Mercurial > libavcodec.hg
annotate resample.c @ 373:3007abcbc510 libavcodec
* Fix a problem with the first sample when down sampling.
* Note that this code needs to be fixed -- the rate conversion from 48000->44100
sounds horrible!
author | philipjsg |
---|---|
date | Thu, 09 May 2002 01:23:49 +0000 |
parents | 5aa6292a1660 |
children | fce0a2520551 |
rev | line source |
---|---|
0 | 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 */ | |
64 | 19 #include "avcodec.h" |
0 | 20 #include <math.h> |
21 | |
22 typedef struct { | |
23 /* fractional resampling */ | |
24 UINT32 incr; /* fractional increment */ | |
25 UINT32 frac; | |
26 int last_sample; | |
27 /* integer down sample */ | |
28 int iratio; /* integer divison ratio */ | |
29 int icount, isum; | |
30 int inv; | |
31 } ReSampleChannelContext; | |
32 | |
33 struct ReSampleContext { | |
34 ReSampleChannelContext channel_ctx[2]; | |
35 float ratio; | |
36 /* channel convert */ | |
37 int input_channels, output_channels, filter_channels; | |
38 }; | |
39 | |
40 | |
41 #define FRAC_BITS 16 | |
42 #define FRAC (1 << FRAC_BITS) | |
43 | |
44 static void init_mono_resample(ReSampleChannelContext *s, float ratio) | |
45 { | |
46 ratio = 1.0 / ratio; | |
47 s->iratio = (int)floor(ratio); | |
48 if (s->iratio == 0) | |
49 s->iratio = 1; | |
50 s->incr = (int)((ratio / s->iratio) * FRAC); | |
373
3007abcbc510
* Fix a problem with the first sample when down sampling.
philipjsg
parents:
64
diff
changeset
|
51 s->frac = FRAC; |
0 | 52 s->last_sample = 0; |
53 s->icount = s->iratio; | |
54 s->isum = 0; | |
55 s->inv = (FRAC / s->iratio); | |
56 } | |
57 | |
58 /* fractional audio resampling */ | |
59 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
60 { | |
61 unsigned int frac, incr; | |
62 int l0, l1; | |
63 short *q, *p, *pend; | |
64 | |
65 l0 = s->last_sample; | |
66 incr = s->incr; | |
67 frac = s->frac; | |
68 | |
69 p = input; | |
70 pend = input + nb_samples; | |
71 q = output; | |
72 | |
73 l1 = *p++; | |
74 for(;;) { | |
75 /* interpolate */ | |
76 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; | |
77 frac = frac + s->incr; | |
78 while (frac >= FRAC) { | |
79 if (p >= pend) | |
80 goto the_end; | |
81 frac -= FRAC; | |
82 l0 = l1; | |
83 l1 = *p++; | |
84 } | |
85 } | |
86 the_end: | |
87 s->last_sample = l1; | |
88 s->frac = frac; | |
89 return q - output; | |
90 } | |
91 | |
92 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
93 { | |
94 short *q, *p, *pend; | |
95 int c, sum; | |
96 | |
97 p = input; | |
98 pend = input + nb_samples; | |
99 q = output; | |
100 | |
101 c = s->icount; | |
102 sum = s->isum; | |
103 | |
104 for(;;) { | |
105 sum += *p++; | |
106 if (--c == 0) { | |
107 *q++ = (sum * s->inv) >> FRAC_BITS; | |
108 c = s->iratio; | |
109 sum = 0; | |
110 } | |
111 if (p >= pend) | |
112 break; | |
113 } | |
114 s->isum = sum; | |
115 s->icount = c; | |
116 return q - output; | |
117 } | |
118 | |
119 /* n1: number of samples */ | |
120 static void stereo_to_mono(short *output, short *input, int n1) | |
121 { | |
122 short *p, *q; | |
123 int n = n1; | |
124 | |
125 p = input; | |
126 q = output; | |
127 while (n >= 4) { | |
128 q[0] = (p[0] + p[1]) >> 1; | |
129 q[1] = (p[2] + p[3]) >> 1; | |
130 q[2] = (p[4] + p[5]) >> 1; | |
131 q[3] = (p[6] + p[7]) >> 1; | |
132 q += 4; | |
133 p += 8; | |
134 n -= 4; | |
135 } | |
136 while (n > 0) { | |
137 q[0] = (p[0] + p[1]) >> 1; | |
138 q++; | |
139 p += 2; | |
140 n--; | |
141 } | |
142 } | |
143 | |
144 /* n1: number of samples */ | |
145 static void mono_to_stereo(short *output, short *input, int n1) | |
146 { | |
147 short *p, *q; | |
148 int n = n1; | |
149 int v; | |
150 | |
151 p = input; | |
152 q = output; | |
153 while (n >= 4) { | |
154 v = p[0]; q[0] = v; q[1] = v; | |
155 v = p[1]; q[2] = v; q[3] = v; | |
156 v = p[2]; q[4] = v; q[5] = v; | |
157 v = p[3]; q[6] = v; q[7] = v; | |
158 q += 8; | |
159 p += 4; | |
160 n -= 4; | |
161 } | |
162 while (n > 0) { | |
163 v = p[0]; q[0] = v; q[1] = v; | |
164 q += 2; | |
165 p += 1; | |
166 n--; | |
167 } | |
168 } | |
169 | |
170 /* XXX: should use more abstract 'N' channels system */ | |
171 static void stereo_split(short *output1, short *output2, short *input, int n) | |
172 { | |
173 int i; | |
174 | |
175 for(i=0;i<n;i++) { | |
176 *output1++ = *input++; | |
