FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
resample.c
Go to the documentation of this file.
1 /*
2  * samplerate conversion for both audio and video
3  * Copyright (c) 2000 Fabrice Bellard
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  * samplerate conversion for both audio and video
25  */
26 
27 #include <string.h>
28 
29 #include "avcodec.h"
30 #include "audioconvert.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/mem.h"
33 #include "libavutil/samplefmt.h"
34 
35 #if FF_API_AVCODEC_RESAMPLE
37 
38 #define MAX_CHANNELS 8
39 
40 struct AVResampleContext;
41 
42 static const char *context_to_name(void *ptr)
43 {
44  return "audioresample";
45 }
46 
47 static const AVOption options[] = {{NULL}};
49  "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
50 };
51 
54  short *temp[MAX_CHANNELS];
55  int temp_len;
56  float ratio;
57  /* channel convert */
60  enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
61  unsigned sample_size[2]; ///< size of one sample in sample_fmt
62  short *buffer[2]; ///< buffers used for conversion to S16
63  unsigned buffer_size[2]; ///< sizes of allocated buffers
64 };
65 
66 /* n1: number of samples */
67 static void stereo_to_mono(short *output, short *input, int n1)
68 {
69  short *p, *q;
70  int n = n1;
71 
72  p = input;
73  q = output;
74  while (n >= 4) {
75  q[0] = (p[0] + p[1]) >> 1;
76  q[1] = (p[2] + p[3]) >> 1;
77  q[2] = (p[4] + p[5]) >> 1;
78  q[3] = (p[6] + p[7]) >> 1;
79  q += 4;
80  p += 8;
81  n -= 4;
82  }
83  while (n > 0) {
84  q[0] = (p[0] + p[1]) >> 1;
85  q++;
86  p += 2;
87  n--;
88  }
89 }
90 
91 /* n1: number of samples */
92 static void mono_to_stereo(short *output, short *input, int n1)
93 {
94  short *p, *q;
95  int n = n1;
96  int v;
97 
98  p = input;
99  q = output;
100  while (n >= 4) {
101  v = p[0]; q[0] = v; q[1] = v;
102  v = p[1]; q[2] = v; q[3] = v;
103  v = p[2]; q[4] = v; q[5] = v;
104  v = p[3]; q[6] = v; q[7] = v;
105  q += 8;
106  p += 4;
107  n -= 4;
108  }
109  while (n > 0) {
110  v = p[0]; q[0] = v; q[1] = v;
111  q += 2;
112  p += 1;
113  n--;
114  }
115 }
116 
117 /*
118 5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
119 - Left = front_left + rear_gain * rear_left + center_gain * center
120 - Right = front_right + rear_gain * rear_right + center_gain * center
121 Where rear_gain is usually around 0.5-1.0 and
122  center_gain is almost always 0.7 (-3 dB)
123 */
124 static void surround_to_stereo(short **output, short *input, int channels, int samples)
125 {
126  int i;
127  short l, r;
128 
129  for (i = 0; i < samples; i++) {
130  int fl,fr,c,rl,rr;
131  fl = input[0];
132  fr = input[1];
133  c = input[2];
134  // lfe = input[3];
135  rl = input[4];
136  rr = input[5];
137 
138  l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
139  r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
140 
141  /* output l & r. */
142  *output[0]++ = l;
143  *output[1]++ = r;
144 
145  /* increment input. */
146  input += channels;
147  }
148 }
149 
150 static void deinterleave(short **output, short *input, int channels, int samples)
151 {
152  int i, j;
153 
154  for (i = 0; i < samples; i++) {
155  for (j = 0; j < channels; j++) {
156  *output[j]++ = *input++;
157  }
158  }
159 }
160 
161 static void interleave(short *output, short **input, int channels, int samples)
162 {
163  int i, j;
164 
165  for (i = 0; i < samples; i++) {
166  for (j = 0; j < channels; j++) {
167  *output++ = *input[j]++;
168  }
169  }
170 }
171 
172 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
173 {
174  int i;
175  short l, r;
176 
177  for (i = 0; i < n; i++) {
178  l = *input1++;
179  r = *input2++;
180  *output++ = l; /* left */
181  *output++ = (l / 2) + (r / 2); /* center */
182  *output++ = r; /* right */
183  *output++ = 0; /* left surround */
184  *output++ = 0; /* right surroud */
185  *output++ = 0; /* low freq */
186  }
187 }
188 
189 #define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
