blendps m6, m4, m0, 1100b
%else
movaps m6, m4
shufps m4, m0, q3210
SWAP 4,6
%endif
movlhps m0, m2
movhlps m4, m2
%if cpuflag(avx)
blendps m2, m5, m1, 1100b
%else
movaps m2, m5
shufps m5, m1, q3210
SWAP 2,5
%endif
movlhps m1, m3
movhlps m5, m3
%7 m0,m6,x,x,m7,m3
%7 m4,m1,x,x,m7,m3
%7 m2,m5,x,x,m7,m3
mov %+ %3 %+ ps [dstq ], m0
mov %+ %3 %+ ps [dstq+16], m6
mov %+ %3 %+ ps [dstq+32], m4
mov %+ %3 %+ ps [dstq+48], m1
mov %+ %3 %+ ps [dstq+64], m2
mov %+ %3 %+ ps [dstq+80], m5
%else ; mmx
SBUTTERFLY dq, 0, 1, 6
SBUTTERFLY dq, 2, 3, 6
SBUTTERFLY dq, 4, 5, 6
movq [dstq ], m0
movq [dstq+ 8], m2
movq [dstq+16], m4
movq [dstq+24], m1
movq [dstq+32], m3
movq [dstq+40], m5
%endif
add srcq, mmsize
add dstq, mmsize*6
sub lend, mmsize/4
jg .loop
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
%macro UNPACK_6CH 8
cglobal unpack_6ch_%2_to_%1_%3, 2, 8, %6, dst, src, dst1, dst2, dst3, dst4, dst5, len
%if ARCH_X86_64
mov lend, r2d
%else
%define lend dword r2m
%endif
mov dst1q, [dstq+1*gprsize]
mov dst2q, [dstq+2*gprsize]
mov dst3q, [dstq+3*gprsize]
mov dst4q, [dstq+4*gprsize]
mov dst5q, [dstq+5*gprsize]
mov dstq, [dstq]
mov srcq, [srcq]
%ifidn %3, a
test dstq, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
test srcq, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
test dst1q, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
test dst2q, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
test dst3q, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
test dst4q, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
test dst5q, mmsize-1
jne unpack_6ch_%2_to_%1_u_int %+ SUFFIX
%else
unpack_6ch_%2_to_%1_u_int %+ SUFFIX:
%endif
sub dst1q, dstq
sub dst2q, dstq
sub dst3q, dstq
sub dst4q, dstq
sub dst5q, dstq
%8 x,x,x,x,m7,x
.loop:
mov%3 m0, [srcq ]
mov%3 m1, [srcq+16]
mov%3 m2, [srcq+32]
mov%3 m3, [srcq+48]
mov%3 m4, [srcq+64]
mov%3 m5, [srcq+80]
SBUTTERFLYPS 0, 3, 6
SBUTTERFLYPS 1, 4, 6
SBUTTERFLYPS 2, 5, 6
SBUTTERFLYPS 0, 4, 6
SBUTTERFLYPS 3, 2, 6
SBUTTERFLYPS 1, 5, 6
SWAP 1, 4
SWAP 2, 3
%7 m0,m1,x,x,m7,m6
%7 m2,m3,x,x,m7,m6
%7 m4,m5,x,x,m7,m6
mov %+ %3 %+ ps [dstq ], m0
mov %+ %3 %+ ps [dstq+dst1q], m1
mov %+ %3 %+ ps [dstq+dst2q], m2
mov %+ %3 %+ ps [dstq+dst3q], m3
mov %+ %3 %+ ps [dstq+dst4q], m4
mov %+ %3 %+ ps [dstq+dst5q], m5
add srcq, mmsize*6
add dstq, mmsize
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
%define PACK_8CH_GPRS (10 * ARCH_X86_64) + ((6 + HAVE_ALIGNED_STACK) * ARCH_X86_32)
%macro PACK_8CH 8
cglobal pack_8ch_%2_to_%1_%3, 2, PACK_8CH_GPRS, %6, ARCH_X86_32*48, dst, src, len, src1, src2, src3, src4, src5, src6, src7
mov dstq, [dstq]
%if ARCH_X86_32
DEFINE_ARGS dst, src, src2, src3, src4, src5, src6
%define lend dword r2m
%define src1q r0q
%define src1m dword [rsp+32]
%if HAVE_ALIGNED_STACK == 0
DEFINE_ARGS dst, src, src2, src3, src5, src6
%define src4q r0q
%define src4m dword [rsp+36]
%endif
%define src7q r0q
%define src7m dword [rsp+40]
mov dstm, dstq
%endif
mov src7q, [srcq+7*gprsize]
mov src6q, [srcq+6*gprsize]
%if ARCH_X86_32
mov src7m, src7q
%endif
mov src5q, [srcq+5*gprsize]
mov src4q, [srcq+4*gprsize]
mov src3q, [srcq+3*gprsize]
%if ARCH_X86_32 && HAVE_ALIGNED_STACK == 0
mov src4m, src4q
%endif
mov src2q, [srcq+2*gprsize]
mov src1q, [srcq+1*gprsize]
mov srcq, [srcq]
%ifidn %3, a
%if ARCH_X86_32
test dstmp, mmsize-1
%else
test dstq, mmsize-1
%endif
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
test srcq, mmsize-1
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
test src1q, mmsize-1
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
test src2q, mmsize-1
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
test src3q, mmsize-1
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
%if ARCH_X86_32 && HAVE_ALIGNED_STACK == 0
test src4m, mmsize-1
%else
test src4q, mmsize-1
%endif
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
test src5q, mmsize-1
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
test src6q, mmsize-1
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
%if ARCH_X86_32
test src7m, mmsize-1
%else
test src7q, mmsize-1
%endif
jne pack_8ch_%2_to_%1_u_int %+ SUFFIX
%else
pack_8ch_%2_to_%1_u_int %+ SUFFIX:
%endif
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
%if ARCH_X86_64 || HAVE_ALIGNED_STACK
sub src4q, srcq
%else
sub src4m, srcq
%endif
sub src5q, srcq
sub src6q, srcq
%if ARCH_X86_64
sub src7q, srcq
%else
mov src1m, src1q
sub src7m, srcq
%endif
%if ARCH_X86_64
%8 x,x,x,x,m9,x
%elifidn %1, int32
%define m9 [flt2p31]
%else
%define m9 [flt2pm31]
%endif
.loop:
mov%3 m0, [srcq ]
mov%3 m1, [srcq+src1q]
mov%3 m2, [srcq+src2q]
%if ARCH_X86_32 && HAVE_ALIGNED_STACK == 0
mov src4q, src4m
%endif
mov%3 m3, [srcq+src3q]
mov%3 m4, [srcq+src4q]
mov%3 m5, [srcq+src5q]
%if ARCH_X86_32
mov src7q, src7m
%endif
mov%3 m6, [srcq+src6q]
mov%3 m7, [srcq+src7q]
%if ARCH_X86_64
TRANSPOSE8x4D 0, 1, 2, 3, 4, 5, 6, 7, 8
%7 m0,m1,x,x,m9,m8
%7 m2,m3,x,x,m9,m8
%7 m4,m5,x,x,m9,m8
%7 m6,m7,x,x,m9,m8
mov%3 [dstq], m0
%else
mov dstq, dstm
TRANSPOSE8x4D 0, 1, 2, 3, 4, 5, 6, 7, [rsp], [rsp+16], 1
%7 m0,m1,x,x,m9,m2
mova m2, [rsp]
mov%3 [dstq], m0
%7 m2,m3,x,x,m9,m0
%7 m4,m5,x,x,m9,m0
%7 m6,m7,x,x,m9,m0
%endif
mov%3 [dstq+16], m1
mov%3 [dstq+32], m2
mov%3 [dstq+48], m3
mov%3 [dstq+64], m4
mov%3 [dstq+80], m5
mov%3 [dstq+96], m6
mov%3 [dstq+112], m7
add srcq, mmsize
add dstq, mmsize*8
%if ARCH_X86_32
mov dstm, dstq
mov src1q, src1m
%endif
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
%macro INT16_TO_INT32_N 6
pxor m2, m2
pxor m3, m3
punpcklwd m2, m1
punpckhwd m3, m1
SWAP 4,0
pxor m0, m0
pxor m1, m1
punpcklwd m0, m4
punpckhwd m1, m4
%endmacro
%macro INT32_TO_INT16_N 6
psrad m0, 16
psrad m1, 16
psrad m2, 16
psrad m3, 16
packssdw m0, m1
packssdw m2, m3
SWAP 1,2
%endmacro
%macro INT32_TO_FLOAT_INIT 6
mova %5, [flt2pm31]
%endmacro
%macro INT32_TO_FLOAT_N 6
cvtdq2ps %1, %1
cvtdq2ps %2, %2
mulps %1, %1, %5
mulps %2, %2, %5
%endmacro
%macro FLOAT_TO_INT32_INIT 6
mova %5, [flt2p31]
%endmacro
%macro FLOAT_TO_INT32_N 6
mulps %1, %5
mulps %2, %5
cvtps2dq %6, %1
cmpps %1, %1, %5, 5
paddd %1, %6
cvtps2dq %6, %2
cmpps %2, %2, %5, 5
paddd %2, %6
%endmacro
%macro INT16_TO_FLOAT_INIT 6
mova m5, [flt2pm31]
%endmacro
%macro INT16_TO_FLOAT_N 6
INT16_TO_INT32_N %1,%2,%3,%4,%5,%6
cvtdq2ps m0, m0
cvtdq2ps m1, m1
cvtdq2ps m2, m2
cvtdq2ps m3, m3
mulps m0, m0, m5
mulps m1, m1, m5
mulps m2, m2, m5
mulps m3, m3, m5
%endmacro
%macro FLOAT_TO_INT16_INIT 6
mova m5, [flt2p15]
%endmacro
%macro FLOAT_TO_INT16_N 6
mulps m0, m5
mulps m1, m5
mulps m2, m5
mulps m3, m5
cvtps2dq m0, m0
