consistent include usage
[libav.git] / libavcodec / resample.c
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1/*
2 * Sample rate convertion for both audio and video
3 * Copyright (c) 2000 Gerard Lantau.
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
9 *
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
14 *
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
18 */
de6d9b64 19#include "avcodec.h"
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20
21typedef struct {
22 /* fractional resampling */
23 UINT32 incr; /* fractional increment */
24 UINT32 frac;
25 int last_sample;
26 /* integer down sample */
27 int iratio; /* integer divison ratio */
28 int icount, isum;
29 int inv;
30} ReSampleChannelContext;
31
32struct ReSampleContext {
33 ReSampleChannelContext channel_ctx[2];
34 float ratio;
35 /* channel convert */
36 int input_channels, output_channels, filter_channels;
37};
38
39
40#define FRAC_BITS 16
41#define FRAC (1 << FRAC_BITS)
42
43static void init_mono_resample(ReSampleChannelContext *s, float ratio)
44{
45 ratio = 1.0 / ratio;
46 s->iratio = (int)floor(ratio);
47 if (s->iratio == 0)
48 s->iratio = 1;
49 s->incr = (int)((ratio / s->iratio) * FRAC);
8170f3dc 50 s->frac = FRAC;
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51 s->last_sample = 0;
52 s->icount = s->iratio;
53 s->isum = 0;
54 s->inv = (FRAC / s->iratio);
55}
56
57/* fractional audio resampling */
58static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
59{
60 unsigned int frac, incr;
61 int l0, l1;
62 short *q, *p, *pend;
63
64 l0 = s->last_sample;
65 incr = s->incr;
66 frac = s->frac;
67
68 p = input;
69 pend = input + nb_samples;
70 q = output;
71
72 l1 = *p++;
73 for(;;) {
74 /* interpolate */
75 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
76 frac = frac + s->incr;
77 while (frac >= FRAC) {
78 if (p >= pend)
79 goto the_end;
80 frac -= FRAC;
81 l0 = l1;
82 l1 = *p++;
83 }
84 }
85 the_end:
86 s->last_sample = l1;
87 s->frac = frac;
88 return q - output;
89}
90
91static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
92{
93 short *q, *p, *pend;
94 int c, sum;
95
96 p = input;
97 pend = input + nb_samples;
98 q = output;
99
100 c = s->icount;
101 sum = s->isum;
102
103 for(;;) {
104 sum += *p++;
105 if (--c == 0) {
106 *q++ = (sum * s->inv) >> FRAC_BITS;
107 c = s->iratio;
108 sum = 0;
109 }
110 if (p >= pend)
111 break;
112 }
113 s->isum = sum;
114 s->icount = c;
115 return q - output;
116}
117
118/* n1: number of samples */
119static void stereo_to_mono(short *output, short *input, int n1)
120{
121 short *p, *q;
122 int n = n1;
123
124 p = input;
125 q = output;
126 while (n >= 4) {
127 q[0] = (p[0] + p[1]) >> 1;
128 q[1] = (p[2] + p[3]) >> 1;
129 q[2] = (p[4] + p[5]) >> 1;
130 q[3] = (p[6] + p[7]) >> 1;
131 q += 4;
132 p += 8;
133 n -= 4;
134 }
135 while (n > 0) {
136 q[0] = (p[0] + p[1]) >> 1;
137 q++;
138 p += 2;
139 n--;
140 }
141}
142
143/* n1: number of samples */
144static void mono_to_stereo(short *output, short *input, int n1)
145{
146 short *p, *q;
147 int n = n1;
148 int v;
149
150 p = input;
151 q = output;
152 while (n >= 4) {
153 v = p[0]; q[0] = v; q[1] = v;
154 v = p[1]; q[2] = v; q[3] = v;
155 v = p[2]; q[4] = v; q[5] = v;
156 v = p[3]; q[6] = v; q[7] = v;
157 q += 8;
158 p += 4;
159 n -= 4;
160 }
161 while (n > 0) {
162 v = p[0]; q[0] = v; q[1] = v;
163 q += 2;
164 p += 1;
165 n--;
166 }
167}
168
169/* XXX: should use more abstract 'N' channels system */
170static void stereo_split(short *output1, short *output2, short *input, int n)
171{
172 int i;
173
174 for(i=0;i<n;i++) {
175 *output1++ = *input++;
176 *output2++ = *input++;
177 }
178}
179
180static void stereo_mux(short *output, short *input1, short *input2, int n)
181{
182 int i;
183
184 for(i=0;i<n;i++) {
185 *output++ = *input1++;
186 *output++ = *input2++;
187 }
188}
189
190static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
191{
1a565432 192 short *buf1;
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193 short *buftmp;
194
6000abfa 195 buf1= (short*)av_malloc( nb_samples * sizeof(short) );
