summaryrefslogtreecommitdiff
path: root/libavcodec/resample.c (plain)
blob: 4c5eb9f10e67234e9f6918fc97f01898b5787e9c
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
36FF_DISABLE_DEPRECATION_WARNINGS
37
38#define MAX_CHANNELS 8
39
40struct AVResampleContext;
41
42static const char *context_to_name(void *ptr)
43{
44 return "audioresample";
45}
46
47static const AVOption options[] = {{NULL}};
48static const AVClass audioresample_context_class = {
49 "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
50};
51
52struct ReSampleContext {
53 struct AVResampleContext *resample_context;
54 short *temp[MAX_CHANNELS];
55 int temp_len;
56 float ratio;
57 /* channel convert */
58 int input_channels, output_channels, filter_channels;
59 AVAudioConvert *convert_ctx[2];
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 */
67static 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 */
92static 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/*
1185.1 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
121Where rear_gain is usually around 0.5-1.0 and
122 center_gain is almost always 0.7 (-3 dB)
123*/
124static 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
150static 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
161static 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
172static 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
192static const uint8_t supported_resampling[MAX_CHANNELS] = {
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
204ReSampleContext *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{
211 ReSampleContext *s;
212
213 if (input_channels > MAX_CHANNELS) {
214 av_log(NULL, AV_LOG_ERROR,
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
242 s->filter_channels = s->input_channels;
243 if (s->output_channels < s->filter_channels)
244 s->filter_channels = s->output_channels;
245
246 s->sample_fmt[0] = sample_fmt_in;
247 s->sample_fmt[1] = sample_fmt_out;
248 s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
249 s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
250
251 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
252 if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
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",
256 av_get_sample_fmt_name(s->sample_fmt[0]));
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",
267 av_get_sample_fmt_name(s->sample_fmt[1]));
268 av_audio_convert_free(s->convert_ctx[0]);
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 ! */
285int 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->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
295 int istride[1] = { s->sample_size[0] };
296 int ostride[1] = { 2 };
297 const void *ibuf[1] = { input };
298 void *obuf[1];
299 unsigned input_size = nb_samples * s->input_channels * 2;
300
301 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
302 av_free(s->buffer[0]);
303 s->buffer_size[0] = input_size;
304 s->buffer[0] = av_malloc(s->buffer_size[0]);
305 if (!s->buffer[0]) {
306 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
307 return 0;
308 }
309 }
310
311 obuf[0] = s->buffer[0];
312
313 if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
314 ibuf, istride, nb_samples * s->input_channels) < 0) {
315 av_log(s->resample_context, AV_LOG_ERROR,
316 "Audio sample format conversion failed\n");
317 return 0;
318 }
319
320 input = s->buffer[0];
321 }
322
323 lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
324
325 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
326 int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
327 s->output_channels;
328 output_bak = output;
329
330 if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
331 av_free(s->buffer[1]);
332 s->buffer_size[1] = out_size;
333 s->buffer[1] = av_malloc(s->buffer_size[1]);
334 if (!s->buffer[1]) {
335 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
336 return 0;
337 }
338 }
339
340 output = s->buffer[1];
341 }
342
343 /* XXX: move those malloc to resample init code */
344 for (i = 0; i < s->filter_channels; i++) {
345 bufin[i] = av_malloc_array((nb_samples + s->temp_len), sizeof(short));
346 bufout[i] = av_malloc_array(lenout, sizeof(short));
347
348 if (!bufin[i] || !bufout[i]) {
349 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
350 nb_samples1 = 0;
351 goto fail;
352 }
353
354 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
355 buftmp2[i] = bufin[i] + s->temp_len;
356 }
357
358 if (s->input_channels == 2 && s->output_channels == 1) {
359 buftmp3[0] = output;
360 stereo_to_mono(buftmp2[0], input, nb_samples);
361 } else if (s->output_channels >= 2 && s->input_channels == 1) {
362 buftmp3[0] = bufout[0];
363 memcpy(buftmp2[0], input, nb_samples * sizeof(short));
364 } else if (s->input_channels == 6 && s->output_channels ==2) {
365 buftmp3[0] = bufout[0];
366 buftmp3[1] = bufout[1];
367 surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
368 } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
369 for (i = 0; i < s->input_channels; i++) {
370 buftmp3[i] = bufout[i];
371 }
372 deinterleave(buftmp2, input, s->input_channels, nb_samples);
373 } else {
374 buftmp3[0] = output;
375 memcpy(buftmp2[0], input, nb_samples * sizeof(short));
376 }
377
378 nb_samples += s->temp_len;
379
380 /* resample each channel */
381 nb_samples1 = 0; /* avoid warning */
382 for (i = 0; i < s->filter_channels; i++) {
383 int consumed;
384 int is_last = i + 1 == s->filter_channels;
385
386 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
387 &consumed, nb_samples, lenout, is_last);
388 s->temp_len = nb_samples - consumed;
389 s->temp[i] = av_realloc_array(s->temp[i], s->temp_len, sizeof(short));
390 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
391 }
392
393 if (s->output_channels == 2 && s->input_channels == 1) {
394 mono_to_stereo(output, buftmp3[0], nb_samples1);
395 } else if (s->output_channels == 6 && s->input_channels == 2) {
396 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
397 } else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
398 (s->output_channels == 2 && s->input_channels == 6)) {
399 interleave(output, buftmp3, s->output_channels, nb_samples1);
400 }
401
402 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
403 int istride[1] = { 2 };
404 int ostride[1] = { s->sample_size[1] };
405 const void *ibuf[1] = { output };
406 void *obuf[1] = { output_bak };
407
408 if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
409 ibuf, istride, nb_samples1 * s->output_channels) < 0) {
410 av_log(s->resample_context, AV_LOG_ERROR,
411 "Audio sample format conversion failed\n");
412 return 0;
413 }
414 }
415
416fail:
417 for (i = 0; i < s->filter_channels; i++) {
418 av_free(bufin[i]);
419 av_free(bufout[i]);
420 }
421
422 return nb_samples1;
423}
424
425void audio_resample_close(ReSampleContext *s)
426{
427 int i;
428 av_resample_close(s->resample_context);
429 for (i = 0; i < s->filter_channels; i++)
430 av_freep(&s->temp[i]);
431 av_freep(&s->buffer[0]);
432 av_freep(&s->buffer[1]);
433 av_audio_convert_free(s->convert_ctx[0]);
434 av_audio_convert_free(s->convert_ctx[1]);
435 av_free(s);
436}
437
438FF_ENABLE_DEPRECATION_WARNINGS
439#endif
440
generated by cgit at 2024-05-17 06:28:15 (GMT)