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