Mercurial > libavcodec.hg
annotate resample.c @ 1125:0980ae063f4e libavcodec
restoring OS/2 compatibility patch by ("Slavik Gnatenko" <miracle9 at newmail dot ru>)
author | michaelni |
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date | Tue, 11 Mar 2003 12:09:13 +0000 |
parents | 1e39f273ecd6 |
children | 300961b1ef4f |
rev | line source |
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0 | 1 /* |
2 * Sample rate convertion for both audio and video | |
429 | 3 * Copyright (c) 2000 Fabrice Bellard. |
0 | 4 * |
429 | 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. | |
0 | 9 * |
429 | 10 * This library is distributed in the hope that it will be useful, |
0 | 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
13 * Lesser General Public License for more details. | |
0 | 14 * |
429 | 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 | |
0 | 18 */ |
1106 | 19 |
20 /** | |
21 * @file resample.c | |
22 * Sample rate convertion for both audio and video. | |
23 */ | |
24 | |
64 | 25 #include "avcodec.h" |
0 | 26 |
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27 #if defined (CONFIG_OS2) |
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28 #define floorf(n) floor(n) |
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29 #endif |
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30 |
0 | 31 typedef struct { |
32 /* fractional resampling */ | |
1064 | 33 uint32_t incr; /* fractional increment */ |
34 uint32_t frac; | |
0 | 35 int last_sample; |
36 /* integer down sample */ | |
37 int iratio; /* integer divison ratio */ | |
38 int icount, isum; | |
39 int inv; | |
40 } ReSampleChannelContext; | |
41 | |
42 struct ReSampleContext { | |
43 ReSampleChannelContext channel_ctx[2]; | |
44 float ratio; | |
45 /* channel convert */ | |
46 int input_channels, output_channels, filter_channels; | |
47 }; | |
48 | |
49 | |
50 #define FRAC_BITS 16 | |
51 #define FRAC (1 << FRAC_BITS) | |
52 | |
53 static void init_mono_resample(ReSampleChannelContext *s, float ratio) | |
54 { | |
55 ratio = 1.0 / ratio; | |
1057 | 56 s->iratio = (int)floorf(ratio); |
0 | 57 if (s->iratio == 0) |
58 s->iratio = 1; | |
59 s->incr = (int)((ratio / s->iratio) * FRAC); | |
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60 s->frac = FRAC; |
0 | 61 s->last_sample = 0; |
62 s->icount = s->iratio; | |
63 s->isum = 0; | |
64 s->inv = (FRAC / s->iratio); | |
65 } | |
66 | |
67 /* fractional audio resampling */ | |
68 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
69 { | |
70 unsigned int frac, incr; | |
71 int l0, l1; | |
72 short *q, *p, *pend; | |
73 | |
74 l0 = s->last_sample; | |
75 incr = s->incr; | |
76 frac = s->frac; | |
77 | |
78 p = input; | |
79 pend = input + nb_samples; | |
80 q = output; | |
81 | |
82 l1 = *p++; | |
83 for(;;) { | |
84 /* interpolate */ | |
85 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; | |
86 frac = frac + s->incr; | |
87 while (frac >= FRAC) { | |
739 | 88 frac -= FRAC; |
0 | 89 if (p >= pend) |
90 goto the_end; | |
91 l0 = l1; | |
92 l1 = *p++; | |
93 } | |
94 } | |
95 the_end: | |
96 s->last_sample = l1; | |
97 s->frac = frac; | |
98 return q - output; | |
99 } | |
100 | |
101 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
102 { | |
103 short *q, *p, *pend; | |
104 int c, sum; | |
105 | |
106 p = input; | |
107 pend = input + nb_samples; | |
108 q = output; | |
109 | |
110 c = s->icount; | |
111 sum = s->isum; | |
112 | |
113 for(;;) { | |
114 sum += *p++; | |
115 if (--c == 0) { | |
116 *q++ = (sum * s->inv) >> FRAC_BITS; | |
117 c = s->iratio; | |
118 sum = 0; | |
119 } | |
120 if (p >= pend) | |
121 break; | |
122 } | |
123 s->isum = sum; | |
124 s->icount = c; | |
125 return q - output; | |
126 } | |
127 | |
128 /* n1: number of samples */ | |
129 static void stereo_to_mono(short *output, short *input, int n1) | |
130 { | |
131 short *p, *q; | |
132 int n = n1; | |
133 | |
134 p = input; | |
135 q = output; | |
136 while (n >= 4) { | |
137 q[0] = (p[0] + p[1]) >> 1; | |
138 q[1] = (p[2] + p[3]) >> 1; | |
139 q[2] = (p[4] + p[5]) >> 1; | |
140 q[3] = (p[6] + p[7]) >> 1; | |
141 q += 4; | |
142 p += 8; | |
143 n -= 4; | |
144 } | |
145 while (n > 0) { | |
146 q[0] = (p[0] + p[1]) >> 1; | |
147 q++; | |
148 p += 2; | |
149 n--; | |
150 } | |
151 } | |
152 | |
153 /* n1: number of samples */ | |
154 static void mono_to_stereo(short *output, short *input, int n1) | |
155 { | |
156 short *p, *q; | |
157 int n = n1; | |
158 int v; | |
159 | |
160 p = input; | |
161 q = output; | |
162 while (n >= 4) { | |
163 v = p[0]; q[0] = v; q[1] = v; | |
164 v = p[1]; q[2] = v; q[3] = v; | |
165 v = p[2]; q[4] = v; q[5] = v; | |
166 v = p[3]; q[6] = v; q[7] = v; | |
167 q += 8; | |
168 p += 4; | |
169 n -= 4; | |
170 } | |
171 while (n > 0) { | |
172 v = p[0]; q[0] = v; q[1] = v; | |
173 q += 2; | |
174 p += 1; | |
175 n--; | |
176 } | |
177 } | |
178 | |
179 /* XXX: should use more abstract 'N' channels system */ | |
180 static void stereo_split(short *output1, short *output2, short *input, int n) | |
