FFmpeg: libavfilter/af_aphaser.c Source File

FFmpeg

[フレーム]

libavfilter

af_aphaser.c

Go to the documentation of this file.

1 /*

3 *

4 * This file is part of FFmpeg.

5 *

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

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

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

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

10 *

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

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

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

14 * Lesser General Public License for more details.

15 *

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

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

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

19 */

21 /**

22 * @file

23 * phaser audio filter

24 */

26 #include "libavutil/avassert.h"

27 #include "libavutil/opt.h"

28 #include "audio.h"

29 #include "avfilter.h"

30 #include "internal.h"

32 enum WaveType {

33 WAVE_SIN,

34 WAVE_TRI,

35 WAVE_NB,

36 };

38 typedef struct AudioPhaserContext {

39 const AVClass *class;

40 double in_gain, out_gain;

41 double delay;

42 double decay;

43 double speed;

45 enum WaveType type;

47 int delay_buffer_length;

48 double *delay_buffer;

50 int modulation_buffer_length;

51 int32_t *modulation_buffer;

53 int delay_pos, modulation_pos;

55 void (*phaser)(struct AudioPhaserContext *p,

56 uint8_t * const *src, uint8_t **dst,

57 int nb_samples, int channels);

58 } AudioPhaserContext;

60 #define OFFSET(x) offsetof(AudioPhaserContext, x)

61 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

63 static const AVOption aphaser_options[] = {

64 { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },

65 { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },

66 { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },

67 { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },

68 { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },

69 { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },

70 { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },

71 { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },

72 { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },

73 { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },

74 { NULL }

75 };

77 AVFILTER_DEFINE_CLASS(aphaser);

79 static av_cold int init(AVFilterContext *ctx)

80 {

81 AudioPhaserContext *p = ctx->priv;

83 if (p->in_gain > (1 - p->decay * p->decay))

84 av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");

85 if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)

86 av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");

88 return 0;

89 }

91 static int query_formats(AVFilterContext *ctx)

92 {

93 AVFilterFormats *formats;

94 AVFilterChannelLayouts *layouts;

95 static const enum AVSampleFormat sample_fmts[] = {

96 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,

97 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,

98 AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,

99 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,

100 AV_SAMPLE_FMT_NONE

101 };

102

103 layouts = ff_all_channel_layouts();

104 if (!layouts)

105 return AVERROR(ENOMEM);

106 ff_set_common_channel_layouts(ctx, layouts);

107

108 formats = ff_make_format_list(sample_fmts);

109 if (!formats)

110 return AVERROR(ENOMEM);

111 ff_set_common_formats(ctx, formats);

112

113 formats = ff_all_samplerates();

114 if (!formats)

115 return AVERROR(ENOMEM);

116 ff_set_common_samplerates(ctx, formats);

117

118 return 0;

119 }

120

121 static void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt,

122 void *table, int table_size,

123 double min, double max, double phase)

124 {

125 uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;

126

127 for (i = 0; i < table_size; i++) {

128 uint32_t point = (i + phase_offset) % table_size;

129 double d;

130

131 switch (wave_type) {

132 case WAVE_SIN:

133 d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;

134 break;

135 case WAVE_TRI:

136 d = (double)point * 2 / table_size;

137 switch (4 * point / table_size) {

138 case 0: d = d + 0.5; break;

139 case 1:

140 case 2: d = 1.5 - d; break;

141 case 3: d = d - 1.5; break;

142 }

143 break;

144 default:

145 av_assert0(0);

146 }

147

148 d = d * (max - min) + min;

149 switch (sample_fmt) {

150 case AV_SAMPLE_FMT_FLT: {

151 float *fp = (float *)table;

152 *fp++ = (float)d;

153 table = fp;

154 continue; }

155 case AV_SAMPLE_FMT_DBL: {

156 double *dp = (double *)table;

157 *dp++ = d;

158 table = dp;

159 continue; }

160 }

161

162 d += d < 0 ? -0.5 : 0.5;

163 switch (sample_fmt) {

164 case AV_SAMPLE_FMT_S16: {

165 int16_t *sp = table;

166 *sp++ = (int16_t)d;

167 table = sp;

168 continue; }

169 case AV_SAMPLE_FMT_S32: {

170 int32_t *ip = table;

171 *ip++ = (int32_t)d;

172 table = ip;

173 continue; }

174 default:

175 av_assert0(0);

176 }

177 }

178 }

179

180 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))

