2 * Copyright (c) 2013 Paul B Mahol
4 * This file is part of FFmpeg.
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.
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.
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
21 #include "libavutil/avassert.h"
22 #include "libavutil/avstring.h"
23 #include "libavutil/opt.h"
24 #include "libavutil/samplefmt.h"
30 typedef struct AudioEchoContext {
32 float in_gain, out_gain;
33 char *delays, *decays;
38 int max_samples, fade_out;
43 void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
44 uint8_t * const *src, uint8_t **dst,
45 int nb_samples, int channels);
48 #define OFFSET(x) offsetof(AudioEchoContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
51 static const AVOption aecho_options[] = {
52 { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
53 { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
54 { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
55 { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
59 AVFILTER_DEFINE_CLASS(aecho);
61 static void count_items(char *item_str, int *nb_items)
66 for (p = item_str; *p; p++) {
73 static void fill_items(char *item_str, int *nb_items, float *items)
75 char *p, *saveptr = NULL;
76 int i, new_nb_items = 0;
79 for (i = 0; i < *nb_items; i++) {
80 char *tstr = av_strtok(p, "|", &saveptr);
83 new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 1;
86 *nb_items = new_nb_items;
89 static av_cold void uninit(AVFilterContext *ctx)
91 AudioEchoContext *s = ctx->priv;
95 av_freep(&s->samples);
98 av_freep(&s->delayptrs[0]);
99 av_freep(&s->delayptrs);
102 static av_cold int init(AVFilterContext *ctx)
104 AudioEchoContext *s = ctx->priv;
105 int nb_delays, nb_decays, i;
107 if (!s->delays || !s->decays) {
108 av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
109 return AVERROR(EINVAL);
112 count_items(s->delays, &nb_delays);
113 count_items(s->decays, &nb_decays);
115 s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
116 s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
117 if (!s->delay || !s->decay)
118 return AVERROR(ENOMEM);
120 fill_items(s->delays, &nb_delays, s->delay);
121 fill_items(s->decays, &nb_decays, s->decay);
123 if (nb_delays != nb_decays) {
124 av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
125 return AVERROR(EINVAL);
128 s->nb_echoes = nb_delays;
130 av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
131 return AVERROR(EINVAL);
134 s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
136 return AVERROR(ENOMEM);
138 for (i = 0; i < nb_delays; i++) {
139 if (s->delay[i] <= 0 || s->delay[i] > 90000) {
140 av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
141 return AVERROR(EINVAL);
143 if (s->decay[i] <= 0 || s->decay[i] > 1) {
144 av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
145 return AVERROR(EINVAL);
149 s->next_pts = AV_NOPTS_VALUE;
151 av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
155 static int query_formats(AVFilterContext *ctx)
157 AVFilterChannelLayouts *layouts;
158 AVFilterFormats *formats;
159 static const enum AVSampleFormat sample_fmts[] = {
160 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
161 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
166 layouts = ff_all_channel_counts();
168 return AVERROR(ENOMEM);
169 ret = ff_set_common_channel_layouts(ctx, layouts);
173 formats = ff_make_format_list(sample_fmts);
175 return AVERROR(ENOMEM);
176 ret = ff_set_common_formats(ctx, formats);
180 formats = ff_all_samplerates();
182 return AVERROR(ENOMEM);
183 return ff_set_common_samplerates(ctx, formats);
186 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
188 #define ECHO(name, type, min, max) \
189 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
190 uint8_t **delayptrs, \
191 uint8_t * const *src, uint8_t **dst, \
192 int nb_samples, int channels) \
194 const double out_gain = ctx->out_gain; \
195 const double in_gain = ctx->in_gain; \
196 const int nb_echoes = ctx->nb_echoes; \
197 const int max_samples = ctx->max_samples; \
198 int i, j, chan, av_uninit(index); \
