FFmpeg: libavcodec/resample2.c Source File

FFmpeg

[フレーム]

libavcodec

resample2.c

Go to the documentation of this file.

1 /*

2 * audio resampling

4 *

5 * This file is part of FFmpeg.

6 *

7 * FFmpeg is free software; you can redistribute it and/or

8 * modify it under the terms of the GNU Lesser General Public

9 * License as published by the Free Software Foundation; either

10 * version 2.1 of the License, or (at your option) any later version.

11 *

12 * FFmpeg is distributed in the hope that it will be useful,

13 * but WITHOUT ANY WARRANTY; without even the implied warranty of

14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

15 * Lesser General Public License for more details.

16 *

17 * You should have received a copy of the GNU Lesser General Public

18 * License along with FFmpeg; if not, write to the Free Software

19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

20 */

22 /**

23 * @file

24 * audio resampling

25 * @author Michael Niedermayer <michaelni@gmx.at>

26 */

28 #include "libavutil/avassert.h"

29 #include "avcodec.h"

30 #include "libavutil/common.h"

32 #if FF_API_AVCODEC_RESAMPLE

34 #ifndef CONFIG_RESAMPLE_HP

35 #define FILTER_SHIFT 15

37 #define FELEM int16_t

38 #define FELEM2 int32_t

39 #define FELEML int64_t

40 #define FELEM_MAX INT16_MAX

41 #define FELEM_MIN INT16_MIN

42 #define WINDOW_TYPE 9

43 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)

44 #define FILTER_SHIFT 30

46 #define FELEM int32_t

47 #define FELEM2 int64_t

48 #define FELEML int64_t

49 #define FELEM_MAX INT32_MAX

50 #define FELEM_MIN INT32_MIN

51 #define WINDOW_TYPE 12

52 #else

53 #define FILTER_SHIFT 0

55 #define FELEM double

56 #define FELEM2 double

57 #define FELEML double

58 #define WINDOW_TYPE 24

59 #endif

62 typedef struct AVResampleContext{

63 const AVClass *av_class;

64 FELEM *filter_bank;

65 int filter_length;

66 int ideal_dst_incr;

67 int dst_incr;

68 int index;

69 int frac;

70 int src_incr;

71 int compensation_distance;

72 int phase_shift;

73 int phase_mask;

74 int linear;

75 }AVResampleContext;

77 /**

78 * 0th order modified bessel function of the first kind.

79 */

80 static double bessel(double x){

81 double v=1;

82 double lastv=0;

83 double t=1;

84 int i;

86 x= x*x/4;

87 for(i=1; v != lastv; i++){

88 lastv=v;

89 t *= x/(i*i);

90 v += t;

91 }

92 return v;

93 }

95 /**

96 * Build a polyphase filterbank.

97 * @param factor resampling factor

98 * @param scale wanted sum of coefficients for each filter

99 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16

100 * @return 0 on success, negative on error

101 */

102 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){

103 int ph, i;

104 double x, y, w;

105 double *tab = av_malloc_array(tap_count, sizeof(*tab));

106 const int center= (tap_count-1)/2;

107

108 if (!tab)

109 return AVERROR(ENOMEM);

110

111 /* if upsampling, only need to interpolate, no filter */

112 if (factor > 1.0)

113 factor = 1.0;

114

115 for(ph=0;ph<phase_count;ph++) {

116 double norm = 0;

117 for(i=0;i<tap_count;i++) {

118 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;

119 if (x == 0) y = 1.0;

120 else y = sin(x) / x;

121 switch(type){

122 case 0:{

123 const float d= -0.5; //first order derivative = -0.5

124 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);

125 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);

126 else y= d*(-4 + 8*x - 5*x*x + x*x*x);

127 break;}

128 case 1:

129 w = 2.0*x / (factor*tap_count) + M_PI;

130 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);

131 break;

132 default:

133 w = 2.0*x / (factor*tap_count*M_PI);

134 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));

135 break;

136 }

137

138 tab[i] = y;

139 norm += y;

140 }

141

142 /* normalize so that an uniform color remains the same */

143 for(i=0;i<tap_count;i++) {

144 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE

145 filter[ph * tap_count + i] = tab[i] / norm;

146 #else

147 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);

148 #endif

149 }

150 }

151 #if 0

152 {

153 #define LEN 1024

154 int j,k;

155 double sine[LEN + tap_count];

156 double filtered[LEN];

157 double maxff=-2, minff=2, maxsf=-2, minsf=2;

158 for(i=0; i<LEN; i++){

159 double ss=0, sf=0, ff=0;

160 for(j=0; j<LEN+tap_count; j++)

161 sine[j]= cos(i*j*M_PI/LEN);

