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audio_convert.c
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1 /*
2  * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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 #include <stdint.h>
23 
24 #include "config.h"
25 #include "libavutil/common.h"
26 #include "libavutil/libm.h"
27 #include "libavutil/log.h"
28 #include "libavutil/mem.h"
29 #include "libavutil/samplefmt.h"
30 #include "audio_convert.h"
31 #include "audio_data.h"
32 #include "dither.h"
33 
38 };
39 
40 typedef void (conv_func_flat)(uint8_t *out, const uint8_t *in, int len);
41 
43  int len, int channels);
44 
45 typedef void (conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len,
46  int channels);
47 
48 struct AudioConvert {
53  int apply_map;
54  int channels;
55  int planes;
56  int ptr_align;
59  const char *func_descr;
60  const char *func_descr_generic;
68 };
69 
71  enum AVSampleFormat in_fmt, int channels,
72  int ptr_align, int samples_align,
73  const char *descr, void *conv)
74 {
75  int found = 0;
76 
77  switch (ac->func_type) {
79  if (av_get_packed_sample_fmt(ac->in_fmt) == in_fmt &&
80  av_get_packed_sample_fmt(ac->out_fmt) == out_fmt) {
81  ac->conv_flat = conv;
82  ac->func_descr = descr;
83  ac->ptr_align = ptr_align;
84  ac->samples_align = samples_align;
85  if (ptr_align == 1 && samples_align == 1) {
86  ac->conv_flat_generic = conv;
87  ac->func_descr_generic = descr;
88  } else {
89  ac->has_optimized_func = 1;
90  }
91  found = 1;
92  }
93  break;
95  if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
96  (!channels || ac->channels == channels)) {
97  ac->conv_interleave = conv;
98  ac->func_descr = descr;
99  ac->ptr_align = ptr_align;
100  ac->samples_align = samples_align;
101  if (ptr_align == 1 && samples_align == 1) {
103  ac->func_descr_generic = descr;
104  } else {
105  ac->has_optimized_func = 1;
106  }
107  found = 1;
108  }
109  break;
111  if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
112  (!channels || ac->channels == channels)) {
113  ac->conv_deinterleave = conv;
114  ac->func_descr = descr;
115  ac->ptr_align = ptr_align;
116  ac->samples_align = samples_align;
117  if (ptr_align == 1 && samples_align == 1) {
119  ac->func_descr_generic = descr;
120  } else {
121  ac->has_optimized_func = 1;
122  }
123  found = 1;
124  }
125  break;
126  }
127  if (found) {
128  av_log(ac->avr, AV_LOG_DEBUG, "audio_convert: found function: %-4s "
129  "to %-4s (%s)\n", av_get_sample_fmt_name(ac->in_fmt),
130  av_get_sample_fmt_name(ac->out_fmt), descr);
131  }
132 }
133 
134 #define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
135 
136 #define CONV_LOOP(otype, expr) \
137  do { \
138  *(otype *)po = expr; \
139  pi += is; \
140  po += os; \
141  } while (po < end); \
142 
143 #define CONV_FUNC_FLAT(ofmt, otype, ifmt, itype, expr) \
144 static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t *in, \
145  int len) \
146 { \
147  int is = sizeof(itype); \
148  int os = sizeof(otype); \
149  const uint8_t *pi = in; \
150  uint8_t *po = out; \
151  uint8_t *end = out + os * len; \
152  CONV_LOOP(otype, expr) \
153 }
154 
155 #define CONV_FUNC_INTERLEAVE(ofmt, otype, ifmt, itype, expr) \
156 static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t **in, \
157  int len, int channels) \
158 { \
159  int ch; \
160  int out_bps = sizeof(otype); \
161  int is = sizeof(itype); \
162  int os = channels * out_bps; \
163  for (ch = 0; ch < channels; ch++) { \
164  const uint8_t *pi = in[ch]; \
165  uint8_t *po = out + ch * out_bps; \
166  uint8_t *end = po + os * len; \
167  CONV_LOOP(otype, expr) \
168  } \
169 }
170 
171 #define CONV_FUNC_DEINTERLEAVE(ofmt, otype, ifmt, itype, expr) \
172 static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t **out, const uint8_t *in, \
173  int len, int channels) \
174 { \
175  int ch; \
176  int in_bps = sizeof(itype); \
177  int is = channels * in_bps; \
178  int os = sizeof(otype); \
179  for (ch = 0; ch < channels; ch++) { \
180  const uint8_t *pi = in + ch * in_bps; \
181  uint8_t *po = out[ch]; \
182  uint8_t *end = po + os * len; \
183  CONV_LOOP(otype, expr) \
184  } \
185 }
186 
187 #define CONV_FUNC_GROUP(ofmt, otype, ifmt, itype, expr) \
188 CONV_FUNC_FLAT( ofmt, otype, ifmt, itype, expr) \
189 CONV_FUNC_INTERLEAVE( ofmt, otype, ifmt ## P, itype, expr) \
