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Regard whitespace Rev 4348 → Rev 4349

/contrib/sdk/sources/ffmpeg/libavresample/Makefile
0,0 → 1,16
NAME = avresample
FFLIBS = avutil
 
HEADERS = avresample.h \
version.h \
 
OBJS = audio_convert.o \
audio_data.o \
audio_mix.o \
audio_mix_matrix.o \
dither.o \
options.o \
resample.o \
utils.o \
 
TESTPROGS = avresample
/contrib/sdk/sources/ffmpeg/libavresample/arm/Makefile
0,0 → 1,2
OBJS += arm/audio_convert_init.o
NEON-OBJS += arm/audio_convert_neon.o
/contrib/sdk/sources/ffmpeg/libavresample/arm/audio_convert_init.c
0,0 → 1,49
/*
* 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 <stdint.h>
 
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/cpu.h"
#include "libavutil/arm/cpu.h"
#include "libavutil/samplefmt.h"
#include "libavresample/audio_convert.h"
 
void ff_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len);
void ff_conv_fltp_to_s16_neon(int16_t *dst, float *const *src,
int len, int channels);
void ff_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src,
int len, int channels);
 
av_cold void ff_audio_convert_init_arm(AudioConvert *ac)
{
int cpu_flags = av_get_cpu_flags();
 
if (have_neon(cpu_flags)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT,
0, 16, 8, "NEON",
ff_conv_flt_to_s16_neon);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
0, 16, 8, "NEON",
ff_conv_fltp_to_s16_neon);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
2, 16, 8, "NEON",
ff_conv_fltp_to_s16_2ch_neon);
}
}
/contrib/sdk/sources/ffmpeg/libavresample/arm/audio_convert_neon.S
0,0 → 1,363
/*
* Copyright (c) 2008 Mans Rullgard <mans@mansr.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 "config.h"
#include "libavutil/arm/asm.S"
 
function ff_conv_flt_to_s16_neon, export=1
subs r2, r2, #8
vld1.32 {q0}, [r1,:128]!
vcvt.s32.f32 q8, q0, #31
vld1.32 {q1}, [r1,:128]!
vcvt.s32.f32 q9, q1, #31
beq 3f
bics r12, r2, #15
beq 2f
1: subs r12, r12, #16
vqrshrn.s32 d4, q8, #16
vld1.32 {q0}, [r1,:128]!
vcvt.s32.f32 q0, q0, #31
vqrshrn.s32 d5, q9, #16
vld1.32 {q1}, [r1,:128]!
vcvt.s32.f32 q1, q1, #31
vqrshrn.s32 d6, q0, #16
vst1.16 {q2}, [r0,:128]!
vqrshrn.s32 d7, q1, #16
vld1.32 {q8}, [r1,:128]!
vcvt.s32.f32 q8, q8, #31
vld1.32 {q9}, [r1,:128]!
vcvt.s32.f32 q9, q9, #31
vst1.16 {q3}, [r0,:128]!
bne 1b
ands r2, r2, #15
beq 3f
2: vld1.32 {q0}, [r1,:128]!
vqrshrn.s32 d4, q8, #16
vcvt.s32.f32 q0, q0, #31
vld1.32 {q1}, [r1,:128]!
vqrshrn.s32 d5, q9, #16
vcvt.s32.f32 q1, q1, #31
vqrshrn.s32 d6, q0, #16
vst1.16 {q2}, [r0,:128]!
vqrshrn.s32 d7, q1, #16
vst1.16 {q3}, [r0,:128]!
bx lr
3: vqrshrn.s32 d4, q8, #16
vqrshrn.s32 d5, q9, #16
vst1.16 {q2}, [r0,:128]!
bx lr
endfunc
 
function ff_conv_fltp_to_s16_2ch_neon, export=1
ldm r1, {r1, r3}
subs r2, r2, #8
vld1.32 {q0}, [r1,:128]!
vcvt.s32.f32 q8, q0, #31
vld1.32 {q1}, [r1,:128]!
vcvt.s32.f32 q9, q1, #31
vld1.32 {q10}, [r3,:128]!
vcvt.s32.f32 q10, q10, #31
vld1.32 {q11}, [r3,:128]!
vcvt.s32.f32 q11, q11, #31
beq 3f
bics r12, r2, #15
beq 2f
1: subs r12, r12, #16
vld1.32 {q0}, [r1,:128]!
vcvt.s32.f32 q0, q0, #31
vsri.32 q10, q8, #16
vld1.32 {q1}, [r1,:128]!
vcvt.s32.f32 q1, q1, #31
vld1.32 {q12}, [r3,:128]!
vcvt.s32.f32 q12, q12, #31
vld1.32 {q13}, [r3,:128]!
vsri.32 q11, q9, #16
vst1.16 {q10}, [r0,:128]!
vcvt.s32.f32 q13, q13, #31
vst1.16 {q11}, [r0,:128]!
vsri.32 q12, q0, #16
vld1.32 {q8}, [r1,:128]!
vsri.32 q13, q1, #16
vst1.16 {q12}, [r0,:128]!
vcvt.s32.f32 q8, q8, #31
vld1.32 {q9}, [r1,:128]!
vcvt.s32.f32 q9, q9, #31
vld1.32 {q10}, [r3,:128]!
vcvt.s32.f32 q10, q10, #31
vld1.32 {q11}, [r3,:128]!
vcvt.s32.f32 q11, q11, #31
vst1.16 {q13}, [r0,:128]!
bne 1b
ands r2, r2, #15
beq 3f
2: vsri.32 q10, q8, #16
vld1.32 {q0}, [r1,:128]!
vcvt.s32.f32 q0, q0, #31
vld1.32 {q1}, [r1,:128]!
vcvt.s32.f32 q1, q1, #31
vld1.32 {q12}, [r3,:128]!
vcvt.s32.f32 q12, q12, #31
vsri.32 q11, q9, #16
vld1.32 {q13}, [r3,:128]!
vcvt.s32.f32 q13, q13, #31
vst1.16 {q10}, [r0,:128]!
vsri.32 q12, q0, #16
vst1.16 {q11}, [r0,:128]!
vsri.32 q13, q1, #16
vst1.16 {q12-q13},[r0,:128]!
bx lr
3: vsri.32 q10, q8, #16
vsri.32 q11, q9, #16
vst1.16 {q10-q11},[r0,:128]!
bx lr
endfunc
 
function ff_conv_fltp_to_s16_neon, export=1
cmp r3, #2
itt lt
ldrlt r1, [r1]
blt ff_conv_flt_to_s16_neon
beq ff_conv_fltp_to_s16_2ch_neon
 
push {r4-r8, lr}
cmp r3, #4
lsl r12, r3, #1
blt 4f
 
@ 4 channels
5: ldm r1!, {r4-r7}
mov lr, r2
mov r8, r0
vld1.32 {q8}, [r4,:128]!
vcvt.s32.f32 q8, q8, #31
vld1.32 {q9}, [r5,:128]!
vcvt.s32.f32 q9, q9, #31
vld1.32 {q10}, [r6,:128]!
vcvt.s32.f32 q10, q10, #31
vld1.32 {q11}, [r7,:128]!
vcvt.s32.f32 q11, q11, #31
6: subs lr, lr, #8
vld1.32 {q0}, [r4,:128]!
vcvt.s32.f32 q0, q0, #31
vsri.32 q9, q8, #16
vld1.32 {q1}, [r5,:128]!
vcvt.s32.f32 q1, q1, #31
vsri.32 q11, q10, #16
vld1.32 {q2}, [r6,:128]!
vcvt.s32.f32 q2, q2, #31
vzip.32 d18, d22
vld1.32 {q3}, [r7,:128]!
vcvt.s32.f32 q3, q3, #31
vzip.32 d19, d23
vst1.16 {d18}, [r8], r12
vsri.32 q1, q0, #16
vst1.16 {d22}, [r8], r12
vsri.32 q3, q2, #16
vst1.16 {d19}, [r8], r12
vzip.32 d2, d6
vst1.16 {d23}, [r8], r12
vzip.32 d3, d7
beq 7f
vld1.32 {q8}, [r4,:128]!
vcvt.s32.f32 q8, q8, #31
vst1.16 {d2}, [r8], r12
vld1.32 {q9}, [r5,:128]!
vcvt.s32.f32 q9, q9, #31
vst1.16 {d6}, [r8], r12
vld1.32 {q10}, [r6,:128]!
vcvt.s32.f32 q10, q10, #31
vst1.16 {d3}, [r8], r12
vld1.32 {q11}, [r7,:128]!
vcvt.s32.f32 q11, q11, #31
vst1.16 {d7}, [r8], r12
b 6b
7: vst1.16 {d2}, [r8], r12
vst1.16 {d6}, [r8], r12
vst1.16 {d3}, [r8], r12
vst1.16 {d7}, [r8], r12
subs r3, r3, #4
it eq
popeq {r4-r8, pc}
cmp r3, #4
add r0, r0, #8
bge 5b
 
@ 2 channels
4: cmp r3, #2
blt 4f
ldm r1!, {r4-r5}
mov lr, r2
mov r8, r0
tst lr, #8
vld1.32 {q8}, [r4,:128]!
vcvt.s32.f32 q8, q8, #31
vld1.32 {q9}, [r5,:128]!
vcvt.s32.f32 q9, q9, #31
vld1.32 {q10}, [r4,:128]!
vcvt.s32.f32 q10, q10, #31
vld1.32 {q11}, [r5,:128]!
vcvt.s32.f32 q11, q11, #31
beq 6f
subs lr, lr, #8
beq 7f
vsri.32 d18, d16, #16
vsri.32 d19, d17, #16
vld1.32 {q8}, [r4,:128]!
vcvt.s32.f32 q8, q8, #31
vst1.32 {d18[0]}, [r8], r12
vsri.32 d22, d20, #16
vst1.32 {d18[1]}, [r8], r12
vsri.32 d23, d21, #16
vst1.32 {d19[0]}, [r8], r12
vst1.32 {d19[1]}, [r8], r12
vld1.32 {q9}, [r5,:128]!
vcvt.s32.f32 q9, q9, #31
vst1.32 {d22[0]}, [r8], r12
vst1.32 {d22[1]}, [r8], r12
vld1.32 {q10}, [r4,:128]!
vcvt.s32.f32 q10, q10, #31
vst1.32 {d23[0]}, [r8], r12
vst1.32 {d23[1]}, [r8], r12
vld1.32 {q11}, [r5,:128]!
vcvt.s32.f32 q11, q11, #31
6: subs lr, lr, #16
vld1.32 {q0}, [r4,:128]!
vcvt.s32.f32 q0, q0, #31
vsri.32 d18, d16, #16
vld1.32 {q1}, [r5,:128]!
vcvt.s32.f32 q1, q1, #31
vsri.32 d19, d17, #16
vld1.32 {q2}, [r4,:128]!
vcvt.s32.f32 q2, q2, #31
vld1.32 {q3}, [r5,:128]!
vcvt.s32.f32 q3, q3, #31
vst1.32 {d18[0]}, [r8], r12
vsri.32 d22, d20, #16
vst1.32 {d18[1]}, [r8], r12
vsri.32 d23, d21, #16
vst1.32 {d19[0]}, [r8], r12
vsri.32 d2, d0, #16
vst1.32 {d19[1]}, [r8], r12
vsri.32 d3, d1, #16
vst1.32 {d22[0]}, [r8], r12
vsri.32 d6, d4, #16
vst1.32 {d22[1]}, [r8], r12
vsri.32 d7, d5, #16
vst1.32 {d23[0]}, [r8], r12
vst1.32 {d23[1]}, [r8], r12
beq 6f
vld1.32 {q8}, [r4,:128]!
vcvt.s32.f32 q8, q8, #31
vst1.32 {d2[0]}, [r8], r12
vst1.32 {d2[1]}, [r8], r12
vld1.32 {q9}, [r5,:128]!
vcvt.s32.f32 q9, q9, #31
vst1.32 {d3[0]}, [r8], r12
vst1.32 {d3[1]}, [r8], r12
vld1.32 {q10}, [r4,:128]!
vcvt.s32.f32 q10, q10, #31
vst1.32 {d6[0]}, [r8], r12
vst1.32 {d6[1]}, [r8], r12
vld1.32 {q11}, [r5,:128]!
vcvt.s32.f32 q11, q11, #31
vst1.32 {d7[0]}, [r8], r12
vst1.32 {d7[1]}, [r8], r12
bgt 6b
6: vst1.32 {d2[0]}, [r8], r12
vst1.32 {d2[1]}, [r8], r12
vst1.32 {d3[0]}, [r8], r12
vst1.32 {d3[1]}, [r8], r12
vst1.32 {d6[0]}, [r8], r12
vst1.32 {d6[1]}, [r8], r12
vst1.32 {d7[0]}, [r8], r12
vst1.32 {d7[1]}, [r8], r12
b 8f
7: vsri.32 d18, d16, #16
vsri.32 d19, d17, #16
vst1.32 {d18[0]}, [r8], r12
vsri.32 d22, d20, #16
vst1.32 {d18[1]}, [r8], r12
vsri.32 d23, d21, #16
vst1.32 {d19[0]}, [r8], r12
vst1.32 {d19[1]}, [r8], r12
vst1.32 {d22[0]}, [r8], r12
vst1.32 {d22[1]}, [r8], r12
vst1.32 {d23[0]}, [r8], r12
vst1.32 {d23[1]}, [r8], r12
8: subs r3, r3, #2
add r0, r0, #4
it eq
popeq {r4-r8, pc}
 
@ 1 channel
4: ldr r4, [r1]
tst r2, #8
mov lr, r2
mov r5, r0
vld1.32 {q0}, [r4,:128]!
vcvt.s32.f32 q0, q0, #31
vld1.32 {q1}, [r4,:128]!
vcvt.s32.f32 q1, q1, #31
bne 8f
6: subs lr, lr, #16
vld1.32 {q2}, [r4,:128]!
vcvt.s32.f32 q2, q2, #31
vld1.32 {q3}, [r4,:128]!
vcvt.s32.f32 q3, q3, #31
vst1.16 {d0[1]}, [r5,:16], r12
vst1.16 {d0[3]}, [r5,:16], r12
vst1.16 {d1[1]}, [r5,:16], r12
vst1.16 {d1[3]}, [r5,:16], r12
vst1.16 {d2[1]}, [r5,:16], r12
vst1.16 {d2[3]}, [r5,:16], r12
vst1.16 {d3[1]}, [r5,:16], r12
vst1.16 {d3[3]}, [r5,:16], r12
beq 7f
vld1.32 {q0}, [r4,:128]!
vcvt.s32.f32 q0, q0, #31
vld1.32 {q1}, [r4,:128]!
vcvt.s32.f32 q1, q1, #31
7: vst1.16 {d4[1]}, [r5,:16], r12
vst1.16 {d4[3]}, [r5,:16], r12
vst1.16 {d5[1]}, [r5,:16], r12
vst1.16 {d5[3]}, [r5,:16], r12
vst1.16 {d6[1]}, [r5,:16], r12
vst1.16 {d6[3]}, [r5,:16], r12
vst1.16 {d7[1]}, [r5,:16], r12
vst1.16 {d7[3]}, [r5,:16], r12
bgt 6b
pop {r4-r8, pc}
8: subs lr, lr, #8
vst1.16 {d0[1]}, [r5,:16], r12
vst1.16 {d0[3]}, [r5,:16], r12
vst1.16 {d1[1]}, [r5,:16], r12
vst1.16 {d1[3]}, [r5,:16], r12
vst1.16 {d2[1]}, [r5,:16], r12
vst1.16 {d2[3]}, [r5,:16], r12
vst1.16 {d3[1]}, [r5,:16], r12
vst1.16 {d3[3]}, [r5,:16], r12
it eq
popeq {r4-r8, pc}
vld1.32 {q0}, [r4,:128]!
vcvt.s32.f32 q0, q0, #31
vld1.32 {q1}, [r4,:128]!
vcvt.s32.f32 q1, q1, #31
b 6b
endfunc
/contrib/sdk/sources/ffmpeg/libavresample/audio_convert.c
0,0 → 1,414
/*
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 <stdint.h>
 
#include "config.h"
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/samplefmt.h"
#include "audio_convert.h"
#include "audio_data.h"
#include "dither.h"
 
enum ConvFuncType {
CONV_FUNC_TYPE_FLAT,
CONV_FUNC_TYPE_INTERLEAVE,
CONV_FUNC_TYPE_DEINTERLEAVE,
};
 
typedef void (conv_func_flat)(uint8_t *out, const uint8_t *in, int len);
 
typedef void (conv_func_interleave)(uint8_t *out, uint8_t *const *in,
int len, int channels);
 
typedef void (conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len,
int channels);
 
struct AudioConvert {
AVAudioResampleContext *avr;
DitherContext *dc;
enum AVSampleFormat in_fmt;
enum AVSampleFormat out_fmt;
int apply_map;
int channels;
int planes;
int ptr_align;
int samples_align;
int has_optimized_func;
const char *func_descr;
const char *func_descr_generic;
enum ConvFuncType func_type;
conv_func_flat *conv_flat;
conv_func_flat *conv_flat_generic;
conv_func_interleave *conv_interleave;
conv_func_interleave *conv_interleave_generic;
conv_func_deinterleave *conv_deinterleave;
conv_func_deinterleave *conv_deinterleave_generic;
};
 
void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt, int channels,
int ptr_align, int samples_align,
const char *descr, void *conv)
{
int found = 0;
 
switch (ac->func_type) {
case CONV_FUNC_TYPE_FLAT:
if (av_get_packed_sample_fmt(ac->in_fmt) == in_fmt &&
av_get_packed_sample_fmt(ac->out_fmt) == out_fmt) {
ac->conv_flat = conv;
ac->func_descr = descr;
ac->ptr_align = ptr_align;
ac->samples_align = samples_align;
if (ptr_align == 1 && samples_align == 1) {
ac->conv_flat_generic = conv;
ac->func_descr_generic = descr;
} else {
ac->has_optimized_func = 1;
}
found = 1;
}
break;
case CONV_FUNC_TYPE_INTERLEAVE:
if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
(!channels || ac->channels == channels)) {
ac->conv_interleave = conv;
ac->func_descr = descr;
ac->ptr_align = ptr_align;
ac->samples_align = samples_align;
if (ptr_align == 1 && samples_align == 1) {
ac->conv_interleave_generic = conv;
ac->func_descr_generic = descr;
} else {
ac->has_optimized_func = 1;
}
found = 1;
}
break;
case CONV_FUNC_TYPE_DEINTERLEAVE:
if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
(!channels || ac->channels == channels)) {
ac->conv_deinterleave = conv;
ac->func_descr = descr;
ac->ptr_align = ptr_align;
ac->samples_align = samples_align;
if (ptr_align == 1 && samples_align == 1) {
ac->conv_deinterleave_generic = conv;
ac->func_descr_generic = descr;
} else {
ac->has_optimized_func = 1;
}
found = 1;
}
break;
}
if (found) {
av_log(ac->avr, AV_LOG_DEBUG, "audio_convert: found function: %-4s "
"to %-4s (%s)\n", av_get_sample_fmt_name(ac->in_fmt),
av_get_sample_fmt_name(ac->out_fmt), descr);
}
}
 
#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
 
#define CONV_LOOP(otype, expr) \
do { \
*(otype *)po = expr; \
pi += is; \
po += os; \
} while (po < end); \
 
#define CONV_FUNC_FLAT(ofmt, otype, ifmt, itype, expr) \
static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t *in, \
int len) \
{ \
int is = sizeof(itype); \
int os = sizeof(otype); \
const uint8_t *pi = in; \
uint8_t *po = out; \
uint8_t *end = out + os * len; \
CONV_LOOP(otype, expr) \
}
 
#define CONV_FUNC_INTERLEAVE(ofmt, otype, ifmt, itype, expr) \
static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t **in, \
int len, int channels) \
{ \
int ch; \
int out_bps = sizeof(otype); \
int is = sizeof(itype); \
int os = channels * out_bps; \
for (ch = 0; ch < channels; ch++) { \
const uint8_t *pi = in[ch]; \
uint8_t *po = out + ch * out_bps; \
uint8_t *end = po + os * len; \
CONV_LOOP(otype, expr) \
} \
}
 
#define CONV_FUNC_DEINTERLEAVE(ofmt, otype, ifmt, itype, expr) \
static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t **out, const uint8_t *in, \
int len, int channels) \
{ \
int ch; \
int in_bps = sizeof(itype); \
int is = channels * in_bps; \
int os = sizeof(otype); \
for (ch = 0; ch < channels; ch++) { \
const uint8_t *pi = in + ch * in_bps; \
uint8_t *po = out[ch]; \
uint8_t *end = po + os * len; \
CONV_LOOP(otype, expr) \
} \
}
 
#define CONV_FUNC_GROUP(ofmt, otype, ifmt, itype, expr) \
CONV_FUNC_FLAT( ofmt, otype, ifmt, itype, expr) \
CONV_FUNC_INTERLEAVE( ofmt, otype, ifmt ## P, itype, expr) \
CONV_FUNC_DEINTERLEAVE(ofmt ## P, otype, ifmt, itype, expr)
 
CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_U8, uint8_t, *(const uint8_t *)pi)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 8)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 24)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0f / (1 << 7)))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0 / (1 << 7)))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t, (*(const int16_t *)pi >> 8) + 0x80)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi << 16)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0f / (1 << 15)))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0 / (1 << 15)))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t, (*(const int32_t *)pi >> 24) + 0x80)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi >> 16)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0f / (1U << 31)))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0 / (1U << 31)))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8( lrintf(*(const float *)pi * (1 << 7)) + 0x80))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16( lrintf(*(const float *)pi * (1 << 15))))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *)pi * (1U << 31))))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8( lrint(*(const double *)pi * (1 << 7)) + 0x80))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16( lrint(*(const double *)pi * (1 << 15))))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *)pi * (1U << 31))))
CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
 
#define SET_CONV_FUNC_GROUP(ofmt, ifmt) \
ff_audio_convert_set_func(ac, ofmt, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt)); \
ff_audio_convert_set_func(ac, ofmt ## P, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt ## P, ifmt)); \
ff_audio_convert_set_func(ac, ofmt, ifmt ## P, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt ## P));
 
static void set_generic_function(AudioConvert *ac)
{
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL)
SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL)
}
 
void ff_audio_convert_free(AudioConvert **ac)
{
if (!*ac)
return;
ff_dither_free(&(*ac)->dc);
av_freep(ac);
}
 
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int sample_rate,
int apply_map)
{
AudioConvert *ac;
int in_planar, out_planar;
 
ac = av_mallocz(sizeof(*ac));
if (!ac)
return NULL;
 
ac->avr = avr;
ac->out_fmt = out_fmt;
ac->in_fmt = in_fmt;
ac->channels = channels;
ac->apply_map = apply_map;
 
if (avr->dither_method != AV_RESAMPLE_DITHER_NONE &&
av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 &&
av_get_bytes_per_sample(in_fmt) > 2) {
ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate,
apply_map);
if (!ac->dc) {
av_free(ac);
return NULL;
}
return ac;
}
 
in_planar = av_sample_fmt_is_planar(in_fmt);
out_planar = av_sample_fmt_is_planar(out_fmt);
 
if (in_planar == out_planar) {
ac->func_type = CONV_FUNC_TYPE_FLAT;
ac->planes = in_planar ? ac->channels : 1;
} else if (in_planar)
ac->func_type = CONV_FUNC_TYPE_INTERLEAVE;
else
ac->func_type = CONV_FUNC_TYPE_DEINTERLEAVE;
 
set_generic_function(ac);
 
if (ARCH_ARM)
ff_audio_convert_init_arm(ac);
if (ARCH_X86)
ff_audio_convert_init_x86(ac);
 
return ac;
}
 
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
{
int use_generic = 1;
int len = in->nb_samples;
int p;
 
if (ac->dc) {
/* dithered conversion */
av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (dithered)\n",
len, av_get_sample_fmt_name(ac->in_fmt),
av_get_sample_fmt_name(ac->out_fmt));
 
return ff_convert_dither(ac->dc, out, in);
}
 
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (ac->has_optimized_func) {
int ptr_align = FFMIN(in->ptr_align, out->ptr_align);
int samples_align = FFMIN(in->samples_align, out->samples_align);
int aligned_len = FFALIGN(len, ac->samples_align);
if (!(ptr_align % ac->ptr_align) && samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (%s)\n", len,
av_get_sample_fmt_name(ac->in_fmt),
av_get_sample_fmt_name(ac->out_fmt),
use_generic ? ac->func_descr_generic : ac->func_descr);
 
if (ac->apply_map) {
ChannelMapInfo *map = &ac->avr->ch_map_info;
 
if (!av_sample_fmt_is_planar(ac->out_fmt)) {
av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n");
return AVERROR(EINVAL);
}
 
if (map->do_remap) {
if (av_sample_fmt_is_planar(ac->in_fmt)) {
conv_func_flat *convert = use_generic ? ac->conv_flat_generic :
ac->conv_flat;
 
for (p = 0; p < ac->planes; p++)
if (map->channel_map[p] >= 0)
convert(out->data[p], in->data[map->channel_map[p]], len);
} else {
uint8_t *data[AVRESAMPLE_MAX_CHANNELS];
conv_func_deinterleave *convert = use_generic ?
ac->conv_deinterleave_generic :
ac->conv_deinterleave;
 
for (p = 0; p < ac->channels; p++)
data[map->input_map[p]] = out->data[p];
 
convert(data, in->data[0], len, ac->channels);
}
}
if (map->do_copy || map->do_zero) {
for (p = 0; p < ac->planes; p++) {
if (map->channel_copy[p])
memcpy(out->data[p], out->data[map->channel_copy[p]],
len * out->stride);
else if (map->channel_zero[p])
av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt);
}
}
} else {
switch (ac->func_type) {
case CONV_FUNC_TYPE_FLAT: {
if (!in->is_planar)
len *= in->channels;
if (use_generic) {
for (p = 0; p < ac->planes; p++)
ac->conv_flat_generic(out->data[p], in->data[p], len);
} else {
for (p = 0; p < ac->planes; p++)
ac->conv_flat(out->data[p], in->data[p], len);
}
break;
}
case CONV_FUNC_TYPE_INTERLEAVE:
if (use_generic)
ac->conv_interleave_generic(out->data[0], in->data, len,
ac->channels);
else
ac->conv_interleave(out->data[0], in->data, len, ac->channels);
break;
case CONV_FUNC_TYPE_DEINTERLEAVE:
if (use_generic)
ac->conv_deinterleave_generic(out->data, in->data[0], len,
ac->channels);
else
ac->conv_deinterleave(out->data, in->data[0], len,
ac->channels);
break;
}
}
 
out->nb_samples = in->nb_samples;
return 0;
}
/contrib/sdk/sources/ffmpeg/libavresample/audio_convert.h
0,0 → 1,102
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
*/
 