cvtps2dq m1, m1
packssdw m0, m1
cvtps2dq m1, m2
cvtps2dq m3, m3
packssdw m1, m3
%endmacro
%macro NOP_N 0-6
%endmacro
INIT_MMX mmx
CONV int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N
CONV int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N
CONV int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N
CONV int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N
PACK_6CH float, float, u, 2, 2, 0, NOP_N, NOP_N
PACK_6CH float, float, a, 2, 2, 0, NOP_N, NOP_N
INIT_XMM sse
PACK_6CH float, float, u, 2, 2, 7, NOP_N, NOP_N
PACK_6CH float, float, a, 2, 2, 7, NOP_N, NOP_N
UNPACK_6CH float, float, u, 2, 2, 7, NOP_N, NOP_N
UNPACK_6CH float, float, a, 2, 2, 7, NOP_N, NOP_N
INIT_XMM sse2
CONV int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N
CONV int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N
CONV int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N
CONV int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N
PACK_2CH int16, int16, u, 1, 1, NOP_N, NOP_N
PACK_2CH int16, int16, a, 1, 1, NOP_N, NOP_N
PACK_2CH int32, int32, u, 2, 2, NOP_N, NOP_N
PACK_2CH int32, int32, a, 2, 2, NOP_N, NOP_N
PACK_2CH int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N
PACK_2CH int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N
PACK_2CH int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N
PACK_2CH int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N
UNPACK_2CH int16, int16, u, 1, 1, NOP_N, NOP_N
UNPACK_2CH int16, int16, a, 1, 1, NOP_N, NOP_N
UNPACK_2CH int32, int32, u, 2, 2, NOP_N, NOP_N
UNPACK_2CH int32, int32, a, 2, 2, NOP_N, NOP_N
UNPACK_2CH int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N
UNPACK_2CH int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N
UNPACK_2CH int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N
UNPACK_2CH int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N
CONV float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
CONV float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
CONV int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
CONV int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
CONV float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
CONV float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
CONV int16, float, u, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT
CONV int16, float, a, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT
PACK_2CH float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_2CH float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_2CH int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_2CH int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_2CH float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
PACK_2CH float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
PACK_2CH int16, float, u, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT
PACK_2CH int16, float, a, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT
UNPACK_2CH float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
UNPACK_2CH float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
UNPACK_2CH int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
UNPACK_2CH int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
UNPACK_2CH float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
UNPACK_2CH float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
UNPACK_2CH int16, float, u, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT
UNPACK_2CH int16, float, a, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT
PACK_6CH float, int32, u, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_6CH float, int32, a, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_6CH int32, float, u, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_6CH int32, float, a, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
UNPACK_6CH float, int32, u, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
UNPACK_6CH float, int32, a, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
UNPACK_6CH int32, float, u, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
UNPACK_6CH int32, float, a, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_8CH float, float, u, 2, 2, 9, NOP_N, NOP_N
PACK_8CH float, float, a, 2, 2, 9, NOP_N, NOP_N
PACK_8CH float, int32, u, 2, 2, 10, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_8CH float, int32, a, 2, 2, 10, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_8CH int32, float, u, 2, 2, 10, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_8CH int32, float, a, 2, 2, 10, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
INIT_XMM ssse3
UNPACK_2CH int16, int16, u, 1, 1, NOP_N, NOP_N
UNPACK_2CH int16, int16, a, 1, 1, NOP_N, NOP_N
UNPACK_2CH int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N
UNPACK_2CH int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N
UNPACK_2CH float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
UNPACK_2CH float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
PACK_6CH float, float, u, 2, 2, 8, NOP_N, NOP_N
PACK_6CH float, float, a, 2, 2, 8, NOP_N, NOP_N
UNPACK_6CH float, float, u, 2, 2, 8, NOP_N, NOP_N
UNPACK_6CH float, float, a, 2, 2, 8, NOP_N, NOP_N
PACK_6CH float, int32, u, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_6CH float, int32, a, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_6CH int32, float, u, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_6CH int32, float, a, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
UNPACK_6CH float, int32, u, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
UNPACK_6CH float, int32, a, 2, 2, 8, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
UNPACK_6CH int32, float, u, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
UNPACK_6CH int32, float, a, 2, 2, 8, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_8CH float, float, u, 2, 2, 9, NOP_N, NOP_N
PACK_8CH float, float, a, 2, 2, 9, NOP_N, NOP_N
PACK_8CH float, int32, u, 2, 2, 10, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_8CH float, int32, a, 2, 2, 10, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
PACK_8CH int32, float, u, 2, 2, 10, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
PACK_8CH int32, float, a, 2, 2, 10, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
INIT_YMM avx
CONV float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
CONV float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT
%endif
%if HAVE_AVX2_EXTERNAL
INIT_YMM avx2
CONV int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