190 CONV_FUNC_DEINTERLEAVE(ofmt ## P, otype, ifmt, itype, expr)
191 
193 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 8)
194 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 24)
195 CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0f / (1 << 7)))
196 CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0 / (1 << 7)))
197 CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t, (*(const int16_t *)pi >> 8) + 0x80)
198 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi)
199 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi << 16)
200 CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0f / (1 << 15)))
201 CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0 / (1 << 15)))
202 CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t, (*(const int32_t *)pi >> 24) + 0x80)
203 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi >> 16)
204 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi)
205 CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0f / (1U << 31)))
206 CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0 / (1U << 31)))
207 CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8( lrintf(*(const float *)pi * (1 << 7)) + 0x80))
208 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16( lrintf(*(const float *)pi * (1 << 15))))
209 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *)pi * (1U << 31))))
210 CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
211 CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
212 CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8( lrint(*(const double *)pi * (1 << 7)) + 0x80))
213 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16( lrint(*(const double *)pi * (1 << 15))))
214 CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *)pi * (1U << 31))))
215 CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
216 CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
217 
218 #define SET_CONV_FUNC_GROUP(ofmt, ifmt) \
219 ff_audio_convert_set_func(ac, ofmt, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt)); \
220 ff_audio_convert_set_func(ac, ofmt ## P, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt ## P, ifmt)); \
221 ff_audio_convert_set_func(ac, ofmt, ifmt ## P, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt ## P));
222 
223 static void set_generic_function(AudioConvert *ac)
224 {
225  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8)
226  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8)
227  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8)
228  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8)
229  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8)
230  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16)
231  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16)
232  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16)
233  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16)
234  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16)
235  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32)
236  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32)
237  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32)
238  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32)
239  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32)
240  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT)
241  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT)
242  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT)
243  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT)
244  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT)
245  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL)
246  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL)
247  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL)
248  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL)
249  SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL)
250 }
251 
253 {
254  if (!*ac)
255  return;
256  ff_dither_free(&(*ac)->dc);
257  av_freep(ac);
258 }
259 
261  enum AVSampleFormat out_fmt,
262  enum AVSampleFormat in_fmt,
263  int channels, int sample_rate,
264  int apply_map)
265 {
266  AudioConvert *ac;
267  int in_planar, out_planar;
268 
269  ac = av_mallocz(sizeof(*ac));
270  if (!ac)
271  return NULL;
272 
273  ac->avr = avr;
274  ac->out_fmt = out_fmt;