#ifndef AVRESAMPLE_AUDIO_CONVERT_H
#define AVRESAMPLE_AUDIO_CONVERT_H
 
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
 
/**
* Set conversion function if the parameters match.
*
* This compares the parameters of the conversion function to the parameters
* in the AudioConvert context. If the parameters do not match, no changes are
* made to the active functions. If the parameters do match and the alignment
* is not constrained, the function is set as the generic conversion function.
* If the parameters match and the alignment is constrained, the function is
* set as the optimized conversion function.
*
* @param ac AudioConvert context
* @param out_fmt output sample format
* @param in_fmt input sample format
* @param channels number of channels, or 0 for any number of channels
* @param ptr_align buffer pointer alignment, in bytes
* @param samples_align buffer size alignment, in samples
* @param descr function type description (e.g. "C" or "SSE")
* @param conv conversion function pointer
*/
void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt, int channels,
int ptr_align, int samples_align,
const char *descr, void *conv);
 
/**
* Allocate and initialize AudioConvert context for sample format conversion.
*
* @param avr AVAudioResampleContext
* @param out_fmt output sample format
* @param in_fmt input sample format
* @param channels number of channels
* @param sample_rate sample rate (used for dithering)
* @param apply_map apply channel map during conversion
* @return newly-allocated AudioConvert context
*/
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int sample_rate,
int apply_map);
 
/**
* Free AudioConvert.
*
* The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
*
* @param ac AudioConvert struct
*/
void ff_audio_convert_free(AudioConvert **ac);
 
/**
* Convert audio data from one sample format to another.
*
* For each call, the alignment of the input and output AudioData buffers are
* examined to determine whether to use the generic or optimized conversion
* function (when available).
*
* The number of samples to convert is determined by in->nb_samples. The output
* buffer must be large enough to handle this many samples. out->nb_samples is
* set by this function before a successful return.
*
* @param ac AudioConvert context
* @param out output audio data
* @param in input audio data
* @return 0 on success, negative AVERROR code on failure
*/
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
 
/* arch-specific initialization functions */
 
void ff_audio_convert_init_arm(AudioConvert *ac);
void ff_audio_convert_init_x86(AudioConvert *ac);
 
#endif /* AVRESAMPLE_AUDIO_CONVERT_H */
/contrib/sdk/sources/ffmpeg/libavresample/audio_data.c
0,0 → 1,372
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 <stdint.h>
#include <string.h>
 
#include "libavutil/mem.h"
#include "audio_data.h"
 
static const AVClass audio_data_class = {
.class_name = "AudioData",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
};
 
/*
* Calculate alignment for data pointers.
*/
static void calc_ptr_alignment(AudioData *a)
{
int p;
int min_align = 128;
 
for (p = 0; p < a->planes; p++) {
int cur_align = 128;
while ((intptr_t)a->data[p] % cur_align)
cur_align >>= 1;
if (cur_align < min_align)
min_align = cur_align;
}
a->ptr_align = min_align;
}
 
int ff_audio_data_set_channels(AudioData *a, int channels)
{
if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS ||
channels > a->allocated_channels)
return AVERROR(EINVAL);
 
a->channels = channels;
a->planes = a->is_planar ? channels : 1;
 
calc_ptr_alignment(a);
 
return 0;
}
 
int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
int nb_samples, enum AVSampleFormat sample_fmt,
int read_only, const char *name)
{
int p;
 
memset(a, 0, sizeof(*a));
a->class = &audio_data_class;
 
if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels);
return AVERROR(EINVAL);
}
 
a->sample_size = av_get_bytes_per_sample(sample_fmt);
if (!a->sample_size) {
av_log(a, AV_LOG_ERROR, "invalid sample format\n");
return AVERROR(EINVAL);
}
a->is_planar = av_sample_fmt_is_planar(sample_fmt);
a->planes = a->is_planar ? channels : 1;
a->stride = a->sample_size * (a->is_planar ? 1 : channels);
 
for (p = 0; p < (a->is_planar ? channels : 1); p++) {
if (!src[p]) {
av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p);
return AVERROR(EINVAL);
}
a->data[p] = src[p];
}
a->allocated_samples = nb_samples * !read_only;
a->nb_samples = nb_samples;
a->sample_fmt = sample_fmt;
a->channels = channels;
a->allocated_channels = channels;
a->read_only = read_only;
a->allow_realloc = 0;
a->name = name ? name : "{no name}";
 
calc_ptr_alignment(a);
a->samples_align = plane_size / a->stride;
 
return 0;
}
 
AudioData *ff_audio_data_alloc(int channels, int nb_samples,
enum AVSampleFormat sample_fmt, const char *name)
{
AudioData *a;
int ret;
 
if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS)
return NULL;
 
a = av_mallocz(sizeof(*a));
if (!a)
return NULL;
 
a->sample_size = av_get_bytes_per_sample(sample_fmt);
if (!a->sample_size) {
av_free(a);
return NULL;
}
a->is_planar = av_sample_fmt_is_planar(sample_fmt);
a->planes = a->is_planar ? channels : 1;
a->stride = a->sample_size * (a->is_planar ? 1 : channels);
 
a->class = &audio_data_class;
a->sample_fmt = sample_fmt;
a->channels = channels;
a->allocated_channels = channels;
a->read_only = 0;
a->allow_realloc = 1;
a->name = name ? name : "{no name}";
 
if (nb_samples > 0) {
ret = ff_audio_data_realloc(a, nb_samples);
if (ret < 0) {
av_free(a);
return NULL;
}
return a;
} else {
calc_ptr_alignment(a);
return a;
}
}
 
int ff_audio_data_realloc(AudioData *a, int nb_samples)
{
int ret, new_buf_size, plane_size, p;
 
/* check if buffer is already large enough */
if (a->allocated_samples >= nb_samples)
return 0;
 
/* validate that the output is not read-only and realloc is allowed */
if (a->read_only || !a->allow_realloc)
return AVERROR(EINVAL);
 
new_buf_size = av_samples_get_buffer_size(&plane_size,
a->allocated_channels, nb_samples,
a->sample_fmt, 0);
if (new_buf_size < 0)
return new_buf_size;
 
/* if there is already data in the buffer and the sample format is planar,
allocate a new buffer and copy the data, otherwise just realloc the
internal buffer and set new data pointers */
if (a->nb_samples > 0 && a->is_planar) {
uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL };
 
ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels,
nb_samples, a->sample_fmt, 0);
if (ret < 0)
return ret;
 
for (p = 0; p < a->planes; p++)
memcpy(new_data[p], a->data[p], a->nb_samples * a->stride);
 
av_freep(&a->buffer);
memcpy(a->data, new_data, sizeof(new_data));
a->buffer = a->data[0];
} else {
av_freep(&a->buffer);
a->buffer = av_malloc(new_buf_size);
if (!a->buffer)
return AVERROR(ENOMEM);
ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer,
a->allocated_channels, nb_samples,
a->sample_fmt, 0);
if (ret < 0)
return ret;
}
a->buffer_size = new_buf_size;
a->allocated_samples = nb_samples;
 
calc_ptr_alignment(a);
a->samples_align = plane_size / a->stride;
 
return 0;
}
 
void ff_audio_data_free(AudioData **a)
{
if (!*a)
return;
av_free((*a)->buffer);
av_freep(a);
}
 
int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
{
int ret, p;
 
/* validate input/output compatibility */
if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
return AVERROR(EINVAL);
 
if (map && !src->is_planar) {
av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n");
return AVERROR(EINVAL);
}
 
/* if the input is empty, just empty the output */
if (!src->nb_samples) {
dst->nb_samples = 0;
return 0;
}
 
/* reallocate output if necessary */
ret = ff_audio_data_realloc(dst, src->nb_samples);
if (ret < 0)
return ret;
 
/* copy data */
if (map) {
if (map->do_remap) {
for (p = 0; p < src->planes; p++) {
if (map->channel_map[p] >= 0)
memcpy(dst->data[p], src->data[map->channel_map[p]],
src->nb_samples * src->stride);
}
}
if (map->do_copy || map->do_zero) {
for (p = 0; p < src->planes; p++) {
if (map->channel_copy[p])
memcpy(dst->data[p], dst->data[map->channel_copy[p]],
src->nb_samples * src->stride);
else if (map->channel_zero[p])
av_samples_set_silence(&dst->data[p], 0, src->nb_samples,
1, dst->sample_fmt);
}
}
} else {
for (p = 0; p < src->planes; p++)
memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
}
 
dst->nb_samples = src->nb_samples;
 
return 0;
}
 
int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
int src_offset, int nb_samples)
{
int ret, p, dst_offset2, dst_move_size;
 
/* validate input/output compatibility */
if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) {
av_log(src, AV_LOG_ERROR, "sample format mismatch\n");
return AVERROR(EINVAL);
}
 
/* validate offsets are within the buffer bounds */
if (dst_offset < 0 || dst_offset > dst->nb_samples ||
src_offset < 0 || src_offset > src->nb_samples) {
av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n",
src_offset, dst_offset);
return AVERROR(EINVAL);
}
 
/* check offsets and sizes to see if we can just do nothing and return */
if (nb_samples > src->nb_samples - src_offset)
nb_samples = src->nb_samples - src_offset;
if (nb_samples <= 0)
return 0;
 
/* validate that the output is not read-only */
if (dst->read_only) {
av_log(dst, AV_LOG_ERROR, "dst is read-only\n");
return AVERROR(EINVAL);
}
 
/* reallocate output if necessary */
ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples);
if (ret < 0) {
av_log(dst, AV_LOG_ERROR, "error reallocating dst\n");
return ret;
}
 
dst_offset2 = dst_offset + nb_samples;
dst_move_size = dst->nb_samples - dst_offset;
 
for (p = 0; p < src->planes; p++) {
if (dst_move_size > 0) {
memmove(dst->data[p] + dst_offset2 * dst->stride,
dst->data[p] + dst_offset * dst->stride,
dst_move_size * dst->stride);
}
memcpy(dst->data[p] + dst_offset * dst->stride,
src->data[p] + src_offset * src->stride,
nb_samples * src->stride);
}
dst->nb_samples += nb_samples;
 
return 0;
}
 
void ff_audio_data_drain(AudioData *a, int nb_samples)
{
if (a->nb_samples <= nb_samples) {
/* drain the whole buffer */
a->nb_samples = 0;
} else {
int p;
int move_offset = a->stride * nb_samples;
int move_size = a->stride * (a->nb_samples - nb_samples);
 
for (p = 0; p < a->planes; p++)
memmove(a->data[p], a->data[p] + move_offset, move_size);
 
a->nb_samples -= nb_samples;
}
}
 
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
int nb_samples)
{
uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS];
int offset_size, p;
 
if (offset >= a->nb_samples)
return 0;
offset_size = offset * a->stride;
for (p = 0; p < a->planes; p++)
offset_data[p] = a->data[p] + offset_size;
 
return av_audio_fifo_write(af, (void **)offset_data, nb_samples);
}
 
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
{
int ret;
 
if (a->read_only)
return AVERROR(EINVAL);
 
ret = ff_audio_data_realloc(a, nb_samples);
if (ret < 0)
return ret;
 
ret = av_audio_fifo_read(af, (void **)a->data, nb_samples);
if (ret >= 0)
a->nb_samples = ret;
return ret;
}
/contrib/sdk/sources/ffmpeg/libavresample/audio_data.h
0,0 → 1,175
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
*/
 
#ifndef AVRESAMPLE_AUDIO_DATA_H
#define AVRESAMPLE_AUDIO_DATA_H
 
#include <stdint.h>
 
#include "libavutil/audio_fifo.h"
#include "libavutil/log.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
 
/**
* Audio buffer used for intermediate storage between conversion phases.
*/
struct AudioData {
const AVClass *class; /**< AVClass for logging */
uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
uint8_t *buffer; /**< data buffer */
unsigned int buffer_size; /**< allocated buffer size */
int allocated_samples; /**< number of samples the buffer can hold */
int nb_samples; /**< current number of samples */
enum AVSampleFormat sample_fmt; /**< sample format */
int channels; /**< channel count */
int allocated_channels; /**< allocated channel count */
int is_planar; /**< sample format is planar */
int planes; /**< number of data planes */
int sample_size; /**< bytes per sample */
int stride; /**< sample byte offset within a plane */
int read_only; /**< data is read-only */
int allow_realloc; /**< realloc is allowed */
int ptr_align; /**< minimum data pointer alignment */
int samples_align; /**< allocated samples alignment */
const char *name; /**< name for debug logging */
};
 
int ff_audio_data_set_channels(AudioData *a, int channels);
 
/**
* Initialize AudioData using a given source.
*
* This does not allocate an internal buffer. It only sets the data pointers
* and audio parameters.
*
* @param a AudioData struct
* @param src source data pointers
* @param plane_size plane size, in bytes.
* This can be 0 if unknown, but that will lead to
* optimized functions not being used in many cases,
* which could slow down some conversions.
* @param channels channel count
* @param nb_samples number of samples in the source data
* @param sample_fmt sample format
* @param read_only indicates if buffer is read only or read/write
* @param name name for debug logging (can be NULL)
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
int nb_samples, enum AVSampleFormat sample_fmt,
int read_only, const char *name);
 
/**
* Allocate AudioData.
*
* This allocates an internal buffer and sets audio parameters.
*
* @param channels channel count
* @param nb_samples number of samples to allocate space for
* @param sample_fmt sample format
* @param name name for debug logging (can be NULL)
* @return newly allocated AudioData struct, or NULL on error
*/
AudioData *ff_audio_data_alloc(int channels, int nb_samples,
enum AVSampleFormat sample_fmt,
const char *name);
 
/**
* Reallocate AudioData.
*
* The AudioData must have been previously allocated with ff_audio_data_alloc().
*
* @param a AudioData struct
* @param nb_samples number of samples to allocate space for
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_realloc(AudioData *a, int nb_samples);
 
/**
* Free AudioData.
*
* The AudioData must have been previously allocated with ff_audio_data_alloc().
*
* @param a AudioData struct
*/
void ff_audio_data_free(AudioData **a);
 
/**
* Copy data from one AudioData to another.
*
* @param out output AudioData
* @param in input AudioData
* @param map channel map, NULL if not remapping
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map);
 
/**
* Append data from one AudioData to the end of another.
*
* @param dst destination AudioData
* @param dst_offset offset, in samples, to start writing, relative to the
* start of dst
* @param src source AudioData
* @param src_offset offset, in samples, to start copying, relative to the
* start of the src
* @param nb_samples number of samples to copy
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
int src_offset, int nb_samples);
 
/**
* Drain samples from the start of the AudioData.
*
* Remaining samples are shifted to the start of the AudioData.
*
* @param a AudioData struct
* @param nb_samples number of samples to drain
*/
void ff_audio_data_drain(AudioData *a, int nb_samples);
 
/**
* Add samples in AudioData to an AVAudioFifo.
*
* @param af Audio FIFO Buffer
* @param a AudioData struct
* @param offset number of samples to skip from the start of the data
* @param nb_samples number of samples to add to the FIFO
* @return number of samples actually added to the FIFO, or
* negative AVERROR code on error
*/
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
int nb_samples);
 
/**
* Read samples from an AVAudioFifo to AudioData.
*
* @param af Audio FIFO Buffer
* @param a AudioData struct
* @param nb_samples number of samples to read from the FIFO
* @return number of samples actually read from the FIFO, or
* negative AVERROR code on error
*/
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
 
#endif /* AVRESAMPLE_AUDIO_DATA_H */
/contrib/sdk/sources/ffmpeg/libavresample/audio_mix.c
0,0 → 1,739
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 <stdint.h>
 
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_mix.h"
 
static const char *coeff_type_names[] = { "q8", "q15", "flt" };
 
struct AudioMix {
AVAudioResampleContext *avr;
enum AVSampleFormat fmt;
enum AVMixCoeffType coeff_type;
uint64_t in_layout;
uint64_t out_layout;
int in_channels;
int out_channels;
 
int ptr_align;
int samples_align;
int has_optimized_func;
const char *func_descr;
const char *func_descr_generic;
mix_func *mix;
mix_func *mix_generic;
 
int in_matrix_channels;
int out_matrix_channels;
int output_zero[AVRESAMPLE_MAX_CHANNELS];
int input_skip[AVRESAMPLE_MAX_CHANNELS];
int output_skip[AVRESAMPLE_MAX_CHANNELS];
int16_t *matrix_q8[AVRESAMPLE_MAX_CHANNELS];
int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
float *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
void **matrix;
};
 
void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
enum AVMixCoeffType coeff_type, int in_channels,
int out_channels, int ptr_align, int samples_align,
const char *descr, void *mix_func)
{
if (fmt == am->fmt && coeff_type == am->coeff_type &&
( in_channels == am->in_matrix_channels || in_channels == 0) &&
(out_channels == am->out_matrix_channels || out_channels == 0)) {
char chan_str[16];
am->mix = mix_func;
am->func_descr = descr;
am->ptr_align = ptr_align;
am->samples_align = samples_align;
if (ptr_align == 1 && samples_align == 1) {
am->mix_generic = mix_func;
am->func_descr_generic = descr;
} else {
am->has_optimized_func = 1;
}
if (in_channels) {
if (out_channels)
snprintf(chan_str, sizeof(chan_str), "[%d to %d] ",
in_channels, out_channels);
else
snprintf(chan_str, sizeof(chan_str), "[%d to any] ",
in_channels);
} else if (out_channels) {
snprintf(chan_str, sizeof(chan_str), "[any to %d] ",
out_channels);
} else {
snprintf(chan_str, sizeof(chan_str), "[any to any] ");
}
av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] "
"[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt),
coeff_type_names[coeff_type], chan_str, descr);
}
}
 
#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c
 
#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \
static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
int len, int out_ch, int in_ch) \
{ \
int i, in, out; \
stype temp[AVRESAMPLE_MAX_CHANNELS]; \
for (i = 0; i < len; i++) { \
for (out = 0; out < out_ch; out++) { \
sumtype sum = 0; \
for (in = 0; in < in_ch; in++) \
sum += samples[in][i] * matrix[out][in]; \
temp[out] = expr; \
} \
for (out = 0; out < out_ch; out++) \
samples[out][i] = temp[out]; \
} \
}
 
MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum)
MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum)))
MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15))
MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8))
 
/* TODO: templatize the channel-specific C functions */
 
static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float *src0 = samples[0];
float *src1 = samples[1];
float *dst = src0;
float m0 = matrix[0][0];
float m1 = matrix[0][1];
 
while (len > 4) {
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
len -= 4;
}
while (len > 0) {
*dst++ = *src0++ * m0 + *src1++ * m1;
len--;
}
}
 
static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len,
int out_ch, int in_ch)
{
int16_t *src0 = samples[0];
int16_t *src1 = samples[1];
int16_t *dst = src0;
float m0 = matrix[0][0];
float m1 = matrix[0][1];
 
while (len > 4) {
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
len -= 4;
}
while (len > 0) {
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
len--;
}
}
 
static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len,
int out_ch, int in_ch)
{
int16_t *src0 = samples[0];
int16_t *src1 = samples[1];
int16_t *dst = src0;
int16_t m0 = matrix[0][0];
int16_t m1 = matrix[0][1];
 
while (len > 4) {
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
len -= 4;
}
while (len > 0) {
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
len--;
}
}
 
static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v;
float *dst0 = samples[0];
float *dst1 = samples[1];
float *src = dst0;
float m0 = matrix[0][0];
float m1 = matrix[1][0];
 
while (len > 4) {
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
len -= 4;
}
while (len > 0) {
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
len--;
}
}
 
static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v0, v1;
float *src0 = samples[0];
float *src1 = samples[1];
float *src2 = samples[2];
float *src3 = samples[3];
float *src4 = samples[4];
float *src5 = samples[5];
float *dst0 = src0;
float *dst1 = src1;
float *m0 = matrix[0];
float *m1 = matrix[1];
 
while (len > 0) {
v0 = *src0++;
v1 = *src1++;
*dst0++ = v0 * m0[0] +
v1 * m0[1] +
*src2 * m0[2] +
*src3 * m0[3] +
*src4 * m0[4] +
*src5 * m0[5];
*dst1++ = v0 * m1[0] +
v1 * m1[1] +
*src2++ * m1[2] +
*src3++ * m1[3] +
*src4++ * m1[4] +
*src5++ * m1[5];
len--;
}
}
 
static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v0, v1;
float *dst0 = samples[0];
float *dst1 = samples[1];
float *dst2 = samples[2];
float *dst3 = samples[3];
float *dst4 = samples[4];
float *dst5 = samples[5];
float *src0 = dst0;
float *src1 = dst1;
 
while (len > 0) {
v0 = *src0++;
v1 = *src1++;
*dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1];
*dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1];
*dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1];
*dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1];
*dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1];
*dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1];
len--;
}
}
 
static av_cold int mix_function_init(AudioMix *am)
{
am->func_descr = am->func_descr_generic = "n/a";
am->mix = am->mix_generic = NULL;
 
/* no need to set a mix function when we're skipping mixing */
if (!am->in_matrix_channels || !am->out_matrix_channels)
return 0;
 
/* any-to-any C versions */
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT));
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT));
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15));
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q8));
 
/* channel-specific C versions */
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c);
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c);
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c);
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c);
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c);
 
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c);
 
if (ARCH_X86)
ff_audio_mix_init_x86(am);
 
if (!am->mix) {
av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] "
"[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt),
coeff_type_names[am->coeff_type], am->in_channels,
am->out_channels);
return AVERROR_PATCHWELCOME;
}
return 0;
}
 
AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr)
{
AudioMix *am;
int ret;
 
am = av_mallocz(sizeof(*am));
if (!am)
return NULL;
am->avr = avr;
 
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"mixing: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
goto error;
}
 
am->fmt = avr->internal_sample_fmt;
am->coeff_type = avr->mix_coeff_type;
am->in_layout = avr->in_channel_layout;
am->out_layout = avr->out_channel_layout;
am->in_channels = avr->in_channels;
am->out_channels = avr->out_channels;
 
/* build matrix if the user did not already set one */
if (avr->mix_matrix) {
ret = ff_audio_mix_set_matrix(am, avr->mix_matrix, avr->in_channels);
if (ret < 0)
goto error;
av_freep(&avr->mix_matrix);
} else {
double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels *
sizeof(*matrix_dbl));
if (!matrix_dbl)
goto error;
 
ret = avresample_build_matrix(avr->in_channel_layout,
avr->out_channel_layout,
avr->center_mix_level,
avr->surround_mix_level,
avr->lfe_mix_level,
avr->normalize_mix_level,
matrix_dbl,
avr->in_channels,
avr->matrix_encoding);
if (ret < 0) {
av_free(matrix_dbl);
goto error;
}
 
ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
if (ret < 0) {
av_log(avr, AV_LOG_ERROR, "error setting mix matrix\n");
av_free(matrix_dbl);
goto error;
}
 
av_free(matrix_dbl);
}
 
return am;
 
error:
av_free(am);
return NULL;
}
 
void ff_audio_mix_free(AudioMix **am_p)
{
AudioMix *am;
 
if (!*am_p)
return;
am = *am_p;
 
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
memset(am->matrix_q8, 0, sizeof(am->matrix_q8 ));
memset(am->matrix_q15, 0, sizeof(am->matrix_q15));
memset(am->matrix_flt, 0, sizeof(am->matrix_flt));
 
av_freep(am_p);
}
 
int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
int i, j;
 
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (am->has_optimized_func) {
int aligned_len = FFALIGN(len, am->samples_align);
if (!(src->ptr_align % am->ptr_align) &&
src->samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_dlog(am->avr, "audio_mix: %d samples - %d to %d channels (%s)\n",
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
 
if (am->in_matrix_channels && am->out_matrix_channels) {
uint8_t **data;
uint8_t *data0[AVRESAMPLE_MAX_CHANNELS];
 
if (am->out_matrix_channels < am->out_channels ||
am->in_matrix_channels < am->in_channels) {
for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) {
if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i])
continue;
data0[j++] = src->data[i];
}
data = data0;
} else {
data = src->data;
}
 
if (use_generic)
am->mix_generic(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
else
am->mix(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
}
 
if (am->out_matrix_channels < am->out_channels) {
for (i = 0; i < am->out_channels; i++)
if (am->output_zero[i])
av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt);
}
 
ff_audio_data_set_channels(src, am->out_channels);
 
return 0;
}
 
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
{
int i, o, i0, o0;
 
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n");
return AVERROR(EINVAL);
}
 
#define GET_MATRIX_CONVERT(suffix, scale) \
if (!am->matrix_ ## suffix[0]) { \
av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \
return AVERROR(EINVAL); \
} \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
if (am->input_skip[i] || am->output_zero[o]) \
matrix[o * stride + i] = 0.0; \
else \
matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \
(scale); \
if (!am->input_skip[i]) \
i0++; \
} \
if (!am->output_zero[o]) \
o0++; \
}
 
switch (am->coeff_type) {
case AV_MIX_COEFF_TYPE_Q8:
GET_MATRIX_CONVERT(q8, 1.0 / 256.0);
break;
case AV_MIX_COEFF_TYPE_Q15:
GET_MATRIX_CONVERT(q15, 1.0 / 32768.0);
break;
case AV_MIX_COEFF_TYPE_FLT:
GET_MATRIX_CONVERT(flt, 1.0);
break;
default:
av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
 
return 0;
}
 
static void reduce_matrix(AudioMix *am, const double *matrix, int stride)
{
int i, o;
 
memset(am->output_zero, 0, sizeof(am->output_zero));
memset(am->input_skip, 0, sizeof(am->input_skip));
memset(am->output_skip, 0, sizeof(am->output_skip));
 