1a565432 196
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197 /* first downsample by an integer factor with averaging filter */
198 if (s->iratio > 1) {
199 buftmp = buf1;
200 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
201 } else {
202 buftmp = input;
203 }
204
205 /* then do a fractional resampling with linear interpolation */
206 if (s->incr != FRAC) {
207 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
208 } else {
209 memcpy(output, buftmp, nb_samples * sizeof(short));
210 }
6000abfa 211 av_free(buf1);
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212 return nb_samples;
213}
214
215ReSampleContext *audio_resample_init(int output_channels, int input_channels,
216 int output_rate, int input_rate)
217{
218 ReSampleContext *s;
219 int i;
220
221 if (output_channels > 2 || input_channels > 2)
222 return NULL;
223
224 s = av_mallocz(sizeof(ReSampleContext));
225 if (!s)
226 return NULL;
227
228 s->ratio = (float)output_rate / (float)input_rate;
229
230 s->input_channels = input_channels;
231 s->output_channels = output_channels;
232
233 s->filter_channels = s->input_channels;
234 if (s->output_channels < s->filter_channels)
235 s->filter_channels = s->output_channels;
236
237 for(i=0;i<s->filter_channels;i++) {
238 init_mono_resample(&s->channel_ctx[i], s->ratio);
239 }
240 return s;
241}
242
243/* resample audio. 'nb_samples' is the number of input samples */
244/* XXX: optimize it ! */
245/* XXX: do it with polyphase filters, since the quality here is
246 HORRIBLE. Return the number of samples available in output */
247int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
248{
249 int i, nb_samples1;
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250 short *bufin[2];
251 short *bufout[2];
de6d9b64 252 short *buftmp2[2], *buftmp3[2];
1a565432 253 int lenout;
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254
255 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
256 /* nothing to do */
257 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
258 return nb_samples;
259 }
260
1a565432 261 /* XXX: move those malloc to resample init code */
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262 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
263 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
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264
265 /* make some zoom to avoid round pb */
266 lenout= (int)(nb_samples * s->ratio) + 16;
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267 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
268 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
1a565432 269
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270 if (s->input_channels == 2 &&
271 s->output_channels == 1) {
272 buftmp2[0] = bufin[0];
273 buftmp3[0] = output;
274 stereo_to_mono(buftmp2[0], input, nb_samples);
275 } else if (s->output_channels == 2 && s->input_channels == 1) {
276 buftmp2[0] = input;
277 buftmp3[0] = bufout[0];
278 } else if (s->output_channels == 2) {
279 buftmp2[0] = bufin[0];
280 buftmp2[1] = bufin[1];
281 buftmp3[0] = bufout[0];
282 buftmp3[1] = bufout[1];
283 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
284 } else {
285 buftmp2[0] = input;
286 buftmp3[0] = output;
287 }
288
289 /* resample each channel */
290 nb_samples1 = 0; /* avoid warning */
291 for(i=0;i<s->filter_channels;i++) {
292 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
293 }
294
295 if (s->output_channels == 2 && s->input_channels == 1) {
296 mono_to_stereo(output, buftmp3[0], nb_samples1);
297 } else if (s->output_channels == 2) {
298 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
299 }
300
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301 av_free(bufin[0]);
302 av_free(bufin[1]);
1a565432 303
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304 av_free(bufout[0]);
305 av_free(bufout[1]);
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306 return nb_samples1;
307}
308
309void audio_resample_close(ReSampleContext *s)
310{
6000abfa 311 av_free(s);
de6d9b64 312}