Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / resample.c
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
36
37 #define MAX_CHANNELS 8
38
39 struct AVResampleContext;
40
41 static const char *context_to_name(void *ptr)
42 {
43 return "audioresample";
44 }
45
46 static const AVOption options[] = {{NULL}};
47 static const AVClass audioresample_context_class = {
48 "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
49 };
50
51 struct ReSampleContext {
52 struct AVResampleContext *resample_context;
53 short *temp[MAX_CHANNELS];
54 int temp_len;
55 float ratio;
56 /* channel convert */
57 int input_channels, output_channels, filter_channels;
58 AVAudioConvert *convert_ctx[2];
59 enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
60 unsigned sample_size[2]; ///< size of one sample in sample_fmt
61 short *buffer[2]; ///< buffers used for conversion to S16
62 unsigned buffer_size[2]; ///< sizes of allocated buffers
63 };
64
65 /* n1: number of samples */
66 static void stereo_to_mono(short *output, short *input, int n1)
67 {
68 short *p, *q;
69 int n = n1;
70
71 p = input;
72 q = output;
73 while (n >= 4) {
74 q[0] = (p[0] + p[1]) >> 1;
75 q[1] = (p[2] + p[3]) >> 1;
76 q[2] = (p[4] + p[5]) >> 1;
77 q[3] = (p[6] + p[7]) >> 1;
78 q += 4;
79 p += 8;
80 n -= 4;
81 }
82 while (n > 0) {
83 q[0] = (p[0] + p[1]) >> 1;
84 q++;
85 p += 2;
86 n--;
87 }
88 }
89
90 /* n1: number of samples */
91 static void mono_to_stereo(short *output, short *input, int n1)
92 {
93 short *p, *q;
94 int n = n1;
95 int v;
96
97 p = input;
98 q = output;
99 while (n >= 4) {
100 v = p[0]; q[0] = v; q[1] = v;
101 v = p[1]; q[2] = v; q[3] = v;
102 v = p[2]; q[4] = v; q[5] = v;
103 v = p[3]; q[6] = v; q[7] = v;
104 q += 8;
105 p += 4;
106 n -= 4;
107 }
108 while (n > 0) {
109 v = p[0]; q[0] = v; q[1] = v;
110 q += 2;
111 p += 1;
112 n--;
113 }
114 }
115
116 /*
117 5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
118 - Left = front_left + rear_gain * rear_left + center_gain * center
119 - Right = front_right + rear_gain * rear_right + center_gain * center
120 Where rear_gain is usually around 0.5-1.0 and
121 center_gain is almost always 0.7 (-3 dB)
122 */
123 static void surround_to_stereo(short **output, short *input, int channels, int samples)
124 {
125 int i;
126 short l, r;
127
128 for (i = 0; i < samples; i++) {
129 int fl,fr,c,rl,rr;
130 fl = input[0];
131 fr = input[1];
132 c = input[2];
133 // lfe = input[3];
134 rl = input[4];
135 rr = input[5];
136
137 l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
138 r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
139
140 /* output l & r. */
141 *output[0]++ = l;
142 *output[1]++ = r;
143
144 /* increment input. */
145 input += channels;
146 }
147 }
148
149 static void deinterleave(short **output, short *input, int channels, int samples)
150 {
151 int i, j;
152
153 for (i = 0; i < samples; i++) {
154 for (j = 0; j < channels; j++) {
155 *output[j]++ = *input++;
156 }
157 }
158 }
159
160 static void interleave(short *output, short **input, int channels, int samples)
161 {
162 int i, j;
163
164 for (i = 0; i < samples; i++) {
165 for (j = 0; j < channels; j++) {
166 *output++ = *input[j]++;
167 }
168 }
169 }
170
171 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
172 {
173 int i;
174 short l, r;
175
176 for (i = 0; i < n; i++) {
177 l = *input1++;
178 r = *input2++;
179 *output++ = l; /* left */
180 *output++ = (l / 2) + (r / 2); /* center */
181 *output++ = r; /* right */
182 *output++ = 0; /* left surround */
183 *output++ = 0; /* right surroud */
184 *output++ = 0; /* low freq */
185 }
186 }
187
188 #define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
189 ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
190
191 static const uint8_t supported_resampling[MAX_CHANNELS] = {
192 // output ch: 1 2 3 4 5 6 7 8
193 SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
194 SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
195 SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
