Mercurial > libavcodec.hg
annotate resample.c @ 1433:46d3fa8501cd libavcodec
ASV2 decoder
author | michaelni |
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date | Mon, 01 Sep 2003 08:10:01 +0000 |
parents | 4d67eb341a0c |
children | daa70b33fc44 |
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" |
1128 | 26 #include "os_support.h" |
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27 |
0 | 28 typedef struct { |
29 /* fractional resampling */ | |
1064 | 30 uint32_t incr; /* fractional increment */ |
31 uint32_t frac; | |
0 | 32 int last_sample; |
33 /* integer down sample */ | |
34 int iratio; /* integer divison ratio */ | |
35 int icount, isum; | |
36 int inv; | |
37 } ReSampleChannelContext; | |
38 | |
39 struct ReSampleContext { | |
40 ReSampleChannelContext channel_ctx[2]; | |
41 float ratio; | |
42 /* channel convert */ | |
43 int input_channels, output_channels, filter_channels; | |
44 }; | |
45 | |
46 | |
47 #define FRAC_BITS 16 | |
48 #define FRAC (1 << FRAC_BITS) | |
49 | |
50 static void init_mono_resample(ReSampleChannelContext *s, float ratio) | |
51 { | |
52 ratio = 1.0 / ratio; | |
1057 | 53 s->iratio = (int)floorf(ratio); |
0 | 54 if (s->iratio == 0) |
55 s->iratio = 1; | |
56 s->incr = (int)((ratio / s->iratio) * FRAC); | |
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57 s->frac = FRAC; |
0 | 58 s->last_sample = 0; |
59 s->icount = s->iratio; | |
60 s->isum = 0; | |
61 s->inv = (FRAC / s->iratio); | |
62 } | |
63 | |
64 /* fractional audio resampling */ | |
65 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
66 { | |
67 unsigned int frac, incr; | |
68 int l0, l1; | |
69 short *q, *p, *pend; | |
70 | |
71 l0 = s->last_sample; | |
72 incr = s->incr; | |
73 frac = s->frac; | |
74 | |
75 p = input; | |
76 pend = input + nb_samples; | |
77 q = output; | |
78 | |
79 l1 = *p++; | |
80 for(;;) { | |
81 /* interpolate */ | |
82 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; | |
83 frac = frac + s->incr; | |
84 while (frac >= FRAC) { | |
739 | 85 frac -= FRAC; |
0 | 86 if (p >= pend) |
87 goto the_end; | |
88 l0 = l1; | |
89 l1 = *p++; | |
90 } | |
91 } | |
92 the_end: | |
93 s->last_sample = l1; | |
94 s->frac = frac; | |
95 return q - output; | |
96 } | |
97 | |
98 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
99 { | |
100 short *q, *p, *pend; | |
101 int c, sum; | |
102 | |
103 p = input; | |
104 pend = input + nb_samples; | |
105 q = output; | |
106 | |
107 c = s->icount; | |
108 sum = s->isum; | |
109 | |
110 for(;;) { | |
111 sum += *p++; | |
112 if (--c == 0) { | |
113 *q++ = (sum * s->inv) >> FRAC_BITS; | |
114 c = s->iratio; | |
115 sum = 0; | |
116 } | |
117 if (p >= pend) | |
118 break; | |
119 } | |
120 s->isum = sum; | |
121 s->icount = c; | |
122 return q - output; | |
123 } | |
124 | |
125 /* n1: number of samples */ | |
126 static void stereo_to_mono(short *output, short *input, int n1) | |
127 { | |
128 short *p, *q; | |
129 int n = n1; | |
130 | |
131 p = input; | |
132 q = output; | |
133 while (n >= 4) { | |
134 q[0] = (p[0] + p[1]) >> 1; | |
135 q[1] = (p[2] + p[3]) >> 1; | |
136 q[2] = (p[4] + p[5]) >> 1; | |
137 q[3] = (p[6] + p[7]) >> 1; | |
138 q += 4; | |
139 p += 8; | |
140 n -= 4; | |
141 } | |
142 while (n > 0) { | |
143 q[0] = (p[0] + p[1]) >> 1; | |
144 q++; | |
145 p += 2; | |
146 n--; | |
147 } | |
148 } | |
149 | |
150 /* n1: number of samples */ | |
151 static void mono_to_stereo(short *output, short *input, int n1) | |
152 { | |
153 short *p, *q; | |
154 int n = n1; | |
155 int v; | |
156 | |
157 p = input; | |
158 q = output; | |
159 while (n >= 4) { | |
160 v = p[0]; q[0] = v; q[1] = v; | |
161 v = p[1]; q[2] = v; q[3] = v; | |
162 v = p[2]; q[4] = v; q[5] = v; | |
163 v = p[3]; q[6] = v; q[7] = v; | |
164 q += 8; | |
165 p += 4; | |
166 n -= 4; | |
167 } | |
168 while (n > 0) { | |
169 v = p[0]; q[0] = v; q[1] = v; | |
170 q += 2; | |
171 p += 1; | |
172 n--; | |
173 } | |
174 } | |
175 | |
176 /* XXX: should use more abstract 'N' channels system */ | |
177 static void stereo_split(short *output1, short *output2, short *input, int n) | |
178 { | |
179 int i; | |
180 | |
181 for(i=0;i<n;i++) { | |
182 *output1++ = *input++; | |
183 *output2++ = *input++; | |
184 } | |
185 } | |
186 | |
187 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
188 { | |
189 int i; | |
190 | |
191 for(i=0;i<n;i++) { | |
192 *output++ = *input1++; | |
193 *output++ = *input2++; | |
194 } | |
195 } | |
196 | |
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197 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) |
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198 { |
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199 int i; |
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200 short l,r; |
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201 |
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202 for(i=0;i<n;i++) { |
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203 l=*input1++; |
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204 r=*input2++; |
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205 *output++ = l; /* left */ |
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206 *output++ = (l/2)+(r/2); /* center */ |
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207 *output++ = r; /* right */ |
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208 *output++ = 0; /* left surround */ |
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209 *output++ = 0; /* right surroud */ |
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210 *output++ = 0; /* low freq */ |
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211 } |
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212 } |
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213 |
0 | 214 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) |
215 { | |
64 | 216 short *buf1; |
0 | 217 short *buftmp; |
218 | |
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219 buf1= (short*)av_malloc( nb_samples * sizeof(short) ); |
64 | 220 |