CONV int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT
%endif

181
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/audio_convert_init.c vendored Normal file
View file

@ -0,0 +1,181 @@
/*
* Copyright (C) 2012 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/x86/cpu.h"
#include "libswresample/swresample_internal.h"
#include "libswresample/audioconvert.h"
#define PROTO(pre, in, out, cap) void ff ## pre ## in## _to_ ##out## _a_ ##cap(uint8_t **dst, const uint8_t **src, int len);
#define PROTO2(pre, out, cap) PROTO(pre, int16, out, cap) PROTO(pre, int32, out, cap) PROTO(pre, float, out, cap)
#define PROTO3(pre, cap) PROTO2(pre, int16, cap) PROTO2(pre, int32, cap) PROTO2(pre, float, cap)
#define PROTO4(pre) PROTO3(pre, mmx) PROTO3(pre, sse) PROTO3(pre, sse2) PROTO3(pre, ssse3) PROTO3(pre, sse4) PROTO3(pre, avx) PROTO3(pre, avx2)
PROTO4(_)
PROTO4(_pack_2ch_)
PROTO4(_pack_6ch_)
PROTO4(_pack_8ch_)
PROTO4(_unpack_2ch_)
PROTO4(_unpack_6ch_)
av_cold void swri_audio_convert_init_x86(struct AudioConvert *ac,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels){
int mm_flags = av_get_cpu_flags();
ac->simd_f= NULL;
//FIXME add memcpy case
#define MULTI_CAPS_FUNC(flag, cap) \
if (EXTERNAL_##flag(mm_flags)) {\
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S16 || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S16P)\
ac->simd_f = ff_int16_to_int32_a_ ## cap;\
if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_S32 || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S32P)\
ac->simd_f = ff_int32_to_int16_a_ ## cap;\
}
MULTI_CAPS_FUNC(MMX, mmx)
MULTI_CAPS_FUNC(SSE2, sse2)
if(EXTERNAL_MMX(mm_flags)) {
if(channels == 6) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_6ch_float_to_float_a_mmx;
}
}
if(EXTERNAL_SSE(mm_flags)) {
if(channels == 6) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_6ch_float_to_float_a_sse;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_6ch_float_to_float_a_sse;
}
}
if(EXTERNAL_SSE2(mm_flags)) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32 || out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_int32_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S16 || out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S16P)
ac->simd_f = ff_int16_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_float_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_float_to_int16_a_sse2;
if(channels == 2) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_2ch_int32_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_S16P)
ac->simd_f = ff_pack_2ch_int16_to_int16_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S16P)
ac->simd_f = ff_pack_2ch_int16_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_2ch_int32_to_int16_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_2ch_int32_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S16)
ac->simd_f = ff_unpack_2ch_int16_to_int16_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S16)
ac->simd_f = ff_unpack_2ch_int16_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_2ch_int32_to_int16_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_2ch_int32_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_pack_2ch_float_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S16P)
ac->simd_f = ff_pack_2ch_int16_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_pack_2ch_float_to_int16_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_2ch_int32_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLT)
ac->simd_f = ff_unpack_2ch_float_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S16)
ac->simd_f = ff_unpack_2ch_int16_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLT)
ac->simd_f = ff_unpack_2ch_float_to_int16_a_sse2;
}
if(channels == 6) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_6ch_int32_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_pack_6ch_float_to_int32_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_6ch_int32_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLT)
ac->simd_f = ff_unpack_6ch_float_to_int32_a_sse2;
}
if(channels == 8) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_8ch_float_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_8ch_int32_to_float_a_sse2;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_pack_8ch_float_to_int32_a_sse2;
}
}
if(EXTERNAL_SSSE3(mm_flags)) {
if(channels == 2) {
if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S16)
ac->simd_f = ff_unpack_2ch_int16_to_int16_a_ssse3;
if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S16)
ac->simd_f = ff_unpack_2ch_int16_to_int32_a_ssse3;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S16)
ac->simd_f = ff_unpack_2ch_int16_to_float_a_ssse3;
}
}
if(EXTERNAL_AVX_FAST(mm_flags)) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32 || out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_int32_to_float_a_avx;
}
if(EXTERNAL_AVX(mm_flags)) {
if(channels == 6) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_6ch_float_to_float_a_avx;
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_6ch_int32_to_float_a_avx;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_pack_6ch_float_to_int32_a_avx;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_6ch_float_to_float_a_avx;
if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32)
ac->simd_f = ff_unpack_6ch_int32_to_float_a_avx;
if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLT)
ac->simd_f = ff_unpack_6ch_float_to_int32_a_avx;
}
if(channels == 8) {
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_8ch_float_to_float_a_avx;
if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P)
ac->simd_f = ff_pack_8ch_int32_to_float_a_avx;
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_pack_8ch_float_to_int32_a_avx;
}
}
if(EXTERNAL_AVX2_FAST(mm_flags)) {
if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLTP)
ac->simd_f = ff_float_to_int32_a_avx2;
}
}

250
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/rematrix.asm vendored Normal file
View file

@ -0,0 +1,250 @@
;******************************************************************************
;* Copyright (c) 2012 Michael Niedermayer
;*
;* This file is part of FFmpeg.
;*
;* FFmpeg is free software; you can redistribute it and/or
;* modify it under the terms of the GNU Lesser General Public
;* License as published by the Free Software Foundation; either
;* version 2.1 of the License, or (at your option) any later version.
;*
;* FFmpeg 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
;* Lesser General Public License for more details.
