10l fix by (Bohdan Horst <nexus at irc dot pl>)
[libav.git] / libavcodec / resample.c
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1/*
2 * Sample rate convertion for both audio and video
ff4ec49e 3 * Copyright (c) 2000 Fabrice Bellard.
de6d9b64 4 *
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5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
de6d9b64 9 *
ff4ec49e 10 * This library is distributed in the hope that it will be useful,
de6d9b64 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
de6d9b64 14 *
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15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
de6d9b64 18 */
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19
20/**
21 * @file resample.c
22 * Sample rate convertion for both audio and video.
23 */
24
de6d9b64 25#include "avcodec.h"
69db4e10 26
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27typedef struct {
28 /* fractional resampling */
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29 uint32_t incr; /* fractional increment */
30 uint32_t frac;
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31 int last_sample;
32 /* integer down sample */
33 int iratio; /* integer divison ratio */
34 int icount, isum;
35 int inv;
36} ReSampleChannelContext;
37
38struct ReSampleContext {
39 ReSampleChannelContext channel_ctx[2];
40 float ratio;
41 /* channel convert */
42 int input_channels, output_channels, filter_channels;
43};
44
45
46#define FRAC_BITS 16
47#define FRAC (1 << FRAC_BITS)
48
49static void init_mono_resample(ReSampleChannelContext *s, float ratio)
50{
51 ratio = 1.0 / ratio;
5c91a675 52 s->iratio = (int)floorf(ratio);
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53 if (s->iratio == 0)
54 s->iratio = 1;
55 s->incr = (int)((ratio / s->iratio) * FRAC);
8170f3dc 56 s->frac = FRAC;
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57 s->last_sample = 0;
58 s->icount = s->iratio;
59 s->isum = 0;
60 s->inv = (FRAC / s->iratio);
61}
62
63/* fractional audio resampling */
64static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
65{
66 unsigned int frac, incr;
67 int l0, l1;
68 short *q, *p, *pend;
69
70 l0 = s->last_sample;
71 incr = s->incr;
72 frac = s->frac;
73
74 p = input;
75 pend = input + nb_samples;
76 q = output;
77
78 l1 = *p++;
79 for(;;) {
80 /* interpolate */
81 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
82 frac = frac + s->incr;
83 while (frac >= FRAC) {
9c89585a 84 frac -= FRAC;
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85 if (p >= pend)
86 goto the_end;
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87 l0 = l1;
88 l1 = *p++;
89 }
90 }
91 the_end:
92 s->last_sample = l1;
93 s->frac = frac;
94 return q - output;
95}
96
97static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
98{
99 short *q, *p, *pend;
100 int c, sum;
101
102 p = input;
103 pend = input + nb_samples;
104 q = output;
105
106 c = s->icount;
107 sum = s->isum;
108
109 for(;;) {
110 sum += *p++;
111 if (--c == 0) {
112 *q++ = (sum * s->inv) >> FRAC_BITS;
113 c = s->iratio;
114 sum = 0;
115 }
116 if (p >= pend)
117 break;
118 }
119 s->isum = sum;
120 s->icount = c;
121 return q - output;
122}
123
124/* n1: number of samples */
125static void stereo_to_mono(short *output, short *input, int n1)
126{
127 short *p, *q;
128 int n = n1;
129
130 p = input;
131 q = output;
132 while (n >= 4) {
133 q[0] = (p[0] + p[1]) >> 1;
134 q[1] = (p[2] + p[3]) >> 1;
135 q[2] = (p[4] + p[5]) >> 1;
136 q[3] = (p[6] + p[7]) >> 1;
137 q += 4;
138 p += 8;
139 n -= 4;
140 }
141 while (n > 0) {
142 q[0] = (p[0] + p[1]) >> 1;
143 q++;
144 p += 2;
145 n--;
146 }
147}
148
149/* n1: number of samples */
150static void mono_to_stereo(short *output, short *input, int n1)
151{
152 short *p, *q;
153 int n = n1;
154 int v;
155
156 p = input;
157 q = output;
158 while (n >= 4) {
159 v = p[0]; q[0] = v; q[1] = v;
160 v = p[1]; q[2] = v; q[3] = v;
161 v = p[2]; q[4] = v; q[5] = v;
162 v = p[3]; q[6] = v; q[7] = v;
163 q += 8;
164 p += 4;
165 n -= 4;
166 }
167 while (n > 0) {
168 v = p[0]; q[0] = v; q[1] = v;
169 q += 2;
170 p += 1;
171 n--;
172 }
173}
174
175/* XXX: should use more abstract 'N' channels system */
176static void stereo_split(short *output1, short *output2, short *input, int n)
177{
178 int i;
179
180 for(i=0;i<n;i++) {
181 *output1++ = *input++;
182 *output2++ = *input++;
183 }
184}
185
186static void stereo_mux(short *output, short *input1, short *input2, int n)
187{
188 int i;
189
190 for(i=0;i<n;i++) {
191 *output++ = *input1++;
192 *output++ = *input2++;
193 }
194}
195
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196static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
197{
198 int i;
199 short l,r;
200
201 for(i=0;i<n;i++) {
202 l=*input1++;
203 r=*input2++;
204 *output++ = l; /* left */
205 *output++ = (l/2)+(r/2); /* center */
206 *output++ = r; /* right */
207 *output++ = 0; /* left surround */
208 *output++ = 0; /* right surroud */
209 *output++ = 0; /* low freq */
210 }
211}
212
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213static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
214{
1a565432 215 short *buf1;
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216 short *buftmp;
217