181

182 #define PHASER_PLANAR(name, type) \

183 static void phaser_## name ##p(AudioPhaserContext *p, \

184 uint8_t * const *src, uint8_t **dst, \

185 int nb_samples, int channels) \

186 { \

187 int i, c, delay_pos, modulation_pos; \

188 \

189 av_assert0(channels > 0); \

190 for (c = 0; c < channels; c++) { \

191 type *s = (type *)src[c]; \

192 type *d = (type *)dst[c]; \

193 double *buffer = p->delay_buffer + \

194 c * p->delay_buffer_length; \

195 \

196 delay_pos = p->delay_pos; \

197 modulation_pos = p->modulation_pos; \

198 \

199 for (i = 0; i < nb_samples; i++, s++, d++) { \

200 double v = *s * p->in_gain + buffer[ \

201 MOD(delay_pos + p->modulation_buffer[ \

202 modulation_pos], \

203 p->delay_buffer_length)] * p->decay; \

204 \

205 modulation_pos = MOD(modulation_pos + 1, \

206 p->modulation_buffer_length); \

207 delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \

208 buffer[delay_pos] = v; \

209 \

210 *d = v * p->out_gain; \

211 } \

212 } \

213 \

214 p->delay_pos = delay_pos; \

215 p->modulation_pos = modulation_pos; \

216 }

217

218 #define PHASER(name, type) \

219 static void phaser_## name (AudioPhaserContext *p, \

220 uint8_t * const *src, uint8_t **dst, \

221 int nb_samples, int channels) \

222 { \

223 int i, c, delay_pos, modulation_pos; \

224 type *s = (type *)src[0]; \

225 type *d = (type *)dst[0]; \

226 double *buffer = p->delay_buffer; \

227 \

228 delay_pos = p->delay_pos; \

229 modulation_pos = p->modulation_pos; \

230 \

231 for (i = 0; i < nb_samples; i++) { \

232 int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \

233 p->delay_buffer_length) * channels; \

234 int npos; \

235 \

236 delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \

237 npos = delay_pos * channels; \

238 for (c = 0; c < channels; c++, s++, d++) { \

239 double v = *s * p->in_gain + buffer[pos + c] * p->decay; \

240 \

241 buffer[npos + c] = v; \

242 \

243 *d = v * p->out_gain; \

244 } \

245 \

246 modulation_pos = MOD(modulation_pos + 1, \

247 p->modulation_buffer_length); \

248 } \

249 \

250 p->delay_pos = delay_pos; \

251 p->modulation_pos = modulation_pos; \

252 }

253

254 PHASER_PLANAR(dbl, double)

255 PHASER_PLANAR(flt, float)

256 PHASER_PLANAR(s16, int16_t)

257 PHASER_PLANAR(s32, int32_t)

258

259 PHASER(dbl, double)

260 PHASER(flt, float)

261 PHASER(s16, int16_t)

262 PHASER(s32, int32_t)

263

264 static int config_output(AVFilterLink *outlink)

265 {

266 AudioPhaserContext *p = outlink->src->priv;

267 AVFilterLink *inlink = outlink->src->inputs[0];

268

269 p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;

270 p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);

271 p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;

272 p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));

273

274 if (!p->modulation_buffer || !p->delay_buffer)

275 return AVERROR(ENOMEM);

276

277 generate_wave_table(p->type, AV_SAMPLE_FMT_S32,

278 p->modulation_buffer, p->modulation_buffer_length,

279 1., p->delay_buffer_length, M_PI / 2.0);

280

281 p->delay_pos = p->modulation_pos = 0;

282

283 switch (inlink->format) {

284 case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;

285 case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;

286 case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;

287 case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;

288 case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;

289 case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;

290 case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;

291 case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;

292 default: av_assert0(0);

293 }

294

295 return 0;

296 }

297

298 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)

299 {

300 AudioPhaserContext *p = inlink->dst->priv;

301 AVFilterLink *outlink = inlink->dst->outputs[0];

302 AVFrame *outbuf;

303

304 if (av_frame_is_writable(inbuf)) {

305 outbuf = inbuf;

306 } else {

307 outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);

308 if (!outbuf)

309 return AVERROR(ENOMEM);

310 av_frame_copy_props(outbuf, inbuf);

311 }

312

313 p->phaser(p, inbuf->extended_data, outbuf->extended_data,

314 outbuf->nb_samples, av_frame_get_channels(outbuf));

315

316 if (inbuf != outbuf)

317 av_frame_free(&inbuf);

318

319 return ff_filter_frame(outlink, outbuf);

320 }

321

322 static av_cold void uninit(AVFilterContext *ctx)

323 {

324 AudioPhaserContext *p = ctx->priv;

325

326 av_freep(&p->delay_buffer);

327 av_freep(&p->modulation_buffer);

328 }

329

330 static const AVFilterPad aphaser_inputs[] = {

331 {

332 .name = "default",

333 .type = AVMEDIA_TYPE_AUDIO,

334 .filter_frame = filter_frame,

335 },

336 { NULL }

337 };

338

339 static const AVFilterPad aphaser_outputs[] = {

340 {

341 .name = "default",

342 .type = AVMEDIA_TYPE_AUDIO,

343 .config_props = config_output,

344 },

345 { NULL }

346 };

347

348 AVFilter avfilter_af_aphaser = {

349 .name = "aphaser",

350 .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),

351 .query_formats = query_formats,

352 .priv_size = sizeof(AudioPhaserContext),

353 .init = init,

354 .uninit = uninit,

355 .inputs = aphaser_inputs,

356 .outputs = aphaser_outputs,

357 .priv_class = &aphaser_class,

358 };

Generated on Sat Jan 25 2014 19:51:59 for FFmpeg by doxygen 1.8.2