181 { | |
182 int i; | |
183 | |
184 for(i=0;i<n;i++) { | |
185 *output1++ = *input++; | |
186 *output2++ = *input++; | |
187 } | |
188 } | |
189 | |
190 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
191 { | |
192 int i; | |
193 | |
194 for(i=0;i<n;i++) { | |
195 *output++ = *input1++; | |
196 *output++ = *input2++; | |
197 } | |
198 } | |
199 | |
200 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
201 { | |
64 | 202 short *buf1; |
0 | 203 short *buftmp; |
204 | |
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205 buf1= (short*)av_malloc( nb_samples * sizeof(short) ); |
64 | 206 |
0 | 207 /* first downsample by an integer factor with averaging filter */ |
208 if (s->iratio > 1) { | |
209 buftmp = buf1; | |
210 nb_samples = integer_downsample(s, buftmp, input, nb_samples); | |
211 } else { | |
212 buftmp = input; | |
213 } | |
214 | |
215 /* then do a fractional resampling with linear interpolation */ | |
216 if (s->incr != FRAC) { | |
217 nb_samples = fractional_resample(s, output, buftmp, nb_samples); | |
218 } else { | |
219 memcpy(output, buftmp, nb_samples * sizeof(short)); | |
220 } | |
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221 av_free(buf1); |
0 | 222 return nb_samples; |
223 } | |
224 | |
225 ReSampleContext *audio_resample_init(int output_channels, int input_channels, | |
226 int output_rate, int input_rate) | |
227 { | |
228 ReSampleContext *s; | |
229 int i; | |
230 | |
231 if (output_channels > 2 || input_channels > 2) | |
232 return NULL; | |
233 | |
234 s = av_mallocz(sizeof(ReSampleContext)); | |
235 if (!s) | |
236 return NULL; | |
237 | |
238 s->ratio = (float)output_rate / (float)input_rate; | |
239 | |
240 s->input_channels = input_channels; | |
241 s->output_channels = output_channels; | |
242 | |
243 s->filter_channels = s->input_channels; | |
244 if (s->output_channels < s->filter_channels) | |
245 s->filter_channels = s->output_channels; | |
246 | |
247 for(i=0;i<s->filter_channels;i++) { | |
248 init_mono_resample(&s->channel_ctx[i], s->ratio); | |
249 } | |
250 return s; | |
251 } | |
252 | |
253 /* resample audio. 'nb_samples' is the number of input samples */ | |
254 /* XXX: optimize it ! */ | |
255 /* XXX: do it with polyphase filters, since the quality here is | |
256 HORRIBLE. Return the number of samples available in output */ | |
257 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
258 { | |
259 int i, nb_samples1; | |
64 | 260 short *bufin[2]; |
261 short *bufout[2]; | |
0 | 262 short *buftmp2[2], *buftmp3[2]; |
64 | 263 int lenout; |
0 | 264 |
265 if (s->input_channels == s->output_channels && s->ratio == 1.0) { | |
266 /* nothing to do */ | |
267 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
268 return nb_samples; | |
269 } | |
270 | |
64 | 271 /* XXX: move those malloc to resample init code */ |
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272 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) ); |
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273 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) ); |
64 | 274 |
275 /* make some zoom to avoid round pb */ | |
276 lenout= (int)(nb_samples * s->ratio) + 16; | |
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277 bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); |
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278 bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); |
64 | 279 |
0 | 280 if (s->input_channels == 2 && |
281 s->output_channels == 1) { | |
282 buftmp2[0] = bufin[0]; | |
283 buftmp3[0] = output; | |
284 stereo_to_mono(buftmp2[0], input, nb_samples); | |
285 } else if (s->output_channels == 2 && s->input_channels == 1) { | |
286 buftmp2[0] = input; | |
287 buftmp3[0] = bufout[0]; | |
288 } else if (s->output_channels == 2) { | |
289 buftmp2[0] = bufin[0]; | |
290 buftmp2[1] = bufin[1]; | |
291 buftmp3[0] = bufout[0]; | |
292 buftmp3[1] = bufout[1]; | |
293 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
294 } else { | |
295 buftmp2[0] = input; | |
296 buftmp3[0] = output; | |
297 } | |
298 | |
299 /* resample each channel */ | |
300 nb_samples1 = 0; /* avoid warning */ | |
301 for(i=0;i<s->filter_channels;i++) { | |
302 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); | |
303 } | |
304 | |
305 if (s->output_channels == 2 && s->input_channels == 1) { | |
306 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
307 } else if (s->output_channels == 2) { | |
308 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
309 } | |
310 | |
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311 av_free(bufin[0]); |
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312 av_free(bufin[1]); |
64 | 313 |
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314 av_free(bufout[0]); |
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315 av_free(bufout[1]); |
0 | 316 return nb_samples1; |
317 } | |
318 | |
319 void audio_resample_close(ReSampleContext *s) | |
320 { | |
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321 av_free(s); |
0 | 322 } |