190  ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
191 
193  // output ch: 1 2 3 4 5 6 7 8
194  SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
195  SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
196  SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
197  SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
198  SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
199  SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
200  SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
201  SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
202 };
203 
204 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
205  int output_rate, int input_rate,
206  enum AVSampleFormat sample_fmt_out,
207  enum AVSampleFormat sample_fmt_in,
208  int filter_length, int log2_phase_count,
209  int linear, double cutoff)
210 {
212 
213  if (input_channels > MAX_CHANNELS) {
215  "Resampling with input channels greater than %d is unsupported.\n",
216  MAX_CHANNELS);
217  return NULL;
218  }
219  if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
220  int i;
221  av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
222  "output channels for %d input channel%s", input_channels,
223  input_channels > 1 ? "s:" : ":");
224  for (i = 0; i < MAX_CHANNELS; i++)
225  if (supported_resampling[input_channels-1] & (1<<i))
226  av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
227  av_log(NULL, AV_LOG_ERROR, "\n");
228  return NULL;
229  }
230 
231  s = av_mallocz(sizeof(ReSampleContext));
232  if (!s) {
233  av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
234  return NULL;
235  }
236 
237  s->ratio = (float)output_rate / (float)input_rate;
238 
239  s->input_channels = input_channels;
240  s->output_channels = output_channels;
241 
243  if (s->output_channels < s->filter_channels)
245 
246  s->sample_fmt[0] = sample_fmt_in;
247  s->sample_fmt[1] = sample_fmt_out;
250 
251  if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
253  s->sample_fmt[0], 1, NULL, 0))) {
254  av_log(s, AV_LOG_ERROR,
255  "Cannot convert %s sample format to s16 sample format\n",
257  av_free(s);
258  return NULL;
259  }
260  }
261 
262  if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
263  if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
264  AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
265  av_log(s, AV_LOG_ERROR,
266  "Cannot convert s16 sample format to %s sample format\n",
269  av_free(s);
270  return NULL;
271  }
272  }
273 
274  s->resample_context = av_resample_init(output_rate, input_rate,
275  filter_length, log2_phase_count,
276  linear, cutoff);
277 
278  *(const AVClass**)s->resample_context = &audioresample_context_class;
279 
280  return s;
281 }
282 
283 /* resample audio. 'nb_samples' is the number of input samples */
284 /* XXX: optimize it ! */
285 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
286 {
287  int i, nb_samples1;
288  short *bufin[MAX_CHANNELS];
289  short *bufout[MAX_CHANNELS];
290  short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
291  short *output_bak = NULL;
292  int lenout;
293 
294  if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
295  /* nothing to do */
296  memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
297  return nb_samples;
298  }
299 
300  if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
301  int istride[1] = { s->sample_size[0] };
302  int ostride[1] = { 2 };
303  const void *ibuf[1] = { input };
304  void *obuf[1];
305  unsigned input_size = nb_samples * s->input_channels * 2;
306 
307  if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
308  av_free(s->buffer[0]);
309  s->buffer_size[0] = input_size;
310  s->buffer[0] = av_malloc(s->buffer_size[0]);
311  if (!s->buffer[0]) {
312  av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
313  return 0;
314  }
315  }
316 
317  obuf[0] = s->buffer[0];
318 
319  if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
320  ibuf, istride, nb_samples * s->input_channels) < 0) {