/* exclude output channels if they can be zeroed instead of mixed */
for (o = 0; o < am->out_channels; o++) {
int zero = 1;
 
/* check if the output is always silent */
for (i = 0; i < am->in_channels; i++) {
if (matrix[o * stride + i] != 0.0) {
zero = 0;
break;
}
}
/* check if the corresponding input channel makes a contribution to
any output channel */
if (o < am->in_channels) {
for (i = 0; i < am->out_channels; i++) {
if (matrix[i * stride + o] != 0.0) {
zero = 0;
break;
}
}
}
if (zero) {
am->output_zero[o] = 1;
am->out_matrix_channels--;
}
}
if (am->out_matrix_channels == 0) {
am->in_matrix_channels = 0;
return;
}
 
/* skip input channels that contribute fully only to the corresponding
output channel */
for (i = 0; i < FFMIN(am->in_channels, am->out_channels); i++) {
int skip = 1;
 
for (o = 0; o < am->out_channels; o++) {
int i0;
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
skip = 0;
break;
}
/* if the input contributes fully to the output, also check that no
other inputs contribute to this output */
if (o == i) {
for (i0 = 0; i0 < am->in_channels; i0++) {
if (i0 != i && matrix[o * stride + i0] != 0.0) {
skip = 0;
break;
}
}
}
}
if (skip) {
am->input_skip[i] = 1;
am->in_matrix_channels--;
}
}
/* skip input channels that do not contribute to any output channel */
for (; i < am->in_channels; i++) {
int contrib = 0;
 
for (o = 0; o < am->out_channels; o++) {
if (matrix[o * stride + i] != 0.0) {
contrib = 1;
break;
}
}
if (!contrib) {
am->input_skip[i] = 1;
am->in_matrix_channels--;
}
}
if (am->in_matrix_channels == 0) {
am->out_matrix_channels = 0;
return;
}
 
/* skip output channels that only get full contribution from the
corresponding input channel */
for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) {
int skip = 1;
int o0;
 
for (i = 0; i < am->in_channels; i++) {
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
skip = 0;
break;
}
}
/* check if the corresponding input channel makes a contribution to
any other output channel */
i = o;
for (o0 = 0; o0 < am->out_channels; o0++) {
if (o0 != i && matrix[o0 * stride + i] != 0.0) {
skip = 0;
break;
}
}
if (skip) {
am->output_skip[o] = 1;
am->out_matrix_channels--;
}
}
if (am->out_matrix_channels == 0) {
am->in_matrix_channels = 0;
return;
}
}
 
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
{
int i, o, i0, o0, ret;
char in_layout_name[128];
char out_layout_name[128];
 
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n");
return AVERROR(EINVAL);
}
 
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
 
am->in_matrix_channels = am->in_channels;
am->out_matrix_channels = am->out_channels;
 
reduce_matrix(am, matrix, stride);
 
#define CONVERT_MATRIX(type, expr) \
am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \
am->in_matrix_channels * \
sizeof(*am->matrix_## type[0])); \
if (!am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
if (am->output_zero[o] || am->output_skip[o]) \
continue; \
if (o0 > 0) \
am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \
am->in_matrix_channels; \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
double v; \
if (am->input_skip[i]) \
continue; \
v = matrix[o * stride + i]; \
am->matrix_## type[o0][i0] = expr; \
i0++; \
} \
o0++; \
} \
am->matrix = (void **)am->matrix_## type;
 
if (am->in_matrix_channels && am->out_matrix_channels) {
switch (am->coeff_type) {
case AV_MIX_COEFF_TYPE_Q8:
CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v)))
break;
case AV_MIX_COEFF_TYPE_Q15:
CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v)))
break;
case AV_MIX_COEFF_TYPE_FLT:
CONVERT_MATRIX(flt, v)
break;
default:
av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
}
 
ret = mix_function_init(am);
if (ret < 0)
return ret;
 
av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name),
am->in_channels, am->in_layout);
av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name),
am->out_channels, am->out_layout);
av_log(am->avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n",
in_layout_name, out_layout_name);
av_log(am->avr, AV_LOG_DEBUG, "matrix size: %d x %d\n",
am->in_matrix_channels, am->out_matrix_channels);
for (o = 0; o < am->out_channels; o++) {
for (i = 0; i < am->in_channels; i++) {
if (am->output_zero[o])
av_log(am->avr, AV_LOG_DEBUG, " (ZERO)");
else if (am->input_skip[i] || am->output_skip[o])
av_log(am->avr, AV_LOG_DEBUG, " (SKIP)");
else
av_log(am->avr, AV_LOG_DEBUG, " %0.3f ",
matrix[o * am->in_channels + i]);
}
av_log(am->avr, AV_LOG_DEBUG, "\n");
}
 
return 0;
}
/contrib/sdk/sources/ffmpeg/libavresample/audio_mix.h
0,0 → 1,94
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
*/
 
#ifndef AVRESAMPLE_AUDIO_MIX_H
#define AVRESAMPLE_AUDIO_MIX_H
 
#include <stdint.h>
 
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
 
typedef void (mix_func)(uint8_t **src, void **matrix, int len, int out_ch,
int in_ch);
 
/**
* Set mixing function if the parameters match.
*
* This compares the parameters of the mixing function to the parameters in the
* AudioMix context. If the parameters do not match, no changes are made to the
* active functions. If the parameters do match and the alignment is not
* constrained, the function is set as the generic mixing function. If the
* parameters match and the alignment is constrained, the function is set as
* the optimized mixing function.
*
* @param am AudioMix context
* @param fmt input/output sample format
* @param coeff_type mixing coefficient type
* @param in_channels number of input channels, or 0 for any number of channels
* @param out_channels number of output channels, or 0 for any number of channels
* @param ptr_align buffer pointer alignment, in bytes
* @param samples_align buffer size alignment, in samples
* @param descr function type description (e.g. "C" or "SSE")
* @param mix_func mixing function pointer
*/
void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
enum AVMixCoeffType coeff_type, int in_channels,
int out_channels, int ptr_align, int samples_align,
const char *descr, void *mix_func);
 
/**
* Allocate and initialize an AudioMix context.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* AudioMix context.
*
* @param avr AVAudioResampleContext
* @return newly-allocated AudioMix context.
*/
AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr);
 
/**
* Free an AudioMix context.
*/
void ff_audio_mix_free(AudioMix **am);
 
/**
* Apply channel mixing to audio data using the current mixing matrix.
*/
int ff_audio_mix(AudioMix *am, AudioData *src);
 
/**
* Get the current mixing matrix.
*/
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride);
 
/**
* Set the current mixing matrix.
*/
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride);
 
/* arch-specific initialization functions */
 
void ff_audio_mix_init_x86(AudioMix *am);
 
#endif /* AVRESAMPLE_AUDIO_MIX_H */
/contrib/sdk/sources/ffmpeg/libavresample/audio_mix_matrix.c
0,0 → 1,289
/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 <stdint.h>
 
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_mix.h"
 
/* channel positions */
#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 STEREO_LEFT 29
#define STEREO_RIGHT 30
#define WIDE_LEFT 31
#define WIDE_RIGHT 32
#define SURROUND_DIRECT_LEFT 33
#define SURROUND_DIRECT_RIGHT 34
#define LOW_FREQUENCY_2 35
 
#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
 
static av_always_inline int even(uint64_t layout)
{
return (!layout || (layout & (layout - 1)));
}
 
static int sane_layout(uint64_t layout)
{
/* check that there is at least 1 front speaker */
if (!(layout & AV_CH_LAYOUT_SURROUND))
return 0;
 
/* check for left/right symmetry */
if (!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT)) ||
!even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT)) ||
!even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT)) ||
!even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER)) ||
!even(layout & (AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT)) ||
!even(layout & (AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT)) ||
!even(layout & (AV_CH_STEREO_LEFT | AV_CH_STEREO_RIGHT)) ||
!even(layout & (AV_CH_WIDE_LEFT | AV_CH_WIDE_RIGHT)) ||
!even(layout & (AV_CH_SURROUND_DIRECT_LEFT | AV_CH_SURROUND_DIRECT_RIGHT)))
return 0;
 
return 1;
}
 
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize,
double *matrix_out, int stride,
enum AVMatrixEncoding matrix_encoding)
{
int i, j, out_i, out_j;
double matrix[64][64] = {{0}};
int64_t unaccounted;
double maxcoef = 0;
int in_channels, out_channels;
 
if ((out_layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == AV_CH_LAYOUT_STEREO_DOWNMIX) {
out_layout = AV_CH_LAYOUT_STEREO;
}
 
unaccounted = in_layout & ~out_layout;
 
in_channels = av_get_channel_layout_nb_channels( in_layout);
out_channels = av_get_channel_layout_nb_channels(out_layout);
 
memset(matrix_out, 0, out_channels * stride * sizeof(*matrix_out));
 
/* check if layouts are supported */
if (!in_layout || in_channels > AVRESAMPLE_MAX_CHANNELS)
return AVERROR(EINVAL);
if (!out_layout || out_channels > AVRESAMPLE_MAX_CHANNELS)
return AVERROR(EINVAL);
 
/* check if layouts are unbalanced or abnormal */
if (!sane_layout(in_layout) || !sane_layout(out_layout))
return AVERROR_PATCHWELCOME;
 
/* route matching input/output channels */
for (i = 0; i < 64; i++) {
if (in_layout & out_layout & (1ULL << i))
matrix[i][i] = 1.0;
}
 
/* mix front center to front left/right */
if (unaccounted & AV_CH_FRONT_CENTER) {
if ((out_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO) {
matrix[FRONT_LEFT ][FRONT_CENTER] += M_SQRT1_2;
matrix[FRONT_RIGHT][FRONT_CENTER] += M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix front left/right to center */
if (unaccounted & AV_CH_LAYOUT_STEREO) {
if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][FRONT_LEFT ] += M_SQRT1_2;
matrix[FRONT_CENTER][FRONT_RIGHT] += M_SQRT1_2;
/* mix left/right/center to center */
if (in_layout & AV_CH_FRONT_CENTER)
matrix[FRONT_CENTER][FRONT_CENTER] = center_mix_level * M_SQRT2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix back center to back, side, or front */
if (unaccounted & AV_CH_BACK_CENTER) {
if (out_layout & AV_CH_BACK_LEFT) {
matrix[BACK_LEFT ][BACK_CENTER] += M_SQRT1_2;
matrix[BACK_RIGHT][BACK_CENTER] += M_SQRT1_2;
} else if (out_layout & AV_CH_SIDE_LEFT) {
matrix[SIDE_LEFT ][BACK_CENTER] += M_SQRT1_2;
matrix[SIDE_RIGHT][BACK_CENTER] += M_SQRT1_2;
} else if (out_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_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix back left/right to back center, side, or front */
if (unaccounted & AV_CH_BACK_LEFT) {
if (out_layout & AV_CH_BACK_CENTER) {
matrix[BACK_CENTER][BACK_LEFT ] += M_SQRT1_2;
matrix[BACK_CENTER][BACK_RIGHT] += M_SQRT1_2;
} else if (out_layout & AV_CH_SIDE_LEFT) {
/* if side channels do not exist in the input, just copy back
channels to side channels, otherwise mix back into side */
if (in_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_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_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
return AVERROR_PATCHWELCOME;
}
/* mix side left/right into back or front */
if (unaccounted & AV_CH_SIDE_LEFT) {
if (out_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_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_layout & AV_CH_BACK_CENTER) {
matrix[BACK_CENTER][SIDE_LEFT ] += M_SQRT1_2;
matrix[BACK_CENTER][SIDE_RIGHT] += M_SQRT1_2;
} else if (out_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_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
return AVERROR_PATCHWELCOME;
}
/* mix left-of-center/right-of-center into front left/right or center */
if (unaccounted & AV_CH_FRONT_LEFT_OF_CENTER) {
if (out_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_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
return AVERROR_PATCHWELCOME;
}
/* mix LFE into front left/right or center */
if (unaccounted & AV_CH_LOW_FREQUENCY) {
if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][LOW_FREQUENCY] += lfe_mix_level;
} else if (out_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
return AVERROR_PATCHWELCOME;
}
 
/* transfer internal matrix to output matrix and calculate maximum
per-channel coefficient sum */
for (out_i = i = 0; out_i < out_channels && i < 64; i++) {
double sum = 0;
for (out_j = j = 0; out_j < in_channels && j < 64; j++) {
matrix_out[out_i * stride + out_j] = matrix[i][j];
sum += fabs(matrix[i][j]);
if (in_layout & (1ULL << j))
out_j++;
}
maxcoef = FFMAX(maxcoef, sum);
if (out_layout & (1ULL << i))
out_i++;
}
 
/* normalize */
if (normalize && maxcoef > 1.0) {
for (i = 0; i < out_channels; i++)
for (j = 0; j < in_channels; j++)
matrix_out[i * stride + j] /= maxcoef;
}
 
return 0;
}
/contrib/sdk/sources/ffmpeg/libavresample/avresample-test.c
0,0 → 1,341
/*
* Copyright (c) 2002 Fabrice Bellard
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 <stdint.h>
#include <stdio.h>
 
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/lfg.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
 
static double dbl_rand(AVLFG *lfg)
{
return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
}
 
#define PUT_FUNC(name, fmt, type, expr) \
static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
int channels, int sample, int ch, \
double v_dbl) \
{ \
type v = expr; \
type **out = (type **)data; \
if (av_sample_fmt_is_planar(sample_fmt)) \
out[ch][sample] = v; \
else \
out[0][sample * channels + ch] = v; \
}
 
PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128))
PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15))))
PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl)
PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl)
 
static void put_sample(void **data, enum AVSampleFormat sample_fmt,
int channels, int sample, int ch, double v_dbl)
{
switch (av_get_packed_sample_fmt(sample_fmt)) {
case AV_SAMPLE_FMT_U8:
put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_S16:
put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_S32:
put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_FLT:
put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_DBL:
put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
break;
}
}
 
static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
int channels, int sample_rate, int nb_samples)
{
int i, ch, k;
double v, f, a, ampa;
double tabf1[AVRESAMPLE_MAX_CHANNELS];
double tabf2[AVRESAMPLE_MAX_CHANNELS];
double taba[AVRESAMPLE_MAX_CHANNELS];
 
#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
 
k = 0;
 
/* 1 second of single freq sine at 1000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
a += M_PI * 1000.0 * 2.0 / sample_rate;
}
 
/* 1 second of varying frequency between 100 and 10000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
a += M_PI * f * 2.0 / sample_rate;
}
 
/* 0.5 second of low amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
 
/* 0.5 second of high amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd);
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
 
/* 1 second of unrelated ramps for each channel */
for (ch = 0; ch < channels; ch++) {
taba[ch] = 0;
tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
}
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
v = sin(taba[ch]) * 0.30;
PUT_SAMPLE
f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
taba[ch] += M_PI * f * 2.0 / sample_rate;
}
}
 
/* 2 seconds of 500 Hz with varying volume */
a = 0;
ampa = 0;
for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
double amp = (1.0 + sin(ampa)) * 0.15;
if (ch & 1)
amp = 0.30 - amp;
v = sin(a) * amp;
PUT_SAMPLE
a += M_PI * 500.0 * 2.0 / sample_rate;
ampa += M_PI * 2.0 / sample_rate;
}
}
}
 
/* formats, rates, and layouts are ordered for priority in testing.
e.g. 'avresample-test 4 2 2' will test all input/output combinations of
S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
 
static const enum AVSampleFormat formats[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_DBL,
};
 
static const int rates[] = {
48000,
44100,
16000
};
 
static const uint64_t layouts[] = {
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_5POINT1,
AV_CH_LAYOUT_7POINT1,
};
 
int main(int argc, char **argv)
{
AVAudioResampleContext *s;
AVLFG rnd;
int ret = 0;
uint8_t *in_buf = NULL;
uint8_t *out_buf = NULL;
unsigned int in_buf_size;
unsigned int out_buf_size;
uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
int in_linesize;
int out_linesize;
uint64_t in_ch_layout;
int in_channels;
enum AVSampleFormat in_fmt;
int in_rate;
uint64_t out_ch_layout;
int out_channels;
enum AVSampleFormat out_fmt;
int out_rate;
int num_formats, num_rates, num_layouts;
int i, j, k, l, m, n;
 
num_formats = 2;
num_rates = 2;
num_layouts = 2;
if (argc > 1) {
if (!av_strncasecmp(argv[1], "-h", 3)) {
av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
"[<num sample rates> [<num channel layouts>]]]\n"
"Default is 2 2 2\n");
return 0;
}
num_formats = strtol(argv[1], NULL, 0);
num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
}
if (argc > 2) {
num_rates = strtol(argv[2], NULL, 0);
num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
}
if (argc > 3) {
num_layouts = strtol(argv[3], NULL, 0);
num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
}
 
av_log_set_level(AV_LOG_DEBUG);
 
av_lfg_init(&rnd, 0xC0FFEE);
 
in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
AV_SAMPLE_FMT_DBLP, 0);
out_buf_size = in_buf_size;
 
in_buf = av_malloc(in_buf_size);
if (!in_buf)
goto end;
out_buf = av_malloc(out_buf_size);
if (!out_buf)
goto end;
 
s = avresample_alloc_context();
if (!s) {
av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
ret = 1;
goto end;
}
 
for (i = 0; i < num_formats; i++) {
in_fmt = formats[i];
for (k = 0; k < num_layouts; k++) {
in_ch_layout = layouts[k];
in_channels = av_get_channel_layout_nb_channels(in_ch_layout);
for (m = 0; m < num_rates; m++) {
in_rate = rates[m];
 
ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
in_channels, in_rate * 6,
in_fmt, 0);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
goto end;
}
audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
 
for (j = 0; j < num_formats; j++) {
out_fmt = formats[j];
for (l = 0; l < num_layouts; l++) {
out_ch_layout = layouts[l];
out_channels = av_get_channel_layout_nb_channels(out_ch_layout);
for (n = 0; n < num_rates; n++) {
out_rate = rates[n];
 
av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
in_channels, out_channels, in_rate, out_rate);
 
ret = av_samples_fill_arrays(out_data, &out_linesize,
out_buf, out_channels,
out_rate * 6, out_fmt, 0);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
goto end;
}
 
av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0);
av_opt_set_int(s, "in_sample_fmt", in_fmt, 0);
av_opt_set_int(s, "in_sample_rate", in_rate, 0);
av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(s, "out_sample_fmt", out_fmt, 0);
av_opt_set_int(s, "out_sample_rate", out_rate, 0);
 
av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
 
ret = avresample_open(s);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "Error opening context\n");
goto end;
}
 
ret = avresample_convert(s, out_data, out_linesize, out_rate * 6,
in_data, in_linesize, in_rate * 6);
if (ret < 0) {
char errbuf[256];
av_strerror(ret, errbuf, sizeof(errbuf));
av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
goto end;
}
av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
in_rate * 6, ret);
if (avresample_get_delay(s) > 0)
av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
avresample_get_delay(s));
if (avresample_available(s) > 0)
av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
avresample_available(s));
av_log(NULL, AV_LOG_INFO, "\n");
 
avresample_close(s);
}
}
}
}
}
}
 
ret = 0;
 
end:
av_freep(&in_buf);
av_freep(&out_buf);
avresample_free(&s);
return ret;
}
/contrib/sdk/sources/ffmpeg/libavresample/avresample.h
0,0 → 1,409
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
*/
 
#ifndef AVRESAMPLE_AVRESAMPLE_H
#define AVRESAMPLE_AVRESAMPLE_H
 
/**
* @file
* @ingroup lavr
* external API header
*/
 
/**
* @defgroup lavr Libavresample
* @{
*
* Libavresample (lavr) is a library that handles audio resampling, sample
* format conversion and mixing.
*
* Interaction with lavr is done through AVAudioResampleContext, which is
* allocated with avresample_alloc_context(). It is opaque, so all parameters
* must be set with the @ref avoptions API.
*
* 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):
* @code
* AVAudioResampleContext *avr = avresample_alloc_context();
* av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
* av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
* av_opt_set_int(avr, "in_sample_rate", 48000, 0);
* av_opt_set_int(avr, "out_sample_rate", 44100, 0);
* av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
* av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
* @endcode
*
* Once the context is initialized, it must be opened with avresample_open(). If
* you need to change the conversion parameters, you must close the context with
* avresample_close(), change the parameters as described above, then reopen it
* again.
*
* The conversion itself is done by repeatedly calling avresample_convert().
* Note that the samples may get buffered in two places in lavr. The first one
* is the output FIFO, where the samples end up if the output buffer is not
* large enough. The data stored in there may be retrieved at any time with
* avresample_read(). The second place is the resampling delay buffer,
* applicable only when resampling is done. The samples in it require more input
* before they can be processed. Their current amount is returned by
* avresample_get_delay(). At the end of conversion the resampling buffer can be
* flushed by calling avresample_convert() with NULL input.
*
* 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_linesize, in_samples;
*
* while (get_input(&input, &in_linesize, &in_samples)) {
* uint8_t *output
* int out_linesize;
* int out_samples = avresample_available(avr) +
* av_rescale_rnd(avresample_get_delay(avr) +
* in_samples, 44100, 48000, AV_ROUND_UP);
* av_samples_alloc(&output, &out_linesize, 2, out_samples,
* AV_SAMPLE_FMT_S16, 0);
* out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
* input, in_linesize, in_samples);
* handle_output(output, out_linesize, out_samples);
* av_freep(&output);
* }
* @endcode
*
* When the conversion is finished and the FIFOs are flushed if required, the
* conversion context and everything associated with it must be freed with
* avresample_free().
*/
 
#include "libavutil/avutil.h"
#include "libavutil/channel_layout.h"
#include "libavutil/dict.h"
#include "libavutil/log.h"
 
#include "libavresample/version.h"
 
#define AVRESAMPLE_MAX_CHANNELS 32
 
typedef struct AVAudioResampleContext AVAudioResampleContext;
 
/** Mixing Coefficient Types */
enum AVMixCoeffType {
AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
AV_MIX_COEFF_TYPE_FLT, /** floating-point */
AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
};
 
/** Resampling Filter Types */
enum AVResampleFilterType {
AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
};
 
enum AVResampleDitherMethod {
AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
};
 
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/
unsigned avresample_version(void);
 
/**
* Return the libavresample build-time configuration.
* @return configure string
*/
const char *avresample_configuration(void);
 
/**
* Return the libavresample license.
*/
const char *avresample_license(void);
 
/**
* Get the AVClass for AVAudioResampleContext.
*
* Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
* without allocating a context.
*
* @see av_opt_find().
*
* @return AVClass for AVAudioResampleContext
*/
const AVClass *avresample_get_class(void);
 
/**
* Allocate AVAudioResampleContext and set options.
*
* @return allocated audio resample context, or NULL on failure
*/
AVAudioResampleContext *avresample_alloc_context(void);
 
/**
* Initialize AVAudioResampleContext.
*
* @param avr audio resample context
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_open(AVAudioResampleContext *avr);
 
/**
* Close AVAudioResampleContext.
*
* This closes the context, but it does not change the parameters. The context
* can be reopened with avresample_open(). It does, however, clear the output
* FIFO and any remaining leftover samples in the resampling delay buffer. If
* there was a custom matrix being used, that is also cleared.
*
* @see avresample_convert()
* @see avresample_set_matrix()
*
* @param avr audio resample context
*/
void avresample_close(AVAudioResampleContext *avr);
 
/**
* Free AVAudioResampleContext and associated AVOption values.
*
* This also calls avresample_close() before freeing.
*
* @param avr audio resample context
*/
void avresample_free(AVAudioResampleContext **avr);
 
/**
* Generate a channel mixing matrix.
*
* This function is the one used internally by libavresample 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 normalize if 1, coefficients will be normalized to prevent
* overflow. if 0, 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)
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize, double *matrix,
int stride, enum AVMatrixEncoding matrix_encoding);
 
/**
* Get the current channel mixing matrix.
*
* If no custom matrix has been previously set or the AVAudioResampleContext is
* not open, an error is returned.
*
* @param avr audio resample context
* @param 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
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
int stride);
 
/**
* Set channel mixing matrix.
*
* Allows for setting a custom mixing matrix, overriding the default matrix
* generated internally during avresample_open(). This function can be called
* anytime on an allocated context, either before or after calling
* avresample_open(), as long as the channel layouts have been set.
* avresample_convert() always uses the current matrix.
* Calling avresample_close() on the context will clear the current matrix.
*
* @see avresample_close()
*
* @param avr audio resample context
* @param 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
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride);
 
/**
* Set a customized input channel mapping.
*
* This function can only be called when the allocated context is not open.
* Also, the input channel layout must have already been set.
*
* Calling avresample_close() on the context will clear the channel mapping.
*
* The map for each input channel specifies the channel index in the source to
* use for that particular channel, or -1 to mute the channel. Source channels
* can be duplicated by using the same index for multiple input channels.
*
* Examples:
*
* Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
* { 1, 2, 0, 5, 3, 4 }
*
* Muting the 3rd channel in 4-channel input:
* { 0, 1, -1, 3 }
*
* Duplicating the left channel of stereo input:
* { 0, 0 }
*
* @param avr audio resample context
* @param channel_map customized input channel mapping
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_channel_mapping(AVAudioResampleContext *avr,
const int *channel_map);
 
/**
* Set compensation for resampling.
*
* This can be called anytime after avresample_open(). If resampling is not
* automatically enabled because of a sample rate conversion, the
* "force_resampling" option must have been set to 1 when opening the context
* in order to use resampling compensation.
*
* @param avr audio resample context
* @param sample_delta compensation delta, in samples
* @param compensation_distance compensation distance, in samples
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
int compensation_distance);
 
