FFmpeg: libavfilter/af_aecho.c Source File

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libavfilter

af_aecho.c

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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 #include "libavutil/avassert.h"

22 #include "libavutil/avstring.h"

23 #include "libavutil/opt.h"

24 #include "libavutil/samplefmt.h"

25 #include "avfilter.h"

26 #include "audio.h"

27 #include "internal.h"

29 typedef struct AudioEchoContext {

30 const AVClass *class;

31 float in_gain, out_gain;

32 char *delays, *decays;

33 float *delay, *decay;

34 int nb_echoes;

35 int delay_index;

36 uint8_t **delayptrs;

37 int max_samples, fade_out;

38 int *samples;

39 int64_t next_pts;

41 void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,

42 uint8_t * const *src, uint8_t **dst,

43 int nb_samples, int channels);

44 } AudioEchoContext;

46 #define OFFSET(x) offsetof(AudioEchoContext, x)

47 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

49 static const AVOption aecho_options[] = {

50 { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },

51 { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },

52 { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },

53 { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },

54 { NULL }

55 };

57 AVFILTER_DEFINE_CLASS(aecho);

59 static void count_items(char *item_str, int *nb_items)

60 {

61 char *p;

63 *nb_items = 1;

64 for (p = item_str; *p; p++) {

65 if (*p == '|')

66 (*nb_items)++;

67 }

69 }

71 static void fill_items(char *item_str, int *nb_items, float *items)

72 {

73 char *p, *saveptr = NULL;

74 int i, new_nb_items = 0;

76 p = item_str;

77 for (i = 0; i < *nb_items; i++) {

78 char *tstr = av_strtok(p, "|", &saveptr);

79 p = NULL;

80 new_nb_items += sscanf(tstr, "%f", &items[i]) == 1;

81 }

83 *nb_items = new_nb_items;

84 }

86 static av_cold void uninit(AVFilterContext *ctx)

87 {

88 AudioEchoContext *s = ctx->priv;

90 av_freep(&s->delay);

91 av_freep(&s->decay);

92 av_freep(&s->samples);

94 if (s->delayptrs)

95 av_freep(&s->delayptrs[0]);

96 av_freep(&s->delayptrs);

97 }

99 static av_cold int init(AVFilterContext *ctx)

100 {

101 AudioEchoContext *s = ctx->priv;

102 int nb_delays, nb_decays, i;

103

104 if (!s->delays || !s->decays) {

105 av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");

106 return AVERROR(EINVAL);

107 }

108

109 count_items(s->delays, &nb_delays);

110 count_items(s->decays, &nb_decays);

111

112 s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));

113 s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));

114 if (!s->delay || !s->decay)

115 return AVERROR(ENOMEM);

116

117 fill_items(s->delays, &nb_delays, s->delay);

118 fill_items(s->decays, &nb_decays, s->decay);

119

120 if (nb_delays != nb_decays) {

121 av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);

122 return AVERROR(EINVAL);

123 }

124

125 s->nb_echoes = nb_delays;

126 if (!s->nb_echoes) {

127 av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");

128 return AVERROR(EINVAL);

129 }

130

131 s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));

132 if (!s->samples)

133 return AVERROR(ENOMEM);

134

135 for (i = 0; i < nb_delays; i++) {

136 if (s->delay[i] <= 0 || s->delay[i] > 90000) {

137 av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);

138 return AVERROR(EINVAL);

139 }

140 if (s->decay[i] <= 0 || s->decay[i] > 1) {

141 av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);

142 return AVERROR(EINVAL);

143 }

144 }

145

146 s->next_pts = AV_NOPTS_VALUE;

147

148 av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);

149 return 0;

150 }

151

152 static int query_formats(AVFilterContext *ctx)

153 {

154 AVFilterChannelLayouts *layouts;

155 AVFilterFormats *formats;

156 static const enum AVSampleFormat sample_fmts[] = {

157 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,

158 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,

159 AV_SAMPLE_FMT_NONE

160 };

161

162 layouts = ff_all_channel_layouts();

163 if (!layouts)

164 return AVERROR(ENOMEM);

165 ff_set_common_channel_layouts(ctx, layouts);

166

167 formats = ff_make_format_list(sample_fmts);

168 if (!formats)

169 return AVERROR(ENOMEM);

170 ff_set_common_formats(ctx, formats);

171

172 formats = ff_all_samplerates();

173 if (!formats)

174 return AVERROR(ENOMEM);

175 ff_set_common_samplerates(ctx, formats);

176

177 return 0;