322  "Audio sample format conversion failed\n");
323  return 0;
324  }
325 
326  input = s->buffer[0];
327  }
328 
329  lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
330 
331  if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
332  int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
333  s->output_channels;
334  output_bak = output;
335 
336  if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
337  av_free(s->buffer[1]);
338  s->buffer_size[1] = out_size;
339  s->buffer[1] = av_malloc(s->buffer_size[1]);
340  if (!s->buffer[1]) {
341  av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
342  return 0;
343  }
344  }
345 
346  output = s->buffer[1];
347  }
348 
349  /* XXX: move those malloc to resample init code */
350  for (i = 0; i < s->filter_channels; i++) {
351  bufin[i] = av_malloc_array((nb_samples + s->temp_len), sizeof(short));
352  bufout[i] = av_malloc_array(lenout, sizeof(short));
353 
354  if (!bufin[i] || !bufout[i]) {
355  av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
356  nb_samples1 = 0;
357  goto fail;
358  }
359 
360  memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
361  buftmp2[i] = bufin[i] + s->temp_len;
362  }
363 
364  if (s->input_channels == 2 && s->output_channels == 1) {
365  buftmp3[0] = output;
366  stereo_to_mono(buftmp2[0], input, nb_samples);
367  } else if (s->output_channels >= 2 && s->input_channels == 1) {
368  buftmp3[0] = bufout[0];
369  memcpy(buftmp2[0], input, nb_samples * sizeof(short));
370  } else if (s->input_channels == 6 && s->output_channels ==2) {
371  buftmp3[0] = bufout[0];
372  buftmp3[1] = bufout[1];
373  surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
374  } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
375  for (i = 0; i < s->input_channels; i++) {
376  buftmp3[i] = bufout[i];
377  }
378  deinterleave(buftmp2, input, s->input_channels, nb_samples);
379  } else {
380  buftmp3[0] = output;
381  memcpy(buftmp2[0], input, nb_samples * sizeof(short));
382  }
383 
384  nb_samples += s->temp_len;
385 
386  /* resample each channel */
387  nb_samples1 = 0; /* avoid warning */
388  for (i = 0; i < s->filter_channels; i++) {
389  int consumed;
390  int is_last = i + 1 == s->filter_channels;
391 
392  nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
393  &consumed, nb_samples, lenout, is_last);
394  s->temp_len = nb_samples - consumed;
395  s->temp[i] = av_realloc_array(s->temp[i], s->temp_len, sizeof(short));
396  memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
397  }
398 
399  if (s->output_channels == 2 && s->input_channels == 1) {
400  mono_to_stereo(output, buftmp3[0], nb_samples1);
401  } else if (s->output_channels == 6 && s->input_channels == 2) {
402  ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
403  } else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
404  (s->output_channels == 2 && s->input_channels == 6)) {
405  interleave(output, buftmp3, s->output_channels, nb_samples1);
406  }
407 
408  if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
409  int istride[1] = { 2 };
410  int ostride[1] = { s->sample_size[1] };
411  const void *ibuf[1] = { output };
412  void *obuf[1] = { output_bak };
413 
414  if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
415  ibuf, istride, nb_samples1 * s->output_channels) < 0) {
417  "Audio sample format conversion failed\n");
418  return 0;
419  }
420  }
421 
422 fail:
423  for (i = 0; i < s->filter_channels; i++) {
424  av_free(bufin[i]);
425  av_free(bufout[i]);
426  }
427 
428  return nb_samples1;
429 }
430 
432 {
433  int i;
435  for (i = 0; i < s->filter_channels; i++)
436  av_freep(&s->temp[i]);
437  av_freep(&s->buffer[0]);
438  av_freep(&s->buffer[1]);
441  av_free(s);
442 }
443 
445 #endif
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:631
static const uint8_t supported_resampling[MAX_CHANNELS]
Definition: resample.c:192
static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
Definition: resample.c:172
AVOption.