/**
* Convert input samples and write them to the output FIFO.
*
* The upper bound on the number of output samples is given by
* avresample_available() + (avresample_get_delay() + number of input samples) *
* output sample rate / input sample rate.
*
* The output data 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 avresample_read().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. avresample_get_delay() tells the number of remaining
* samples. To get this data as output, call avresample_convert() with NULL
* input.
*
* At the end of the conversion process, there may be data remaining in the
* internal FIFO buffer. avresample_available() tells the number of remaining
* samples. To get this data as output, either call avresample_convert() with
* NULL input or call avresample_read().
*
* @see avresample_available()
* @see avresample_read()
* @see avresample_get_delay()
*
* @param avr audio resample context
* @param output output data pointers
* @param out_plane_size output plane size, in bytes.
* This can be 0 if unknown, but that will lead to
* optimized functions not being used directly on the
* output, which could slow down some conversions.
* @param out_samples maximum number of samples that the output buffer can hold
* @param input input data pointers
* @param in_plane_size input plane size, in bytes
* This can be 0 if unknown, but that will lead to
* optimized functions not being used directly on the
* input, which could slow down some conversions.
* @param in_samples number of input samples to convert
* @return number of samples written to the output buffer,
* not including converted samples added to the internal
* output FIFO
*/
int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
int out_plane_size, int out_samples, uint8_t **input,
int in_plane_size, int in_samples);
 
/**
* Return the number of samples currently in the resampling delay buffer.
*
* When resampling, there may be a delay between the input and output. Any
* unconverted samples in each call are stored internally in a delay buffer.
* This function allows the user to determine the current number of samples in
* the delay buffer, which can be useful for synchronization.
*
* @see avresample_convert()
*
* @param avr audio resample context
* @return number of samples currently in the resampling delay buffer
*/
int avresample_get_delay(AVAudioResampleContext *avr);
 
/**
* Return the number of available samples in the output FIFO.
*
* During conversion, if the user does not specify an output buffer or
* specifies an output buffer that is smaller than what is needed, remaining
* samples that are not written to the output are stored to an internal FIFO
* buffer. The samples in the FIFO can be read with avresample_read() or
* avresample_convert().
*
* @see avresample_read()
* @see avresample_convert()
*
* @param avr audio resample context
* @return number of samples available for reading
*/
int avresample_available(AVAudioResampleContext *avr);
 
/**
* Read samples from the output FIFO.
*
* During conversion, if the user does not specify an output buffer or
* specifies an output buffer that is smaller than what is needed, remaining
* samples that are not written to the output are stored to an internal FIFO
* buffer. This function can be used to read samples from that internal FIFO.
*
* @see avresample_available()
* @see avresample_convert()
*
* @param avr audio resample context
* @param output output data pointers. May be NULL, in which case
* nb_samples of data is discarded from output FIFO.
* @param nb_samples number of samples to read from the FIFO
* @return the number of samples written to output
*/
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
 
/**
* @}
*/
 
#endif /* AVRESAMPLE_AVRESAMPLE_H */
/contrib/sdk/sources/ffmpeg/libavresample/dither.c
0,0 → 1,440
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* Triangular with Noise Shaping is based on opusfile.
* Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors
*
* 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
* Dithered Audio Sample Quantization
*
* Converts from dbl, flt, or s32 to s16 using dithering.
*/
 
#include <math.h>
#include <stdint.h>
 
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/lfg.h"
#include "libavutil/mem.h"
#include "libavutil/samplefmt.h"
#include "audio_convert.h"
#include "dither.h"
#include "internal.h"
 
typedef struct DitherState {
int mute;
unsigned int seed;
AVLFG lfg;
float *noise_buf;
int noise_buf_size;
int noise_buf_ptr;
float dither_a[4];
float dither_b[4];
} DitherState;
 
struct DitherContext {
DitherDSPContext ddsp;
enum AVResampleDitherMethod method;
int apply_map;
ChannelMapInfo *ch_map_info;
 
int mute_dither_threshold; // threshold for disabling dither
int mute_reset_threshold; // threshold for resetting noise shaping
const float *ns_coef_b; // noise shaping coeffs
const float *ns_coef_a; // noise shaping coeffs
 
int channels;
DitherState *state; // dither states for each channel
 
AudioData *flt_data; // input data in fltp
AudioData *s16_data; // dithered output in s16p
AudioConvert *ac_in; // converter for input to fltp
AudioConvert *ac_out; // converter for s16p to s16 (if needed)
 
void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
int samples_align;
};
 
/* mute threshold, in seconds */
#define MUTE_THRESHOLD_SEC 0.000333
 
/* scale factor for 16-bit output.
The signal is attenuated slightly to avoid clipping */
#define S16_SCALE 32753.0f
 
/* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */
#define LFG_SCALE (1.0f / (2.0f * INT32_MAX))
 
/* noise shaping coefficients */
 
static const float ns_48_coef_b[4] = {
2.2374f, -0.7339f, -0.1251f, -0.6033f
};
 
static const float ns_48_coef_a[4] = {
0.9030f, 0.0116f, -0.5853f, -0.2571f
};
 
static const float ns_44_coef_b[4] = {
2.2061f, -0.4707f, -0.2534f, -0.6213f
};
 
static const float ns_44_coef_a[4] = {
1.0587f, 0.0676f, -0.6054f, -0.2738f
};
 
static void dither_int_to_float_rectangular_c(float *dst, int *src, int len)
{
int i;
for (i = 0; i < len; i++)
dst[i] = src[i] * LFG_SCALE;
}
 
static void dither_int_to_float_triangular_c(float *dst, int *src0, int len)
{
int i;
int *src1 = src0 + len;
 
for (i = 0; i < len; i++) {
float r = src0[i] * LFG_SCALE;
r += src1[i] * LFG_SCALE;
dst[i] = r;
}
}
 
static void quantize_c(int16_t *dst, const float *src, float *dither, int len)
{
int i;
for (i = 0; i < len; i++)
dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i]));
}
 
#define SQRT_1_6 0.40824829046386301723f
 
static void dither_highpass_filter(float *src, int len)
{
int i;
 
/* filter is from libswresample in FFmpeg */
for (i = 0; i < len - 2; i++)
src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6;
}
 
static int generate_dither_noise(DitherContext *c, DitherState *state,
int min_samples)
{
int i;
int nb_samples = FFALIGN(min_samples, 16) + 16;
int buf_samples = nb_samples *
(c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2);
unsigned int *noise_buf_ui;
 
av_freep(&state->noise_buf);
state->noise_buf_size = state->noise_buf_ptr = 0;
 
state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf));
if (!state->noise_buf)
return AVERROR(ENOMEM);
state->noise_buf_size = FFALIGN(min_samples, 16);
noise_buf_ui = (unsigned int *)state->noise_buf;
 
av_lfg_init(&state->lfg, state->seed);
for (i = 0; i < buf_samples; i++)
noise_buf_ui[i] = av_lfg_get(&state->lfg);
 
c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples);
 
if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP)
dither_highpass_filter(state->noise_buf, nb_samples);
 
return 0;
}
 
static void quantize_triangular_ns(DitherContext *c, DitherState *state,
int16_t *dst, const float *src,
int nb_samples)
{
int i, j;
float *dither = &state->noise_buf[state->noise_buf_ptr];
 
if (state->mute > c->mute_reset_threshold)
memset(state->dither_a, 0, sizeof(state->dither_a));
 
for (i = 0; i < nb_samples; i++) {
float err = 0;
float sample = src[i] * S16_SCALE;
 
for (j = 0; j < 4; j++) {
err += c->ns_coef_b[j] * state->dither_b[j] -
c->ns_coef_a[j] * state->dither_a[j];
}
for (j = 3; j > 0; j--) {
state->dither_a[j] = state->dither_a[j - 1];
state->dither_b[j] = state->dither_b[j - 1];
}
state->dither_a[0] = err;
sample -= err;
 
if (state->mute > c->mute_dither_threshold) {
dst[i] = av_clip_int16(lrintf(sample));
state->dither_b[0] = 0;
} else {
dst[i] = av_clip_int16(lrintf(sample + dither[i]));
state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f);
}
 
state->mute++;
if (src[i])
state->mute = 0;
}
}
 
static int convert_samples(DitherContext *c, int16_t **dst, float * const *src,
int channels, int nb_samples)
{
int ch, ret;
int aligned_samples = FFALIGN(nb_samples, 16);
 
for (ch = 0; ch < channels; ch++) {
DitherState *state = &c->state[ch];
 
if (state->noise_buf_size < aligned_samples) {
ret = generate_dither_noise(c, state, nb_samples);
if (ret < 0)
return ret;
} else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) {
state->noise_buf_ptr = 0;
}
 
if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples);
} else {
c->quantize(dst[ch], src[ch],
&state->noise_buf[state->noise_buf_ptr],
FFALIGN(nb_samples, c->samples_align));
}
 
state->noise_buf_ptr += aligned_samples;
}
 
return 0;
}
 
int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
{
int ret;
AudioData *flt_data;
 
/* output directly to dst if it is planar */
if (dst->sample_fmt == AV_SAMPLE_FMT_S16P)
c->s16_data = dst;
else {
/* make sure s16_data is large enough for the output */
ret = ff_audio_data_realloc(c->s16_data, src->nb_samples);
if (ret < 0)
return ret;
}
 
if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
/* make sure flt_data is large enough for the input */
ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
if (ret < 0)
return ret;
flt_data = c->flt_data;
}
 
if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
/* convert input samples to fltp and scale to s16 range */
ret = ff_audio_convert(c->ac_in, flt_data, src);
if (ret < 0)
return ret;
} else if (c->apply_map) {
ret = ff_audio_data_copy(flt_data, src, c->ch_map_info);
if (ret < 0)
return ret;
} else {
flt_data = src;
}
 
/* check alignment and padding constraints */
if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
int ptr_align = FFMIN(flt_data->ptr_align, c->s16_data->ptr_align);
int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align);
int aligned_len = FFALIGN(src->nb_samples, c->ddsp.samples_align);
 
if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) {
c->quantize = c->ddsp.quantize;
c->samples_align = c->ddsp.samples_align;
} else {
c->quantize = quantize_c;
c->samples_align = 1;
}
}
 
ret = convert_samples(c, (int16_t **)c->s16_data->data,
(float * const *)flt_data->data, src->channels,
src->nb_samples);
if (ret < 0)
return ret;
 
c->s16_data->nb_samples = src->nb_samples;
 
/* interleave output to dst if needed */
if (dst->sample_fmt == AV_SAMPLE_FMT_S16) {
ret = ff_audio_convert(c->ac_out, dst, c->s16_data);
if (ret < 0)
return ret;
} else
c->s16_data = NULL;
 
return 0;
}
 
void ff_dither_free(DitherContext **cp)
{
DitherContext *c = *cp;
int ch;
 
if (!c)
return;
ff_audio_data_free(&c->flt_data);
ff_audio_data_free(&c->s16_data);
ff_audio_convert_free(&c->ac_in);
ff_audio_convert_free(&c->ac_out);
for (ch = 0; ch < c->channels; ch++)
av_free(c->state[ch].noise_buf);
av_free(c->state);
av_freep(cp);
}
 
static av_cold void dither_init(DitherDSPContext *ddsp,
enum AVResampleDitherMethod method)
{
ddsp->quantize = quantize_c;
ddsp->ptr_align = 1;
ddsp->samples_align = 1;
 
if (method == AV_RESAMPLE_DITHER_RECTANGULAR)
ddsp->dither_int_to_float = dither_int_to_float_rectangular_c;
else
ddsp->dither_int_to_float = dither_int_to_float_triangular_c;
 
if (ARCH_X86)
ff_dither_init_x86(ddsp, method);
}
 
DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int sample_rate, int apply_map)
{
AVLFG seed_gen;
DitherContext *c;
int ch;
 
if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 ||
av_get_bytes_per_sample(in_fmt) <= 2) {
av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n",
av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt));
return NULL;
}
 
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
 
c->apply_map = apply_map;
if (apply_map)
c->ch_map_info = &avr->ch_map_info;
 
if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
sample_rate != 48000 && sample_rate != 44100) {
av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
"for triangular_ns dither. using triangular_hp instead.\n");
avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP;
}
c->method = avr->dither_method;
dither_init(&c->ddsp, c->method);
 
if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
if (sample_rate == 48000) {
c->ns_coef_b = ns_48_coef_b;
c->ns_coef_a = ns_48_coef_a;
} else {
c->ns_coef_b = ns_44_coef_b;
c->ns_coef_a = ns_44_coef_a;
}
}
 
/* Either s16 or s16p output format is allowed, but s16p is used
internally, so we need to use a temp buffer and interleave if the output
format is s16 */
if (out_fmt != AV_SAMPLE_FMT_S16P) {
c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P,
"dither s16 buffer");
if (!c->s16_data)
goto fail;
 
c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
channels, sample_rate, 0);
if (!c->ac_out)
goto fail;
}
 
if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
"dither flt buffer");
if (!c->flt_data)
goto fail;
}
if (in_fmt != AV_SAMPLE_FMT_FLTP) {
c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
channels, sample_rate, c->apply_map);
if (!c->ac_in)
goto fail;
}
 
c->state = av_mallocz(channels * sizeof(*c->state));
if (!c->state)
goto fail;
c->channels = channels;
 
/* calculate thresholds for turning off dithering during periods of
silence to avoid replacing digital silence with quiet dither noise */
c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC);
c->mute_reset_threshold = c->mute_dither_threshold * 4;
 
/* initialize dither states */
av_lfg_init(&seed_gen, 0xC0FFEE);
for (ch = 0; ch < channels; ch++) {
DitherState *state = &c->state[ch];
state->mute = c->mute_reset_threshold + 1;
state->seed = av_lfg_get(&seed_gen);
generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2));
}
 
return c;
 
fail:
ff_dither_free(&c);
return NULL;
}
/contrib/sdk/sources/ffmpeg/libavresample/dither.h
0,0 → 1,93
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
*/
 
#ifndef AVRESAMPLE_DITHER_H
#define AVRESAMPLE_DITHER_H
 
#include "avresample.h"
#include "audio_data.h"
 
typedef struct DitherContext DitherContext;
 
typedef struct DitherDSPContext {
/**
* Convert samples from flt to s16 with added dither noise.
*
* @param dst destination float array, range -0.5 to 0.5
* @param src source int array, range INT_MIN to INT_MAX.
* @param dither float dither noise array
* @param len number of samples
*/
void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
 
int ptr_align; ///< src and dst constraits for quantize()
int samples_align; ///< len constraits for quantize()
 
/**
* Convert dither noise from int to float with triangular distribution.
*
* @param dst destination float array, range -0.5 to 0.5
* constraints: 32-byte aligned
* @param src0 source int array, range INT_MIN to INT_MAX.
* the array size is len * 2
* constraints: 32-byte aligned
* @param len number of output noise samples
* constraints: multiple of 16
*/
void (*dither_int_to_float)(float *dst, int *src0, int len);
} DitherDSPContext;
 
/**
* Allocate and initialize a DitherContext.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* DitherContext.
*
* @param avr AVAudioResampleContext
* @return newly-allocated DitherContext
*/
DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int sample_rate, int apply_map);
 
/**
* Free a DitherContext.
*
* @param c DitherContext
*/
void ff_dither_free(DitherContext **c);
 
/**
* Convert audio sample format with dithering.
*
* @param c DitherContext
* @param dst destination audio data
* @param src source audio data
* @return 0 if ok, negative AVERROR code on failure
*/
int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src);
 
/* arch-specific initialization functions */
 
void ff_dither_init_x86(DitherDSPContext *ddsp,
enum AVResampleDitherMethod method);
 
#endif /* AVRESAMPLE_DITHER_H */
/contrib/sdk/sources/ffmpeg/libavresample/internal.h
0,0 → 1,110
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
*/
 
#ifndef AVRESAMPLE_INTERNAL_H
#define AVRESAMPLE_INTERNAL_H
 
#include "libavutil/audio_fifo.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
 
typedef struct AudioData AudioData;
typedef struct AudioConvert AudioConvert;
typedef struct AudioMix AudioMix;
typedef struct ResampleContext ResampleContext;
 
enum RemapPoint {
REMAP_NONE,
REMAP_IN_COPY,
REMAP_IN_CONVERT,
REMAP_OUT_COPY,
REMAP_OUT_CONVERT,
};
 
typedef struct ChannelMapInfo {
int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */
int do_remap; /**< remap needed */
int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */
int do_copy; /**< copy needed */
int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */
int do_zero; /**< zeroing needed */
int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */
} ChannelMapInfo;
 
struct AVAudioResampleContext {
const AVClass *av_class; /**< AVClass for logging and AVOptions */
 
uint64_t in_channel_layout; /**< input channel layout */
enum AVSampleFormat in_sample_fmt; /**< input sample format */
int in_sample_rate; /**< input sample rate */
uint64_t out_channel_layout; /**< output channel layout */
enum AVSampleFormat out_sample_fmt; /**< output sample format */
int out_sample_rate; /**< output sample rate */
enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
double center_mix_level; /**< center mix level */
double surround_mix_level; /**< surround mix level */
double lfe_mix_level; /**< lfe mix level */
int normalize_mix_level; /**< enable mix level normalization */
int force_resampling; /**< force resampling */
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 */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
enum AVResampleFilterType filter_type; /**< resampling filter type */
int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
enum AVResampleDitherMethod dither_method; /**< dither method */
 
int in_channels; /**< number of input channels */
int out_channels; /**< number of output channels */
int resample_channels; /**< number of channels used for resampling */
int downmix_needed; /**< downmixing is needed */
int upmix_needed; /**< upmixing is needed */
int mixing_needed; /**< either upmixing or downmixing is needed */
int resample_needed; /**< resampling is needed */
int in_convert_needed; /**< input sample format conversion is needed */
int out_convert_needed; /**< output sample format conversion is needed */
int in_copy_needed; /**< input data copy is needed */
 
AudioData *in_buffer; /**< buffer for converted input */
AudioData *resample_out_buffer; /**< buffer for output from resampler */
AudioData *out_buffer; /**< buffer for converted output */
AVAudioFifo *out_fifo; /**< FIFO for output samples */
 
AudioConvert *ac_in; /**< input sample format conversion context */
AudioConvert *ac_out; /**< output sample format conversion context */
ResampleContext *resample; /**< resampling context */
AudioMix *am; /**< channel mixing context */
enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
 
/**
* mix matrix
* only used if avresample_set_matrix() is called before avresample_open()
*/
double *mix_matrix;
 
int use_channel_map;
enum RemapPoint remap_point;
ChannelMapInfo ch_map_info;
};
 
#endif /* AVRESAMPLE_INTERNAL_H */
/contrib/sdk/sources/ffmpeg/libavresample/libavresample.v
0,0 → 1,4
LIBAVRESAMPLE_$MAJOR {
global: av*;
local: *;
};
/contrib/sdk/sources/ffmpeg/libavresample/options.c
0,0 → 1,111
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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/mathematics.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_mix.h"
 
/**
* @file
* Options definition for AVAudioResampleContext.
*/
 
#define OFFSET(x) offsetof(AVAudioResampleContext, x)
#define PARAM AV_OPT_FLAG_AUDIO_PARAM
 
static const AVOption avresample_options[] = {
{ "in_channel_layout", "Input Channel Layout", OFFSET(in_channel_layout), AV_OPT_TYPE_INT64, { .i64 = 0 }, INT64_MIN, INT64_MAX, PARAM },
{ "in_sample_fmt", "Input Sample Format", OFFSET(in_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
{ "in_sample_rate", "Input Sample Rate", OFFSET(in_sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1, INT_MAX, PARAM },
{ "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { .i64 = 0 }, INT64_MIN, INT64_MAX, PARAM },
{ "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
{ "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1, INT_MAX, PARAM },
{ "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM, "internal_sample_fmt" },
{"u8" , "8-bit unsigned integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_U8 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"s16", "16-bit signed integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S16 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"s32", "32-bit signed integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"flt", "32-bit float", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"dbl", "64-bit double", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"u8p" , "8-bit unsigned integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_U8P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"s16p", "16-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S16P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"s32p", "32-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S32P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"fltp", "32-bit float planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_FLTP }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{"dblp", "64-bit double planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_DBLP }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
{ "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { .i64 = AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
{ "q8", "16-bit 8.8 Fixed-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_Q8 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "flt", "Floating-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "center_mix_level", "Center Mix Level", OFFSET(center_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = M_SQRT1_2 }, -32.0, 32.0, PARAM },
{ "surround_mix_level", "Surround Mix Level", OFFSET(surround_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = M_SQRT1_2 }, -32.0, 32.0, PARAM },
{ "lfe_mix_level", "LFE Mix Level", OFFSET(lfe_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -32.0, 32.0, PARAM },
{ "normalize_mix_level", "Normalize Mix Level", OFFSET(normalize_mix_level), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, PARAM },
{ "force_resampling", "Force Resampling", OFFSET(force_resampling), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, PARAM },
{ "filter_size", "Resampling Filter Size", OFFSET(filter_size), AV_OPT_TYPE_INT, { .i64 = 16 }, 0, 32, /* ??? */ PARAM },
{ "phase_shift", "Resampling Phase Shift", OFFSET(phase_shift), AV_OPT_TYPE_INT, { .i64 = 10 }, 0, 30, /* ??? */ PARAM },
{ "linear_interp", "Use Linear Interpolation", OFFSET(linear_interp), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, PARAM },
{ "cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { .dbl = 0.8 }, 0.0, 1.0, PARAM },
/* duplicate option in order to work with avconv */
{ "resample_cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { .dbl = 0.8 }, 0.0, 1.0, PARAM },
{ "matrix_encoding", "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", "None", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "filter_type", "Filter Type", OFFSET(filter_type), AV_OPT_TYPE_INT, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER }, AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_KAISER, PARAM, "filter_type" },
{ "cubic", "Cubic", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser", "Kaiser Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser_beta", "Kaiser Window Beta", OFFSET(kaiser_beta), AV_OPT_TYPE_INT, { .i64 = 9 }, 2, 16, PARAM },
{ "dither_method", "Dither Method", OFFSET(dither_method), AV_OPT_TYPE_INT, { .i64 = AV_RESAMPLE_DITHER_NONE }, 0, AV_RESAMPLE_DITHER_NB-1, PARAM, "dither_method"},
{"none", "No Dithering", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_NONE }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"rectangular", "Rectangular Dither", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_RECTANGULAR }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"triangular", "Triangular Dither", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"triangular_hp", "Triangular Dither With High Pass", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_HP }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"triangular_ns", "Triangular Dither With Noise Shaping", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_NS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{ NULL },
};
 
static const AVClass av_resample_context_class = {
.class_name = "AVAudioResampleContext",
.item_name = av_default_item_name,
.option = avresample_options,
.version = LIBAVUTIL_VERSION_INT,
};
 
AVAudioResampleContext *avresample_alloc_context(void)
{
AVAudioResampleContext *avr;
 
avr = av_mallocz(sizeof(*avr));
if (!avr)
return NULL;
 
avr->av_class = &av_resample_context_class;
av_opt_set_defaults(avr);
 
return avr;
}
 
const AVClass *avresample_get_class(void)
{
return &av_resample_context_class;
}
/contrib/sdk/sources/ffmpeg/libavresample/resample.c
0,0 → 1,469
/*
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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/common.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "internal.h"
#include "resample.h"
#include "audio_data.h"
 
struct ResampleContext {
AVAudioResampleContext *avr;
AudioData *buffer;
uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
enum AVResampleFilterType filter_type;
int kaiser_beta;
double factor;
void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
int dst_index, const void *src0, int src_size,
int index, int frac);
};
 
 
/* double template */
#define CONFIG_RESAMPLE_DBL
#include "resample_template.c"
#undef CONFIG_RESAMPLE_DBL
 
/* float template */
#define CONFIG_RESAMPLE_FLT
#include "resample_template.c"
#undef CONFIG_RESAMPLE_FLT
 
/* s32 template */
#define CONFIG_RESAMPLE_S32
#include "resample_template.c"
#undef CONFIG_RESAMPLE_S32
 
/* s16 template */
#include "resample_template.c"
 
 
/* 0th order modified bessel function of the first kind. */
static double bessel(double x)
{
double v = 1;
double lastv = 0;
double t = 1;
int i;
 
x = x * x / 4;
for (i = 1; v != lastv; i++) {
lastv = v;
t *= x / (i * i);
v += t;
}
return v;
}
 
/* Build a polyphase filterbank. */
static int build_filter(ResampleContext *c)
{
int ph, i;
double x, y, w, factor;
double *tab;
int tap_count = c->filter_length;
int phase_count = 1 << c->phase_shift;
const int center = (tap_count - 1) / 2;
 
tab = av_malloc(tap_count * sizeof(*tab));
if (!tab)
return AVERROR(ENOMEM);
 
/* if upsampling, only need to interpolate, no filter */
factor = FFMIN(c->factor, 1.0);
 
for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
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 y = sin(x) / x;
switch (c->filter_type) {
case AV_RESAMPLE_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 AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0 * x / (factor * tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos( w) +
0.1365995 * cos(2 * w) -
0.0106411 * cos(3 * w);
break;
case AV_RESAMPLE_FILTER_TYPE_KAISER:
w = 2.0 * x / (factor * tap_count * M_PI);
y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
break;
}
 
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for (i = 0; i < tap_count; i++)
tab[i] = tab[i] / norm;
 
c->set_filter(c->filter_bank, tab, ph, tap_count);
}
 
av_free(tab);
return 0;
}
 
ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
{
ResampleContext *c;
int out_rate = avr->out_sample_rate;
int in_rate = avr->in_sample_rate;
double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
int phase_count = 1 << avr->phase_shift;
int felem_size;
 
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"resampling: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
return NULL;
}
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
 
c->avr = avr;
c->phase_shift = avr->phase_shift;
c->phase_mask = phase_count - 1;
c->linear = avr->linear_interp;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
c->filter_type = avr->filter_type;
c->kaiser_beta = avr->kaiser_beta;
 
switch (avr->internal_sample_fmt) {
case AV_SAMPLE_FMT_DBLP:
c->resample_one = resample_one_dbl;
c->set_filter = set_filter_dbl;
break;
case AV_SAMPLE_FMT_FLTP:
c->resample_one = resample_one_flt;
c->set_filter = set_filter_flt;
break;
case AV_SAMPLE_FMT_S32P:
c->resample_one = resample_one_s32;
c->set_filter = set_filter_s32;
break;
case AV_SAMPLE_FMT_S16P:
c->resample_one = resample_one_s16;
c->set_filter = set_filter_s16;
break;
}
 
felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
if (!c->filter_bank)
goto error;
 
if (build_filter(c) < 0)
goto error;
 
memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
c->filter_bank, (c->filter_length - 1) * felem_size);
memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
&c->filter_bank[(c->filter_length - 1) * felem_size], 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;
c->ideal_dst_incr = c->dst_incr;
 
c->index = -phase_count * ((c->filter_length - 1) / 2);
c->frac = 0;
 
