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1 /*
2  * Copyright (c) 2013 Paul B Mahol
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  */
20 
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 "filters.h"
28 #include "internal.h"
29 
30 typedef struct AudioEchoContext {
31     const AVClass *class;
32     float in_gain, out_gain;
33     char *delays, *decays;
34     float *delay, *decay;
35     int nb_echoes;
36     int delay_index;
37     uint8_t **delayptrs;
38     int max_samples, fade_out;
39     int *samples;
40     int eof;
41     int64_t next_pts;
42 
43     void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
44                          uint8_t * const *src, uint8_t **dst,
45                          int nb_samples, int channels);
46 } AudioEchoContext;
47 
48 #define OFFSET(x) offsetof(AudioEchoContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
50 
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 },
56     { NULL }
57 };
58 
59 AVFILTER_DEFINE_CLASS(aecho);
60 
count_items(char * item_str,int * nb_items)61 static void count_items(char *item_str, int *nb_items)
62 {
63     char *p;
64 
65     *nb_items = 1;
66     for (p = item_str; *p; p++) {
67         if (*p == '|')
68             (*nb_items)++;
69     }
70 
71 }
72 
fill_items(char * item_str,int * nb_items,float * items)73 static void fill_items(char *item_str, int *nb_items, float *items)
74 {
75     char *p, *saveptr = NULL;
76     int i, new_nb_items = 0;
77 
78     p = item_str;
79     for (i = 0; i < *nb_items; i++) {
80         char *tstr = av_strtok(p, "|", &saveptr);
81         p = NULL;
82         if (tstr)
83             new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 1;
84     }
85 
86     *nb_items = new_nb_items;
87 }
88 
uninit(AVFilterContext * ctx)89 static av_cold void uninit(AVFilterContext *ctx)
90 {
91     AudioEchoContext *s = ctx->priv;
92 
93     av_freep(&s->delay);
94     av_freep(&s->decay);
95     av_freep(&s->samples);
96 
97     if (s->delayptrs)
98         av_freep(&s->delayptrs[0]);
99     av_freep(&s->delayptrs);
100 }
101 
init(AVFilterContext * ctx)102 static av_cold int init(AVFilterContext *ctx)
103 {
104     AudioEchoContext *s = ctx->priv;
105     int nb_delays, nb_decays, i;
106 
107     if (!s->delays || !s->decays) {
108         av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
109         return AVERROR(EINVAL);
110     }
111 
112     count_items(s->delays, &nb_delays);
113     count_items(s->decays, &nb_decays);
114 
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);
119 
120     fill_items(s->delays, &nb_delays, s->delay);
121     fill_items(s->decays, &nb_decays, s->decay);
122 
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);
126     }
127 
128     s->nb_echoes = nb_delays;
129     if (!s->nb_echoes) {
130         av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
131         return AVERROR(EINVAL);
132     }
133 
134     s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
135     if (!s->samples)
136         return AVERROR(ENOMEM);
137 
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);
142         }
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);
146         }
147     }
148 
149     s->next_pts = AV_NOPTS_VALUE;
150 
151     av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
152     return 0;
153 }
154 
query_formats(AVFilterContext * ctx)155 static int query_formats(AVFilterContext *ctx)
156 {
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,
162         AV_SAMPLE_FMT_NONE
163     };
164     int ret;
165 
166     layouts = ff_all_channel_counts();
167     if (!layouts)
168         return AVERROR(ENOMEM);
169     ret = ff_set_common_channel_layouts(ctx, layouts);
170     if (ret < 0)
171         return ret;
172 
173     formats = ff_make_format_list(sample_fmts);
174     if (!formats)
175         return AVERROR(ENOMEM);
176     ret = ff_set_common_formats(ctx, formats);
177     if (ret < 0)
178         return ret;
179 
180     formats = ff_all_samplerates();
181     if (!formats)
182         return AVERROR(ENOMEM);
183     return ff_set_common_samplerates(ctx, formats);
184 }
185 
186 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
187 
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)          \
193 {                                                                           \
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);                                       \
199                                                                             \
200     av_assert1(channels > 0); /* would corrupt delay_index */               \
201                                                                             \
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];                               \
206                                                                             \
207         index = ctx->delay_index;                                           \
208         for (i = 0; i < nb_samples; i++, s++, d++) {                        \
209             double out, in;                                                 \
210                                                                             \
211             in = *s;                                                        \
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];                            \
