avfilter/af_afftdn: add alias for sample_noise end
[ffmpeg.git] / libavfilter / af_adelay.c
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/avstring.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "filters.h"
27 #include "internal.h"
28
29 typedef struct ChanDelay {
30     int delay;
31     unsigned delay_index;
32     unsigned index;
33     uint8_t *samples;
34 } ChanDelay;
35
36 typedef struct AudioDelayContext {
37     const AVClass *class;
38     char *delays;
39     ChanDelay *chandelay;
40     int nb_delays;
41     int block_align;
42     int64_t padding;
43     int64_t max_delay;
44     int64_t next_pts;
45     int eof;
46
47     void (*delay_channel)(ChanDelay *d, int nb_samples,
48                           const uint8_t *src, uint8_t *dst);
49 } AudioDelayContext;
50
51 #define OFFSET(x) offsetof(AudioDelayContext, x)
52 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
53
54 static const AVOption adelay_options[] = {
55     { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
56     { NULL }
57 };
58
59 AVFILTER_DEFINE_CLASS(adelay);
60
61 static int query_formats(AVFilterContext *ctx)
62 {
63     AVFilterChannelLayouts *layouts;
64     AVFilterFormats *formats;
65     static const enum AVSampleFormat sample_fmts[] = {
66         AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
67         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
68         AV_SAMPLE_FMT_NONE
69     };
70     int ret;
71
72     layouts = ff_all_channel_counts();
73     if (!layouts)
74         return AVERROR(ENOMEM);
75     ret = ff_set_common_channel_layouts(ctx, layouts);
76     if (ret < 0)
77         return ret;
78
79     formats = ff_make_format_list(sample_fmts);
80     if (!formats)
81         return AVERROR(ENOMEM);
82     ret = ff_set_common_formats(ctx, formats);
83     if (ret < 0)
84         return ret;
85
86     formats = ff_all_samplerates();
87     if (!formats)
88         return AVERROR(ENOMEM);
89     return ff_set_common_samplerates(ctx, formats);
90 }
91
92 #define DELAY(name, type, fill)                                           \
93 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples,       \
94                                       const uint8_t *ssrc, uint8_t *ddst) \
95 {                                                                         \
96     const type *src = (type *)ssrc;                                       \
97     type *dst = (type *)ddst;                                             \
98     type *samples = (type *)d->samples;                                   \
99                                                                           \
100     while (nb_samples) {                                                  \
101         if (d->delay_index < d->delay) {                                  \
102             const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
103                                                                           \
104             memcpy(&samples[d->delay_index], src, len * sizeof(type));    \
105             memset(dst, fill, len * sizeof(type));                        \
106             d->delay_index += len;                                        \
107             src += len;                                                   \
108             dst += len;                                                   \
109             nb_samples -= len;                                            \
110         } else {                                                          \
111             *dst = samples[d->index];                                     \
112             samples[d->index] = *src;                                     \
113             nb_samples--;                                                 \
114             d->index++;                                                   \
115             src++, dst++;                                                 \
116             d->index = d->index >= d->delay ? 0 : d->index;               \
117         }                                                                 \
118     }                                                                     \
119 }
120
121 DELAY(u8,  uint8_t, 0x80)
122 DELAY(s16, int16_t, 0)
123 DELAY(s32, int32_t, 0)
124 DELAY(flt, float,   0)
125 DELAY(dbl, double,  0)
126
127 static int config_input(AVFilterLink *inlink)
128 {
129     AVFilterContext *ctx = inlink->dst;
130     AudioDelayContext *s = ctx->priv;
131     char *p, *arg, *saveptr = NULL;
132     int i;
133
134     s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
135     if (!s->chandelay)
136         return AVERROR(ENOMEM);
137     s->nb_delays = inlink->channels;
138     s->block_align = av_get_bytes_per_sample(inlink->format);
139
140     p = s->delays;
141     for (i = 0; i < s->nb_delays; i++) {
142         ChanDelay *d = &s->chandelay[i];
143         float delay;
144         char type = 0;
145         int ret;
146
147         if (!(arg = av_strtok(p, "|", &saveptr)))
148             break;
149
150         p = NULL;
151
152         ret = sscanf(arg, "%d%c", &d->delay, &type);
153         if (ret != 2 || type != 'S') {
154             sscanf(arg, "%f", &delay);
155             d->delay = delay * inlink->sample_rate / 1000.0;
156         }
157
158         if (d->delay < 0) {
159             av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
160             return AVERROR(EINVAL);
161         }
162     }
163
164     s->padding = s->chandelay[0].delay;
165     for (i = 1; i < s->nb_delays; i++) {
166         ChanDelay *d = &s->chandelay[i];
167