/* allocate internal buffer */
c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
avr->internal_sample_fmt,
"resample buffer");
if (!c->buffer)
goto error;
 
av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
av_get_sample_fmt_name(avr->internal_sample_fmt),
avr->in_sample_rate, avr->out_sample_rate);
 
return c;
 
error:
ff_audio_data_free(&c->buffer);
av_free(c->filter_bank);
av_free(c);
return NULL;
}
 
void ff_audio_resample_free(ResampleContext **c)
{
if (!*c)
return;
ff_audio_data_free(&(*c)->buffer);
av_free((*c)->filter_bank);
av_freep(c);
}
 
int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
int compensation_distance)
{
ResampleContext *c;
AudioData *fifo_buf = NULL;
int ret = 0;
 
if (compensation_distance < 0)
return AVERROR(EINVAL);
if (!compensation_distance && sample_delta)
return AVERROR(EINVAL);
 
if (!avr->resample_needed) {
#if FF_API_RESAMPLE_CLOSE_OPEN
/* if resampling was not enabled previously, re-initialize the
AVAudioResampleContext and force resampling */
int fifo_samples;
int restore_matrix = 0;
double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
 
/* buffer any remaining samples in the output FIFO before closing */
fifo_samples = av_audio_fifo_size(avr->out_fifo);
if (fifo_samples > 0) {
fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
avr->out_sample_fmt, NULL);
if (!fifo_buf)
return AVERROR(EINVAL);
ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
fifo_samples);
if (ret < 0)
goto reinit_fail;
}
/* save the channel mixing matrix */
if (avr->am) {
ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
if (ret < 0)
goto reinit_fail;
restore_matrix = 1;
}
 
/* close the AVAudioResampleContext */
avresample_close(avr);
 
avr->force_resampling = 1;
 
/* restore the channel mixing matrix */
if (restore_matrix) {
ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
if (ret < 0)
goto reinit_fail;
}
 
/* re-open the AVAudioResampleContext */
ret = avresample_open(avr);
if (ret < 0)
goto reinit_fail;
 
/* restore buffered samples to the output FIFO */
if (fifo_samples > 0) {
ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
fifo_samples);
if (ret < 0)
goto reinit_fail;
ff_audio_data_free(&fifo_buf);
}
#else
av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
return AVERROR(EINVAL);
#endif
}
c = avr->resample;
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;
}
return 0;
 
reinit_fail:
ff_audio_data_free(&fifo_buf);
return ret;
}
 
static int resample(ResampleContext *c, void *dst, const void *src,
int *consumed, int src_size, int dst_size, int update_ctx)
{
int dst_index;
int index = c->index;
int frac = c->frac;
int dst_incr_frac = c->dst_incr % c->src_incr;
int dst_incr = c->dst_incr / c->src_incr;
int compensation_distance = c->compensation_distance;
 
if (!dst != !src)
return AVERROR(EINVAL);
 
if (compensation_distance == 0 && c->filter_length == 1 &&
c->phase_shift == 0) {
int64_t index2 = ((int64_t)index) << 32;
int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
dst_size = FFMIN(dst_size,
(src_size-1-index) * (int64_t)c->src_incr /
c->dst_incr);
 
if (dst) {
for(dst_index = 0; dst_index < dst_size; dst_index++) {
c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
index2 += incr;
}
} else {
dst_index = dst_size;
}
index += dst_index * dst_incr;
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
} else {
for (dst_index = 0; dst_index < dst_size; dst_index++) {
int sample_index = index >> c->phase_shift;
 
if (sample_index + c->filter_length > src_size ||
-sample_index >= src_size)
break;
 
if (dst)
c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
 
frac += dst_incr_frac;
index += dst_incr;
if (frac >= c->src_incr) {
frac -= c->src_incr;
index++;
}
if (dst_index + 1 == compensation_distance) {
compensation_distance = 0;
dst_incr_frac = c->ideal_dst_incr % c->src_incr;
dst_incr = c->ideal_dst_incr / c->src_incr;
}
}
}
if (consumed)
*consumed = FFMAX(index, 0) >> c->phase_shift;
 
if (update_ctx) {
if (index >= 0)
index &= c->phase_mask;
 
if (compensation_distance) {
compensation_distance -= dst_index;
if (compensation_distance <= 0)
return AVERROR_BUG;
}
c->frac = frac;
c->index = index;
c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
c->compensation_distance = compensation_distance;
}
 
return dst_index;
}
 
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
{
int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
int ret = AVERROR(EINVAL);
 
in_samples = src ? src->nb_samples : 0;
in_leftover = c->buffer->nb_samples;
 
/* add input samples to the internal buffer */
if (src) {
ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
if (ret < 0)
return ret;
} else if (!in_leftover) {
/* no remaining samples to flush */
return 0;
} else {
/* TODO: pad buffer to flush completely */
}
 
/* calculate output size and reallocate output buffer if needed */
/* TODO: try to calculate this without the dummy resample() run */
if (!dst->read_only && dst->allow_realloc) {
out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
INT_MAX, 0);
ret = ff_audio_data_realloc(dst, out_samples);
if (ret < 0) {
av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
return ret;
}
}
 
/* resample each channel plane */
for (ch = 0; ch < c->buffer->channels; ch++) {
out_samples = resample(c, (void *)dst->data[ch],
(const void *)c->buffer->data[ch], &consumed,
c->buffer->nb_samples, dst->allocated_samples,
ch + 1 == c->buffer->channels);
}
if (out_samples < 0) {
av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
return out_samples;
}
 
/* drain consumed samples from the internal buffer */
ff_audio_data_drain(c->buffer, consumed);
 
av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
in_samples, in_leftover, out_samples, c->buffer->nb_samples);
 
dst->nb_samples = out_samples;
return 0;
}
 
int avresample_get_delay(AVAudioResampleContext *avr)
{
if (!avr->resample_needed || !avr->resample)
return 0;
 
return avr->resample->buffer->nb_samples;
}
/contrib/sdk/sources/ffmpeg/libavresample/resample.h
0,0 → 1,67
/*
* Copyright (c) 2004 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 AVRESAMPLE_RESAMPLE_H
#define AVRESAMPLE_RESAMPLE_H
 
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
 
/**
* Allocate and initialize a ResampleContext.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* ResampleContext.
*
* @param avr AVAudioResampleContext
* @return newly-allocated ResampleContext
*/
ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
 
/**
* Free a ResampleContext.
*
* @param c ResampleContext
*/
void ff_audio_resample_free(ResampleContext **c);
 
/**
* Resample audio data.
*
* Changes the sample rate.
*
* @par
* All samples in the source data may not be consumed depending on the
* resampling parameters and the size of the output buffer. The unconsumed
* samples are automatically added to the start of the source in the next call.
* If the destination data can be reallocated, that may be done in this function
* in order to fit all available output. If it cannot be reallocated, fewer
* input samples will be consumed in order to have the output fit in the
* destination data buffers.
*
* @param c ResampleContext
* @param dst destination audio data
* @param src source audio data
* @return 0 on success, negative AVERROR code on failure
*/
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src);
 
#endif /* AVRESAMPLE_RESAMPLE_H */
/contrib/sdk/sources/ffmpeg/libavresample/resample_template.c
0,0 → 1,102
/*
* Copyright (c) 2004 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
*/
 
#if defined(CONFIG_RESAMPLE_DBL)
#define SET_TYPE(func) func ## _dbl
#define FELEM double
#define FELEM2 double
#define FELEML double
#define OUT(d, v) d = v
#define DBL_TO_FELEM(d, v) d = v
#elif defined(CONFIG_RESAMPLE_FLT)
#define SET_TYPE(func) func ## _flt
#define FELEM float
#define FELEM2 float
#define FELEML float
#define OUT(d, v) d = v
#define DBL_TO_FELEM(d, v) d = v
#elif defined(CONFIG_RESAMPLE_S32)
#define SET_TYPE(func) func ## _s32
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
#else
#define SET_TYPE(func) func ## _s16
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
#endif
 
static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter,
void *dst0, int dst_index, const void *src0,
int src_size, int index, int frac)
{
FELEM *dst = dst0;
const FELEM *src = src0;
 
if (no_filter) {
dst[dst_index] = src[index];
} else {
int i;
int sample_index = index >> c->phase_shift;
FELEM2 val = 0;
FELEM *filter = ((FELEM *)c->filter_bank) +
c->filter_length * (index & c->phase_mask);
 
if (sample_index < 0) {
for (i = 0; i < c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] *
(FELEM2)filter[i];
} else if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[abs(sample_index + i)] * (FELEM2)filter[i];
v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
}
val += (v2 - val) * (FELEML)frac / c->src_incr;
} else {
for (i = 0; i < c->filter_length; i++)
val += src[sample_index + i] * (FELEM2)filter[i];
}
 
OUT(dst[dst_index], val);
}
}
 
static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
int tap_count)
{
int i;
FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
for (i = 0; i < tap_count; i++) {
DBL_TO_FELEM(filter[i], tab[i]);
}
}
 
#undef SET_TYPE
#undef FELEM
#undef FELEM2
#undef FELEML
#undef OUT
#undef DBL_TO_FELEM
/contrib/sdk/sources/ffmpeg/libavresample/utils.c
0,0 → 1,635
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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/common.h"
#include "libavutil/dict.h"
// #include "libavutil/error.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
 
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_convert.h"
#include "audio_mix.h"
#include "resample.h"
 
int avresample_open(AVAudioResampleContext *avr)
{
int ret;
 
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
avr->in_channel_layout);
return AVERROR(EINVAL);
}
avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
avr->out_channel_layout);
return AVERROR(EINVAL);
}
avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
avr->downmix_needed = avr->in_channels > avr->out_channels;
avr->upmix_needed = avr->out_channels > avr->in_channels ||
(!avr->downmix_needed && (avr->mix_matrix ||
avr->in_channel_layout != avr->out_channel_layout));
avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
 
/* set resampling parameters */
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
 
/* select internal sample format if not specified by the user */
if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
(avr->mixing_needed || avr->resample_needed)) {
enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
av_get_bytes_per_sample(out_fmt));
if (max_bps <= 2) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
} else {
if (max_bps <= 4) {
if (in_fmt == AV_SAMPLE_FMT_S32P ||
out_fmt == AV_SAMPLE_FMT_S32P) {
if (in_fmt == AV_SAMPLE_FMT_FLTP ||
out_fmt == AV_SAMPLE_FMT_FLTP) {
/* if one is s32 and the other is flt, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
} else {
/* if one is s32 and the other is s32, s16, or u8, use s32 */
avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
}
} else {
/* if one is flt and the other is flt, s16 or u8, use flt */
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
} else {
/* if either is dbl, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
}
}
av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
}
 
/* treat all mono as planar for easier comparison */
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
if (avr->out_channels == 1)
avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
 
/* we may need to add an extra conversion in order to remap channels if
the output format is not planar */
if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
!av_sample_fmt_is_planar(avr->out_sample_fmt)) {
avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
}
 
/* set sample format conversion parameters */
if (avr->resample_needed || avr->mixing_needed)
avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
else
avr->in_convert_needed = avr->use_channel_map &&
!av_sample_fmt_is_planar(avr->out_sample_fmt);
 
if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
else
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
 
avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
(avr->use_channel_map && avr->resample_needed));
 
if (avr->use_channel_map) {
if (avr->in_copy_needed) {
avr->remap_point = REMAP_IN_COPY;
av_dlog(avr, "remap channels during in_copy\n");
} else if (avr->in_convert_needed) {
avr->remap_point = REMAP_IN_CONVERT;
av_dlog(avr, "remap channels during in_convert\n");
} else if (avr->out_convert_needed) {
avr->remap_point = REMAP_OUT_CONVERT;
av_dlog(avr, "remap channels during out_convert\n");
} else {
avr->remap_point = REMAP_OUT_COPY;
av_dlog(avr, "remap channels during out_copy\n");
}
 
#ifdef DEBUG
{
int ch;
av_dlog(avr, "output map: ");
if (avr->ch_map_info.do_remap)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "copy map: ");
if (avr->ch_map_info.do_copy)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "zero map: ");
if (avr->ch_map_info.do_zero)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "input map: ");
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]);
av_dlog(avr, "\n");
}
#endif
} else
avr->remap_point = REMAP_NONE;
 
/* allocate buffers */
if (avr->in_copy_needed || avr->in_convert_needed) {
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
0, avr->internal_sample_fmt,
"in_buffer");
if (!avr->in_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
0, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->out_convert_needed) {
avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
avr->out_sample_fmt, "out_buffer");
if (!avr->out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
1024);
if (!avr->out_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
 
/* setup contexts */
if (avr->in_convert_needed) {
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
avr->in_sample_fmt, avr->in_channels,
avr->in_sample_rate,
avr->remap_point == REMAP_IN_CONVERT);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->out_convert_needed) {
enum AVSampleFormat src_fmt;
if (avr->in_convert_needed)
src_fmt = avr->internal_sample_fmt;
else
src_fmt = avr->in_sample_fmt;
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
avr->out_channels,
avr->out_sample_rate,
avr->remap_point == REMAP_OUT_CONVERT);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->resample_needed) {
avr->resample = ff_audio_resample_init(avr);
if (!avr->resample) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
avr->am = ff_audio_mix_alloc(avr);
if (!avr->am) {
ret = AVERROR(ENOMEM);
goto error;
}
}
 
return 0;
 
error:
avresample_close(avr);
return ret;
}
 
void avresample_close(AVAudioResampleContext *avr)
{
ff_audio_data_free(&avr->in_buffer);
ff_audio_data_free(&avr->resample_out_buffer);
ff_audio_data_free(&avr->out_buffer);
av_audio_fifo_free(avr->out_fifo);
avr->out_fifo = NULL;
ff_audio_convert_free(&avr->ac_in);
ff_audio_convert_free(&avr->ac_out);
ff_audio_resample_free(&avr->resample);
ff_audio_mix_free(&avr->am);
av_freep(&avr->mix_matrix);
 
avr->use_channel_map = 0;
}
 
void avresample_free(AVAudioResampleContext **avr)
{
if (!*avr)
return;
avresample_close(*avr);
av_opt_free(*avr);
av_freep(avr);
}
 
static int handle_buffered_output(AVAudioResampleContext *avr,
AudioData *output, AudioData *converted)
{
int ret;
 
if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
(converted && output->allocated_samples < converted->nb_samples)) {
if (converted) {
/* if there are any samples in the output FIFO or if the
user-supplied output buffer is not large enough for all samples,
we add to the output FIFO */
av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
converted->nb_samples);
if (ret < 0)
return ret;
}
 
/* if the user specified an output buffer, read samples from the output
FIFO to the user output */
if (output && output->allocated_samples > 0) {
av_dlog(avr, "[FIFO] read from out_fifo to output\n");
av_dlog(avr, "[end conversion]\n");
return ff_audio_data_read_from_fifo(avr->out_fifo, output,
output->allocated_samples);
}
} else if (converted) {
/* copy directly to output if it is large enough or there is not any
data in the output FIFO */
av_dlog(avr, "[copy] %s to output\n", converted->name);
output->nb_samples = 0;
ret = ff_audio_data_copy(output, converted,
avr->remap_point == REMAP_OUT_COPY ?
&avr->ch_map_info : NULL);
if (ret < 0)
return ret;
av_dlog(avr, "[end conversion]\n");
return output->nb_samples;
}
av_dlog(avr, "[end conversion]\n");
return 0;
}
 
int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
uint8_t **output, int out_plane_size,
int out_samples, uint8_t **input,
int in_plane_size, int in_samples)
{
AudioData input_buffer;
AudioData output_buffer;
AudioData *current_buffer;
int ret, direct_output;
 
/* reset internal buffers */
if (avr->in_buffer) {
avr->in_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->in_buffer,
avr->in_buffer->allocated_channels);
}
if (avr->resample_out_buffer) {
avr->resample_out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->resample_out_buffer,
avr->resample_out_buffer->allocated_channels);
}
if (avr->out_buffer) {
avr->out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->out_buffer,
avr->out_buffer->allocated_channels);
}
 
av_dlog(avr, "[start conversion]\n");
 
/* initialize output_buffer with output data */
direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
if (output) {
ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
avr->out_channels, out_samples,
avr->out_sample_fmt, 0, "output");
if (ret < 0)
return ret;
output_buffer.nb_samples = 0;
}
 
if (input) {
/* initialize input_buffer with input data */
ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
avr->in_channels, in_samples,
avr->in_sample_fmt, 1, "input");
if (ret < 0)
return ret;
current_buffer = &input_buffer;
 
if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
!avr->out_convert_needed && direct_output && out_samples >= in_samples) {
/* in some rare cases we can copy input to output and upmix
directly in the output buffer */
av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
ret = ff_audio_data_copy(&output_buffer, current_buffer,
avr->remap_point == REMAP_OUT_COPY ?
&avr->ch_map_info : NULL);
if (ret < 0)
return ret;
current_buffer = &output_buffer;
} else if (avr->remap_point == REMAP_OUT_COPY &&
(!direct_output || out_samples < in_samples)) {
/* if remapping channels during output copy, we may need to
* use an intermediate buffer in order to remap before adding
* samples to the output fifo */
av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name);
ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
&avr->ch_map_info);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
} else if (avr->in_copy_needed || avr->in_convert_needed) {
/* if needed, copy or convert input to in_buffer, and downmix if
applicable */
if (avr->in_convert_needed) {
ret = ff_audio_data_realloc(avr->in_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
current_buffer);
if (ret < 0)
return ret;
} else {
av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
avr->remap_point == REMAP_IN_COPY ?
&avr->ch_map_info : NULL);
if (ret < 0)
return ret;
}
ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
if (avr->downmix_needed) {
av_dlog(avr, "[downmix] in_buffer\n");
ret = ff_audio_mix(avr->am, avr->in_buffer);
if (ret < 0)
return ret;
}
current_buffer = avr->in_buffer;
}
} else {
/* flush resampling buffer and/or output FIFO if input is NULL */
if (!avr->resample_needed)
return handle_buffered_output(avr, output ? &output_buffer : NULL,
NULL);
current_buffer = NULL;
}
 
if (avr->resample_needed) {
AudioData *resample_out;
 
if (!avr->out_convert_needed && direct_output && out_samples > 0)
resample_out = &output_buffer;
else
resample_out = avr->resample_out_buffer;
av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
resample_out->name);
ret = ff_audio_resample(avr->resample, resample_out,
current_buffer);
if (ret < 0)
return ret;
 
/* if resampling did not produce any samples, just return 0 */
if (resample_out->nb_samples == 0) {
av_dlog(avr, "[end conversion]\n");
return 0;
}
 
current_buffer = resample_out;
}
 
if (avr->upmix_needed) {
av_dlog(avr, "[upmix] %s\n", current_buffer->name);
ret = ff_audio_mix(avr->am, current_buffer);
if (ret < 0)
return ret;
}
 
/* if we resampled or upmixed directly to output, return here */
if (current_buffer == &output_buffer) {
av_dlog(avr, "[end conversion]\n");
return current_buffer->nb_samples;
}
 
if (avr->out_convert_needed) {
if (direct_output && out_samples >= current_buffer->nb_samples) {
/* convert directly to output */
av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
if (ret < 0)
return ret;
 
av_dlog(avr, "[end conversion]\n");
return output_buffer.nb_samples;
} else {
ret = ff_audio_data_realloc(avr->out_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
current_buffer);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
}
}
 
return handle_buffered_output(avr, output ? &output_buffer : NULL,
current_buffer);
}
 
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
int stride)
{
int in_channels, out_channels, i, o;
 
if (avr->am)
return ff_audio_mix_get_matrix(avr->am, matrix, stride);
 
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
 
if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
return AVERROR(EINVAL);
}
 
if (!avr->mix_matrix) {
av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
return AVERROR(EINVAL);
}
 
for (o = 0; o < out_channels; o++)
for (i = 0; i < in_channels; i++)
matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i];
 
return 0;
}
 
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride)
{
int in_channels, out_channels, i, o;
 
if (avr->am)
return ff_audio_mix_set_matrix(avr->am, matrix, stride);
 
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
 
if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
return AVERROR(EINVAL);
}
 
if (avr->mix_matrix)
av_freep(&avr->mix_matrix);
avr->mix_matrix = av_malloc(in_channels * out_channels *
sizeof(*avr->mix_matrix));
if (!avr->mix_matrix)
return AVERROR(ENOMEM);
 
for (o = 0; o < out_channels; o++)
for (i = 0; i < in_channels; i++)
avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i];
 
return 0;
}
 
int avresample_set_channel_mapping(AVAudioResampleContext *avr,
const int *channel_map)
{
ChannelMapInfo *info = &avr->ch_map_info;
int in_channels, ch, i;
 
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
return AVERROR(EINVAL);
}
 
memset(info, 0, sizeof(*info));
memset(info->input_map, -1, sizeof(info->input_map));
 
for (ch = 0; ch < in_channels; ch++) {
if (channel_map[ch] >= in_channels) {
av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
return AVERROR(EINVAL);
}
if (channel_map[ch] < 0) {
info->channel_zero[ch] = 1;
info->channel_map[ch] = -1;
info->do_zero = 1;
} else if (info->input_map[channel_map[ch]] >= 0) {
info->channel_copy[ch] = info->input_map[channel_map[ch]];
info->channel_map[ch] = -1;
info->do_copy = 1;
} else {
info->channel_map[ch] = channel_map[ch];
info->input_map[channel_map[ch]] = ch;
info->do_remap = 1;
}
}
/* Fill-in unmapped input channels with unmapped output channels.
This is used when remapping during conversion from interleaved to
planar format. */
for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
while (ch < in_channels && info->input_map[ch] >= 0)
ch++;
while (i < in_channels && info->channel_map[i] >= 0)
i++;
if (ch >= in_channels || i >= in_channels)
break;
info->input_map[ch] = i;
}
 
avr->use_channel_map = 1;
return 0;
}
 
int avresample_available(AVAudioResampleContext *avr)
{
return av_audio_fifo_size(avr->out_fifo);
}
 
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
{
if (!output)
return av_audio_fifo_drain(avr->out_fifo, nb_samples);
return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
}
 
unsigned avresample_version(void)
{
return LIBAVRESAMPLE_VERSION_INT;
}
 
const char *avresample_license(void)
{
#define LICENSE_PREFIX "libavresample license: "
return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
}
 
const char *avresample_configuration(void)
{
return FFMPEG_CONFIGURATION;
}
/contrib/sdk/sources/ffmpeg/libavresample/version.h
0,0 → 1,52
/*
* 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 AVRESAMPLE_VERSION_H
#define AVRESAMPLE_VERSION_H
 
/**
* @file
* @ingroup lavr
* Libavresample version macros.
*/
 
#define LIBAVRESAMPLE_VERSION_MAJOR 1
#define LIBAVRESAMPLE_VERSION_MINOR 1
#define LIBAVRESAMPLE_VERSION_MICRO 0
 
#define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
LIBAVRESAMPLE_VERSION_MINOR, \
LIBAVRESAMPLE_VERSION_MICRO)
#define LIBAVRESAMPLE_VERSION AV_VERSION(LIBAVRESAMPLE_VERSION_MAJOR, \
LIBAVRESAMPLE_VERSION_MINOR, \
LIBAVRESAMPLE_VERSION_MICRO)
#define LIBAVRESAMPLE_BUILD LIBAVRESAMPLE_VERSION_INT
 
#define LIBAVRESAMPLE_IDENT "Lavr" AV_STRINGIFY(LIBAVRESAMPLE_VERSION)
 
/**
* FF_API_* defines may be placed below to indicate public API that will be
* dropped at a future version bump. The defines themselves are not part of
* the public API and may change, break or disappear at any time.
*/
 
#ifndef FF_API_RESAMPLE_CLOSE_OPEN
#define FF_API_RESAMPLE_CLOSE_OPEN (LIBAVRESAMPLE_VERSION_MAJOR < 2)
#endif
 
#endif /* AVRESAMPLE_VERSION_H */
/contrib/sdk/sources/ffmpeg/libavresample/x86/Makefile
0,0 → 1,7
OBJS += x86/audio_convert_init.o \
x86/audio_mix_init.o \
x86/dither_init.o \
 
YASM-OBJS += x86/audio_convert.o \
x86/audio_mix.o \
x86/dither.o \
/contrib/sdk/sources/ffmpeg/libavresample/x86/audio_convert.asm
0,0 → 1,1261
;******************************************************************************
;* x86 optimized Format Conversion Utils
;* Copyright (c) 2008 Loren Merritt
;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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"
%include "util.asm"
 
SECTION_RODATA 32
 
pf_s32_inv_scale: times 8 dd 0x30000000
pf_s32_scale: times 8 dd 0x4f000000
pf_s32_clip: times 8 dd 0x4effffff
pf_s16_inv_scale: times 4 dd 0x38000000
pf_s16_scale: times 4 dd 0x47000000
pb_shuf_unpack_even: db -1, -1, 0, 1, -1, -1, 2, 3, -1, -1, 8, 9, -1, -1, 10, 11
pb_shuf_unpack_odd: db -1, -1, 4, 5, -1, -1, 6, 7, -1, -1, 12, 13, -1, -1, 14, 15
pb_interleave_words: SHUFFLE_MASK_W 0, 4, 1, 5, 2, 6, 3, 7
pb_deinterleave_words: SHUFFLE_MASK_W 0, 2, 4, 6, 1, 3, 5, 7
pw_zero_even: times 4 dw 0x0000, 0xffff
 
SECTION_TEXT
 
;------------------------------------------------------------------------------
; void ff_conv_s16_to_s32(int32_t *dst, const int16_t *src, int len);
;------------------------------------------------------------------------------
 
INIT_XMM sse2
cglobal conv_s16_to_s32, 3,3,3, dst, src, len
lea lenq, [2*lend]
lea dstq, [dstq+2*lenq]
add srcq, lenq
neg lenq
.loop:
mova m2, [srcq+lenq]
pxor m0, m0
pxor m1, m1
punpcklwd m0, m2
punpckhwd m1, m2
mova [dstq+2*lenq ], m0
mova [dstq+2*lenq+mmsize], m1
add lenq, mmsize
jl .loop
REP_RET
 
