FFmpeg
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 "libavutil/avstring.h"
29 #include "libavutil/avassert.h"
30 #include "libavutil/libm.h"
31 #include "libavutil/samplefmt.h"
32 #include "audioconvert.h"
33 
34 
35 #define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
36 
37 //FIXME rounding ?
38 #define CONV_FUNC(ofmt, otype, ifmt, expr)\
39 static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)\
40 {\
41  uint8_t *end2 = end - 3*os;\
42  while(po < end2){\
43  *(otype*)po = expr; pi += is; po += os;\
44  *(otype*)po = expr; pi += is; po += os;\
45  *(otype*)po = expr; pi += is; po += os;\
46  *(otype*)po = expr; pi += is; po += os;\
47  }\
48  while(po < end){\
49  *(otype*)po = expr; pi += is; po += os;\
50  }\
51 }
52 
53 //FIXME put things below under ifdefs so we do not waste space for cases no codec will need
54 CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
55 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<8)
56 CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<24)
57 CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8 , (uint64_t)((*(const uint8_t*)pi - 0x80U))<<56)
58 CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7)))
59 CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
60 CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
61 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
62 CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi * (1 << 16))
63 CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16, (uint64_t)(*(const int16_t*)pi)<<48)
64 CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15)))
65 CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
66 CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
67 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
69 CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32, (uint64_t)(*(const int32_t*)pi)<<32)
70 CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31)))
71 CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
72 CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S64, (*(const int64_t*)pi>>56) + 0x80)
73 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi>>48)
74 CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi>>32)
75 CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S64, *(const int64_t*)pi)
76 CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S64, *(const int64_t*)pi*(1.0f/ (UINT64_C(1)<<63)))
77 CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S64, *(const int64_t*)pi*(1.0 / (UINT64_C(1)<<63)))
78 CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
79 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
81 CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float*)pi * (UINT64_C(1)<<63)))
82 CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
83 CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
84 CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
85 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
87 CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double*)pi * (UINT64_C(1)<<63)))
88 CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
89 CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
90 
91 #define FMT_PAIR_FUNC(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME(out, in)
92 
93 static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = {
130 };
131 
132 static void cpy1(uint8_t **dst, const uint8_t **src, int len){
133  memcpy(*dst, *src, len);
134 }
135 static void cpy2(uint8_t **dst, const uint8_t **src, int len){
136  memcpy(*dst, *src, 2*len);
137 }
138 static void cpy4(uint8_t **dst, const uint8_t **src, int len){
139  memcpy(*dst, *src, 4*len);
140 }
141 static void cpy8(uint8_t **dst, const uint8_t **src, int len){
142  memcpy(*dst, *src, 8*len);
143 }
144 
146  enum AVSampleFormat in_fmt,
147  int channels, const int *ch_map,
148  int flags)
149 {
150  AudioConvert *ctx;
151  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)];
152 
153  if (!f)
154  return NULL;
155  ctx = av_mallocz(sizeof(*ctx));
156  if (!ctx)
157  return NULL;
158 
159  if(channels == 1){
160  in_fmt = av_get_planar_sample_fmt( in_fmt);
161  out_fmt = av_get_planar_sample_fmt(out_fmt);
162  }
163 
164  ctx->channels = channels;
165  ctx->conv_f = f;
166  ctx->ch_map = ch_map;
167  if (in_fmt == AV_SAMPLE_FMT_U8 || in_fmt == AV_SAMPLE_FMT_U8P)
168  memset(ctx->silence, 0x80, sizeof(ctx->silence));
169 
170  if(out_fmt == in_fmt && !ch_map) {
171  switch(av_get_bytes_per_sample(in_fmt)){
172  case 1:ctx->simd_f = cpy1; break;
173  case 2:ctx->simd_f = cpy2; break;
174  case 4:ctx->simd_f = cpy4; break;
175  case 8:ctx->simd_f = cpy8; break;
176  }
177  }
178 
179 #if ARCH_X86 && HAVE_X86ASM && HAVE_MMX
180  swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);
181 #elif ARCH_ARM
182  swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);
183 #elif ARCH_AARCH64
184  swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);
185 #endif
186 
187  return ctx;
188 }
189 
191 {
192  av_freep(ctx);
193 }
194 
196 {
197  int ch;
198  int off=0;