;*
;* You should have received a copy of the GNU Lesser General Public
;* License along with FFmpeg; if not, write to the Free Software
;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
;******************************************************************************
%include "libavutil/x86/x86util.asm"
SECTION_RODATA 32
dw1: times 8 dd 1
w1 : times 16 dw 1
SECTION .text
%macro MIX2_FLT 1
cglobal mix_2_1_%1_float, 7, 7, 6, out, in1, in2, coeffp, index1, index2, len
%ifidn %1, a
test in1q, mmsize-1
jne mix_2_1_float_u_int %+ SUFFIX
test in2q, mmsize-1
jne mix_2_1_float_u_int %+ SUFFIX
test outq, mmsize-1
jne mix_2_1_float_u_int %+ SUFFIX
%else
mix_2_1_float_u_int %+ SUFFIX:
%endif
VBROADCASTSS m4, [coeffpq + 4*index1q]
VBROADCASTSS m5, [coeffpq + 4*index2q]
shl lend , 2
add in1q , lenq
add in2q , lenq
add outq , lenq
neg lenq
.next:
%ifidn %1, a
mulps m0, m4, [in1q + lenq ]
mulps m1, m5, [in2q + lenq ]
mulps m2, m4, [in1q + lenq + mmsize]
mulps m3, m5, [in2q + lenq + mmsize]
%else
movu m0, [in1q + lenq ]
movu m1, [in2q + lenq ]
movu m2, [in1q + lenq + mmsize]
movu m3, [in2q + lenq + mmsize]
mulps m0, m0, m4
mulps m1, m1, m5
mulps m2, m2, m4
mulps m3, m3, m5
%endif
addps m0, m0, m1
addps m2, m2, m3
mov%1 [outq + lenq ], m0
mov%1 [outq + lenq + mmsize], m2
add lenq, mmsize*2
jl .next
REP_RET
%endmacro
%macro MIX1_FLT 1
cglobal mix_1_1_%1_float, 5, 5, 3, out, in, coeffp, index, len
%ifidn %1, a
test inq, mmsize-1
jne mix_1_1_float_u_int %+ SUFFIX
test outq, mmsize-1
jne mix_1_1_float_u_int %+ SUFFIX
%else
mix_1_1_float_u_int %+ SUFFIX:
%endif
VBROADCASTSS m2, [coeffpq + 4*indexq]
shl lenq , 2
add inq , lenq
add outq , lenq
neg lenq
.next:
%ifidn %1, a
mulps m0, m2, [inq + lenq ]
mulps m1, m2, [inq + lenq + mmsize]
%else
movu m0, [inq + lenq ]
movu m1, [inq + lenq + mmsize]
mulps m0, m0, m2
mulps m1, m1, m2
%endif
mov%1 [outq + lenq ], m0
mov%1 [outq + lenq + mmsize], m1
add lenq, mmsize*2
jl .next
REP_RET
%endmacro
%macro MIX1_INT16 1
cglobal mix_1_1_%1_int16, 5, 5, 6, out, in, coeffp, index, len
%ifidn %1, a
test inq, mmsize-1
jne mix_1_1_int16_u_int %+ SUFFIX
test outq, mmsize-1
jne mix_1_1_int16_u_int %+ SUFFIX
%else
mix_1_1_int16_u_int %+ SUFFIX:
%endif
movd m4, [coeffpq + 4*indexq]
SPLATW m5, m4
psllq m4, 32
psrlq m4, 48
mova m0, [w1]
psllw m0, m4
psrlw m0, 1
punpcklwd m5, m0
add lenq , lenq
add inq , lenq
add outq , lenq
neg lenq
.next:
mov%1 m0, [inq + lenq ]
mov%1 m2, [inq + lenq + mmsize]
mova m1, m0
mova m3, m2
punpcklwd m0, [w1]
punpckhwd m1, [w1]
punpcklwd m2, [w1]
punpckhwd m3, [w1]
pmaddwd m0, m5
pmaddwd m1, m5
pmaddwd m2, m5
pmaddwd m3, m5
psrad m0, m4
psrad m1, m4
psrad m2, m4
psrad m3, m4
packssdw m0, m1
packssdw m2, m3
mov%1 [outq + lenq ], m0
mov%1 [outq + lenq + mmsize], m2
add lenq, mmsize*2
jl .next
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
%macro MIX2_INT16 1
cglobal mix_2_1_%1_int16, 7, 7, 8, out, in1, in2, coeffp, index1, index2, len
%ifidn %1, a
test in1q, mmsize-1
jne mix_2_1_int16_u_int %+ SUFFIX
test in2q, mmsize-1
jne mix_2_1_int16_u_int %+ SUFFIX
test outq, mmsize-1
jne mix_2_1_int16_u_int %+ SUFFIX
%else
mix_2_1_int16_u_int %+ SUFFIX:
%endif
movd m4, [coeffpq + 4*index1q]
movd m6, [coeffpq + 4*index2q]
SPLATW m5, m4
SPLATW m6, m6
psllq m4, 32
psrlq m4, 48
mova m7, [dw1]
pslld m7, m4
psrld m7, 1
punpcklwd m5, m6
add lend , lend
add in1q , lenq
add in2q , lenq
add outq , lenq
neg lenq
.next:
mov%1 m0, [in1q + lenq ]
mov%1 m2, [in2q + lenq ]
mova m1, m0
punpcklwd m0, m2
punpckhwd m1, m2
mov%1 m2, [in1q + lenq + mmsize]
mov%1 m6, [in2q + lenq + mmsize]
mova m3, m2
punpcklwd m2, m6
punpckhwd m3, m6
pmaddwd m0, m5
pmaddwd m1, m5
pmaddwd m2, m5
pmaddwd m3, m5
paddd m0, m7
paddd m1, m7
paddd m2, m7
paddd m3, m7
psrad m0, m4
psrad m1, m4
psrad m2, m4
psrad m3, m4
packssdw m0, m1
packssdw m2, m3
mov%1 [outq + lenq ], m0
mov%1 [outq + lenq + mmsize], m2
add lenq, mmsize*2
jl .next
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
INIT_MMX mmx
MIX1_INT16 u
MIX1_INT16 a
MIX2_INT16 u
MIX2_INT16 a
INIT_XMM sse
MIX2_FLT u
MIX2_FLT a
MIX1_FLT u
MIX1_FLT a
INIT_XMM sse2
MIX1_INT16 u
MIX1_INT16 a
MIX2_INT16 u
MIX2_INT16 a
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
MIX2_FLT u
MIX2_FLT a
MIX1_FLT u
MIX1_FLT a
%endif

90
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/rematrix_init.c vendored Normal file
View file

@ -0,0 +1,90 @@
/*
* Copyright (C) 2012 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/x86/cpu.h"
#include "libswresample/swresample_internal.h"
#define D(type, simd) \
mix_1_1_func_type ff_mix_1_1_a_## type ## _ ## simd;\
mix_2_1_func_type ff_mix_2_1_a_## type ## _ ## simd;
D(float, sse)
D(float, avx)
D(int16, mmx)
D(int16, sse2)
av_cold int swri_rematrix_init_x86(struct SwrContext *s){
#if HAVE_X86ASM
int mm_flags = av_get_cpu_flags();
int nb_in = s->used_ch_count;
int nb_out = s->out.ch_count;
int num = nb_in * nb_out;
int i,j;
s->mix_1_1_simd = NULL;
s->mix_2_1_simd = NULL;