;------------------------------------------------------------------------------
; void ff_conv_s16_to_flt(float *dst, const int16_t *src, int len);
;------------------------------------------------------------------------------
 
%macro CONV_S16_TO_FLT 0
cglobal conv_s16_to_flt, 3,3,3, dst, src, len
lea lenq, [2*lend]
add srcq, lenq
lea dstq, [dstq + 2*lenq]
neg lenq
mova m2, [pf_s16_inv_scale]
ALIGN 16
.loop:
mova m0, [srcq+lenq]
S16_TO_S32_SX 0, 1
cvtdq2ps m0, m0
cvtdq2ps m1, m1
mulps m0, m2
mulps m1, m2
mova [dstq+2*lenq ], m0
mova [dstq+2*lenq+mmsize], m1
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16_TO_FLT
INIT_XMM sse4
CONV_S16_TO_FLT
 
;------------------------------------------------------------------------------
; void ff_conv_s32_to_s16(int16_t *dst, const int32_t *src, int len);
;------------------------------------------------------------------------------
 
%macro CONV_S32_TO_S16 0
cglobal conv_s32_to_s16, 3,3,4, dst, src, len
lea lenq, [2*lend]
lea srcq, [srcq+2*lenq]
add dstq, lenq
neg lenq
.loop:
mova m0, [srcq+2*lenq ]
mova m1, [srcq+2*lenq+ mmsize]
mova m2, [srcq+2*lenq+2*mmsize]
mova m3, [srcq+2*lenq+3*mmsize]
psrad m0, 16
psrad m1, 16
psrad m2, 16
psrad m3, 16
packssdw m0, m1
packssdw m2, m3
mova [dstq+lenq ], m0
mova [dstq+lenq+mmsize], m2
add lenq, mmsize*2
jl .loop
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
 
INIT_MMX mmx
CONV_S32_TO_S16
INIT_XMM sse2
CONV_S32_TO_S16
 
;------------------------------------------------------------------------------
; void ff_conv_s32_to_flt(float *dst, const int32_t *src, int len);
;------------------------------------------------------------------------------
 
%macro CONV_S32_TO_FLT 0
cglobal conv_s32_to_flt, 3,3,3, dst, src, len
lea lenq, [4*lend]
add srcq, lenq
add dstq, lenq
neg lenq
mova m0, [pf_s32_inv_scale]
ALIGN 16
.loop:
cvtdq2ps m1, [srcq+lenq ]
cvtdq2ps m2, [srcq+lenq+mmsize]
mulps m1, m1, m0
mulps m2, m2, m0
mova [dstq+lenq ], m1
mova [dstq+lenq+mmsize], m2
add lenq, mmsize*2
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S32_TO_FLT
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
CONV_S32_TO_FLT
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_flt_to_s16(int16_t *dst, const float *src, int len);
;------------------------------------------------------------------------------
 
INIT_XMM sse2
cglobal conv_flt_to_s16, 3,3,5, dst, src, len
lea lenq, [2*lend]
lea srcq, [srcq+2*lenq]
add dstq, lenq
neg lenq
mova m4, [pf_s16_scale]
.loop:
mova m0, [srcq+2*lenq ]
mova m1, [srcq+2*lenq+1*mmsize]
mova m2, [srcq+2*lenq+2*mmsize]
mova m3, [srcq+2*lenq+3*mmsize]
mulps m0, m4
mulps m1, m4
mulps m2, m4
mulps m3, m4
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
packssdw m0, m1
packssdw m2, m3
mova [dstq+lenq ], m0
mova [dstq+lenq+mmsize], m2
add lenq, mmsize*2
jl .loop
REP_RET
 
;------------------------------------------------------------------------------
; void ff_conv_flt_to_s32(int32_t *dst, const float *src, int len);
;------------------------------------------------------------------------------
 
%macro CONV_FLT_TO_S32 0
cglobal conv_flt_to_s32, 3,3,6, dst, src, len
lea lenq, [lend*4]
add srcq, lenq
add dstq, lenq
neg lenq
mova m4, [pf_s32_scale]
mova m5, [pf_s32_clip]
.loop:
mulps m0, m4, [srcq+lenq ]
mulps m1, m4, [srcq+lenq+1*mmsize]
mulps m2, m4, [srcq+lenq+2*mmsize]
mulps m3, m4, [srcq+lenq+3*mmsize]
minps m0, m0, m5
minps m1, m1, m5
minps m2, m2, m5
minps m3, m3, m5
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
mova [dstq+lenq ], m0
mova [dstq+lenq+1*mmsize], m1
mova [dstq+lenq+2*mmsize], m2
mova [dstq+lenq+3*mmsize], m3
add lenq, mmsize*4
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_FLT_TO_S32
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
CONV_FLT_TO_S32
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16p_to_s16_2ch(int16_t *dst, int16_t *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16P_TO_S16_2CH 0
cglobal conv_s16p_to_s16_2ch, 3,4,5, dst, src0, len, src1
mov src1q, [src0q+gprsize]
mov src0q, [src0q ]
lea lenq, [2*lend]
add src0q, lenq
add src1q, lenq
lea dstq, [dstq+2*lenq]
neg lenq
.loop:
mova m0, [src0q+lenq ]
mova m1, [src1q+lenq ]
mova m2, [src0q+lenq+mmsize]
mova m3, [src1q+lenq+mmsize]
SBUTTERFLY2 wd, 0, 1, 4
SBUTTERFLY2 wd, 2, 3, 4
mova [dstq+2*lenq+0*mmsize], m0
mova [dstq+2*lenq+1*mmsize], m1
mova [dstq+2*lenq+2*mmsize], m2
mova [dstq+2*lenq+3*mmsize], m3
add lenq, 2*mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16P_TO_S16_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16P_TO_S16_2CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16p_to_s16_6ch(int16_t *dst, int16_t *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
;------------------------------------------------------------------------------
; NOTE: In the 6-channel functions, len could be used as an index on x86-64
; instead of just a counter, which would avoid incrementing the
; pointers, but the extra complexity and amount of code is not worth
; the small gain. On x86-32 there are not enough registers to use len
; as an index without keeping two of the pointers on the stack and
; loading them in each iteration.
;------------------------------------------------------------------------------
 
%macro CONV_S16P_TO_S16_6CH 0
%if ARCH_X86_64
cglobal conv_s16p_to_s16_6ch, 3,8,7, dst, src0, len, src1, src2, src3, src4, src5
%else
cglobal conv_s16p_to_s16_6ch, 2,7,7, dst, src0, src1, src2, src3, src4, src5
%define lend dword r2m
%endif
mov src1q, [src0q+1*gprsize]
mov src2q, [src0q+2*gprsize]
mov src3q, [src0q+3*gprsize]
mov src4q, [src0q+4*gprsize]
mov src5q, [src0q+5*gprsize]
mov src0q, [src0q]
sub src1q, src0q
sub src2q, src0q
sub src3q, src0q
sub src4q, src0q
sub src5q, src0q
.loop:
%if cpuflag(sse2slow)
movq m0, [src0q ] ; m0 = 0, 6, 12, 18, x, x, x, x
movq m1, [src0q+src1q] ; m1 = 1, 7, 13, 19, x, x, x, x
movq m2, [src0q+src2q] ; m2 = 2, 8, 14, 20, x, x, x, x
movq m3, [src0q+src3q] ; m3 = 3, 9, 15, 21, x, x, x, x
movq m4, [src0q+src4q] ; m4 = 4, 10, 16, 22, x, x, x, x
movq m5, [src0q+src5q] ; m5 = 5, 11, 17, 23, x, x, x, x
; unpack words:
punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
punpcklwd m2, m3 ; m2 = 4, 5, 10, 11, 16, 17, 22, 23
punpcklwd m4, m5 ; m4 = 2, 3, 8, 9, 14, 15, 20, 21
; blend dwords
shufps m1, m0, m2, q2020 ; m1 = 0, 1, 12, 13, 2, 3, 14, 15
shufps m0, m4, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17
shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
; shuffle dwords
pshufd m0, m0, q1302 ; m0 = 4, 5, 6, 7, 16, 17, 18, 19
pshufd m1, m1, q3120 ; m1 = 0, 1, 2, 3, 12, 13, 14, 15
pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23
movq [dstq+0*mmsize/2], m1
movq [dstq+1*mmsize/2], m0
movq [dstq+2*mmsize/2], m2
movhps [dstq+3*mmsize/2], m1
movhps [dstq+4*mmsize/2], m0
movhps [dstq+5*mmsize/2], m2
add src0q, mmsize/2
add dstq, mmsize*3
sub lend, mmsize/4
%else
mova m0, [src0q ] ; m0 = 0, 6, 12, 18, 24, 30, 36, 42
mova m1, [src0q+src1q] ; m1 = 1, 7, 13, 19, 25, 31, 37, 43
mova m2, [src0q+src2q] ; m2 = 2, 8, 14, 20, 26, 32, 38, 44
mova m3, [src0q+src3q] ; m3 = 3, 9, 15, 21, 27, 33, 39, 45
mova m4, [src0q+src4q] ; m4 = 4, 10, 16, 22, 28, 34, 40, 46
mova m5, [src0q+src5q] ; m5 = 5, 11, 17, 23, 29, 35, 41, 47
; unpack words:
SBUTTERFLY2 wd, 0, 1, 6 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
; m1 = 24, 25, 30, 31, 36, 37, 42, 43
SBUTTERFLY2 wd, 2, 3, 6 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21
; m3 = 26, 27, 32, 33, 38, 39, 44, 45
SBUTTERFLY2 wd, 4, 5, 6 ; m4 = 4, 5, 10, 11, 16, 17, 22, 23
; m5 = 28, 29, 34, 35, 40, 41, 46, 47
; blend dwords
shufps m6, m0, m2, q2020 ; m6 = 0, 1, 12, 13, 2, 3, 14, 15
shufps m0, m4, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17
shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
SWAP 4,6 ; m4 = 0, 1, 12, 13, 2, 3, 14, 15
shufps m6, m1, m3, q2020 ; m6 = 24, 25, 36, 37, 26, 27, 38, 39
shufps m1, m5, q2031 ; m1 = 30, 31, 42, 43, 28, 29, 40, 41
shufps m3, m5, q3131 ; m3 = 32, 33, 44, 45, 34, 35, 46, 47
SWAP 5,6 ; m5 = 24, 25, 36, 37, 26, 27, 38, 39
; shuffle dwords
pshufd m0, m0, q1302 ; m0 = 4, 5, 6, 7, 16, 17, 18, 19
pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23
pshufd m4, m4, q3120 ; m4 = 0, 1, 2, 3, 12, 13, 14, 15
pshufd m1, m1, q1302 ; m1 = 28, 29, 30, 31, 40, 41, 42, 43
pshufd m3, m3, q3120 ; m3 = 32, 33, 34, 35, 44, 45, 46, 47
pshufd m5, m5, q3120 ; m5 = 24, 25, 26, 27, 36, 37, 38, 39
; shuffle qwords
punpcklqdq m6, m4, m0 ; m6 = 0, 1, 2, 3, 4, 5, 6, 7
punpckhqdq m0, m2 ; m0 = 16, 17, 18, 19, 20, 21, 22, 23
shufps m2, m4, q3210 ; m2 = 8, 9, 10, 11, 12, 13, 14, 15
SWAP 4,6 ; m4 = 0, 1, 2, 3, 4, 5, 6, 7
punpcklqdq m6, m5, m1 ; m6 = 24, 25, 26, 27, 28, 29, 30, 31
punpckhqdq m1, m3 ; m1 = 40, 41, 42, 43, 44, 45, 46, 47
shufps m3, m5, q3210 ; m3 = 32, 33, 34, 35, 36, 37, 38, 39
SWAP 5,6 ; m5 = 24, 25, 26, 27, 28, 29, 30, 31
mova [dstq+0*mmsize], m4
mova [dstq+1*mmsize], m2
mova [dstq+2*mmsize], m0
mova [dstq+3*mmsize], m5
mova [dstq+4*mmsize], m3
mova [dstq+5*mmsize], m1
add src0q, mmsize
add dstq, mmsize*6
sub lend, mmsize/2
%endif
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16P_TO_S16_6CH
INIT_XMM sse2slow
CONV_S16P_TO_S16_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16P_TO_S16_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16p_to_flt_2ch(float *dst, int16_t *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16P_TO_FLT_2CH 0
cglobal conv_s16p_to_flt_2ch, 3,4,6, dst, src0, len, src1
lea lenq, [2*lend]
mov src1q, [src0q+gprsize]
mov src0q, [src0q ]
lea dstq, [dstq+4*lenq]
add src0q, lenq
add src1q, lenq
neg lenq
mova m5, [pf_s32_inv_scale]
.loop:
mova m2, [src0q+lenq] ; m2 = 0, 2, 4, 6, 8, 10, 12, 14
mova m4, [src1q+lenq] ; m4 = 1, 3, 5, 7, 9, 11, 13, 15
SBUTTERFLY2 wd, 2, 4, 3 ; m2 = 0, 1, 2, 3, 4, 5, 6, 7
; m4 = 8, 9, 10, 11, 12, 13, 14, 15
pxor m3, m3
punpcklwd m0, m3, m2 ; m0 = 0, 1, 2, 3
punpckhwd m1, m3, m2 ; m1 = 4, 5, 6, 7
punpcklwd m2, m3, m4 ; m2 = 8, 9, 10, 11
punpckhwd m3, m4 ; m3 = 12, 13, 14, 15
cvtdq2ps m0, m0
cvtdq2ps m1, m1
cvtdq2ps m2, m2
cvtdq2ps m3, m3
mulps m0, m5
mulps m1, m5
mulps m2, m5
mulps m3, m5
mova [dstq+4*lenq ], m0
mova [dstq+4*lenq+ mmsize], m1
mova [dstq+4*lenq+2*mmsize], m2
mova [dstq+4*lenq+3*mmsize], m3
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16P_TO_FLT_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16P_TO_FLT_2CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16p_to_flt_6ch(float *dst, int16_t *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16P_TO_FLT_6CH 0
%if ARCH_X86_64
cglobal conv_s16p_to_flt_6ch, 3,8,8, dst, src, len, src1, src2, src3, src4, src5
%else
cglobal conv_s16p_to_flt_6ch, 2,7,8, dst, src, src1, src2, src3, src4, src5
%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]
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
sub src4q, srcq
sub src5q, srcq
mova m7, [pf_s32_inv_scale]
%if cpuflag(ssse3)
%define unpack_even m6
mova m6, [pb_shuf_unpack_even]
%if ARCH_X86_64
%define unpack_odd m8
mova m8, [pb_shuf_unpack_odd]
%else
%define unpack_odd [pb_shuf_unpack_odd]
%endif
%endif
.loop:
movq m0, [srcq ] ; m0 = 0, 6, 12, 18, x, x, x, x
movq m1, [srcq+src1q] ; m1 = 1, 7, 13, 19, x, x, x, x
movq m2, [srcq+src2q] ; m2 = 2, 8, 14, 20, x, x, x, x
movq m3, [srcq+src3q] ; m3 = 3, 9, 15, 21, x, x, x, x
movq m4, [srcq+src4q] ; m4 = 4, 10, 16, 22, x, x, x, x
movq m5, [srcq+src5q] ; m5 = 5, 11, 17, 23, x, x, x, x
; unpack words:
punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
punpcklwd m2, m3 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21
punpcklwd m4, m5 ; m4 = 4, 5, 10, 11, 16, 17, 22, 23
; blend dwords
shufps m1, m4, m0, q3120 ; m1 = 4, 5, 16, 17, 6, 7, 18, 19
shufps m0, m2, q2020 ; m0 = 0, 1, 12, 13, 2, 3, 14, 15
shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
%if cpuflag(ssse3)
pshufb m3, m0, unpack_odd ; m3 = 12, 13, 14, 15
pshufb m0, unpack_even ; m0 = 0, 1, 2, 3
pshufb m4, m1, unpack_odd ; m4 = 16, 17, 18, 19
pshufb m1, unpack_even ; m1 = 4, 5, 6, 7
pshufb m5, m2, unpack_odd ; m5 = 20, 21, 22, 23
pshufb m2, unpack_even ; m2 = 8, 9, 10, 11
%else
; shuffle dwords
pshufd m0, m0, q3120 ; m0 = 0, 1, 2, 3, 12, 13, 14, 15
pshufd m1, m1, q3120 ; m1 = 4, 5, 6, 7, 16, 17, 18, 19
pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23
pxor m6, m6 ; convert s16 in m0-m2 to s32 in m0-m5
punpcklwd m3, m6, m0 ; m3 = 0, 1, 2, 3
punpckhwd m4, m6, m0 ; m4 = 12, 13, 14, 15
punpcklwd m0, m6, m1 ; m0 = 4, 5, 6, 7
punpckhwd m5, m6, m1 ; m5 = 16, 17, 18, 19
punpcklwd m1, m6, m2 ; m1 = 8, 9, 10, 11
punpckhwd m6, m2 ; m6 = 20, 21, 22, 23
SWAP 6,2,1,0,3,4,5 ; swap registers 3,0,1,4,5,6 to 0,1,2,3,4,5
%endif
cvtdq2ps m0, m0 ; convert s32 to float
cvtdq2ps m1, m1
cvtdq2ps m2, m2
cvtdq2ps m3, m3
cvtdq2ps m4, m4
cvtdq2ps m5, m5
mulps m0, m7 ; scale float from s32 range to [-1.0,1.0]
mulps m1, m7
mulps m2, m7
mulps m3, m7
mulps m4, m7
mulps m5, m7
mova [dstq ], m0
mova [dstq+ mmsize], m1
mova [dstq+2*mmsize], m2
mova [dstq+3*mmsize], m3
mova [dstq+4*mmsize], m4
mova [dstq+5*mmsize], m5
add srcq, mmsize/2
add dstq, mmsize*6
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16P_TO_FLT_6CH
INIT_XMM ssse3
CONV_S16P_TO_FLT_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16P_TO_FLT_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_fltp_to_s16_2ch(int16_t *dst, float *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLTP_TO_S16_2CH 0
cglobal conv_fltp_to_s16_2ch, 3,4,3, dst, src0, len, src1
lea lenq, [4*lend]
mov src1q, [src0q+gprsize]
mov src0q, [src0q ]
add dstq, lenq
add src0q, lenq
add src1q, lenq
neg lenq
mova m2, [pf_s16_scale]
%if cpuflag(ssse3)
mova m3, [pb_interleave_words]
%endif
.loop:
mulps m0, m2, [src0q+lenq] ; m0 = 0, 2, 4, 6
mulps m1, m2, [src1q+lenq] ; m1 = 1, 3, 5, 7
cvtps2dq m0, m0
cvtps2dq m1, m1
%if cpuflag(ssse3)
packssdw m0, m1 ; m0 = 0, 2, 4, 6, 1, 3, 5, 7
pshufb m0, m3 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
%else
packssdw m0, m0 ; m0 = 0, 2, 4, 6, x, x, x, x
packssdw m1, m1 ; m1 = 1, 3, 5, 7, x, x, x, x
punpcklwd m0, m1 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
%endif
mova [dstq+lenq], m0
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_FLTP_TO_S16_2CH
INIT_XMM ssse3
CONV_FLTP_TO_S16_2CH
 
;------------------------------------------------------------------------------
; void ff_conv_fltp_to_s16_6ch(int16_t *dst, float *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLTP_TO_S16_6CH 0
%if ARCH_X86_64
cglobal conv_fltp_to_s16_6ch, 3,8,7, dst, src, len, src1, src2, src3, src4, src5
%else
cglobal conv_fltp_to_s16_6ch, 2,7,7, dst, src, src1, src2, src3, src4, src5
%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]
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
sub src4q, srcq
sub src5q, srcq
movaps xmm6, [pf_s16_scale]
.loop:
%if cpuflag(sse2)
mulps m0, m6, [srcq ]
mulps m1, m6, [srcq+src1q]
mulps m2, m6, [srcq+src2q]
mulps m3, m6, [srcq+src3q]
mulps m4, m6, [srcq+src4q]
mulps m5, m6, [srcq+src5q]
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
cvtps2dq m4, m4
cvtps2dq m5, m5
packssdw m0, m3 ; m0 = 0, 6, 12, 18, 3, 9, 15, 21
packssdw m1, m4 ; m1 = 1, 7, 13, 19, 4, 10, 16, 22
packssdw m2, m5 ; m2 = 2, 8, 14, 20, 5, 11, 17, 23
; unpack words:
movhlps m3, m0 ; m3 = 3, 9, 15, 21, x, x, x, x
punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
punpckhwd m1, m2 ; m1 = 4, 5, 10, 11, 16, 17, 22, 23
punpcklwd m2, m3 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21
; blend dwords:
shufps m3, m0, m2, q2020 ; m3 = 0, 1, 12, 13, 2, 3, 14, 15
shufps m0, m1, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17
shufps m2, m1, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
; shuffle dwords:
shufps m1, m2, m3, q3120 ; m1 = 8, 9, 10, 11, 12, 13, 14, 15
shufps m3, m0, q0220 ; m3 = 0, 1, 2, 3, 4, 5, 6, 7
shufps m0, m2, q3113 ; m0 = 16, 17, 18, 19, 20, 21, 22, 23
mova [dstq+0*mmsize], m3
mova [dstq+1*mmsize], m1
mova [dstq+2*mmsize], m0
%else ; sse
movlps xmm0, [srcq ]
movlps xmm1, [srcq+src1q]
movlps xmm2, [srcq+src2q]
movlps xmm3, [srcq+src3q]
movlps xmm4, [srcq+src4q]
movlps xmm5, [srcq+src5q]
mulps xmm0, xmm6
mulps xmm1, xmm6
mulps xmm2, xmm6
mulps xmm3, xmm6
mulps xmm4, xmm6
mulps xmm5, xmm6
cvtps2pi mm0, xmm0
cvtps2pi mm1, xmm1
cvtps2pi mm2, xmm2
cvtps2pi mm3, xmm3
cvtps2pi mm4, xmm4
cvtps2pi mm5, xmm5
packssdw mm0, mm3 ; m0 = 0, 6, 3, 9
packssdw mm1, mm4 ; m1 = 1, 7, 4, 10
packssdw mm2, mm5 ; m2 = 2, 8, 5, 11
; unpack words
pshufw mm3, mm0, q1032 ; m3 = 3, 9, 0, 6
punpcklwd mm0, mm1 ; m0 = 0, 1, 6, 7
punpckhwd mm1, mm2 ; m1 = 4, 5, 10, 11
punpcklwd mm2, mm3 ; m2 = 2, 3, 8, 9
; unpack dwords
pshufw mm3, mm0, q1032 ; m3 = 6, 7, 0, 1
punpckldq mm0, mm2 ; m0 = 0, 1, 2, 3 (final)
punpckhdq mm2, mm1 ; m2 = 8, 9, 10, 11 (final)
punpckldq mm1, mm3 ; m1 = 4, 5, 6, 7 (final)
mova [dstq+0*mmsize], mm0
mova [dstq+1*mmsize], mm1
mova [dstq+2*mmsize], mm2
%endif
add srcq, mmsize
add dstq, mmsize*3
sub lend, mmsize/4
jg .loop
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
 
INIT_MMX sse
CONV_FLTP_TO_S16_6CH
INIT_XMM sse2
CONV_FLTP_TO_S16_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLTP_TO_S16_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_fltp_to_flt_2ch(float *dst, float *const *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLTP_TO_FLT_2CH 0
cglobal conv_fltp_to_flt_2ch, 3,4,5, dst, src0, len, src1
mov src1q, [src0q+gprsize]
mov src0q, [src0q]
lea lenq, [4*lend]
add src0q, lenq
add src1q, lenq
lea dstq, [dstq+2*lenq]
neg lenq
.loop:
mova m0, [src0q+lenq ]
mova m1, [src1q+lenq ]
mova m2, [src0q+lenq+mmsize]
mova m3, [src1q+lenq+mmsize]
SBUTTERFLYPS 0, 1, 4
SBUTTERFLYPS 2, 3, 4
mova [dstq+2*lenq+0*mmsize], m0
mova [dstq+2*lenq+1*mmsize], m1
mova [dstq+2*lenq+2*mmsize], m2
mova [dstq+2*lenq+3*mmsize], m3
add lenq, 2*mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse
CONV_FLTP_TO_FLT_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLTP_TO_FLT_2CH
%endif
 
;-----------------------------------------------------------------------------
; void ff_conv_fltp_to_flt_6ch(float *dst, float *const *src, int len,
; int channels);
;-----------------------------------------------------------------------------
 
%macro CONV_FLTP_TO_FLT_6CH 0
cglobal conv_fltp_to_flt_6ch, 2,8,7, 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]
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
sub src4q, srcq
sub src5q, srcq
.loop:
mova m0, [srcq ]
mova m1, [srcq+src1q]
mova m2, [srcq+src2q]
mova m3, [srcq+src3q]
mova m4, [srcq+src4q]
mova m5, [srcq+src5q]
%if cpuflag(sse4)
SBUTTERFLYPS 0, 1, 6
SBUTTERFLYPS 2, 3, 6
SBUTTERFLYPS 4, 5, 6
 
blendps m6, m4, m0, 1100b
movlhps m0, m2
movhlps m4, m2
blendps m2, m5, m1, 1100b
movlhps m1, m3
movhlps m5, m3
 
movaps [dstq ], m0
movaps [dstq+16], m6
movaps [dstq+32], m4
movaps [dstq+48], m1
movaps [dstq+64], m2
movaps [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
 