0 | 221 /* first downsample by an integer factor with averaging filter */ |
222 if (s->iratio > 1) { | |
223 buftmp = buf1; | |
224 nb_samples = integer_downsample(s, buftmp, input, nb_samples); | |
225 } else { | |
226 buftmp = input; | |
227 } | |
228 | |
229 /* then do a fractional resampling with linear interpolation */ | |
230 if (s->incr != FRAC) { | |
231 nb_samples = fractional_resample(s, output, buftmp, nb_samples); | |
232 } else { | |
233 memcpy(output, buftmp, nb_samples * sizeof(short)); | |
234 } | |
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235 av_free(buf1); |
0 | 236 return nb_samples; |
237 } | |
238 | |
239 ReSampleContext *audio_resample_init(int output_channels, int input_channels, | |
240 int output_rate, int input_rate) | |
241 { | |
242 ReSampleContext *s; | |
243 int i; | |
244 | |
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245 if ( input_channels > 2) |
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246 { |
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247 printf("Resampling with input channels greater than 2 unsupported."); |
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248 return NULL; |
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249 } |
0 | 250 |
251 s = av_mallocz(sizeof(ReSampleContext)); | |
252 if (!s) | |
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253 { |
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254 printf("Can't allocate memory for resample context."); |
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255 return NULL; |
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256 } |
0 | 257 |
258 s->ratio = (float)output_rate / (float)input_rate; | |
259 | |
260 s->input_channels = input_channels; | |
261 s->output_channels = output_channels; | |
262 | |
263 s->filter_channels = s->input_channels; | |
264 if (s->output_channels < s->filter_channels) | |
265 s->filter_channels = s->output_channels; | |
266 | |
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267 /* |
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268 * ac3 output is the only case where filter_channels could be greater than 2. |
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269 * input channels can't be greater than 2, so resample the 2 channels and then |
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270 * expand to 6 channels after the resampling. |
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271 */ |
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272 if(s->filter_channels>2) |
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273 s->filter_channels = 2; |
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274 |
0 | 275 for(i=0;i<s->filter_channels;i++) { |
276 init_mono_resample(&s->channel_ctx[i], s->ratio); | |
277 } | |
278 return s; | |
279 } | |
280 | |
281 /* resample audio. 'nb_samples' is the number of input samples */ | |
282 /* XXX: optimize it ! */ | |
283 /* XXX: do it with polyphase filters, since the quality here is | |
284 HORRIBLE. Return the number of samples available in output */ | |
285 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
286 { | |
287 int i, nb_samples1; | |
64 | 288 short *bufin[2]; |
289 short *bufout[2]; | |
0 | 290 short *buftmp2[2], *buftmp3[2]; |
64 | 291 int lenout; |
0 | 292 |
293 if (s->input_channels == s->output_channels && s->ratio == 1.0) { | |
294 /* nothing to do */ | |
295 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
296 return nb_samples; | |
297 } | |
298 | |
64 | 299 /* XXX: move those malloc to resample init code */ |
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300 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) ); |
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301 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) ); |
64 | 302 |
303 /* make some zoom to avoid round pb */ | |
304 lenout= (int)(nb_samples * s->ratio) + 16; | |
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305 bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); |
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306 bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); |
64 | 307 |
0 | 308 if (s->input_channels == 2 && |
309 s->output_channels == 1) { | |
310 buftmp2[0] = bufin[0]; | |
311 buftmp3[0] = output; | |
312 stereo_to_mono(buftmp2[0], input, nb_samples); | |
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313 } else if (s->output_channels >= 2 && s->input_channels == 1) { |
0 | 314 buftmp2[0] = input; |
315 buftmp3[0] = bufout[0]; | |
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316 } else if (s->output_channels >= 2) { |
0 | 317 buftmp2[0] = bufin[0]; |
318 buftmp2[1] = bufin[1]; | |
319 buftmp3[0] = bufout[0]; | |
320 buftmp3[1] = bufout[1]; | |
321 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
322 } else { | |
323 buftmp2[0] = input; | |
324 buftmp3[0] = output; | |
325 } | |
326 | |
327 /* resample each channel */ | |
328 nb_samples1 = 0; /* avoid warning */ | |
329 for(i=0;i<s->filter_channels;i++) { | |
330 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); | |
331 } | |
332 | |
333 if (s->output_channels == 2 && s->input_channels == 1) { | |
334 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
335 } else if (s->output_channels == 2) { | |
336 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
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337 } else if (s->output_channels == 6) { |
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338 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); |
0 | 339 } |
340 | |
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341 av_free(bufin[0]); |
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342 av_free(bufin[1]); |
64 | 343 |
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344 av_free(bufout[0]); |
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345 av_free(bufout[1]); |
0 | 346 return nb_samples1; |
347 } | |
348 | |
349 void audio_resample_close(ReSampleContext *s) | |
350 { | |
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351 av_free(s); |
0 | 352 } |