275  ac->in_fmt = in_fmt;
276  ac->channels = channels;
277  ac->apply_map = apply_map;
278 
280  av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 &&
281  av_get_bytes_per_sample(in_fmt) > 2) {
282  ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate,
283  apply_map);
284  if (!ac->dc) {
285  av_free(ac);
286  return NULL;
287  }
288  return ac;
289  }
290 
291  in_planar = ff_sample_fmt_is_planar(in_fmt, channels);
292  out_planar = ff_sample_fmt_is_planar(out_fmt, channels);
293 
294  if (in_planar == out_planar) {
296  ac->planes = in_planar ? ac->channels : 1;
297  } else if (in_planar)
299  else
301 
302  set_generic_function(ac);
303 
304  if (ARCH_AARCH64)
306  if (ARCH_ARM)
308  if (ARCH_X86)
310 
311  return ac;
312 }
313 
315 {
316  int use_generic = 1;
317  int len = in->nb_samples;
318  int p;
319 
320  if (ac->dc) {
321  /* dithered conversion */
322  av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n",
323  len, av_get_sample_fmt_name(ac->in_fmt),
325 
326  return ff_convert_dither(ac->dc, out, in);
327  }
328 
329  /* determine whether to use the optimized function based on pointer and
330  samples alignment in both the input and output */
332  int ptr_align = FFMIN(in->ptr_align, out->ptr_align);
333  int samples_align = FFMIN(in->samples_align, out->samples_align);
334  int aligned_len = FFALIGN(len, ac->samples_align);
335  if (!(ptr_align % ac->ptr_align) && samples_align >= aligned_len) {
336  len = aligned_len;
337  use_generic = 0;
338  }
339  }
340  av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (%s)\n", len,
343  use_generic ? ac->func_descr_generic : ac->func_descr);
344 
345  if (ac->apply_map) {
346  ChannelMapInfo *map = &ac->avr->ch_map_info;
347 
348  if (!ff_sample_fmt_is_planar(ac->out_fmt, ac->channels)) {
349  av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n");
350  return AVERROR(EINVAL);
351  }
352 
353  if (map->do_remap) {
354  if (ff_sample_fmt_is_planar(ac->in_fmt, ac->channels)) {
355  conv_func_flat *convert = use_generic ? ac->conv_flat_generic :
356  ac->conv_flat;
357 
358  for (p = 0; p < ac->planes; p++)
359  if (map->channel_map[p] >= 0)
360  convert(out->data[p], in->data[map->channel_map[p]], len);
361  } else {
362  uint8_t *data[AVRESAMPLE_MAX_CHANNELS];
363  conv_func_deinterleave *convert = use_generic ?
365  ac->conv_deinterleave;
366 
367  for (p = 0; p < ac->channels; p++)
368  data[map->input_map[p]] = out->data[p];
369 
370  convert(data, in->data[0], len, ac->channels);
371  }
372  }
373  if (map->do_copy || map->do_zero) {
374  for (p = 0; p < ac->planes; p++) {
375  if (map->channel_copy[p])
376  memcpy(out->data[p], out->data[map->channel_copy[p]],
377  len * out->stride);
378  else if (map->channel_zero[p])
379  av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt);
380  }
381  }
382  } else {
383  switch (ac->func_type) {
384  case CONV_FUNC_TYPE_FLAT: {
385  if (!in->is_planar)
386  len *= in->channels;
387  if (use_generic) {
388  for (p = 0; p < ac->planes; p++)
389  ac->conv_flat_generic(out->data[p], in->data[p], len);
390  } else {
391  for (p = 0; p < ac->planes; p++)
392  ac->conv_flat(out->data[p], in->data[p], len);
393  }
394  break;
395  }
397  if (use_generic)
398  ac->conv_interleave_generic(out->data[0], in->data, len,
399  ac->channels);
400  else
401  ac->conv_interleave(out->data[0], in->data, len, ac->channels);
402  break;
404  if (use_generic)
405  ac->conv_deinterleave_generic(out->data, in->data[0], len,
406  ac->channels);
407  else
408  ac->conv_deinterleave(out->data, in->data[0], len,
409  ac->channels);
410  break;
411  }
412  }
413 
414  out->nb_samples = in->nb_samples;
415  return 0;
416 }
AVAudioResampleContext * avr
Definition: audio_convert.c:49
#define NULL
Definition: coverity.c:32
void ff_dither_free(DitherContext **cp)
Free a DitherContext.
Definition: dither.c:312
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static int conv(int samples, float **pcm, char *buf, int channels)
Definition: libvorbisdec.c:119
int input_map[AVRESAMPLE_MAX_CHANNELS]
dest index of each input channel
Definition: internal.h:50
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
conv_func_deinterleave * conv_deinterleave
Definition: audio_convert.c:66
memory handling functions
DitherContext * dc
Definition: audio_convert.c:50
Do not use dithering.