INIT_MMX mmx
CONV_FLTP_TO_FLT_6CH
INIT_XMM sse4
CONV_FLTP_TO_FLT_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLTP_TO_FLT_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16_to_s16p_2ch(int16_t *const *dst, int16_t *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16_TO_S16P_2CH 0
cglobal conv_s16_to_s16p_2ch, 3,4,4, dst0, src, len, dst1
lea lenq, [2*lend]
mov dst1q, [dst0q+gprsize]
mov dst0q, [dst0q ]
lea srcq, [srcq+2*lenq]
add dst0q, lenq
add dst1q, lenq
neg lenq
%if cpuflag(ssse3)
mova m3, [pb_deinterleave_words]
%endif
.loop:
mova m0, [srcq+2*lenq ] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
mova m1, [srcq+2*lenq+mmsize] ; m1 = 8, 9, 10, 11, 12, 13, 14, 15
%if cpuflag(ssse3)
pshufb m0, m3 ; m0 = 0, 2, 4, 6, 1, 3, 5, 7
pshufb m1, m3 ; m1 = 8, 10, 12, 14, 9, 11, 13, 15
SBUTTERFLY2 qdq, 0, 1, 2 ; m0 = 0, 2, 4, 6, 8, 10, 12, 14
; m1 = 1, 3, 5, 7, 9, 11, 13, 15
%else ; sse2
pshuflw m0, m0, q3120 ; m0 = 0, 2, 1, 3, 4, 5, 6, 7
pshufhw m0, m0, q3120 ; m0 = 0, 2, 1, 3, 4, 6, 5, 7
pshuflw m1, m1, q3120 ; m1 = 8, 10, 9, 11, 12, 13, 14, 15
pshufhw m1, m1, q3120 ; m1 = 8, 10, 9, 11, 12, 14, 13, 15
DEINT2_PS 0, 1, 2 ; m0 = 0, 2, 4, 6, 8, 10, 12, 14
; m1 = 1, 3, 5, 7, 9, 11, 13, 15
%endif
mova [dst0q+lenq], m0
mova [dst1q+lenq], m1
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16_TO_S16P_2CH
INIT_XMM ssse3
CONV_S16_TO_S16P_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16_TO_S16P_2CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16_to_s16p_6ch(int16_t *const *dst, int16_t *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16_TO_S16P_6CH 0
%if ARCH_X86_64
cglobal conv_s16_to_s16p_6ch, 3,8,5, dst, src, len, dst1, dst2, dst3, dst4, dst5
%else
cglobal conv_s16_to_s16p_6ch, 2,7,5, dst, src, dst1, dst2, dst3, dst4, dst5
%define lend dword r2m
%endif
mov dst1q, [dstq+ gprsize]
mov dst2q, [dstq+2*gprsize]
mov dst3q, [dstq+3*gprsize]
mov dst4q, [dstq+4*gprsize]
mov dst5q, [dstq+5*gprsize]
mov dstq, [dstq ]
sub dst1q, dstq
sub dst2q, dstq
sub dst3q, dstq
sub dst4q, dstq
sub dst5q, dstq
.loop:
mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
mova m3, [srcq+1*mmsize] ; m3 = 8, 9, 10, 11, 12, 13, 14, 15
mova m2, [srcq+2*mmsize] ; m2 = 16, 17, 18, 19, 20, 21, 22, 23
PALIGNR m1, m3, m0, 12, m4 ; m1 = 6, 7, 8, 9, 10, 11, x, x
shufps m3, m2, q1032 ; m3 = 12, 13, 14, 15, 16, 17, 18, 19
psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x
SBUTTERFLY2 wd, 0, 1, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9
; m1 = 4, 10, 5, 11, x, x, x, x
SBUTTERFLY2 wd, 3, 2, 4 ; m3 = 12, 18, 13, 19, 14, 20, 15, 21
; m2 = 16, 22, 17, 23, x, x, x, x
SBUTTERFLY2 dq, 0, 3, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19
; m3 = 2, 8, 14, 20, 3, 9, 15, 21
punpckldq m1, m2 ; m1 = 4, 10, 16, 22, 5, 11, 17, 23
movq [dstq ], m0
movhps [dstq+dst1q], m0
movq [dstq+dst2q], m3
movhps [dstq+dst3q], m3
movq [dstq+dst4q], m1
movhps [dstq+dst5q], m1
add srcq, mmsize*3
add dstq, mmsize/2
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16_TO_S16P_6CH
INIT_XMM ssse3
CONV_S16_TO_S16P_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16_TO_S16P_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16_to_fltp_2ch(float *const *dst, int16_t *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16_TO_FLTP_2CH 0
cglobal conv_s16_to_fltp_2ch, 3,4,5, dst0, src, len, dst1
lea lenq, [4*lend]
mov dst1q, [dst0q+gprsize]
mov dst0q, [dst0q ]
add srcq, lenq
add dst0q, lenq
add dst1q, lenq
neg lenq
mova m3, [pf_s32_inv_scale]
mova m4, [pw_zero_even]
.loop:
mova m1, [srcq+lenq]
pslld m0, m1, 16
pand m1, m4
cvtdq2ps m0, m0
cvtdq2ps m1, m1
mulps m0, m0, m3
mulps m1, m1, m3
mova [dst0q+lenq], m0
mova [dst1q+lenq], m1
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16_TO_FLTP_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16_TO_FLTP_2CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_s16_to_fltp_6ch(float *const *dst, int16_t *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_S16_TO_FLTP_6CH 0
%if ARCH_X86_64
cglobal conv_s16_to_fltp_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5
%else
cglobal conv_s16_to_fltp_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5
%define lend dword r2m
%endif
mov dst1q, [dstq+ gprsize]
mov dst2q, [dstq+2*gprsize]
mov dst3q, [dstq+3*gprsize]
mov dst4q, [dstq+4*gprsize]
mov dst5q, [dstq+5*gprsize]
mov dstq, [dstq ]
sub dst1q, dstq
sub dst2q, dstq
sub dst3q, dstq
sub dst4q, dstq
sub dst5q, dstq
mova m6, [pf_s16_inv_scale]
.loop:
mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
mova m3, [srcq+1*mmsize] ; m3 = 8, 9, 10, 11, 12, 13, 14, 15
mova m2, [srcq+2*mmsize] ; m2 = 16, 17, 18, 19, 20, 21, 22, 23
PALIGNR m1, m3, m0, 12, m4 ; m1 = 6, 7, 8, 9, 10, 11, x, x
shufps m3, m2, q1032 ; m3 = 12, 13, 14, 15, 16, 17, 18, 19
psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x
SBUTTERFLY2 wd, 0, 1, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9
; m1 = 4, 10, 5, 11, x, x, x, x
SBUTTERFLY2 wd, 3, 2, 4 ; m3 = 12, 18, 13, 19, 14, 20, 15, 21
; m2 = 16, 22, 17, 23, x, x, x, x
SBUTTERFLY2 dq, 0, 3, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19
; m3 = 2, 8, 14, 20, 3, 9, 15, 21
punpckldq m1, m2 ; m1 = 4, 10, 16, 22, 5, 11, 17, 23
S16_TO_S32_SX 0, 2 ; m0 = 0, 6, 12, 18
; m2 = 1, 7, 13, 19
S16_TO_S32_SX 3, 4 ; m3 = 2, 8, 14, 20
; m4 = 3, 9, 15, 21
S16_TO_S32_SX 1, 5 ; m1 = 4, 10, 16, 22
; m5 = 5, 11, 17, 23
SWAP 1,2,3,4
cvtdq2ps m0, m0
cvtdq2ps m1, m1
cvtdq2ps m2, m2
cvtdq2ps m3, m3
cvtdq2ps m4, m4
cvtdq2ps m5, m5
mulps m0, m6
mulps m1, m6
mulps m2, m6
mulps m3, m6
mulps m4, m6
mulps m5, m6
mova [dstq ], m0
mova [dstq+dst1q], m1
mova [dstq+dst2q], m2
mova [dstq+dst3q], m3
mova [dstq+dst4q], m4
mova [dstq+dst5q], m5
add srcq, mmsize*3
add dstq, mmsize
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_S16_TO_FLTP_6CH
INIT_XMM ssse3
CONV_S16_TO_FLTP_6CH
INIT_XMM sse4
CONV_S16_TO_FLTP_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_S16_TO_FLTP_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_flt_to_s16p_2ch(int16_t *const *dst, float *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLT_TO_S16P_2CH 0
cglobal conv_flt_to_s16p_2ch, 3,4,6, dst0, src, len, dst1
lea lenq, [2*lend]
mov dst1q, [dst0q+gprsize]
mov dst0q, [dst0q ]
lea srcq, [srcq+4*lenq]
add dst0q, lenq
add dst1q, lenq
neg lenq
mova m5, [pf_s16_scale]
.loop:
mova m0, [srcq+4*lenq ]
mova m1, [srcq+4*lenq+ mmsize]
mova m2, [srcq+4*lenq+2*mmsize]
mova m3, [srcq+4*lenq+3*mmsize]
DEINT2_PS 0, 1, 4
DEINT2_PS 2, 3, 4
mulps m0, m0, m5
mulps m1, m1, m5
mulps m2, m2, m5
mulps m3, m3, m5
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
packssdw m0, m2
packssdw m1, m3
mova [dst0q+lenq], m0
mova [dst1q+lenq], m1
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_FLT_TO_S16P_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLT_TO_S16P_2CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_flt_to_s16p_6ch(int16_t *const *dst, float *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLT_TO_S16P_6CH 0
%if ARCH_X86_64
cglobal conv_flt_to_s16p_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5
%else
cglobal conv_flt_to_s16p_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5
%define lend dword r2m
%endif
mov dst1q, [dstq+ gprsize]
mov dst2q, [dstq+2*gprsize]
mov dst3q, [dstq+3*gprsize]
mov dst4q, [dstq+4*gprsize]
mov dst5q, [dstq+5*gprsize]
mov dstq, [dstq ]
sub dst1q, dstq
sub dst2q, dstq
sub dst3q, dstq
sub dst4q, dstq
sub dst5q, dstq
mova m6, [pf_s16_scale]
.loop:
mulps m0, m6, [srcq+0*mmsize]
mulps m3, m6, [srcq+1*mmsize]
mulps m1, m6, [srcq+2*mmsize]
mulps m4, m6, [srcq+3*mmsize]
mulps m2, m6, [srcq+4*mmsize]
mulps m5, m6, [srcq+5*mmsize]
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
cvtps2dq m4, m4
cvtps2dq m5, m5
packssdw m0, m3 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
packssdw m1, m4 ; m1 = 8, 9, 10, 11, 12, 13, 14, 15
packssdw m2, m5 ; m2 = 16, 17, 18, 19, 20, 21, 22, 23
PALIGNR m3, m1, m0, 12, m4 ; m3 = 6, 7, 8, 9, 10, 11, x, x
shufps m1, m2, q1032 ; m1 = 12, 13, 14, 15, 16, 17, 18, 19
psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x
SBUTTERFLY2 wd, 0, 3, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9
; m3 = 4, 10, 5, 11, x, x, x, x
SBUTTERFLY2 wd, 1, 2, 4 ; m1 = 12, 18, 13, 19, 14, 20, 15, 21
; m2 = 16, 22, 17, 23, x, x, x, x
SBUTTERFLY2 dq, 0, 1, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19
; m1 = 2, 8, 14, 20, 3, 9, 15, 21
punpckldq m3, m2 ; m3 = 4, 10, 16, 22, 5, 11, 17, 23
movq [dstq ], m0
movhps [dstq+dst1q], m0
movq [dstq+dst2q], m1
movhps [dstq+dst3q], m1
movq [dstq+dst4q], m3
movhps [dstq+dst5q], m3
add srcq, mmsize*6
add dstq, mmsize/2
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_FLT_TO_S16P_6CH
INIT_XMM ssse3
CONV_FLT_TO_S16P_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLT_TO_S16P_6CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_flt_to_fltp_2ch(float *const *dst, float *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLT_TO_FLTP_2CH 0
cglobal conv_flt_to_fltp_2ch, 3,4,3, dst0, src, len, dst1
lea lenq, [4*lend]
mov dst1q, [dst0q+gprsize]
mov dst0q, [dst0q ]
lea srcq, [srcq+2*lenq]
add dst0q, lenq
add dst1q, lenq
neg lenq
.loop:
mova m0, [srcq+2*lenq ]
mova m1, [srcq+2*lenq+mmsize]
DEINT2_PS 0, 1, 2
mova [dst0q+lenq], m0
mova [dst1q+lenq], m1
add lenq, mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse
CONV_FLT_TO_FLTP_2CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLT_TO_FLTP_2CH
%endif
 
;------------------------------------------------------------------------------
; void ff_conv_flt_to_fltp_6ch(float *const *dst, float *src, int len,
; int channels);
;------------------------------------------------------------------------------
 
%macro CONV_FLT_TO_FLTP_6CH 0
%if ARCH_X86_64
cglobal conv_flt_to_fltp_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5
%else
cglobal conv_flt_to_fltp_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5
%define lend dword r2m
%endif
mov dst1q, [dstq+ gprsize]
mov dst2q, [dstq+2*gprsize]
mov dst3q, [dstq+3*gprsize]
mov dst4q, [dstq+4*gprsize]
mov dst5q, [dstq+5*gprsize]
mov dstq, [dstq ]
sub dst1q, dstq
sub dst2q, dstq
sub dst3q, dstq
sub dst4q, dstq
sub dst5q, dstq
.loop:
mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3
mova m1, [srcq+1*mmsize] ; m1 = 4, 5, 6, 7
mova m2, [srcq+2*mmsize] ; m2 = 8, 9, 10, 11
mova m3, [srcq+3*mmsize] ; m3 = 12, 13, 14, 15
mova m4, [srcq+4*mmsize] ; m4 = 16, 17, 18, 19
mova m5, [srcq+5*mmsize] ; m5 = 20, 21, 22, 23
 
SBUTTERFLY2 dq, 0, 3, 6 ; m0 = 0, 12, 1, 13
; m3 = 2, 14, 3, 15
SBUTTERFLY2 dq, 1, 4, 6 ; m1 = 4, 16, 5, 17
; m4 = 6, 18, 7, 19
SBUTTERFLY2 dq, 2, 5, 6 ; m2 = 8, 20, 9, 21
; m5 = 10, 22, 11, 23
SBUTTERFLY2 dq, 0, 4, 6 ; m0 = 0, 6, 12, 18
; m4 = 1, 7, 13, 19
SBUTTERFLY2 dq, 3, 2, 6 ; m3 = 2, 8, 14, 20
; m2 = 3, 9, 15, 21
SBUTTERFLY2 dq, 1, 5, 6 ; m1 = 4, 10, 16, 22
; m5 = 5, 11, 17, 23
mova [dstq ], m0
mova [dstq+dst1q], m4
mova [dstq+dst2q], m3
mova [dstq+dst3q], m2
mova [dstq+dst4q], m1
mova [dstq+dst5q], m5
add srcq, mmsize*6
add dstq, mmsize
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
CONV_FLT_TO_FLTP_6CH
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
CONV_FLT_TO_FLTP_6CH
%endif
/contrib/sdk/sources/ffmpeg/libavresample/x86/audio_convert_init.c
0,0 → 1,263
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 "config.h"
#include "libavutil/cpu.h"
#include "libavutil/x86/cpu.h"
#include "libavresample/audio_convert.h"
 
/* flat conversions */
 
void ff_conv_s16_to_s32_sse2(int16_t *dst, const int32_t *src, int len);
 
void ff_conv_s16_to_flt_sse2(float *dst, const int16_t *src, int len);
void ff_conv_s16_to_flt_sse4(float *dst, const int16_t *src, int len);
 
void ff_conv_s32_to_s16_mmx (int16_t *dst, const int32_t *src, int len);
void ff_conv_s32_to_s16_sse2(int16_t *dst, const int32_t *src, int len);
 
void ff_conv_s32_to_flt_sse2(float *dst, const int32_t *src, int len);
void ff_conv_s32_to_flt_avx (float *dst, const int32_t *src, int len);
 
void ff_conv_flt_to_s16_sse2(int16_t *dst, const float *src, int len);
 
void ff_conv_flt_to_s32_sse2(int32_t *dst, const float *src, int len);
void ff_conv_flt_to_s32_avx (int32_t *dst, const float *src, int len);
 
/* interleave conversions */
 
void ff_conv_s16p_to_s16_2ch_sse2(int16_t *dst, int16_t *const *src,
int len, int channels);
void ff_conv_s16p_to_s16_2ch_avx (int16_t *dst, int16_t *const *src,
int len, int channels);
 
void ff_conv_s16p_to_s16_6ch_sse2(int16_t *dst, int16_t *const *src,
int len, int channels);
void ff_conv_s16p_to_s16_6ch_sse2slow(int16_t *dst, int16_t *const *src,
int len, int channels);
void ff_conv_s16p_to_s16_6ch_avx (int16_t *dst, int16_t *const *src,
int len, int channels);
 
void ff_conv_s16p_to_flt_2ch_sse2(float *dst, int16_t *const *src,
int len, int channels);
void ff_conv_s16p_to_flt_2ch_avx (float *dst, int16_t *const *src,
int len, int channels);
 
void ff_conv_s16p_to_flt_6ch_sse2 (float *dst, int16_t *const *src,
int len, int channels);
void ff_conv_s16p_to_flt_6ch_ssse3(float *dst, int16_t *const *src,
int len, int channels);
void ff_conv_s16p_to_flt_6ch_avx (float *dst, int16_t *const *src,
int len, int channels);
 
void ff_conv_fltp_to_s16_2ch_sse2 (int16_t *dst, float *const *src,
int len, int channels);
void ff_conv_fltp_to_s16_2ch_ssse3(int16_t *dst, float *const *src,
int len, int channels);
 
void ff_conv_fltp_to_s16_6ch_sse (int16_t *dst, float *const *src,
int len, int channels);
void ff_conv_fltp_to_s16_6ch_sse2(int16_t *dst, float *const *src,
int len, int channels);
void ff_conv_fltp_to_s16_6ch_avx (int16_t *dst, float *const *src,
int len, int channels);
 
void ff_conv_fltp_to_flt_2ch_sse(float *dst, float *const *src, int len,
int channels);
void ff_conv_fltp_to_flt_2ch_avx(float *dst, float *const *src, int len,
int channels);
 
void ff_conv_fltp_to_flt_6ch_mmx (float *dst, float *const *src, int len,
int channels);
void ff_conv_fltp_to_flt_6ch_sse4(float *dst, float *const *src, int len,
int channels);
void ff_conv_fltp_to_flt_6ch_avx (float *dst, float *const *src, int len,
int channels);
 
/* deinterleave conversions */
 
void ff_conv_s16_to_s16p_2ch_sse2(int16_t *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_s16p_2ch_ssse3(int16_t *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_s16p_2ch_avx (int16_t *const *dst, int16_t *src,
int len, int channels);
 
void ff_conv_s16_to_s16p_6ch_sse2 (int16_t *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_s16p_6ch_ssse3(int16_t *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_s16p_6ch_avx (int16_t *const *dst, int16_t *src,
int len, int channels);
 
void ff_conv_s16_to_fltp_2ch_sse2(float *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_fltp_2ch_avx (float *const *dst, int16_t *src,
int len, int channels);
 
void ff_conv_s16_to_fltp_6ch_sse2 (float *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_fltp_6ch_ssse3(float *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_fltp_6ch_sse4 (float *const *dst, int16_t *src,
int len, int channels);
void ff_conv_s16_to_fltp_6ch_avx (float *const *dst, int16_t *src,
int len, int channels);
 
void ff_conv_flt_to_s16p_2ch_sse2(int16_t *const *dst, float *src,
int len, int channels);
void ff_conv_flt_to_s16p_2ch_avx (int16_t *const *dst, float *src,
int len, int channels);
 
void ff_conv_flt_to_s16p_6ch_sse2 (int16_t *const *dst, float *src,
int len, int channels);
void ff_conv_flt_to_s16p_6ch_ssse3(int16_t *const *dst, float *src,
int len, int channels);
void ff_conv_flt_to_s16p_6ch_avx (int16_t *const *dst, float *src,
int len, int channels);
 
void ff_conv_flt_to_fltp_2ch_sse(float *const *dst, float *src, int len,
int channels);
void ff_conv_flt_to_fltp_2ch_avx(float *const *dst, float *src, int len,
int channels);
 
void ff_conv_flt_to_fltp_6ch_sse2(float *const *dst, float *src, int len,
int channels);
void ff_conv_flt_to_fltp_6ch_avx (float *const *dst, float *src, int len,
int channels);
 
av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
{
int cpu_flags = av_get_cpu_flags();
 
if (EXTERNAL_MMX(cpu_flags)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
0, 1, 8, "MMX", ff_conv_s32_to_s16_mmx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
6, 1, 4, "MMX", ff_conv_fltp_to_flt_6ch_mmx);
}
if (EXTERNAL_SSE(cpu_flags)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
6, 1, 2, "SSE", ff_conv_fltp_to_s16_6ch_sse);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
2, 16, 8, "SSE", ff_conv_fltp_to_flt_2ch_sse);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
2, 16, 4, "SSE", ff_conv_flt_to_fltp_2ch_sse);
}
if (EXTERNAL_SSE2(cpu_flags)) {
if (!(cpu_flags & AV_CPU_FLAG_SSE2SLOW)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
0, 16, 16, "SSE2", ff_conv_s32_to_s16_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
6, 16, 8, "SSE2", ff_conv_s16p_to_s16_6ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
6, 16, 4, "SSE2", ff_conv_fltp_to_s16_6ch_sse2);
} else {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
6, 1, 4, "SSE2SLOW", ff_conv_s16p_to_s16_6ch_sse2slow);
}
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16,
0, 16, 8, "SSE2", ff_conv_s16_to_s32_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
0, 16, 8, "SSE2", ff_conv_s16_to_flt_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32,
0, 16, 8, "SSE2", ff_conv_s32_to_flt_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT,
0, 16, 16, "SSE2", ff_conv_flt_to_s16_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT,
0, 16, 16, "SSE2", ff_conv_flt_to_s32_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
2, 16, 16, "SSE2", ff_conv_s16p_to_s16_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
2, 16, 8, "SSE2", ff_conv_s16p_to_flt_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
6, 16, 4, "SSE2", ff_conv_s16p_to_flt_6ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
2, 16, 4, "SSE2", ff_conv_fltp_to_s16_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
2, 16, 8, "SSE2", ff_conv_s16_to_s16p_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
6, 16, 4, "SSE2", ff_conv_s16_to_s16p_6ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
2, 16, 8, "SSE2", ff_conv_s16_to_fltp_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
6, 16, 4, "SSE2", ff_conv_s16_to_fltp_6ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
2, 16, 8, "SSE2", ff_conv_flt_to_s16p_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
6, 16, 4, "SSE2", ff_conv_flt_to_s16p_6ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
6, 16, 4, "SSE2", ff_conv_flt_to_fltp_6ch_sse2);
}
if (EXTERNAL_SSSE3(cpu_flags)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
6, 16, 4, "SSSE3", ff_conv_s16p_to_flt_6ch_ssse3);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
2, 16, 4, "SSSE3", ff_conv_fltp_to_s16_2ch_ssse3);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
2, 16, 8, "SSSE3", ff_conv_s16_to_s16p_2ch_ssse3);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
6, 16, 4, "SSSE3", ff_conv_s16_to_s16p_6ch_ssse3);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
6, 16, 4, "SSSE3", ff_conv_s16_to_fltp_6ch_ssse3);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
6, 16, 4, "SSSE3", ff_conv_flt_to_s16p_6ch_ssse3);
}
if (EXTERNAL_SSE4(cpu_flags)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
0, 16, 8, "SSE4", ff_conv_s16_to_flt_sse4);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
6, 16, 4, "SSE4", ff_conv_fltp_to_flt_6ch_sse4);
}
if (EXTERNAL_AVX(cpu_flags)) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32,
0, 32, 16, "AVX", ff_conv_s32_to_flt_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT,
0, 32, 32, "AVX", ff_conv_flt_to_s32_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
2, 16, 16, "AVX", ff_conv_s16p_to_s16_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
6, 16, 8, "AVX", ff_conv_s16p_to_s16_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
2, 16, 8, "AVX", ff_conv_s16p_to_flt_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
6, 16, 4, "AVX", ff_conv_s16p_to_flt_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
6, 16, 4, "AVX", ff_conv_fltp_to_s16_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
6, 16, 4, "AVX", ff_conv_fltp_to_flt_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
2, 16, 8, "AVX", ff_conv_s16_to_s16p_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
6, 16, 4, "AVX", ff_conv_s16_to_s16p_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
2, 16, 8, "AVX", ff_conv_s16_to_fltp_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
6, 16, 4, "AVX", ff_conv_s16_to_fltp_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
2, 16, 8, "AVX", ff_conv_flt_to_s16p_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
6, 16, 4, "AVX", ff_conv_flt_to_s16p_6ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
2, 16, 4, "AVX", ff_conv_flt_to_fltp_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
6, 16, 4, "AVX", ff_conv_flt_to_fltp_6ch_avx);
}
}
/contrib/sdk/sources/ffmpeg/libavresample/x86/audio_mix.asm
0,0 → 1,511
;******************************************************************************
;* x86 optimized channel mixing
;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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"
%include "util.asm"
 
SECTION_TEXT
 
;-----------------------------------------------------------------------------
; void ff_mix_2_to_1_fltp_flt(float **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
 
%macro MIX_2_TO_1_FLTP_FLT 0
cglobal mix_2_to_1_fltp_flt, 3,4,6, src, matrix, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq ]
sub src1q, srcq
mov matrixq, [matrixq ]
VBROADCASTSS m4, [matrixq ]
VBROADCASTSS m5, [matrixq+4]
ALIGN 16
.loop:
mulps m0, m4, [srcq ]
mulps m1, m5, [srcq+src1q ]
mulps m2, m4, [srcq+ mmsize]
mulps m3, m5, [srcq+src1q+mmsize]
addps m0, m0, m1
addps m2, m2, m3
mova [srcq ], m0
mova [srcq+mmsize], m2
add srcq, mmsize*2
sub lend, mmsize*2/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse
MIX_2_TO_1_FLTP_FLT
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
MIX_2_TO_1_FLTP_FLT
%endif
 
;-----------------------------------------------------------------------------
; void ff_mix_2_to_1_s16p_flt(int16_t **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
 