6000abfa 218 buf1= (short*)av_malloc( nb_samples * sizeof(short) );
1a565432 219
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220 /* first downsample by an integer factor with averaging filter */
221 if (s->iratio > 1) {
222 buftmp = buf1;
223 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
224 } else {
225 buftmp = input;
226 }
227
228 /* then do a fractional resampling with linear interpolation */
229 if (s->incr != FRAC) {
230 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
231 } else {
232 memcpy(output, buftmp, nb_samples * sizeof(short));
233 }
6000abfa 234 av_free(buf1);
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235 return nb_samples;
236}
237
238ReSampleContext *audio_resample_init(int output_channels, int input_channels,
239 int output_rate, int input_rate)
240{
241 ReSampleContext *s;
242 int i;
243
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244 if ( input_channels > 2)
245 {
9b879566 246 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
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247 return NULL;
248 }
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249
250 s = av_mallocz(sizeof(ReSampleContext));
251 if (!s)
743739d2 252 {
9b879566 253 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
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254 return NULL;
255 }
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256
257 s->ratio = (float)output_rate / (float)input_rate;
258
259 s->input_channels = input_channels;
260 s->output_channels = output_channels;
261
262 s->filter_channels = s->input_channels;
263 if (s->output_channels < s->filter_channels)
264 s->filter_channels = s->output_channels;
265
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266/*
267 * ac3 output is the only case where filter_channels could be greater than 2.
268 * input channels can't be greater than 2, so resample the 2 channels and then
269 * expand to 6 channels after the resampling.
270 */
271 if(s->filter_channels>2)
272 s->filter_channels = 2;
273
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274 for(i=0;i<s->filter_channels;i++) {
275 init_mono_resample(&s->channel_ctx[i], s->ratio);
276 }
277 return s;
278}
279
280/* resample audio. 'nb_samples' is the number of input samples */
281/* XXX: optimize it ! */
282/* XXX: do it with polyphase filters, since the quality here is
283 HORRIBLE. Return the number of samples available in output */
284int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
285{
286 int i, nb_samples1;
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287 short *bufin[2];
288 short *bufout[2];
de6d9b64 289 short *buftmp2[2], *buftmp3[2];
1a565432 290 int lenout;
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291
292 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
293 /* nothing to do */
294 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
295 return nb_samples;
296 }
297
1a565432 298 /* XXX: move those malloc to resample init code */
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299 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
300 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
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301
302 /* make some zoom to avoid round pb */
303 lenout= (int)(nb_samples * s->ratio) + 16;
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304 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
305 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
1a565432 306
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307 if (s->input_channels == 2 &&
308 s->output_channels == 1) {
309 buftmp2[0] = bufin[0];
310 buftmp3[0] = output;
311 stereo_to_mono(buftmp2[0], input, nb_samples);
743739d2 312 } else if (s->output_channels >= 2 && s->input_channels == 1) {
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313 buftmp2[0] = input;
314 buftmp3[0] = bufout[0];
743739d2 315 } else if (s->output_channels >= 2) {
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316 buftmp2[0] = bufin[0];
317 buftmp2[1] = bufin[1];
318 buftmp3[0] = bufout[0];
319 buftmp3[1] = bufout[1];
320 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
321 } else {
322 buftmp2[0] = input;
323 buftmp3[0] = output;
324 }
325
326 /* resample each channel */
327 nb_samples1 = 0; /* avoid warning */
328 for(i=0;i<s->filter_channels;i++) {
329 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
330 }
331
332 if (s->output_channels == 2 && s->input_channels == 1) {
333 mono_to_stereo(output, buftmp3[0], nb_samples1);
334 } else if (s->output_channels == 2) {
335 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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336 } else if (s->output_channels == 6) {
337 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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338 }
339
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340 av_free(bufin[0]);
341 av_free(bufin[1]);
1a565432 342
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343 av_free(bufout[0]);
344 av_free(bufout[1]);
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345 return nb_samples1;
346}
347
348void audio_resample_close(ReSampleContext *s)
349{
6000abfa 350 av_free(s);
de6d9b64 351}