Definition: opt.h:245
AVAudioConvert * convert_ctx[2]
Definition: resample.c:59
static void deinterleave(short **output, short *input, int channels, int samples)
Definition: resample.c:150
static int linear(InterplayACMContext *s, unsigned ind, unsigned col)
Definition: interplayacm.c:115
#define LIBAVUTIL_VERSION_INT
Definition: version.h:70
memory handling functions
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
Definition: resample.c:285
static void stereo_to_mono(short *output, short *input, int n1)
Definition: resample.c:67
attribute_deprecated int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx)
Resample an array of samples using a previously configured context.
Definition: resample2.c:237
static const char * context_to_name(void *ptr)
Definition: resample.c:42
uint8_t
#define av_malloc(s)
static void surround_to_stereo(short **output, short *input, int channels, int samples)
Definition: resample.c:124
AVOptions.
static void mono_to_stereo(short *output, short *input, int n1)
Definition: resample.c:92
attribute_deprecated void av_resample_close(struct AVResampleContext *c)
Definition: resample2.c:226
#define av_log(a,...)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int av_audio_convert(AVAudioConvert *ctx, void *const out[6], const int out_stride[6], const void *const in[6], const int in_stride[6], int len)
Convert between audio sample formats.
Definition: audioconvert.c:63
enum AVSampleFormat sample_fmt[2]
input and output sample format
Definition: resample.c:60
const char * r
Definition: vf_curves.c:107
int out_size
Definition: movenc-test.c:55
static const AVClass audioresample_context_class
Definition: resample.c:48
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
#define fail()
Definition: checkasm.h:80
short * buffer[2]
buffers used for conversion to S16
Definition: resample.c:62
unsigned buffer_size[2]
sizes of allocated buffers
Definition: resample.c:63
int filter_channels
Definition: resample.c:58
short * temp[MAX_CHANNELS]
Definition: resample.c:54
int n
Definition: avisynth_c.h:547
#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8)
Definition: resample.c:189
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
int output_channels
Definition: resample.c:58
ReSampleContext * av_audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate, enum AVSampleFormat sample_fmt_out, enum AVSampleFormat sample_fmt_in, int filter_length, int log2_phase_count, int linear, double cutoff)
Initialize audio resampling context.
Definition: resample.c:204
Describe the class of an AVClass context structure.
Definition: log.h:67
void audio_resample_close(ReSampleContext *s)
Free resample context.
Definition: resample.c:431
unsigned sample_size[2]
size of one sample in sample_fmt
Definition: resample.c:61
static void interleave(short *output, short **input, int channels, int samples)
Definition: resample.c:161
int input_channels
Definition: resample.c:58
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:104
Audio format conversion routines This interface is deprecated and will be dropped in a future version...
attribute_deprecated struct AVResampleContext * av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff)
Initialize an audio resampler.
Definition: resample2.c:192
#define FF_DISABLE_DEPRECATION_WARNINGS
Definition: internal.h:80
AVAudioConvert * av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels, enum AVSampleFormat in_fmt, int in_channels, const float *matrix, int flags)
Create an audio sample format converter context.
Definition: audioconvert.c:42
signed 16 bits
Definition: samplefmt.h:62
static double c[64]
static const AVOption options[]
Definition: resample.c:47
struct AVResampleContext * resample_context
Definition: resample.c:53
void * av_realloc_array(void *ptr, size_t nmemb, size_t size)
Definition: mem.c:208
#define av_free(p)
#define FF_ENABLE_DEPRECATION_WARNINGS
Definition: internal.h:81
#define av_freep(p)
#define av_malloc_array(a, b)
void av_audio_convert_free(AVAudioConvert *ctx)
Free audio sample format converter context.
Definition: audioconvert.c:58
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:252
#define MAX_CHANNELS
Definition: resample.c:38
GLuint buffer
Definition: opengl_enc.c:102