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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 |