217             }                                                               \
218             out *= out_gain;                                                \
219                                                                             \
220             *d = av_clipd(out, min, max);                                   \
221             dbuf[index] = in;                                               \
222                                                                             \
223             index = MOD(index + 1, max_samples);                            \
224         }                                                                   \
225     }                                                                       \
226     ctx->delay_index = index;                                               \
227 }
228 
229 ECHO(dbl, double,  -1.0,      1.0      )
230 ECHO(flt, float,   -1.0,      1.0      )
ECHO(s16,int16_t,INT16_MIN,INT16_MAX)231 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
232 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
233 
234 static int config_output(AVFilterLink *outlink)
235 {
236     AVFilterContext *ctx = outlink->src;
237     AudioEchoContext *s = ctx->priv;
238     float volume = 1.0;
239     int i;
240 
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];
245     }
246 
247     if (s->max_samples <= 0) {
248         av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
249         return AVERROR(EINVAL);
250     }
251     s->fade_out = s->max_samples;
252 
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);
256 
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;
262     }
263 
264 
265     if (s->delayptrs)
266         av_freep(&s->delayptrs[0]);
267     av_freep(&s->delayptrs);
268 
269     return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
270                                               outlink->channels,
271                                               s->max_samples,
272                                               outlink->format, 0);
273 }
274 
filter_frame(AVFilterLink * inlink,AVFrame * frame)275 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
276 {
277     AVFilterContext *ctx = inlink->dst;
278     AudioEchoContext *s = ctx->priv;
279     AVFrame *out_frame;
280 
281     if (av_frame_is_writable(frame)) {
282         out_frame = frame;
283     } else {
284         out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
285         if (!out_frame) {
286             av_frame_free(&frame);
287             return AVERROR(ENOMEM);
288         }
289         av_frame_copy_props(out_frame, frame);
290     }
291 
292     s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
293                     frame->nb_samples, inlink->channels);
294 
295     s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
296 
297     if (frame != out_frame)
298         av_frame_free(&frame);
299 
300     return ff_filter_frame(ctx->outputs[0], out_frame);
301 }
302 
request_frame(AVFilterLink * outlink)303 static int request_frame(AVFilterLink *outlink)
304 {
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);
309 
310     if (!frame)
311         return AVERROR(ENOMEM);
312     s->fade_out -= nb_samples;
313 
314     av_samples_set_silence(frame->extended_data, 0,
315                            frame->nb_samples,
316                            outlink->channels,
317                            frame->format);
318 
319     s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
320                     frame->nb_samples, outlink->channels);
321 
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);
325 
326     return ff_filter_frame(outlink, frame);
327 }
328 
activate(AVFilterContext * ctx)329 static int activate(AVFilterContext *ctx)
330 {
331     AVFilterLink *inlink = ctx->inputs[0];
332     AVFilterLink *outlink = ctx->outputs[0];
333     AudioEchoContext *s = ctx->priv;
334     AVFrame *in;
335     int ret, status;
336     int64_t pts;
337 
338     FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
339 
340     ret = ff_inlink_consume_frame(inlink, &in);
341     if (ret < 0)
342         return ret;
343     if (ret > 0)
344         return filter_frame(inlink, in);
345 
346     if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
347         if (status == AVERROR_EOF)
348             s->eof = 1;
349     }
350 
351     if (s->eof && s->fade_out <= 0) {
352         ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
353         return 0;
354     }
355 
356     if (!s->eof)
357         FF_FILTER_FORWARD_WANTED(outlink, inlink);
358 
359     return request_frame(outlink);
360 }
361 
362 static const AVFilterPad aecho_inputs[] = {
363     {
364         .name         = "default",
365         .type         = AVMEDIA_TYPE_AUDIO,
366     },
367     { NULL }
368 };
369 
370 static const AVFilterPad aecho_outputs[] = {
371     {
372         .name          = "default",
373         .config_props  = config_output,
374         .type          = AVMEDIA_TYPE_AUDIO,
375     },
376     { NULL }
377 };
378 
379 AVFilter ff_af_aecho = {
380     .name          = "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,
385     .init          = init,
386     .activate      = activate,
387     .uninit        = uninit,
388     .inputs        = aecho_inputs,
389     .outputs       = aecho_outputs,
390 };
391