Definition: avresample.h:124
int do_zero
zeroing needed
Definition: internal.h:49
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
Convert audio data from one sample format to another.
void( conv_func_interleave)(uint8_t *out, uint8_t *const *in, int len, int channels)
Definition: audio_convert.c:42
void ff_audio_convert_init_x86(AudioConvert *ac)
int channel_zero[AVRESAMPLE_MAX_CHANNELS]
dest index to zero
Definition: internal.h:48
#define FFALIGN(x, a)
Definition: common.h:71
int nb_samples
current number of samples
Definition: audio_data.h:43
conv_func_interleave * conv_interleave
Definition: audio_convert.c:64
uint8_t
AV_SAMPLE_FMT_U8
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels)
Definition: audio_data.c:51
#define llrintf(x)
Definition: libm.h:117
#define lrintf(x)
Definition: libm_mips.h:70
signed 32 bits
Definition: samplefmt.h:63
AudioConvert * ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map)
Allocate and initialize AudioConvert context for sample format conversion.
#define U(x)
Definition: vp56_arith.h:37
enum AVResampleDitherMethod dither_method
dither method
Definition: internal.h:75
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:235
int stride
sample byte offset within a plane
Definition: audio_data.h:50
#define AVERROR(e)
Definition: error.h:43
void( conv_func_flat)(uint8_t *out, const uint8_t *in, int len)
Definition: audio_convert.c:40
int channels
channel count
Definition: audio_data.h:45
av_cold void ff_audio_convert_init_arm(AudioConvert *ac)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
void ff_audio_convert_free(AudioConvert **ac)
Free AudioConvert.
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:47
void( conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len, int channels)
Definition: audio_convert.c:45
DitherContext * ff_dither_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map)
Allocate and initialize a DitherContext.
Definition: dither.c:345
int is_planar
sample format is planar
Definition: audio_data.h:47
int channel_copy[AVRESAMPLE_MAX_CHANNELS]
dest index to copy from
Definition: internal.h:46
#define SET_CONV_FUNC_GROUP(ofmt, ifmt)
int has_optimized_func
Definition: audio_convert.c:58
conv_func_flat * conv_flat
Definition: audio_convert.c:62
#define FFMIN(a, b)
Definition: common.h:66
int do_remap
remap needed
Definition: internal.h:45
int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
Convert audio sample format with dithering.
Definition: dither.c:242
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int32_t
static int convert(uint8_t x)
Definition: xbmdec.c:29
#define CONV_FUNC_GROUP(ofmt, otype, ifmt, itype, expr)
conv_func_interleave * conv_interleave_generic
Definition: audio_convert.c:65
ChannelMapInfo ch_map_info
Definition: internal.h:107
sample_rate
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
const char * func_descr
Definition: audio_convert.c:59
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
Definition: audio_data.h:39
#define AVRESAMPLE_MAX_CHANNELS
Definition: avresample.h:104
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
#define llrint(x)
Definition: libm.h:112
Replacements for frequently missing libm functions.
static av_always_inline av_const long int lrint(double x)
Definition: libm.h:148
conv_func_flat * conv_flat_generic
Definition: audio_convert.c:63
int channel_map[AVRESAMPLE_MAX_CHANNELS]
source index of each output channel, -1 if not remapped
Definition: internal.h:44
ConvFuncType
Definition: audio_convert.c:34
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:104
int samples_align
allocated samples alignment
Definition: audio_data.h:54
av_log(ac->avr, AV_LOG_TRACE,"%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)
const char * func_descr_generic
Definition: audio_convert.c:60
common internal and external API header
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition: samplefmt.c:73
signed 16 bits
Definition: samplefmt.h:62
int do_copy
copy needed
Definition: internal.h:47
#define av_free(p)
int len
enum ConvFuncType func_type
Definition: audio_convert.c:61
enum AVSampleFormat out_fmt
Definition: audio_convert.c:52
conv_func_deinterleave * conv_deinterleave_generic
Definition: audio_convert.c:67
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)
Set conversion function if the parameters match.
Definition: audio_convert.c:70
enum AVSampleFormat in_fmt
Definition: audio_convert.c:51
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
av_cold void ff_audio_convert_init_aarch64(AudioConvert *ac)
#define av_freep(p)
int ptr_align
minimum data pointer alignment
Definition: audio_data.h:53
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:250