if (s->midbuf.fmt == AV_SAMPLE_FMT_S16P){
if(EXTERNAL_MMX(mm_flags)) {
s->mix_1_1_simd = ff_mix_1_1_a_int16_mmx;
s->mix_2_1_simd = ff_mix_2_1_a_int16_mmx;
}
if(EXTERNAL_SSE2(mm_flags)) {
s->mix_1_1_simd = ff_mix_1_1_a_int16_sse2;
s->mix_2_1_simd = ff_mix_2_1_a_int16_sse2;
}
s->native_simd_matrix = av_mallocz_array(num, 2 * sizeof(int16_t));
s->native_simd_one = av_mallocz(2 * sizeof(int16_t));
if (!s->native_simd_matrix || !s->native_simd_one)
return AVERROR(ENOMEM);
for(i=0; i<nb_out; i++){
int sh = 0;
for(j=0; j<nb_in; j++)
sh = FFMAX(sh, FFABS(((int*)s->native_matrix)[i * nb_in + j]));
sh = FFMAX(av_log2(sh) - 14, 0);
for(j=0; j<nb_in; j++) {
((int16_t*)s->native_simd_matrix)[2*(i * nb_in + j)+1] = 15 - sh;
((int16_t*)s->native_simd_matrix)[2*(i * nb_in + j)] =
((((int*)s->native_matrix)[i * nb_in + j]) + (1<<sh>>1)) >> sh;
}
}
((int16_t*)s->native_simd_one)[1] = 14;
((int16_t*)s->native_simd_one)[0] = 16384;
} else if(s->midbuf.fmt == AV_SAMPLE_FMT_FLTP){
if(EXTERNAL_SSE(mm_flags)) {
s->mix_1_1_simd = ff_mix_1_1_a_float_sse;
s->mix_2_1_simd = ff_mix_2_1_a_float_sse;
}
if(EXTERNAL_AVX_FAST(mm_flags)) {
s->mix_1_1_simd = ff_mix_1_1_a_float_avx;
s->mix_2_1_simd = ff_mix_2_1_a_float_avx;
}
s->native_simd_matrix = av_mallocz_array(num, sizeof(float));
s->native_simd_one = av_mallocz(sizeof(float));
if (!s->native_simd_matrix || !s->native_simd_one)
return AVERROR(ENOMEM);
memcpy(s->native_simd_matrix, s->native_matrix, num * sizeof(float));
memcpy(s->native_simd_one, s->native_one, sizeof(float));
}
#endif
return 0;
}

619
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/resample.asm vendored Normal file
View file

@ -0,0 +1,619 @@
;******************************************************************************
;* Copyright (c) 2012 Michael Niedermayer
;* Copyright (c) 2014 James Almer <jamrial <at> gmail.com>
;* Copyright (c) 2014 Ronald S. Bultje <rsbultje@gmail.com>
;*
;* This file is part of FFmpeg.
;*
;* FFmpeg is free software; you can redistribute it and/or
;* modify it under the terms of the GNU Lesser General Public
;* License as published by the Free Software Foundation; either
;* version 2.1 of the License, or (at your option) any later version.
;*
;* FFmpeg 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
;* Lesser General Public License for more details.
;*
;* You should have received a copy of the GNU Lesser General Public
;* License along with FFmpeg; if not, write to the Free Software
;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
;******************************************************************************
%include "libavutil/x86/x86util.asm"
%if ARCH_X86_64
%define pointer resq
%else
%define pointer resd
%endif
struc ResampleContext
.av_class: pointer 1
.filter_bank: pointer 1
.filter_length: resd 1
.filter_alloc: resd 1
.ideal_dst_incr: resd 1
.dst_incr: resd 1
.dst_incr_div: resd 1
.dst_incr_mod: resd 1
.index: resd 1
.frac: resd 1
.src_incr: resd 1
.compensation_distance: resd 1
.phase_count: resd 1
; there's a few more here but we only care about the first few
endstruc
SECTION_RODATA
pf_1: dd 1.0
pdbl_1: dq 1.0
pd_0x4000: dd 0x4000
SECTION .text
; FIXME remove unneeded variables (index_incr, phase_mask)
%macro RESAMPLE_FNS 3-5 ; format [float or int16], bps, log2_bps, float op suffix [s or d], 1.0 constant
; int resample_common_$format(ResampleContext *ctx, $format *dst,
; const $format *src, int size, int update_ctx)
%if ARCH_X86_64 ; unix64 and win64
cglobal resample_common_%1, 0, 15, 2, ctx, dst, src, phase_count, index, frac, \
dst_incr_mod, size, min_filter_count_x4, \
min_filter_len_x4, dst_incr_div, src_incr, \
phase_mask, dst_end, filter_bank
; use red-zone for variable storage
%define ctx_stackq [rsp-0x8]
%define src_stackq [rsp-0x10]
%if WIN64
%define update_context_stackd r4m
%else ; unix64
%define update_context_stackd [rsp-0x14]
%endif
; load as many variables in registers as possible; for the rest, store
; on stack so that we have 'ctx' available as one extra register
mov sized, r3d
%if UNIX64
mov update_context_stackd, r4d
%endif
mov indexd, [ctxq+ResampleContext.index]
mov fracd, [ctxq+ResampleContext.frac]
mov dst_incr_modd, [ctxq+ResampleContext.dst_incr_mod]
mov filter_bankq, [ctxq+ResampleContext.filter_bank]
mov src_incrd, [ctxq+ResampleContext.src_incr]
mov ctx_stackq, ctxq
mov min_filter_len_x4d, [ctxq+ResampleContext.filter_length]
mov dst_incr_divd, [ctxq+ResampleContext.dst_incr_div]
shl min_filter_len_x4d, %3
lea dst_endq, [dstq+sizeq*%2]
%if UNIX64
mov ecx, [ctxq+ResampleContext.phase_count]
mov edi, [ctxq+ResampleContext.filter_alloc]
DEFINE_ARGS filter_alloc, dst, src, phase_count, index, frac, dst_incr_mod, \
filter, min_filter_count_x4, min_filter_len_x4, dst_incr_div, \
src_incr, phase_mask, dst_end, filter_bank
%elif WIN64
mov R9d, [ctxq+ResampleContext.filter_alloc]
mov ecx, [ctxq+ResampleContext.phase_count]
DEFINE_ARGS phase_count, dst, src, filter_alloc, index, frac, dst_incr_mod, \