196 SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
197 SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
198 SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
199 SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
200 SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
201 };
202
203 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
204 int output_rate, int input_rate,
205 enum AVSampleFormat sample_fmt_out,
206 enum AVSampleFormat sample_fmt_in,
207 int filter_length, int log2_phase_count,
208 int linear, double cutoff)
209 {
210 ReSampleContext *s;
211
212 if (input_channels > MAX_CHANNELS) {
213 av_log(NULL, AV_LOG_ERROR,
214 "Resampling with input channels greater than %d is unsupported.\n",
215 MAX_CHANNELS);
216 return NULL;
217 }
218 if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
219 int i;
220 av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
221 "output channels for %d input channel%s", input_channels,
222 input_channels > 1 ? "s:" : ":");
223 for (i = 0; i < MAX_CHANNELS; i++)
224 if (supported_resampling[input_channels-1] & (1<<i))
225 av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
226 av_log(NULL, AV_LOG_ERROR, "\n");
227 return NULL;
228 }
229
230 s = av_mallocz(sizeof(ReSampleContext));
231 if (!s) {
232 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
233 return NULL;
234 }
235
236 s->ratio = (float)output_rate / (float)input_rate;
237
238 s->input_channels = input_channels;
239 s->output_channels = output_channels;
240
241 s->filter_channels = s->input_channels;
242 if (s->output_channels < s->filter_channels)
243 s->filter_channels = s->output_channels;
244
245 s->sample_fmt[0] = sample_fmt_in;
246 s->sample_fmt[1] = sample_fmt_out;
247 s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
248 s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
249
250 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
251 if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
252 s->sample_fmt[0], 1, NULL, 0))) {
253 av_log(s, AV_LOG_ERROR,
254 "Cannot convert %s sample format to s16 sample format\n",
255 av_get_sample_fmt_name(s->sample_fmt[0]));
256 av_free(s);
257 return NULL;
258 }
259 }
260
261 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
262 if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
263 AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
264 av_log(s, AV_LOG_ERROR,
265 "Cannot convert s16 sample format to %s sample format\n",
266 av_get_sample_fmt_name(s->sample_fmt[1]));
267 av_audio_convert_free(s->convert_ctx[0]);
268 av_free(s);
269 return NULL;
270 }
271 }
272
273 s->resample_context = av_resample_init(output_rate, input_rate,
274 filter_length, log2_phase_count,
275 linear, cutoff);
276
277 *(const AVClass**)s->resample_context = &audioresample_context_class;
278
279 return s;
280 }
281
282 /* resample audio. 'nb_samples' is the number of input samples */
283 /* XXX: optimize it ! */
284 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
285 {
286 int i, nb_samples1;
287 short *bufin[MAX_CHANNELS];
288 short *bufout[MAX_CHANNELS];
289 short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
290 short *output_bak = NULL;
291 int lenout;
292
293 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
294 /* nothing to do */
295 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
296 return nb_samples;
297 }
298
299 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
300 int istride[1] = { s->sample_size[0] };
301 int ostride[1] = { 2 };
302 const void *ibuf[1] = { input };
303 void *obuf[1];
304 unsigned input_size = nb_samples * s->input_channels * 2;
305
306 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
307 av_free(s->buffer[0]);
308 s->buffer_size[0] = input_size;
309 s->buffer[0] = av_malloc(s->buffer_size[0]);
310 if (!s->buffer[0]) {
311 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
312 return 0;
313 }
314 }
315
316 obuf[0] = s->buffer[0];
317