200 av_assert1(channels > 0); /* would corrupt delay_index */ \
202 for (chan = 0; chan < channels; chan++) { \
203 const type *s = (type *)src[chan]; \
204 type *d = (type *)dst[chan]; \
205 type *dbuf = (type *)delayptrs[chan]; \
207 index = ctx->delay_index; \
208 for (i = 0; i < nb_samples; i++, s++, d++) { \
212 out = in * in_gain; \
213 for (j = 0; j < nb_echoes; j++) { \
214 int ix = index + max_samples - ctx->samples[j]; \
215 ix = MOD(ix, max_samples); \
216 out += dbuf[ix] * ctx->decay[j]; \
220 *d = av_clipd(out, min, max); \
223 index = MOD(index + 1, max_samples); \
226 ctx->delay_index = index; \
229 ECHO(dbl, double, -1.0, 1.0 )
230 ECHO(flt, float, -1.0, 1.0 )
231 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
232 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
234 static int config_output(AVFilterLink *outlink)
236 AVFilterContext *ctx = outlink->src;
237 AudioEchoContext *s = ctx->priv;
241 for (i = 0; i < s->nb_echoes; i++) {
242 s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
243 s->max_samples = FFMAX(s->max_samples, s->samples[i]);
244 volume += s->decay[i];
247 if (s->max_samples <= 0) {
248 av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
249 return AVERROR(EINVAL);
251 s->fade_out = s->max_samples;
253 if (volume * s->in_gain * s->out_gain > 1.0)
254 av_log(ctx, AV_LOG_WARNING,
255 "out_gain %f can cause saturation of output\n", s->out_gain);
257 switch (outlink->format) {
258 case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
259 case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
260 case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
261 case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
266 av_freep(&s->delayptrs[0]);
267 av_freep(&s->delayptrs);
269 return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
275 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
277 AVFilterContext *ctx = inlink->dst;
278 AudioEchoContext *s = ctx->priv;
281 if (av_frame_is_writable(frame)) {
284 out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
286 av_frame_free(&frame);
287 return AVERROR(ENOMEM);
289 av_frame_copy_props(out_frame, frame);
292 s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
293 frame->nb_samples, inlink->channels);
295 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
297 if (frame != out_frame)
298 av_frame_free(&frame);
300 return ff_filter_frame(ctx->outputs[0], out_frame);
303 static int request_frame(AVFilterLink *outlink)
305 AVFilterContext *ctx = outlink->src;
306 AudioEchoContext *s = ctx->priv;
307 int nb_samples = FFMIN(s->fade_out, 2048);
308 AVFrame *frame = ff_get_audio_buffer(outlink, nb_samples);
311 return AVERROR(ENOMEM);
312 s->fade_out -= nb_samples;
314 av_samples_set_silence(frame->extended_data, 0,
319 s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
320 frame->nb_samples, outlink->channels);
322 frame->pts = s->next_pts;
323 if (s->next_pts != AV_NOPTS_VALUE)
324 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
326 return ff_filter_frame(outlink, frame);
329 static int activate(AVFilterContext *ctx)
331 AVFilterLink *inlink = ctx->inputs[0];
332 AVFilterLink *outlink = ctx->outputs[0];
333 AudioEchoContext *s = ctx->priv;
338 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
340 ret = ff_inlink_consume_frame(inlink, &in);
344 return filter_frame(inlink, in);
346 if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
347 if (status == AVERROR_EOF)
351 if (s->eof && s->fade_out <= 0) {
352 ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
357 FF_FILTER_FORWARD_WANTED(outlink, inlink);
359 return request_frame(outlink);
362 static const AVFilterPad aecho_inputs[] = {
365 .type = AVMEDIA_TYPE_AUDIO,
370 static const AVFilterPad aecho_outputs[] = {
373 .config_props = config_output,
374 .type = AVMEDIA_TYPE_AUDIO,
379 AVFilter ff_af_aecho = {
381 .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
382 .query_formats = query_formats,
383 .priv_size = sizeof(AudioEchoContext),
384 .priv_class = &aecho_class,
386 .activate = activate,
388 .inputs = aecho_inputs,
389 .outputs = aecho_outputs,