filter, min_filter_count_x4, min_filter_len_x4, dst_incr_div, \
src_incr, phase_mask, dst_end, filter_bank
%endif
neg min_filter_len_x4q
sub filter_bankq, min_filter_len_x4q
sub srcq, min_filter_len_x4q
mov src_stackq, srcq
%else ; x86-32
cglobal resample_common_%1, 1, 7, 2, ctx, phase_count, dst, frac, \
index, min_filter_length_x4, filter_bank
; push temp variables to stack
%define ctx_stackq r0mp
%define src_stackq r2mp
%define update_context_stackd r4m
mov dstq, r1mp
mov r3, r3mp
lea r3, [dstq+r3*%2]
PUSH dword [ctxq+ResampleContext.dst_incr_div]
PUSH dword [ctxq+ResampleContext.dst_incr_mod]
PUSH dword [ctxq+ResampleContext.filter_alloc]
PUSH r3
PUSH dword [ctxq+ResampleContext.phase_count] ; unneeded replacement for phase_mask
PUSH dword [ctxq+ResampleContext.src_incr]
mov min_filter_length_x4d, [ctxq+ResampleContext.filter_length]
mov indexd, [ctxq+ResampleContext.index]
shl min_filter_length_x4d, %3
mov fracd, [ctxq+ResampleContext.frac]
neg min_filter_length_x4q
mov filter_bankq, [ctxq+ResampleContext.filter_bank]
sub r2mp, min_filter_length_x4q
sub filter_bankq, min_filter_length_x4q
PUSH min_filter_length_x4q
PUSH filter_bankq
mov phase_countd, [ctxq+ResampleContext.phase_count]
DEFINE_ARGS src, phase_count, dst, frac, index, min_filter_count_x4, filter
%define filter_bankq dword [rsp+0x0]
%define min_filter_length_x4q dword [rsp+0x4]
%define src_incrd dword [rsp+0x8]
%define phase_maskd dword [rsp+0xc]
%define dst_endq dword [rsp+0x10]
%define filter_allocd dword [rsp+0x14]
%define dst_incr_modd dword [rsp+0x18]
%define dst_incr_divd dword [rsp+0x1c]
mov srcq, r2mp
%endif
.loop:
mov filterd, filter_allocd
imul filterd, indexd
%if ARCH_X86_64
mov min_filter_count_x4q, min_filter_len_x4q
lea filterq, [filter_bankq+filterq*%2]
%else ; x86-32
mov min_filter_count_x4q, filter_bankq
lea filterq, [min_filter_count_x4q+filterq*%2]
mov min_filter_count_x4q, min_filter_length_x4q
%endif
%ifidn %1, int16
movd m0, [pd_0x4000]
%else ; float/double
xorps m0, m0, m0
%endif
align 16
.inner_loop:
movu m1, [srcq+min_filter_count_x4q*1]
%ifidn %1, int16
%if cpuflag(xop)
vpmadcswd m0, m1, [filterq+min_filter_count_x4q*1], m0
%else
pmaddwd m1, [filterq+min_filter_count_x4q*1]
paddd m0, m1
%endif
%else ; float/double
%if cpuflag(fma4) || cpuflag(fma3)
fmaddp%4 m0, m1, [filterq+min_filter_count_x4q*1], m0
%else
mulp%4 m1, m1, [filterq+min_filter_count_x4q*1]
addp%4 m0, m0, m1
%endif ; cpuflag
%endif
add min_filter_count_x4q, mmsize
js .inner_loop
%ifidn %1, int16
HADDD m0, m1
psrad m0, 15
add fracd, dst_incr_modd
packssdw m0, m0
add indexd, dst_incr_divd
movd [dstq], m0
%else ; float/double
; horizontal sum & store
%if mmsize == 32
vextractf128 xm1, m0, 0x1
addp%4 xm0, xm1
%endif
movhlps xm1, xm0
%ifidn %1, float
addps xm0, xm1
shufps xm1, xm0, xm0, q0001
%endif
add fracd, dst_incr_modd
addp%4 xm0, xm1
add indexd, dst_incr_divd
movs%4 [dstq], xm0
%endif
cmp fracd, src_incrd
jl .skip
sub fracd, src_incrd
inc indexd
%if UNIX64
DEFINE_ARGS filter_alloc, dst, src, phase_count, index, frac, dst_incr_mod, \
index_incr, min_filter_count_x4, min_filter_len_x4, dst_incr_div, \
src_incr, phase_mask, dst_end, filter_bank
%elif WIN64
DEFINE_ARGS phase_count, dst, src, filter_alloc, index, frac, dst_incr_mod, \
index_incr, min_filter_count_x4, min_filter_len_x4, dst_incr_div, \
src_incr, phase_mask, dst_end, filter_bank
%else ; x86-32
DEFINE_ARGS src, phase_count, dst, frac, index, index_incr
%endif
.skip:
add dstq, %2
cmp indexd, phase_countd
jb .index_skip
.index_while:
sub indexd, phase_countd
lea srcq, [srcq+%2]
cmp indexd, phase_countd
jnb .index_while
.index_skip:
cmp dstq, dst_endq
jne .loop
%if ARCH_X86_64
DEFINE_ARGS ctx, dst, src, phase_count, index, frac
%else ; x86-32
DEFINE_ARGS src, ctx, update_context, frac, index
%endif
cmp dword update_context_stackd, 0
jz .skip_store
; strictly speaking, the function should always return the consumed
; number of bytes; however, we only use the value if update_context
; is true, so let's just leave it uninitialized otherwise
mov ctxq, ctx_stackq
movifnidn rax, srcq
mov [ctxq+ResampleContext.frac ], fracd
sub rax, src_stackq
mov [ctxq+ResampleContext.index], indexd
shr rax, %3
.skip_store:
%if ARCH_X86_32
ADD rsp, 0x20
%endif
RET
; int resample_linear_$format(ResampleContext *ctx, float *dst,
; const float *src, int size, int update_ctx)
%if ARCH_X86_64 ; unix64 and win64
%if UNIX64
cglobal resample_linear_%1, 0, 15, 5, ctx, dst, phase_mask, phase_count, index, frac, \
size, dst_incr_mod, min_filter_count_x4, \
min_filter_len_x4, dst_incr_div, src_incr, \
src, dst_end, filter_bank
mov srcq, r2mp
%else ; win64
cglobal resample_linear_%1, 0, 15, 5, ctx, phase_mask, src, phase_count, index, frac, \
size, dst_incr_mod, min_filter_count_x4, \
min_filter_len_x4, dst_incr_div, src_incr, \
dst, dst_end, filter_bank
mov dstq, r1mp
%endif
; use red-zone for variable storage
%define ctx_stackq [rsp-0x8]