177 *output2++ = *input++; | |
178 } | |
179 } | |
180 | |
181 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
182 { | |
183 int i; | |
184 | |
185 for(i=0;i<n;i++) { | |
186 *output++ = *input1++; | |
187 *output++ = *input2++; | |
188 } | |
189 } | |
190 | |
191 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
192 { | |
64 | 193 short *buf1; |
0 | 194 short *buftmp; |
195 | |
64 | 196 buf1= (short*) malloc( nb_samples * sizeof(short) ); |
197 | |
0 | 198 /* first downsample by an integer factor with averaging filter */ |
199 if (s->iratio > 1) { | |
200 buftmp = buf1; | |
201 nb_samples = integer_downsample(s, buftmp, input, nb_samples); | |
202 } else { | |
203 buftmp = input; | |
204 } | |
205 | |
206 /* then do a fractional resampling with linear interpolation */ | |
207 if (s->incr != FRAC) { | |
208 nb_samples = fractional_resample(s, output, buftmp, nb_samples); | |
209 } else { | |
210 memcpy(output, buftmp, nb_samples * sizeof(short)); | |
211 } | |
64 | 212 free(buf1); |
0 | 213 return nb_samples; |
214 } | |
215 | |
216 ReSampleContext *audio_resample_init(int output_channels, int input_channels, | |
217 int output_rate, int input_rate) | |
218 { | |
219 ReSampleContext *s; | |
220 int i; | |
221 | |
222 if (output_channels > 2 || input_channels > 2) | |
223 return NULL; | |
224 | |
225 s = av_mallocz(sizeof(ReSampleContext)); | |
226 if (!s) | |
227 return NULL; | |
228 | |
229 s->ratio = (float)output_rate / (float)input_rate; | |
230 | |
231 s->input_channels = input_channels; | |
232 s->output_channels = output_channels; | |
233 | |
234 s->filter_channels = s->input_channels; | |
235 if (s->output_channels < s->filter_channels) | |
236 s->filter_channels = s->output_channels; | |
237 | |
238 for(i=0;i<s->filter_channels;i++) { | |
239 init_mono_resample(&s->channel_ctx[i], s->ratio); | |
240 } | |
241 return s; | |
242 } | |
243 | |
244 /* resample audio. 'nb_samples' is the number of input samples */ | |
245 /* XXX: optimize it ! */ | |
246 /* XXX: do it with polyphase filters, since the quality here is | |
247 HORRIBLE. Return the number of samples available in output */ | |
248 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
249 { | |
250 int i, nb_samples1; | |
64 | 251 short *bufin[2]; |
252 short *bufout[2]; | |
0 | 253 short *buftmp2[2], *buftmp3[2]; |
64 | 254 int lenout; |
0 | 255 |
256 if (s->input_channels == s->output_channels && s->ratio == 1.0) { | |
257 /* nothing to do */ | |
258 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
259 return nb_samples; | |
260 } | |
261 | |
64 | 262 /* XXX: move those malloc to resample init code */ |
263 bufin[0]= (short*) malloc( nb_samples * sizeof(short) ); | |
264 bufin[1]= (short*) malloc( nb_samples * sizeof(short) ); | |
265 | |
266 /* make some zoom to avoid round pb */ | |
267 lenout= (int)(nb_samples * s->ratio) + 16; | |
268 bufout[0]= (short*) malloc( lenout * sizeof(short) ); | |
269 bufout[1]= (short*) malloc( lenout * sizeof(short) ); | |
270 | |
0 | 271 if (s->input_channels == 2 && |
272 s->output_channels == 1) { | |
273 buftmp2[0] = bufin[0]; | |
274 buftmp3[0] = output; | |
275 stereo_to_mono(buftmp2[0], input, nb_samples); | |
276 } else if (s->output_channels == 2 && s->input_channels == 1) { | |
277 buftmp2[0] = input; | |
278 buftmp3[0] = bufout[0]; | |
279 } else if (s->output_channels == 2) { | |
280 buftmp2[0] = bufin[0]; | |
281 buftmp2[1] = bufin[1]; | |
282 buftmp3[0] = bufout[0]; | |
283 buftmp3[1] = bufout[1]; | |
284 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
285 } else { | |
286 buftmp2[0] = input; | |
287 buftmp3[0] = output; | |
288 } | |
289 | |
290 /* resample each channel */ | |
291 nb_samples1 = 0; /* avoid warning */ | |
292 for(i=0;i<s->filter_channels;i++) { | |
293 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); | |
294 } | |
295 | |
296 if (s->output_channels == 2 && s->input_channels == 1) { | |
297 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
298 } else if (s->output_channels == 2) { | |
299 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
300 } | |
301 | |
64 | 302 free(bufin[0]); |
303 free(bufin[1]); | |
304 | |
305 free(bufout[0]); | |
306 free(bufout[1]); | |
0 | 307 return nb_samples1; |
308 } | |
309 | |
310 void audio_resample_close(ReSampleContext *s) | |
311 { | |
312 free(s); | |
313 } |