318 if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
319 ibuf, istride, nb_samples * s->input_channels) < 0) {
320 av_log(s->resample_context, AV_LOG_ERROR,
321 "Audio sample format conversion failed\n");
322 return 0;
323 }
324
325 input = s->buffer[0];
326 }
327
328 lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
329
330 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
331 int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
332 s->output_channels;
333 output_bak = output;
334
335 if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
336 av_free(s->buffer[1]);
337 s->buffer_size[1] = out_size;
338 s->buffer[1] = av_malloc(s->buffer_size[1]);
339 if (!s->buffer[1]) {
340 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
341 return 0;
342 }
343 }
344
345 output = s->buffer[1];
346 }
347
348 /* XXX: move those malloc to resample init code */
349 for (i = 0; i < s->filter_channels; i++) {
350 bufin[i] = av_malloc_array((nb_samples + s->temp_len), sizeof(short));
351 bufout[i] = av_malloc_array(lenout, sizeof(short));
352
353 if (!bufin[i] || !bufout[i]) {
354 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
355 nb_samples1 = 0;
356 goto fail;
357 }
358
359 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
360 buftmp2[i] = bufin[i] + s->temp_len;
361 }
362
363 if (s->input_channels == 2 && s->output_channels == 1) {
364 buftmp3[0] = output;
365 stereo_to_mono(buftmp2[0], input, nb_samples);
366 } else if (s->output_channels >= 2 && s->input_channels == 1) {
367 buftmp3[0] = bufout[0];
368 memcpy(buftmp2[0], input, nb_samples * sizeof(short));
369 } else if (s->input_channels == 6 && s->output_channels ==2) {
370 buftmp3[0] = bufout[0];
371 buftmp3[1] = bufout[1];
372 surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
373 } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
374 for (i = 0; i < s->input_channels; i++) {
375 buftmp3[i] = bufout[i];
376 }
377 deinterleave(buftmp2, input, s->input_channels, nb_samples);
378 } else {
379 buftmp3[0] = output;
380 memcpy(buftmp2[0], input, nb_samples * sizeof(short));
381 }
382
383 nb_samples += s->temp_len;
384
385 /* resample each channel */
386 nb_samples1 = 0; /* avoid warning */
387 for (i = 0; i < s->filter_channels; i++) {
388 int consumed;
389 int is_last = i + 1 == s->filter_channels;
390
391 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
392 &consumed, nb_samples, lenout, is_last);
393 s->temp_len = nb_samples - consumed;
394 s->temp[i] = av_realloc_array(s->temp[i], s->temp_len, sizeof(short));
395 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
396 }
397
398 if (s->output_channels == 2 && s->input_channels == 1) {
399 mono_to_stereo(output, buftmp3[0], nb_samples1);
400 } else if (s->output_channels == 6 && s->input_channels == 2) {
401 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
402 } else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
403 (s->output_channels == 2 && s->input_channels == 6)) {
404 interleave(output, buftmp3, s->output_channels, nb_samples1);
405 }
406
407 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
408 int istride[1] = { 2 };
409 int ostride[1] = { s->sample_size[1] };
410 const void *ibuf[1] = { output };
411 void *obuf[1] = { output_bak };
412
413 if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
414 ibuf, istride, nb_samples1 * s->output_channels) < 0) {
415 av_log(s->resample_context, AV_LOG_ERROR,
416 "Audio sample format conversion failed\n");
417 return 0;
418 }
419 }
420
421 fail:
422 for (i = 0; i < s->filter_channels; i++) {
423 av_free(bufin[i]);
424 av_free(bufout[i]);
425 }
426
427 return nb_samples1;
428 }
429
430 void audio_resample_close(ReSampleContext *s)
431 {
432 int i;
433 av_resample_close(s->resample_context);
434 for (i = 0; i < s->filter_channels; i++)
435 av_freep(&s->temp[i]);
436 av_freep(&s->buffer[0]);
437 av_freep(&s->buffer[1]);
438 av_audio_convert_free(s->convert_ctx[0]);
439 av_audio_convert_free(s->convert_ctx[1]);
440 av_free(s);
441 }
442
443 #endif