%define src_stackq [rsp-0x10]
%define phase_mask_stackd [rsp-0x14]
%if WIN64
%define update_context_stackd r4m
%else ; unix64
%define update_context_stackd [rsp-0x18]
%endif
; load as many variables in registers as possible; for the rest, store
; on stack so that we have 'ctx' available as one extra register
mov sized, r3d
%if UNIX64
mov update_context_stackd, r4d
%endif
mov indexd, [ctxq+ResampleContext.index]
mov fracd, [ctxq+ResampleContext.frac]
mov dst_incr_modd, [ctxq+ResampleContext.dst_incr_mod]
mov filter_bankq, [ctxq+ResampleContext.filter_bank]
mov src_incrd, [ctxq+ResampleContext.src_incr]
mov ctx_stackq, ctxq
mov min_filter_len_x4d, [ctxq+ResampleContext.filter_length]
%ifidn %1, int16
movd m4, [pd_0x4000]
%else ; float/double
cvtsi2s%4 xm0, src_incrd
movs%4 xm4, [%5]
divs%4 xm4, xm0
%endif
mov dst_incr_divd, [ctxq+ResampleContext.dst_incr_div]
shl min_filter_len_x4d, %3
lea dst_endq, [dstq+sizeq*%2]
%if UNIX64
mov ecx, [ctxq+ResampleContext.phase_count]
mov edi, [ctxq+ResampleContext.filter_alloc]
DEFINE_ARGS filter_alloc, dst, filter2, phase_count, index, frac, filter1, \
dst_incr_mod, min_filter_count_x4, min_filter_len_x4, \
dst_incr_div, src_incr, src, dst_end, filter_bank
%elif WIN64
mov R9d, [ctxq+ResampleContext.filter_alloc]
mov ecx, [ctxq+ResampleContext.phase_count]
DEFINE_ARGS phase_count, filter2, src, filter_alloc, index, frac, filter1, \
dst_incr_mod, min_filter_count_x4, min_filter_len_x4, \
dst_incr_div, src_incr, dst, dst_end, filter_bank
%endif
neg min_filter_len_x4q
sub filter_bankq, min_filter_len_x4q
sub srcq, min_filter_len_x4q
mov src_stackq, srcq
%else ; x86-32
cglobal resample_linear_%1, 1, 7, 5, ctx, min_filter_length_x4, filter2, \
frac, index, dst, filter_bank
; push temp variables to stack
%define ctx_stackq r0mp
%define src_stackq r2mp
%define update_context_stackd r4m
mov dstq, r1mp
mov r3, r3mp
lea r3, [dstq+r3*%2]
PUSH dword [ctxq+ResampleContext.dst_incr_div]
PUSH r3
mov r3, dword [ctxq+ResampleContext.filter_alloc]
PUSH dword [ctxq+ResampleContext.dst_incr_mod]
PUSH r3
shl r3, %3
PUSH r3
mov r3, dword [ctxq+ResampleContext.src_incr]
PUSH dword [ctxq+ResampleContext.phase_count] ; unneeded replacement of phase_mask
PUSH r3d
%ifidn %1, int16
movd m4, [pd_0x4000]
%else ; float/double
cvtsi2s%4 xm0, r3d
movs%4 xm4, [%5]
divs%4 xm4, xm0
%endif
mov min_filter_length_x4d, [ctxq+ResampleContext.filter_length]
mov indexd, [ctxq+ResampleContext.index]
shl min_filter_length_x4d, %3
mov fracd, [ctxq+ResampleContext.frac]
neg min_filter_length_x4q
mov filter_bankq, [ctxq+ResampleContext.filter_bank]
sub r2mp, min_filter_length_x4q
sub filter_bankq, min_filter_length_x4q
PUSH min_filter_length_x4q
PUSH filter_bankq
PUSH dword [ctxq+ResampleContext.phase_count]
DEFINE_ARGS filter1, min_filter_count_x4, filter2, frac, index, dst, src
%define phase_count_stackd dword [rsp+0x0]
%define filter_bankq dword [rsp+0x4]
%define min_filter_length_x4q dword [rsp+0x8]
%define src_incrd dword [rsp+0xc]
%define phase_mask_stackd dword [rsp+0x10]
%define filter_alloc_x4q dword [rsp+0x14]
%define filter_allocd dword [rsp+0x18]
%define dst_incr_modd dword [rsp+0x1c]
%define dst_endq dword [rsp+0x20]
%define dst_incr_divd dword [rsp+0x24]
mov srcq, r2mp
%endif
.loop:
mov filter1d, filter_allocd
imul filter1d, indexd
%if ARCH_X86_64
mov min_filter_count_x4q, min_filter_len_x4q
lea filter1q, [filter_bankq+filter1q*%2]
lea filter2q, [filter1q+filter_allocq*%2]
%else ; x86-32
mov min_filter_count_x4q, filter_bankq
lea filter1q, [min_filter_count_x4q+filter1q*%2]
mov min_filter_count_x4q, min_filter_length_x4q
mov filter2q, filter1q
add filter2q, filter_alloc_x4q
%endif
%ifidn %1, int16
mova m0, m4
mova m2, m4
%else ; float/double
xorps m0, m0, m0
xorps m2, m2, m2
%endif
align 16
.inner_loop:
movu m1, [srcq+min_filter_count_x4q*1]
%ifidn %1, int16
%if cpuflag(xop)
vpmadcswd m2, m1, [filter2q+min_filter_count_x4q*1], m2
vpmadcswd m0, m1, [filter1q+min_filter_count_x4q*1], m0
%else
pmaddwd m3, m1, [filter2q+min_filter_count_x4q*1]
pmaddwd m1, [filter1q+min_filter_count_x4q*1]
paddd m2, m3
paddd m0, m1
%endif ; cpuflag
%else ; float/double
%if cpuflag(fma4) || cpuflag(fma3)
fmaddp%4 m2, m1, [filter2q+min_filter_count_x4q*1], m2
fmaddp%4 m0, m1, [filter1q+min_filter_count_x4q*1], m0
%else
mulp%4 m3, m1, [filter2q+min_filter_count_x4q*1]
mulp%4 m1, m1, [filter1q+min_filter_count_x4q*1]
addp%4 m2, m2, m3
addp%4 m0, m0, m1
%endif ; cpuflag
%endif
add min_filter_count_x4q, mmsize
js .inner_loop
%ifidn %1, int16
%if mmsize == 16
%if cpuflag(xop)
vphadddq m2, m2
vphadddq m0, m0
%endif
pshufd m3, m2, q0032
pshufd m1, m0, q0032
paddd m2, m3
paddd m0, m1
%endif
%if notcpuflag(xop)
PSHUFLW m3, m2, q0032
PSHUFLW m1, m0, q0032
paddd m2, m3
paddd m0, m1
%endif
psubd m2, m0
; This is probably a really bad idea on atom and other machines with a
; long transfer latency between GPRs and XMMs (atom). However, it does
; make the clip a lot simpler...