178 }

179

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

181

182 #define ECHO(name, type, min, max) \

183 static void echo_samples_## name ##p(AudioEchoContext *ctx, \

184 uint8_t **delayptrs, \

185 uint8_t * const *src, uint8_t **dst, \

186 int nb_samples, int channels) \

187 { \

188 const double out_gain = ctx->out_gain; \

189 const double in_gain = ctx->in_gain; \

190 const int nb_echoes = ctx->nb_echoes; \

191 const int max_samples = ctx->max_samples; \

192 int i, j, chan, av_uninit(index); \

193 \

194 av_assert1(channels > 0); /* would corrupt delay_index */ \

195 \

196 for (chan = 0; chan < channels; chan++) { \

197 const type *s = (type *)src[chan]; \

198 type *d = (type *)dst[chan]; \

199 type *dbuf = (type *)delayptrs[chan]; \

200 \

201 index = ctx->delay_index; \

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

203 double out, in; \

204 \

205 in = *s; \

206 out = in * in_gain; \

207 for (j = 0; j < nb_echoes; j++) { \

208 int ix = index + max_samples - ctx->samples[j]; \

209 ix = MOD(ix, max_samples); \

210 out += dbuf[ix] * ctx->decay[j]; \

211 } \

212 out *= out_gain; \

213 \

214 *d = av_clipd(out, min, max); \

215 dbuf[index] = in; \

216 \

217 index = MOD(index + 1, max_samples); \

218 } \

219 } \

220 ctx->delay_index = index; \

221 }

222

223 ECHO(dbl, double, -1.0, 1.0 )

224 ECHO(flt, float, -1.0, 1.0 )

225 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)

226 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)

227

228 static int config_output(AVFilterLink *outlink)

229 {

230 AVFilterContext *ctx = outlink->src;

231 AudioEchoContext *s = ctx->priv;

232 float volume = 1.0;

233 int i;

234

235 for (i = 0; i < s->nb_echoes; i++) {

236 s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;

237 s->max_samples = FFMAX(s->max_samples, s->samples[i]);

238 volume += s->decay[i];

239 }

240

241 if (s->max_samples <= 0) {

242 av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");

243 return AVERROR(EINVAL);

244 }

245 s->fade_out = s->max_samples;

246

247 if (volume * s->in_gain * s->out_gain > 1.0)

248 av_log(ctx, AV_LOG_WARNING,

249 "out_gain %f can cause saturation of output\n", s->out_gain);

250

251 switch (outlink->format) {

252 case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;

253 case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;

254 case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;

255 case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;

256 }

257

258

259 if (s->delayptrs)

260 av_freep(&s->delayptrs[0]);

261 av_freep(&s->delayptrs);

262

263 return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,

264 outlink->channels,

265 s->max_samples,

266 outlink->format, 0);

267 }

268

269 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)

270 {

271 AVFilterContext *ctx = inlink->dst;

272 AudioEchoContext *s = ctx->priv;

273 AVFrame *out_frame;

274

275 if (av_frame_is_writable(frame)) {

276 out_frame = frame;

277 } else {

278 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);

279 if (!out_frame)

280 return AVERROR(ENOMEM);

281 av_frame_copy_props(out_frame, frame);

282 }

283

284 s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,

285 frame->nb_samples, inlink->channels);

286

287 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);

288

289 if (frame != out_frame)

290 av_frame_free(&frame);

291

292 return ff_filter_frame(ctx->outputs[0], out_frame);

293 }

294

295 static int request_frame(AVFilterLink *outlink)

296 {

297 AVFilterContext *ctx = outlink->src;

298 AudioEchoContext *s = ctx->priv;

299 int ret;

300

301 ret = ff_request_frame(ctx->inputs[0]);

302

303 if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {

304 int nb_samples = FFMIN(s->fade_out, 2048);

305 AVFrame *frame;

306

307 frame = ff_get_audio_buffer(outlink, nb_samples);

308 if (!frame)

309 return AVERROR(ENOMEM);

310 s->fade_out -= nb_samples;

311

312 av_samples_set_silence(frame->extended_data, 0,

313 frame->nb_samples,

314 outlink->channels,

315 frame->format);

316

317 s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,

318 frame->nb_samples, outlink->channels);

319

320 frame->pts = s->next_pts;

321 if (s->next_pts != AV_NOPTS_VALUE)

322 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);

323

324 return ff_filter_frame(outlink, frame);

325 }

326

327 return ret;

328 }

329

330 static const AVFilterPad aecho_inputs[] = {

331 {

332 .name = "default",

333 .type = AVMEDIA_TYPE_AUDIO,

334 .filter_frame = filter_frame,

335 },

336 { NULL }

337 };

338

339 static const AVFilterPad aecho_outputs[] = {

340 {

341 .name = "default",

342 .request_frame = request_frame,

343 .config_props = config_output,

344 .type = AVMEDIA_TYPE_AUDIO,

345 },

346 { NULL }

347 };

348

349 AVFilter ff_af_aecho = {

350 .name = "aecho",

351 .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),

352 .query_formats = query_formats,

353 .priv_size = sizeof(AudioEchoContext),

354 .priv_class = &aecho_class,

355 .init = init,

356 .uninit = uninit,

357 .inputs = aecho_inputs,

358 .outputs = aecho_outputs,

359 };

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