%macro MIX_2_TO_1_S16P_FLT 0
cglobal mix_2_to_1_s16p_flt, 3,4,6, src, matrix, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq]
sub src1q, srcq
mov matrixq, [matrixq ]
VBROADCASTSS m4, [matrixq ]
VBROADCASTSS m5, [matrixq+4]
ALIGN 16
.loop:
mova m0, [srcq ]
mova m2, [srcq+src1q]
S16_TO_S32_SX 0, 1
S16_TO_S32_SX 2, 3
cvtdq2ps m0, m0
cvtdq2ps m1, m1
cvtdq2ps m2, m2
cvtdq2ps m3, m3
mulps m0, m4
mulps m1, m4
mulps m2, m5
mulps m3, m5
addps m0, m2
addps m1, m3
cvtps2dq m0, m0
cvtps2dq m1, m1
packssdw m0, m1
mova [srcq], m0
add srcq, mmsize
sub lend, mmsize/2
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
MIX_2_TO_1_S16P_FLT
INIT_XMM sse4
MIX_2_TO_1_S16P_FLT
 
;-----------------------------------------------------------------------------
; void ff_mix_2_to_1_s16p_q8(int16_t **src, int16_t **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
 
INIT_XMM sse2
cglobal mix_2_to_1_s16p_q8, 3,4,6, src, matrix, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq]
sub src1q, srcq
mov matrixq, [matrixq]
movd m4, [matrixq]
movd m5, [matrixq]
SPLATW m4, m4, 0
SPLATW m5, m5, 1
pxor m0, m0
punpcklwd m4, m0
punpcklwd m5, m0
ALIGN 16
.loop:
mova m0, [srcq ]
mova m2, [srcq+src1q]
punpckhwd m1, m0, m0
punpcklwd m0, m0
punpckhwd m3, m2, m2
punpcklwd m2, m2
pmaddwd m0, m4
pmaddwd m1, m4
pmaddwd m2, m5
pmaddwd m3, m5
paddd m0, m2
paddd m1, m3
psrad m0, 8
psrad m1, 8
packssdw m0, m1
mova [srcq], m0
add srcq, mmsize
sub lend, mmsize/2
jg .loop
REP_RET
 
;-----------------------------------------------------------------------------
; void ff_mix_1_to_2_fltp_flt(float **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
 
%macro MIX_1_TO_2_FLTP_FLT 0
cglobal mix_1_to_2_fltp_flt, 3,5,4, src0, matrix0, len, src1, matrix1
mov src1q, [src0q+gprsize]
mov src0q, [src0q]
sub src1q, src0q
mov matrix1q, [matrix0q+gprsize]
mov matrix0q, [matrix0q]
VBROADCASTSS m2, [matrix0q]
VBROADCASTSS m3, [matrix1q]
ALIGN 16
.loop:
mova m0, [src0q]
mulps m1, m0, m3
mulps m0, m0, m2
mova [src0q ], m0
mova [src0q+src1q], m1
add src0q, mmsize
sub lend, mmsize/4
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse
MIX_1_TO_2_FLTP_FLT
%if HAVE_AVX_EXTERNAL
INIT_YMM avx
MIX_1_TO_2_FLTP_FLT
%endif
 
;-----------------------------------------------------------------------------
; void ff_mix_1_to_2_s16p_flt(int16_t **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
 
%macro MIX_1_TO_2_S16P_FLT 0
cglobal mix_1_to_2_s16p_flt, 3,5,6, src0, matrix0, len, src1, matrix1
mov src1q, [src0q+gprsize]
mov src0q, [src0q]
sub src1q, src0q
mov matrix1q, [matrix0q+gprsize]
mov matrix0q, [matrix0q]
VBROADCASTSS m4, [matrix0q]
VBROADCASTSS m5, [matrix1q]
ALIGN 16
.loop:
mova m0, [src0q]
S16_TO_S32_SX 0, 2
cvtdq2ps m0, m0
cvtdq2ps m2, m2
mulps m1, m0, m5
mulps m0, m0, m4
mulps m3, m2, m5
mulps m2, m2, m4
cvtps2dq m0, m0
cvtps2dq m1, m1
cvtps2dq m2, m2
cvtps2dq m3, m3
packssdw m0, m2
packssdw m1, m3
mova [src0q ], m0
mova [src0q+src1q], m1
add src0q, mmsize
sub lend, mmsize/2
jg .loop
REP_RET
%endmacro
 
INIT_XMM sse2
MIX_1_TO_2_S16P_FLT
INIT_XMM sse4
MIX_1_TO_2_S16P_FLT
%if HAVE_AVX_EXTERNAL
INIT_XMM avx
MIX_1_TO_2_S16P_FLT
%endif
 
;-----------------------------------------------------------------------------
; void ff_mix_3_8_to_1_2_fltp/s16p_flt(float/int16_t **src, float **matrix,
; int len, int out_ch, int in_ch);
;-----------------------------------------------------------------------------
 
%macro MIX_3_8_TO_1_2_FLT 3 ; %1 = in channels, %2 = out channels, %3 = s16p or fltp
; define some names to make the code clearer
%assign in_channels %1
%assign out_channels %2
%assign stereo out_channels - 1
%ifidn %3, s16p
%assign is_s16 1
%else
%assign is_s16 0
%endif
 
; determine how many matrix elements must go on the stack vs. mmregs
%assign matrix_elements in_channels * out_channels
%if is_s16
%if stereo
%assign needed_mmregs 7
%else
%assign needed_mmregs 5
%endif
%else
%if stereo
%assign needed_mmregs 4
%else
%assign needed_mmregs 3
%endif
%endif
%assign matrix_elements_mm num_mmregs - needed_mmregs
%if matrix_elements < matrix_elements_mm
%assign matrix_elements_mm matrix_elements
%endif
%if matrix_elements_mm < matrix_elements
%assign matrix_elements_stack matrix_elements - matrix_elements_mm
%else
%assign matrix_elements_stack 0
%endif
%assign matrix_stack_size matrix_elements_stack * mmsize
 
%assign needed_stack_size -1 * matrix_stack_size
%if ARCH_X86_32 && in_channels >= 7
%assign needed_stack_size needed_stack_size - 16
%endif
 
cglobal mix_%1_to_%2_%3_flt, 3,in_channels+2,needed_mmregs+matrix_elements_mm, needed_stack_size, src0, src1, len, src2, src3, src4, src5, src6, src7
 
; define src pointers on stack if needed
%if matrix_elements_stack > 0 && ARCH_X86_32 && in_channels >= 7
%define src5m [rsp+matrix_stack_size+0]
%define src6m [rsp+matrix_stack_size+4]
%define src7m [rsp+matrix_stack_size+8]
%endif
 
; load matrix pointers
%define matrix0q r1q
%define matrix1q r3q
%if stereo
mov matrix1q, [matrix0q+gprsize]
%endif
mov matrix0q, [matrix0q]
 
; define matrix coeff names
%assign %%i 0
%assign %%j needed_mmregs
%rep in_channels
%if %%i >= matrix_elements_mm
CAT_XDEFINE mx_stack_0_, %%i, 1
CAT_XDEFINE mx_0_, %%i, [rsp+(%%i-matrix_elements_mm)*mmsize]
%else
CAT_XDEFINE mx_stack_0_, %%i, 0
CAT_XDEFINE mx_0_, %%i, m %+ %%j
%assign %%j %%j+1
%endif
%assign %%i %%i+1
%endrep
%if stereo
%assign %%i 0
%rep in_channels
%if in_channels + %%i >= matrix_elements_mm
CAT_XDEFINE mx_stack_1_, %%i, 1
CAT_XDEFINE mx_1_, %%i, [rsp+(in_channels+%%i-matrix_elements_mm)*mmsize]
%else
CAT_XDEFINE mx_stack_1_, %%i, 0
CAT_XDEFINE mx_1_, %%i, m %+ %%j
%assign %%j %%j+1
%endif
%assign %%i %%i+1
%endrep
%endif
 
; load/splat matrix coeffs
%assign %%i 0
%rep in_channels
%if mx_stack_0_ %+ %%i
VBROADCASTSS m0, [matrix0q+4*%%i]
mova mx_0_ %+ %%i, m0
%else
VBROADCASTSS mx_0_ %+ %%i, [matrix0q+4*%%i]
%endif
%if stereo
%if mx_stack_1_ %+ %%i
VBROADCASTSS m0, [matrix1q+4*%%i]
mova mx_1_ %+ %%i, m0
%else
VBROADCASTSS mx_1_ %+ %%i, [matrix1q+4*%%i]
%endif
%endif
%assign %%i %%i+1
%endrep
 
; load channel pointers to registers as offsets from the first channel pointer
%if ARCH_X86_64
movsxd lenq, r2d
%endif
shl lenq, 2-is_s16
%assign %%i 1
%rep (in_channels - 1)
%if ARCH_X86_32 && in_channels >= 7 && %%i >= 5
mov src5q, [src0q+%%i*gprsize]
add src5q, lenq
mov src %+ %%i %+ m, src5q
%else
mov src %+ %%i %+ q, [src0q+%%i*gprsize]
add src %+ %%i %+ q, lenq
%endif
%assign %%i %%i+1
%endrep
mov src0q, [src0q]
add src0q, lenq
neg lenq
.loop:
; for x86-32 with 7-8 channels we do not have enough gp registers for all src
; pointers, so we have to load some of them from the stack each time
%define copy_src_from_stack ARCH_X86_32 && in_channels >= 7 && %%i >= 5
%if is_s16
; mix with s16p input
mova m0, [src0q+lenq]
S16_TO_S32_SX 0, 1
cvtdq2ps m0, m0
cvtdq2ps m1, m1
%if stereo
mulps m2, m0, mx_1_0
mulps m3, m1, mx_1_0
%endif
mulps m0, m0, mx_0_0
mulps m1, m1, mx_0_0
%assign %%i 1
%rep (in_channels - 1)
%if copy_src_from_stack
%define src_ptr src5q
%else
%define src_ptr src %+ %%i %+ q
%endif
%if stereo
%if copy_src_from_stack
mov src_ptr, src %+ %%i %+ m
%endif
mova m4, [src_ptr+lenq]
S16_TO_S32_SX 4, 5
cvtdq2ps m4, m4
cvtdq2ps m5, m5
FMULADD_PS m2, m4, mx_1_ %+ %%i, m2, m6
FMULADD_PS m3, m5, mx_1_ %+ %%i, m3, m6
FMULADD_PS m0, m4, mx_0_ %+ %%i, m0, m4
FMULADD_PS m1, m5, mx_0_ %+ %%i, m1, m5
%else
%if copy_src_from_stack
mov src_ptr, src %+ %%i %+ m
%endif
mova m2, [src_ptr+lenq]
S16_TO_S32_SX 2, 3
cvtdq2ps m2, m2
cvtdq2ps m3, m3
FMULADD_PS m0, m2, mx_0_ %+ %%i, m0, m4
FMULADD_PS m1, m3, mx_0_ %+ %%i, m1, m4
%endif
%assign %%i %%i+1
%endrep
%if stereo
cvtps2dq m2, m2
cvtps2dq m3, m3
packssdw m2, m3
mova [src1q+lenq], m2
%endif
cvtps2dq m0, m0
cvtps2dq m1, m1
packssdw m0, m1
mova [src0q+lenq], m0
%else
; mix with fltp input
%if stereo || mx_stack_0_0
mova m0, [src0q+lenq]
%endif
%if stereo
mulps m1, m0, mx_1_0
%endif
%if stereo || mx_stack_0_0
mulps m0, m0, mx_0_0
%else
mulps m0, mx_0_0, [src0q+lenq]
%endif
%assign %%i 1
%rep (in_channels - 1)
%if copy_src_from_stack
%define src_ptr src5q
mov src_ptr, src %+ %%i %+ m
%else
%define src_ptr src %+ %%i %+ q
%endif
; avoid extra load for mono if matrix is in a mm register
%if stereo || mx_stack_0_ %+ %%i
mova m2, [src_ptr+lenq]
%endif
%if stereo
FMULADD_PS m1, m2, mx_1_ %+ %%i, m1, m3
%endif
%if stereo || mx_stack_0_ %+ %%i
FMULADD_PS m0, m2, mx_0_ %+ %%i, m0, m2
%else
FMULADD_PS m0, mx_0_ %+ %%i, [src_ptr+lenq], m0, m1
%endif
%assign %%i %%i+1
%endrep
mova [src0q+lenq], m0
%if stereo
mova [src1q+lenq], m1
%endif
%endif
 
add lenq, mmsize
jl .loop
; zero ymm high halves
%if mmsize == 32
vzeroupper
%endif
RET
%endmacro
 
%macro MIX_3_8_TO_1_2_FLT_FUNCS 0
%assign %%i 3
%rep 6
INIT_XMM sse
MIX_3_8_TO_1_2_FLT %%i, 1, fltp
MIX_3_8_TO_1_2_FLT %%i, 2, fltp
INIT_XMM sse2
MIX_3_8_TO_1_2_FLT %%i, 1, s16p
MIX_3_8_TO_1_2_FLT %%i, 2, s16p
INIT_XMM sse4
MIX_3_8_TO_1_2_FLT %%i, 1, s16p
MIX_3_8_TO_1_2_FLT %%i, 2, s16p
; do not use ymm AVX or FMA4 in x86-32 for 6 or more channels due to stack alignment issues
%if HAVE_AVX_EXTERNAL
%if ARCH_X86_64 || %%i < 6
INIT_YMM avx
%else
INIT_XMM avx
%endif
MIX_3_8_TO_1_2_FLT %%i, 1, fltp
MIX_3_8_TO_1_2_FLT %%i, 2, fltp
INIT_XMM avx
MIX_3_8_TO_1_2_FLT %%i, 1, s16p
MIX_3_8_TO_1_2_FLT %%i, 2, s16p
%endif
%if HAVE_FMA4_EXTERNAL
%if ARCH_X86_64 || %%i < 6
INIT_YMM fma4
%else
INIT_XMM fma4
%endif
MIX_3_8_TO_1_2_FLT %%i, 1, fltp
MIX_3_8_TO_1_2_FLT %%i, 2, fltp
INIT_XMM fma4
MIX_3_8_TO_1_2_FLT %%i, 1, s16p
MIX_3_8_TO_1_2_FLT %%i, 2, s16p
%endif
%assign %%i %%i+1
%endrep
%endmacro
 
MIX_3_8_TO_1_2_FLT_FUNCS
/contrib/sdk/sources/ffmpeg/libavresample/x86/audio_mix_init.c
0,0 → 1,215
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 "config.h"
#include "libavutil/cpu.h"
#include "libavutil/x86/cpu.h"
#include "libavresample/audio_mix.h"
 
void ff_mix_2_to_1_fltp_flt_sse(float **src, float **matrix, int len,
int out_ch, int in_ch);
void ff_mix_2_to_1_fltp_flt_avx(float **src, float **matrix, int len,
int out_ch, int in_ch);
 
void ff_mix_2_to_1_s16p_flt_sse2(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
void ff_mix_2_to_1_s16p_flt_sse4(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
 
void ff_mix_2_to_1_s16p_q8_sse2(int16_t **src, int16_t **matrix,
int len, int out_ch, int in_ch);
 
void ff_mix_1_to_2_fltp_flt_sse(float **src, float **matrix, int len,
int out_ch, int in_ch);
void ff_mix_1_to_2_fltp_flt_avx(float **src, float **matrix, int len,
int out_ch, int in_ch);
 
void ff_mix_1_to_2_s16p_flt_sse2(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
void ff_mix_1_to_2_s16p_flt_sse4(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
void ff_mix_1_to_2_s16p_flt_avx (int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
 
#define DEFINE_MIX_3_8_TO_1_2(chan) \
void ff_mix_ ## chan ## _to_1_fltp_flt_sse(float **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_fltp_flt_sse(float **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
\
void ff_mix_ ## chan ## _to_1_s16p_flt_sse2(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_s16p_flt_sse2(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
\
void ff_mix_ ## chan ## _to_1_s16p_flt_sse4(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_s16p_flt_sse4(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
\
void ff_mix_ ## chan ## _to_1_fltp_flt_avx(float **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_fltp_flt_avx(float **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
\
void ff_mix_ ## chan ## _to_1_s16p_flt_avx(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_s16p_flt_avx(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
\
void ff_mix_ ## chan ## _to_1_fltp_flt_fma4(float **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_fltp_flt_fma4(float **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
\
void ff_mix_ ## chan ## _to_1_s16p_flt_fma4(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch); \
void ff_mix_ ## chan ## _to_2_s16p_flt_fma4(int16_t **src, \
float **matrix, int len, \
int out_ch, int in_ch);
 
DEFINE_MIX_3_8_TO_1_2(3)
DEFINE_MIX_3_8_TO_1_2(4)
DEFINE_MIX_3_8_TO_1_2(5)
DEFINE_MIX_3_8_TO_1_2(6)
DEFINE_MIX_3_8_TO_1_2(7)
DEFINE_MIX_3_8_TO_1_2(8)
 
#define SET_MIX_3_8_TO_1_2(chan) \
if (EXTERNAL_SSE(cpu_flags)) { \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, 16, 4, "SSE", \
ff_mix_ ## chan ## _to_1_fltp_flt_sse); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, 16, 4, "SSE", \
ff_mix_## chan ##_to_2_fltp_flt_sse); \
} \
if (EXTERNAL_SSE2(cpu_flags)) { \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, 16, 8, "SSE2", \
ff_mix_ ## chan ## _to_1_s16p_flt_sse2); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, 16, 8, "SSE2", \
ff_mix_ ## chan ## _to_2_s16p_flt_sse2); \
} \
if (EXTERNAL_SSE4(cpu_flags)) { \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, 16, 8, "SSE4", \
ff_mix_ ## chan ## _to_1_s16p_flt_sse4); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, 16, 8, "SSE4", \
ff_mix_ ## chan ## _to_2_s16p_flt_sse4); \
} \
if (EXTERNAL_AVX(cpu_flags)) { \
int ptr_align = 32; \
int smp_align = 8; \
if (ARCH_X86_32 || chan >= 6) { \
ptr_align = 16; \
smp_align = 4; \
} \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, ptr_align, smp_align, "AVX", \
ff_mix_ ## chan ## _to_1_fltp_flt_avx); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, ptr_align, smp_align, "AVX", \
ff_mix_ ## chan ## _to_2_fltp_flt_avx); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, 16, 8, "AVX", \
ff_mix_ ## chan ## _to_1_s16p_flt_avx); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, 16, 8, "AVX", \
ff_mix_ ## chan ## _to_2_s16p_flt_avx); \
} \
if (EXTERNAL_FMA4(cpu_flags)) { \
int ptr_align = 32; \
int smp_align = 8; \
if (ARCH_X86_32 || chan >= 6) { \
ptr_align = 16; \
smp_align = 4; \
} \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, ptr_align, smp_align, "FMA4", \
ff_mix_ ## chan ## _to_1_fltp_flt_fma4); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, ptr_align, smp_align, "FMA4", \
ff_mix_ ## chan ## _to_2_fltp_flt_fma4); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 1, 16, 8, "FMA4", \
ff_mix_ ## chan ## _to_1_s16p_flt_fma4); \
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
chan, 2, 16, 8, "FMA4", \
ff_mix_ ## chan ## _to_2_s16p_flt_fma4); \
}
 
av_cold void ff_audio_mix_init_x86(AudioMix *am)
{
#if HAVE_YASM
int cpu_flags = av_get_cpu_flags();
 
if (EXTERNAL_SSE(cpu_flags)) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE", ff_mix_2_to_1_fltp_flt_sse);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 4, "SSE", ff_mix_1_to_2_fltp_flt_sse);
}
if (EXTERNAL_SSE2(cpu_flags)) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_flt_sse2);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_q8_sse2);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 8, "SSE2", ff_mix_1_to_2_s16p_flt_sse2);
}
if (EXTERNAL_SSE4(cpu_flags)) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE4", ff_mix_2_to_1_s16p_flt_sse4);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 8, "SSE4", ff_mix_1_to_2_s16p_flt_sse4);
}
if (EXTERNAL_AVX(cpu_flags)) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 32, 16, "AVX", ff_mix_2_to_1_fltp_flt_avx);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 32, 8, "AVX", ff_mix_1_to_2_fltp_flt_avx);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
1, 2, 16, 8, "AVX", ff_mix_1_to_2_s16p_flt_avx);
}
 
SET_MIX_3_8_TO_1_2(3)
SET_MIX_3_8_TO_1_2(4)
SET_MIX_3_8_TO_1_2(5)
SET_MIX_3_8_TO_1_2(6)
SET_MIX_3_8_TO_1_2(7)
SET_MIX_3_8_TO_1_2(8)
#endif /* HAVE_YASM */
}
/contrib/sdk/sources/ffmpeg/libavresample/x86/dither.asm
0,0 → 1,117
;******************************************************************************
;* x86 optimized dithering format conversion
;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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"
 
SECTION_RODATA 32
 
; 1.0f / (2.0f * INT32_MAX)
pf_dither_scale: times 8 dd 2.32830643762e-10
 
pf_s16_scale: times 4 dd 32753.0
 
SECTION_TEXT
 
;------------------------------------------------------------------------------
; void ff_quantize(int16_t *dst, float *src, float *dither, int len);
;------------------------------------------------------------------------------
 
INIT_XMM sse2
cglobal quantize, 4,4,3, dst, src, dither, len
lea lenq, [2*lend]
add dstq, lenq
lea srcq, [srcq+2*lenq]
lea ditherq, [ditherq+2*lenq]
neg lenq
mova m2, [pf_s16_scale]
.loop:
mulps m0, m2, [srcq+2*lenq]
mulps m1, m2, [srcq+2*lenq+mmsize]
addps m0, [ditherq+2*lenq]
addps m1, [ditherq+2*lenq+mmsize]
cvtps2dq m0, m0
cvtps2dq m1, m1
packssdw m0, m1
mova [dstq+lenq], m0
add lenq, mmsize
jl .loop
REP_RET
 
;------------------------------------------------------------------------------
; void ff_dither_int_to_float_rectangular(float *dst, int *src, int len)
;------------------------------------------------------------------------------
 
%macro DITHER_INT_TO_FLOAT_RECTANGULAR 0
cglobal dither_int_to_float_rectangular, 3,3,3, dst, src, len
lea lenq, [4*lend]
add srcq, lenq
add dstq, lenq
neg lenq
mova m0, [pf_dither_scale]
.loop:
cvtdq2ps m1, [srcq+lenq]
cvtdq2ps m2, [srcq+lenq+mmsize]
mulps m1, m1, m0
mulps m2, m2, m0
mova [dstq+lenq], m1
mova [dstq+lenq+mmsize], m2
add lenq, 2*mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
DITHER_INT_TO_FLOAT_RECTANGULAR
INIT_YMM avx
DITHER_INT_TO_FLOAT_RECTANGULAR
 
;------------------------------------------------------------------------------
; void ff_dither_int_to_float_triangular(float *dst, int *src0, int len)
;------------------------------------------------------------------------------
 
%macro DITHER_INT_TO_FLOAT_TRIANGULAR 0
cglobal dither_int_to_float_triangular, 3,4,5, dst, src0, len, src1
lea lenq, [4*lend]
lea src1q, [src0q+2*lenq]
add src0q, lenq
add dstq, lenq
neg lenq
mova m0, [pf_dither_scale]
.loop:
cvtdq2ps m1, [src0q+lenq]
cvtdq2ps m2, [src0q+lenq+mmsize]
cvtdq2ps m3, [src1q+lenq]
cvtdq2ps m4, [src1q+lenq+mmsize]
addps m1, m1, m3
addps m2, m2, m4
mulps m1, m1, m0
mulps m2, m2, m0
mova [dstq+lenq], m1
mova [dstq+lenq+mmsize], m2
add lenq, 2*mmsize
jl .loop
REP_RET
%endmacro
 
INIT_XMM sse2
DITHER_INT_TO_FLOAT_TRIANGULAR
INIT_YMM avx
DITHER_INT_TO_FLOAT_TRIANGULAR
/contrib/sdk/sources/ffmpeg/libavresample/x86/dither_init.c
0,0 → 1,60
/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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 "config.h"
#include "libavutil/cpu.h"
#include "libavutil/x86/cpu.h"
#include "libavresample/dither.h"
 
void ff_quantize_sse2(int16_t *dst, const float *src, float *dither, int len);
 
void ff_dither_int_to_float_rectangular_sse2(float *dst, int *src, int len);
void ff_dither_int_to_float_rectangular_avx(float *dst, int *src, int len);
 
void ff_dither_int_to_float_triangular_sse2(float *dst, int *src0, int len);
void ff_dither_int_to_float_triangular_avx(float *dst, int *src0, int len);
 
av_cold void ff_dither_init_x86(DitherDSPContext *ddsp,
enum AVResampleDitherMethod method)
{
int cpu_flags = av_get_cpu_flags();
 
if (EXTERNAL_SSE2(cpu_flags)) {
ddsp->quantize = ff_quantize_sse2;
ddsp->ptr_align = 16;
ddsp->samples_align = 8;
}
 
if (method == AV_RESAMPLE_DITHER_RECTANGULAR) {
if (EXTERNAL_SSE2(cpu_flags)) {
ddsp->dither_int_to_float = ff_dither_int_to_float_rectangular_sse2;
}
if (EXTERNAL_AVX(cpu_flags)) {
ddsp->dither_int_to_float = ff_dither_int_to_float_rectangular_avx;
}
} else {
if (EXTERNAL_SSE2(cpu_flags)) {
ddsp->dither_int_to_float = ff_dither_int_to_float_triangular_sse2;
}
if (EXTERNAL_AVX(cpu_flags)) {
ddsp->dither_int_to_float = ff_dither_int_to_float_triangular_avx;
}
}
}
/contrib/sdk/sources/ffmpeg/libavresample/x86/util.asm
0,0 → 1,41
;******************************************************************************
;* x86 utility macros for libavresample
;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@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
;******************************************************************************
 
%macro S16_TO_S32_SX 2 ; src/low dst, high dst
%if cpuflag(sse4)
pmovsxwd m%2, m%1
psrldq m%1, 8
pmovsxwd m%1, m%1
SWAP %1, %2
%else
mova m%2, m%1
punpckhwd m%2, m%2
punpcklwd m%1, m%1
psrad m%2, 16
psrad m%1, 16
%endif
%endmacro
 
%macro DEINT2_PS 3 ; src0/even dst, src1/odd dst, temp
shufps m%3, m%1, m%2, q3131
shufps m%1, m%2, q2020
SWAP %2,%3
%endmacro