movd eax, m2
add indexd, dst_incr_divd
imul fracd
idiv src_incrd
movd m1, eax
add fracd, dst_incr_modd
paddd m0, m1
psrad m0, 15
packssdw m0, m0
movd [dstq], m0
; note that for imul/idiv, I need to move filter to edx/eax for each:
; - 32bit: eax=r0[filter1], edx=r2[filter2]
; - win64: eax=r6[filter1], edx=r1[todo]
; - unix64: eax=r6[filter1], edx=r2[todo]
%else ; float/double
; val += (v2 - val) * (FELEML) frac / c->src_incr;
%if mmsize == 32
vextractf128 xm1, m0, 0x1
vextractf128 xm3, m2, 0x1
addp%4 xm0, xm1
addp%4 xm2, xm3
%endif
cvtsi2s%4 xm1, fracd
subp%4 xm2, xm0
mulp%4 xm1, xm4
shufp%4 xm1, xm1, q0000
%if cpuflag(fma4) || cpuflag(fma3)
fmaddp%4 xm0, xm2, xm1, xm0
%else
mulp%4 xm2, xm1
addp%4 xm0, xm2
%endif ; cpuflag
; horizontal sum & store
movhlps xm1, xm0
%ifidn %1, float
addps xm0, xm1
shufps xm1, xm0, xm0, q0001
%endif
add fracd, dst_incr_modd
addp%4 xm0, xm1
add indexd, dst_incr_divd
movs%4 [dstq], xm0
%endif
cmp fracd, src_incrd
jl .skip
sub fracd, src_incrd
inc indexd
%if UNIX64
DEFINE_ARGS filter_alloc, dst, filter2, phase_count, index, frac, index_incr, \
dst_incr_mod, min_filter_count_x4, min_filter_len_x4, \
dst_incr_div, src_incr, src, dst_end, filter_bank
%elif WIN64
DEFINE_ARGS phase_count, filter2, src, filter_alloc, index, frac, index_incr, \
dst_incr_mod, min_filter_count_x4, min_filter_len_x4, \
dst_incr_div, src_incr, dst, dst_end, filter_bank
%else ; x86-32
DEFINE_ARGS filter1, phase_count, index_incr, frac, index, dst, src
%endif
.skip:
%if ARCH_X86_32
mov phase_countd, phase_count_stackd
%endif
add dstq, %2
cmp indexd, phase_countd
jb .index_skip
.index_while:
sub indexd, phase_countd
lea srcq, [srcq+%2]
cmp indexd, phase_countd
jnb .index_while
.index_skip:
cmp dstq, dst_endq
jne .loop
%if UNIX64
DEFINE_ARGS ctx, dst, filter2, phase_count, index, frac, index_incr, \
dst_incr_mod, min_filter_count_x4, min_filter_len_x4, \
dst_incr_div, src_incr, src, dst_end, filter_bank
%elif WIN64
DEFINE_ARGS ctx, filter2, src, phase_count, index, frac, index_incr, \
dst_incr_mod, min_filter_count_x4, min_filter_len_x4, \
dst_incr_div, src_incr, dst, dst_end, filter_bank
%else ; x86-32
DEFINE_ARGS filter1, ctx, update_context, frac, index, dst, src
%endif
cmp dword update_context_stackd, 0
jz .skip_store
; strictly speaking, the function should always return the consumed
; number of bytes; however, we only use the value if update_context
; is true, so let's just leave it uninitialized otherwise
mov ctxq, ctx_stackq
movifnidn rax, srcq
mov [ctxq+ResampleContext.frac ], fracd
sub rax, src_stackq
mov [ctxq+ResampleContext.index], indexd
shr rax, %3
.skip_store:
%if ARCH_X86_32
ADD rsp, 0x28
%endif
RET
%endmacro
INIT_XMM sse
RESAMPLE_FNS float, 4, 2, s, pf_1
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
RESAMPLE_FNS float, 4, 2, s, pf_1
%endif
%if HAVE_FMA3_EXTERNAL
INIT_YMM fma3
RESAMPLE_FNS float, 4, 2, s, pf_1
%endif
%if HAVE_FMA4_EXTERNAL
INIT_XMM fma4
RESAMPLE_FNS float, 4, 2, s, pf_1
%endif
%if ARCH_X86_32
INIT_MMX mmxext
RESAMPLE_FNS int16, 2, 1
%endif
INIT_XMM sse2
RESAMPLE_FNS int16, 2, 1
%if HAVE_XOP_EXTERNAL
INIT_XMM xop
RESAMPLE_FNS int16, 2, 1
%endif
INIT_XMM sse2
RESAMPLE_FNS double, 8, 3, d, pdbl_1
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
RESAMPLE_FNS double, 8, 3, d, pdbl_1
%endif
%if HAVE_FMA3_EXTERNAL
INIT_YMM fma3
RESAMPLE_FNS double, 8, 3, d, pdbl_1
%endif

100
trunk/3rdparty/ffmpeg-4-fit/libswresample/x86/resample_init.c vendored Normal file
View file

@ -0,0 +1,100 @@
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/x86/cpu.h"
#include "libswresample/resample.h"
#define RESAMPLE_FUNCS(type, opt) \
int ff_resample_common_##type##_##opt(ResampleContext *c, void *dst, \
const void *src, int sz, int upd); \
int ff_resample_linear_##type##_##opt(ResampleContext *c, void *dst, \
const void *src, int sz, int upd)
RESAMPLE_FUNCS(int16, mmxext);
RESAMPLE_FUNCS(int16, sse2);
RESAMPLE_FUNCS(int16, xop);
RESAMPLE_FUNCS(float, sse);
RESAMPLE_FUNCS(float, avx);
RESAMPLE_FUNCS(float, fma3);
RESAMPLE_FUNCS(float, fma4);
RESAMPLE_FUNCS(double, sse2);
RESAMPLE_FUNCS(double, avx);
RESAMPLE_FUNCS(double, fma3);
av_cold void swri_resample_dsp_x86_init(ResampleContext *c)
{
int av_unused mm_flags = av_get_cpu_flags();
switch(c->format){
case AV_SAMPLE_FMT_S16P:
if (ARCH_X86_32 && EXTERNAL_MMXEXT(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_int16_mmxext;
c->dsp.resample_common = ff_resample_common_int16_mmxext;
}
if (EXTERNAL_SSE2(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_int16_sse2;
c->dsp.resample_common = ff_resample_common_int16_sse2;
}
if (EXTERNAL_XOP(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_int16_xop;
c->dsp.resample_common = ff_resample_common_int16_xop;
}
break;
case AV_SAMPLE_FMT_FLTP:
if (EXTERNAL_SSE(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_float_sse;
c->dsp.resample_common = ff_resample_common_float_sse;
}
if (EXTERNAL_AVX_FAST(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_float_avx;
c->dsp.resample_common = ff_resample_common_float_avx;
}
if (EXTERNAL_FMA3_FAST(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_float_fma3;
c->dsp.resample_common = ff_resample_common_float_fma3;
}
if (EXTERNAL_FMA4(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_float_fma4;
c->dsp.resample_common = ff_resample_common_float_fma4;
}
break;
case AV_SAMPLE_FMT_DBLP:
if (EXTERNAL_SSE2(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_double_sse2;
c->dsp.resample_common = ff_resample_common_double_sse2;
}
if (EXTERNAL_AVX_FAST(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_double_avx;
c->dsp.resample_common = ff_resample_common_double_avx;
}
if (EXTERNAL_FMA3_FAST(mm_flags)) {
c->dsp.resample_linear = ff_resample_linear_double_fma3;
c->dsp.resample_common = ff_resample_common_double_fma3;
}
break;
}
}