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audioconvert.c
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1/*
2 * audio conversion
3 * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
23 * @file
24 * audio conversion
25 * @author Michael Niedermayer <michaelni@gmx.at>
26 */
27
28#include <math.h>
29
30#include "libavutil/avassert.h"
31#include "libavutil/internal.h"
32#include "libavutil/mem.h"
33#include "libavutil/samplefmt.h"
34#include "audioconvert.h"
35
36
37#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
38
39//FIXME rounding ?
40#define CONV_FUNC(ofmt, otype, ifmt, expr)\
41static void CONV_FUNC_NAME(ofmt, ifmt)(DSDContext *st, uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)\
42{\
43 uint8_t *end2 = end - 3*os;\
44 while(po < end2){\
45 *(otype*)po = expr; pi += is; po += os;\
46 *(otype*)po = expr; pi += is; po += os;\
47 *(otype*)po = expr; pi += is; po += os;\
48 *(otype*)po = expr; pi += is; po += os;\
49 }\
50 while(po < end){\
51 *(otype*)po = expr; pi += is; po += os;\
52 }\
53}
54
55//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
56CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
57CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<8)
58CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<24)
59CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8 , (uint64_t)((*(const uint8_t*)pi - 0x80U))<<56)
60CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7)))
61CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
62CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
63CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
64CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi * (1 << 16))
65CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16, (uint64_t)(*(const int16_t*)pi)<<48)
66CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15)))
67CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
68CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
69CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
71CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32, (uint64_t)(*(const int32_t*)pi)<<32)
72CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31)))
73CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
74CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S64, (*(const int64_t*)pi>>56) + 0x80)
75CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi>>48)
78CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S64, *(const int64_t*)pi*(1.0f/ (UINT64_C(1)<<63)))
79CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S64, *(const int64_t*)pi*(1.0 / (UINT64_C(1)<<63)))
80CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
81CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
82CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
83CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float*)pi * (UINT64_C(1)<<63)))
84CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
85CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
86CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
87CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
89CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double*)pi * (UINT64_C(1)<<63)))
90CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
91CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
92CONV_FUNC(AV_SAMPLE_FMT_DSD, uint8_t, AV_SAMPLE_FMT_DSD, *(const uint8_t*)pi)
93
94static void CONV_FUNC_NAME(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DSD)(DSDContext *st, uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)
95{
96 swri_dsd2pcm_translate(st, (end - po) / os, pi, is, (float *)po, os / sizeof(float));
97}
98
99#define FMT_PAIR_FUNC(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME(out, in)
100
101static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = {
140};
141
142static void cpy1(uint8_t **dst, const uint8_t **src, int len){
143 memcpy(*dst, *src, len);
144}
145static void cpy2(uint8_t **dst, const uint8_t **src, int len){
146 memcpy(*dst, *src, 2*len);
147}
148static void cpy4(uint8_t **dst, const uint8_t **src, int len){
149 memcpy(*dst, *src, 4*len);
150}
151static void cpy8(uint8_t **dst, const uint8_t **src, int len){
152 memcpy(*dst, *src, 8*len);
153}
154
156 enum AVSampleFormat in_fmt,
157 int channels, const int *ch_map,
158 int flags)
159{
161 conv_func_type *f = fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt) + AV_SAMPLE_FMT_NB*av_get_packed_sample_fmt(in_fmt)];
162
163 if (!f)
164 return NULL;
165 ctx = av_mallocz(sizeof(*ctx));
166 if (!ctx)
167 return NULL;
168
169 if(channels == 1){
170 in_fmt = av_get_planar_sample_fmt( in_fmt);
171 out_fmt = av_get_planar_sample_fmt(out_fmt);
172 }
173
174 ctx->channels = channels;
175 ctx->conv_f = f;
176 ctx->ch_map = ch_map;
177 if (in_fmt == AV_SAMPLE_FMT_U8 || in_fmt == AV_SAMPLE_FMT_U8P)
178 memset(ctx->silence, 0x80, sizeof(ctx->silence));
179 if (in_fmt == AV_SAMPLE_FMT_DSD) {
181 memset(ctx->silence, 0x69, sizeof(ctx->silence));
182 for (int ch = 0; ch < FF_ARRAY_ELEMS(ctx->dsd_state); ch++)
183 memset(ctx->dsd_state[ch].buf, 0x69, sizeof(ctx->dsd_state[ch].buf));
184 }
185
186 if(out_fmt == in_fmt && !ch_map) {
187 switch(av_get_bytes_per_sample(in_fmt)){
188 case 1:ctx->simd_f = cpy1; break;