199  const int os= (out->planar ? 1 :out->ch_count) *out->bps;
200  unsigned misaligned = 0;
201 
202  av_assert0(ctx->channels == out->ch_count);
203 
204  if (ctx->in_simd_align_mask) {
205  int planes = in->planar ? in->ch_count : 1;
206  unsigned m = 0;
207  for (ch = 0; ch < planes; ch++)
208  m |= (intptr_t)in->ch[ch];
209  misaligned |= m & ctx->in_simd_align_mask;
210  }
211  if (ctx->out_simd_align_mask) {
212  int planes = out->planar ? out->ch_count : 1;
213  unsigned m = 0;
214  for (ch = 0; ch < planes; ch++)
215  m |= (intptr_t)out->ch[ch];
216  misaligned |= m & ctx->out_simd_align_mask;
217  }
218 
219  //FIXME optimize common cases
220 
221  if(ctx->simd_f && !ctx->ch_map && !misaligned){
222  off = len&~15;
223  av_assert1(off>=0);
224  av_assert1(off<=len);
225  av_assert2(ctx->channels == SWR_CH_MAX || !in->ch[ctx->channels]);
226  if(off>0){
227  if(out->planar == in->planar){
228  int planes = out->planar ? out->ch_count : 1;
229  for(ch=0; ch<planes; ch++){
230  ctx->simd_f(out->ch+ch, (const uint8_t **)in->ch+ch, off * (out->planar ? 1 :out->ch_count));
231  }
232  }else{
233  ctx->simd_f(out->ch, (const uint8_t **)in->ch, off);
234  }
235  }
236  if(off == len)
237  return 0;
238  }
239 
240  for(ch=0; ch<ctx->channels; ch++){
241  const int ich= ctx->ch_map ? ctx->ch_map[ch] : ch;
242  const int is= ich < 0 ? 0 : (in->planar ? 1 : in->ch_count) * in->bps;
243  const uint8_t *pi= ich < 0 ? ctx->silence : in->ch[ich];
244  uint8_t *end, *po = out->ch[ch];
245  if(!po)
246  continue;
247  end = po + os * len;
248  ctx->conv_f(po+off*os, pi+off*is, is, os, end);
249  }
250  return 0;
251 }
libm.h
out
FILE * out
Definition: movenc.c:54
is
The official guide to swscale for confused that is
Definition: swscale.txt:28
AudioData::bps
int bps
bytes per sample
Definition: swresample_internal.h:49
AudioData
Definition: swresample_internal.h:45
U
#define U(x)
Definition: vp56_arith.h:37
swri_audio_convert_init_x86
void swri_audio_convert_init_x86(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
Definition: audio_convert_init.c:37
samplefmt.h
av_get_planar_sample_fmt
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:86
avassert.h
lrint
#define lrint
Definition: tablegen.h:53
llrintf
#define llrintf(x)
Definition: libm.h:399
CONV_FUNC
#define CONV_FUNC(ofmt, otype, ifmt, expr)
Definition: audioconvert.c:38
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AudioData::planar
int planar
1 if planar audio, 0 otherwise
Definition: swresample_internal.h:51
ctx
AVFormatContext * ctx
Definition: movenc.c:48
channels
channels
Definition: aptx.h:32
swri_audio_convert
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
Convert between audio sample formats.
AudioData::ch
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel
Definition: swresample_internal.h:46
av_clip_int16
#define av_clip_int16
Definition: common.h:110
NULL
#define NULL
Definition: coverity.c:32
AudioData::ch_count
int ch_count
number of channels
Definition: swresample_internal.h:48
planes
static const struct @328 planes[]
swri_audio_convert_init_arm
av_cold void swri_audio_convert_init_arm(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
Definition: audio_convert_init.c:48
f
f
Definition: af_crystalizer.c:122
swri_audio_convert_init_aarch64
av_cold void swri_audio_convert_init_aarch64(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels)
Definition: audio_convert_init.c:48
AV_SAMPLE_FMT_NB
@ AV_SAMPLE_FMT_NB
Number of sample formats. DO NOT USE if linking dynamically.
Definition: samplefmt.h:71
AV_SAMPLE_FMT_U8P
@ AV_SAMPLE_FMT_U8P
unsigned 8 bits, planar
Definition: samplefmt.h:63
av_clipl_int32
#define av_clipl_int32
Definition: common.h:113
swri_audio_convert_alloc
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.
av_assert2
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
lrintf
#define lrintf(x)
Definition: libm_mips.h:72
av_get_bytes_per_sample
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:108
av_assert1
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:264
len
int len
Definition: vorbis_enc_data.h:426
FMT_PAIR_FUNC
#define FMT_PAIR_FUNC(out, in)
SWR_CH_MAX
#define SWR_CH_MAX
Definition: af_amerge.c:35
AudioConvert
Definition: audioconvert.h:38
swri_audio_convert_free
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
av_get_packed_sample_fmt
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition: samplefmt.c:77
av_clip_uint8
#define av_clip_uint8
Definition: common.h:101
AV_SAMPLE_FMT_U8
AV_SAMPLE_FMT_U8
Definition: audioconvert.c:56
llrint
#define llrint(x)
Definition: libm.h:394
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
int32_t
int32_t
Definition: audioconvert.c:56
conv_func_type
void() conv_func_type(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)
Definition: audioconvert.h:35
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:61
avstring.h
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:59
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
AV_SAMPLE_FMT_S64
@ AV_SAMPLE_FMT_S64
signed 64 bits
Definition: samplefmt.h:68
audioconvert.h