168         s->padding = FFMIN(s->padding, d->delay);
169     }
170
171     if (s->padding) {
172         for (i = 0; i < s->nb_delays; i++) {
173             ChanDelay *d = &s->chandelay[i];
174
175             d->delay -= s->padding;
176         }
177     }
178
179     for (i = 0; i < s->nb_delays; i++) {
180         ChanDelay *d = &s->chandelay[i];
181
182         if (!d->delay)
183             continue;
184
185         d->samples = av_malloc_array(d->delay, s->block_align);
186         if (!d->samples)
187             return AVERROR(ENOMEM);
188
189         s->max_delay = FFMAX(s->max_delay, d->delay);
190     }
191
192     switch (inlink->format) {
193     case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
194     case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
195     case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
196     case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
197     case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
198     }
199
200     return 0;
201 }
202
203 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
204 {
205     AVFilterContext *ctx = inlink->dst;
206     AudioDelayContext *s = ctx->priv;
207     AVFrame *out_frame;
208     int i;
209
210     if (ctx->is_disabled || !s->delays)
211         return ff_filter_frame(ctx->outputs[0], frame);
212
213     out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
214     if (!out_frame) {
215         av_frame_free(&frame);
216         return AVERROR(ENOMEM);
217     }
218     av_frame_copy_props(out_frame, frame);
219
220     for (i = 0; i < s->nb_delays; i++) {
221         ChanDelay *d = &s->chandelay[i];
222         const uint8_t *src = frame->extended_data[i];
223         uint8_t *dst = out_frame->extended_data[i];
224
225         if (!d->delay)
226             memcpy(dst, src, frame->nb_samples * s->block_align);
227         else
228             s->delay_channel(d, frame->nb_samples, src, dst);
229     }
230
231     out_frame->pts = s->next_pts;
232     s->next_pts += av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
233     av_frame_free(&frame);
234     return ff_filter_frame(ctx->outputs[0], out_frame);
235 }
236
237 static int activate(AVFilterContext *ctx)
238 {
239     AVFilterLink *inlink = ctx->inputs[0];
240     AVFilterLink *outlink = ctx->outputs[0];
241     AudioDelayContext *s = ctx->priv;
242     AVFrame *frame = NULL;
243     int ret, status;
244     int64_t pts;
245
246     FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
247
248     if (s->padding) {
249         int nb_samples = FFMIN(s->padding, 2048);
250
251         frame = ff_get_audio_buffer(outlink, nb_samples);
252         if (!frame)
253             return AVERROR(ENOMEM);
254         s->padding -= nb_samples;
255
256         av_samples_set_silence(frame->extended_data, 0,
257                                frame->nb_samples,
258                                outlink->channels,
259                                frame->format);
260
261         frame->pts = s->next_pts;
262         if (s->next_pts != AV_NOPTS_VALUE)
263             s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
264
265         return ff_filter_frame(outlink, frame);
266     }
267
268     ret = ff_inlink_consume_frame(inlink, &frame);
269     if (ret < 0)
270         return ret;
271
272     if (ret > 0)
273         return filter_frame(inlink, frame);
274
275     if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
276         if (status == AVERROR_EOF)
277             s->eof = 1;
278     }
279
280     if (s->eof && s->max_delay) {
281         int nb_samples = FFMIN(s->max_delay, 2048);
282
283         frame = ff_get_audio_buffer(outlink, nb_samples);
284         if (!frame)
285             return AVERROR(ENOMEM);
286         s->max_delay -= nb_samples;
287
288         av_samples_set_silence(frame->extended_data, 0,
289                                frame->nb_samples,
290                                outlink->channels,
291                                frame->format);
292
293         frame->pts = s->next_pts;
294         return filter_frame(inlink, frame);
295     }
296
297     if (s->eof && s->max_delay == 0) {
298         ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
299         return 0;
300     }
301
302     if (!s->eof)
303         FF_FILTER_FORWARD_WANTED(outlink, inlink);
304
305     return FFERROR_NOT_READY;
306 }
307
308 static av_cold void uninit(AVFilterContext *ctx)
309 {
310     AudioDelayContext *s = ctx->priv;
311
312     if (s->chandelay) {
313         for (int i = 0; i < s->nb_delays; i++)
314             av_freep(&s->chandelay[i].samples);
315     }
316     av_freep(&s->chandelay);
317 }
318
319 static const AVFilterPad adelay_inputs[] = {
320     {
321         .name         = "default",
322         .type         = AVMEDIA_TYPE_AUDIO,
323         .config_props = config_input,
324     },
325     { NULL }
326 };
327
328 static const AVFilterPad adelay_outputs[] = {
329     {
330         .name = "default",
331         .type = AVMEDIA_TYPE_AUDIO,
332     },
333     { NULL }
334 };
335
336 AVFilter ff_af_adelay = {
337     .name          = "adelay",
338     .description   = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
339     .query_formats = query_formats,
340     .priv_size     = sizeof(AudioDelayContext),
341     .priv_class    = &adelay_class,
342     .activate      = activate,
343     .uninit        = uninit,
344     .inputs        = adelay_inputs,
345     .outputs       = adelay_outputs,
346     .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
347 };