189 case 2:ctx->simd_f = cpy2; break;
190 case 4:ctx->simd_f = cpy4; break;
191 case 8:ctx->simd_f = cpy8; break;
192 }
193 }
194
195#if ARCH_X86 && HAVE_X86ASM
196 swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);
197#elif ARCH_ARM
198 swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);
199#elif ARCH_AARCH64
201#endif
202
203 return ctx;
204}
205
207{
208 av_freep(ctx);
209}
210
212{
213 int ch;
214 int off=0;
215 const int os= (out->planar ? 1 :out->ch_count) *out->bps;
216 unsigned misaligned = 0;
217
218 av_assert0(ctx->channels == out->ch_count);
219
220 if (ctx->in_simd_align_mask) {
221 int planes = in->planar ? in->ch_count : 1;
222 unsigned m = 0;
223 for (ch = 0; ch < planes; ch++)
224 m |= (intptr_t)in->ch[ch];
225 misaligned |= m & ctx->in_simd_align_mask;
226 }
227 if (ctx->out_simd_align_mask) {
228 int planes = out->planar ? out->ch_count : 1;
229 unsigned m = 0;
230 for (ch = 0; ch < planes; ch++)
231 m |= (intptr_t)out->ch[ch];
232 misaligned |= m & ctx->out_simd_align_mask;
233 }
234
235 //FIXME optimize common cases
236
237 if(ctx->simd_f && !ctx->ch_map && !misaligned){
238 off = len&~15;
239 av_assert1(off>=0);
240 av_assert1(off<=len);
241 av_assert2(ctx->channels == SWR_CH_MAX || !in->ch[ctx->channels]);
242 if(off>0){
243 if(out->planar == in->planar){
244 int planes = out->planar ? out->ch_count : 1;
245 for(ch=0; ch<planes; ch++){
246 ctx->simd_f(out->ch+ch, (const uint8_t **)in->ch+ch, off * (out->planar ? 1 :out->ch_count));
247 }
248 }else{
249 ctx->simd_f(out->ch, (const uint8_t **)in->ch, off);
250 }
251 }
252 if(off == len)
253 return 0;
254 }
255
256 for(ch=0; ch<ctx->channels; ch++){
257 const int ich= ctx->ch_map ? ctx->ch_map[ch] : ch;
258 const int is= ich < 0 ? 0 : (in->planar ? 1 : in->ch_count) * in->bps;
259 const uint8_t *pi= ich < 0 ? ctx->silence : in->ch[ich];
260 uint8_t *end, *po = out->ch[ch];
261 if(!po)
262 continue;
263 end = po + os * len;
264 ctx->conv_f(&ctx->dsd_state[ch], po+off*os, pi+off*is, is, os, end);
265 }
266 return 0;
267}
av_cold void swri_audio_convert_init_aarch64(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition dsp.h:87
#define SWR_CH_MAX
Definition af_amerge.c:37
static const int ch_map[SC_NB]
Definition af_surround.c:37
static FILE * out
static AVFormatContext * ctx
channels
Definition aptx.h:31
av_cold void swri_audio_convert_init_arm(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
#define U(x)
Definition vpx_arith.h:37
#define FMT_PAIR_FUNC(out, in)
#define CONV_FUNC_NAME(dst_fmt, src_fmt)
#define CONV_FUNC(ofmt, otype, ifmt, expr)
int32_t
Audio format conversion routines.
AudioConvert * swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags)
Create an audio sample format converter context.
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
Convert between audio sample formats.
void conv_func_type(DSDContext *st, uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition avassert.h:68
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition avassert.h:58
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition avassert.h:42
#define is(width, name, range_min, range_max, subs,...)
Definition cbs_h264.c:78
#define flags(name, subs,...)
Definition cbs_h264.c:74
#define f(width, name)
Definition cbs_vp8.c:236
#define av_clipl_int32
Definition common.h:118
#define av_clip_int16
Definition common.h:115
#define av_clip_uint8
Definition common.h:106
#define NULL
Definition coverity.c:32
long long int64_t
Definition coverity.c:34
av_cold void swri_dsd2pcm_init(void)
Definition dsd2pcm.c:89
void swri_dsd2pcm_translate(DSDContext *s, size_t samples, const uint8_t *src, ptrdiff_t src_stride, float *dst, ptrdiff_t dst_stride)
Convert one channel of MSB-first DSD data (one byte = 8 samples) to float PCM at 1/8th of the DSD bit...
Definition dsd2pcm.c:95
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition samplefmt.c:109
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition samplefmt.c:78
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition samplefmt.c:87
AVSampleFormat
Audio sample formats.
Definition samplefmt.h:55
@ AV_SAMPLE_FMT_FLT
float
Definition samplefmt.h:60
@ AV_SAMPLE_FMT_S64
signed 64 bits
Definition samplefmt.h:68
@ AV_SAMPLE_FMT_U8P
unsigned 8 bits, planar
Definition samplefmt.h:63
@ AV_SAMPLE_FMT_DSD
DSD (Direct Stream Digital) bitstream, interleaved.
Definition samplefmt.h:77
@ AV_SAMPLE_FMT_NB
Number of sample formats. DO NOT USE if linking dynamically.
Definition samplefmt.h:79
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition samplefmt.h:59
@ AV_SAMPLE_FMT_U8
unsigned 8 bits
Definition samplefmt.h:57
@ AV_SAMPLE_FMT_DBL
double
Definition samplefmt.h:61
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition samplefmt.h:58
common internal API header
#define lrintf(x)
Definition libm_mips.h:74
static const struct @257111027162314367033347246032313251342043035002 planes[]
Memory handling functions.
#define FF_ARRAY_ELEMS(a)
Per-channel buffer.
Definition dsd2pcm.h:41
void swri_audio_convert_init_x86(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
#define lrint
Definition tablegen.h:53
#define llrint
Definition tablegen.h:47
#define av_mallocz(s)
#define av_freep(p)
#define src
Definition vp8dsp.c:248
int len