162 for(j=0; j<LEN; j++){

163 double sum=0;

164 ph=0;

165 for(k=0; k<tap_count; k++)

166 sum += filter[ph * tap_count + k] * sine[k+j];

167 filtered[j]= sum / (1<<FILTER_SHIFT);

168 ss+= sine[j + center] * sine[j + center];

169 ff+= filtered[j] * filtered[j];

170 sf+= sine[j + center] * filtered[j];

171 }

172 ss= sqrt(2*ss/LEN);

173 ff= sqrt(2*ff/LEN);

174 sf= 2*sf/LEN;

175 maxff= FFMAX(maxff, ff);

176 minff= FFMIN(minff, ff);

177 maxsf= FFMAX(maxsf, sf);

178 minsf= FFMIN(minsf, sf);

179 if(i%11==0){

180 av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);

181 minff=minsf= 2;

182 maxff=maxsf= -2;

183 }

184 }

185 }

186 #endif

187

188 av_free(tab);

189 return 0;

190 }

191

192 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){

193 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));

194 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);

195 int phase_count= 1<<phase_shift;

196

197 if (!c)

198 return NULL;

199

200 c->phase_shift= phase_shift;

201 c->phase_mask= phase_count-1;

202 c->linear= linear;

203

204 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);

205 c->filter_bank= av_mallocz_array(c->filter_length, (phase_count+1)*sizeof(FELEM));

206 if (!c->filter_bank)

207 goto error;

208 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))

209 goto error;

210 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));

211 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];

212

213 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))

214 goto error;

215 c->ideal_dst_incr= c->dst_incr;

216

217 c->index= -phase_count*((c->filter_length-1)/2);

218

219 return c;

220 error:

221 av_free(c->filter_bank);

222 av_free(c);

223 return NULL;

224 }

225

226 void av_resample_close(AVResampleContext *c){

227 av_freep(&c->filter_bank);

228 av_freep(&c);

229 }

230

231 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){

232 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;

233 c->compensation_distance= compensation_distance;

234 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;

235 }

236

237 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){

238 int dst_index, i;

239 int index= c->index;

240 int frac= c->frac;

241 int dst_incr_frac= c->dst_incr % c->src_incr;

242 int dst_incr= c->dst_incr / c->src_incr;

243 int compensation_distance= c->compensation_distance;

244

245 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){

246 int64_t index2= ((int64_t)index)<<32;

247 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;

248 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);

249

250 for(dst_index=0; dst_index < dst_size; dst_index++){

251 dst[dst_index] = src[index2>>32];

252 index2 += incr;

253 }

254 index += dst_index * dst_incr;

255 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;

256 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;

257 }else{

258 for(dst_index=0; dst_index < dst_size; dst_index++){

259 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);

260 int sample_index= index >> c->phase_shift;

261 FELEM2 val=0;

262

263 if(sample_index < 0){

264 for(i=0; i<c->filter_length; i++)

265 val += src[FFABS(sample_index + i) % src_size] * filter[i];

266 }else if(sample_index + c->filter_length > src_size){

267 break;

268 }else if(c->linear){

269 FELEM2 v2=0;

270 for(i=0; i<c->filter_length; i++){

271 val += src[sample_index + i] * (FELEM2)filter[i];

272 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];

273 }

274 val+=(v2-val)*(FELEML)frac / c->src_incr;

275 }else{

276 for(i=0; i<c->filter_length; i++){

277 val += src[sample_index + i] * (FELEM2)filter[i];

278 }

279 }

280

281 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE

282 dst[dst_index] = av_clip_int16(lrintf(val));

283 #else

284 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;

285 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;

286 #endif

287

288 frac += dst_incr_frac;

289 index += dst_incr;

290 if(frac >= c->src_incr){

291 frac -= c->src_incr;

292 index++;

293 }

294

295 if(dst_index + 1 == compensation_distance){

296 compensation_distance= 0;

297 dst_incr_frac= c->ideal_dst_incr % c->src_incr;

298 dst_incr= c->ideal_dst_incr / c->src_incr;

299 }

300 }

301 }

302 *consumed= FFMAX(index, 0) >> c->phase_shift;

303 if(index>=0) index &= c->phase_mask;

304

305 if(compensation_distance){

306 compensation_distance -= dst_index;

307 av_assert2(compensation_distance > 0);

308 }

309 if(update_ctx){

310 c->frac= frac;

311 c->index= index;

312 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;

313 c->compensation_distance= compensation_distance;

314 }

315

316 return dst_index;

317 }

318

319 #endif

Generated on Sun Mar 8 2015 02:34:59 for FFmpeg by doxygen 1.8.2