Merge commit 'a6653787a4b1e89170ff7962312dd110769c83a3'
[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
68     layouts = ff_all_channel_layouts();
69     if (!layouts)
70         return AVERROR(ENOMEM);
71     ff_set_common_channel_layouts(ctx, layouts);
72
73     formats = ff_make_format_list(sample_fmts);
74     if (!formats)
75         return AVERROR(ENOMEM);
76     ff_set_common_formats(ctx, formats);
77
78     formats = ff_all_samplerates();
79     if (!formats)
80         return AVERROR(ENOMEM);
81     ff_set_common_samplerates(ctx, formats);
82
83     return 0;
84 }
85
86 #define DELAY(name, type, fill)                                           \
87 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples,       \
88                                       const uint8_t *ssrc, uint8_t *ddst) \
89 {                                                                         \
90     const type *src = (type *)ssrc;                                       \
91     type *dst = (type *)ddst;                                             \
92     type *samples = (type *)d->samples;                                   \
93                                                                           \
94     while (nb_samples) {                                                  \
95         if (d->delay_index < d->delay) {                                  \
96             const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
97                                                                           \
98             memcpy(&samples[d->delay_index], src, len * sizeof(type));    \
99             memset(dst, fill, len * sizeof(type));                        \
100             d->delay_index += len;                                        \
101             src += len;                                                   \
102             dst += len;                                                   \
103             nb_samples -= len;                                            \
104         } else {                                                          \
105             *dst = samples[d->index];                                     \
106             samples[d->index] = *src;                                     \
107             nb_samples--;                                                 \
108             d->index++;                                                   \
109             src++, dst++;                                                 \
110             d->index = d->index >= d->delay ? 0 : d->index;               \
111         }                                                                 \
112     }                                                                     \
113 }
114
115 DELAY(u8,  uint8_t, 0x80)
116 DELAY(s16, int16_t, 0)
117 DELAY(s32, int32_t, 0)
118 DELAY(flt, float,   0)
119 DELAY(dbl, double,  0)
120
121 static int config_input(AVFilterLink *inlink)
122 {
123     AVFilterContext *ctx = inlink->dst;
124     AudioDelayContext *s = ctx->priv;
125     char *p, *arg, *saveptr = NULL;
126     int i;
127
128     s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
129     if (!s->chandelay)
130         return AVERROR(ENOMEM);
131     s->nb_delays = inlink->channels;
132     s->block_align = av_get_bytes_per_sample(inlink->format);
133
134     p = s->delays;
135     for (i = 0; i < s->nb_delays; i++) {
136         ChanDelay *d = &s->chandelay[i];
137         float delay;
138
139         if (!(arg = av_strtok(p, "|", &saveptr)))
140             break;
141
142         p = NULL;
143         sscanf(arg, "%f", &delay);
144
145         d->delay = delay * inlink->sample_rate / 1000.0;
146         if (d->delay < 0) {
147             av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
148             return AVERROR(EINVAL);
149         }
150     }
151
152     for (i = 0; i < s->nb_delays; i++) {
153         ChanDelay *d = &s->chandelay[i];
154
155         if (!d->delay)
156             continue;
157
158         d->samples = av_malloc_array(d->delay, s->block_align);
159         if (!d->samples)
160             return AVERROR(ENOMEM);
161
162         s->max_delay = FFMAX(s->max_delay, d->delay);
163     }
164
165     if (!s->max_delay) {
166         av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n");
167         return AVERROR(EINVAL);
168     }
169
170     switch (inlink->format) {
171     case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
172     case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
173     case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
174     case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
175     case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
176     }
177
178     return 0;
179 }
180
181 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
182 {
183     AVFilterContext *ctx = inlink->dst;
184     AudioDelayContext *s = ctx->priv;
185     AVFrame *out_frame;
186     int i;
187
188     if (ctx->is_disabled || !s->delays)
189         return ff_filter_frame(ctx->outputs[0], frame);
190
191     out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
192     if (!out_frame)
193         return AVERROR(ENOMEM);
194     av_frame_copy_props(out_frame, frame);
195
196     for (i = 0; i < s->nb_delays; i++) {
197         ChanDelay *d = &s->chandelay[i];
198         const uint8_t *src = frame->extended_data[i];
199         uint8_t *dst = out_frame->extended_data[i];
200
201         if (!d->delay)
202             memcpy(dst, src, frame->nb_samples * s->block_align);
203         else
204             s->delay_channel(d, frame->nb_samples, src, dst);
205     }
206
207     s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
208     av_frame_free(&frame);
209     return ff_filter_frame(ctx->outputs[0], out_frame);
210 }
211
212 static int request_frame(AVFilterLink *outlink)
213 {
214     AVFilterContext *ctx = outlink->src;
215     AudioDelayContext *s = ctx->priv;
216     int ret;
217
218     ret = ff_request_frame(ctx->inputs[0]);
219     if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
220         int nb_samples = FFMIN(s->max_delay, 2048);
221         AVFrame *frame;
222
223         frame = ff_get_audio_buffer(outlink, nb_samples);
224         if (!frame)
225             return AVERROR(ENOMEM);
226         s->max_delay -= nb_samples;
227
228         av_samples_set_silence(frame->extended_data, 0,
229                                frame->nb_samples,
230                                outlink->channels,
231                                frame->format);
232
233         frame->pts = s->next_pts;
234         if (s->next_pts != AV_NOPTS_VALUE)
235             s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
236
237         ret = filter_frame(ctx->inputs[0], frame);
238     }
239
240     return ret;
241 }
242
243 static av_cold void uninit(AVFilterContext *ctx)
244 {
245     AudioDelayContext *s = ctx->priv;
246     int i;
247
248     for (i = 0; i < s->nb_delays; i++)
249         av_freep(&s->chandelay[i].samples);
250     av_freep(&s->chandelay);
251 }
252
253 static const AVFilterPad adelay_inputs[] = {
254     {
255         .name         = "default",
256         .type         = AVMEDIA_TYPE_AUDIO,
257         .config_props = config_input,
258         .filter_frame = filter_frame,
259     },
260     { NULL }
261 };
262
263 static const AVFilterPad adelay_outputs[] = {
264     {
265         .name          = "default",
266         .request_frame = request_frame,
267         .type          = AVMEDIA_TYPE_AUDIO,
268     },
269     { NULL }
270 };
271
272 AVFilter ff_af_adelay = {
273     .name          = "adelay",
274     .description   = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
275     .query_formats = query_formats,
276     .priv_size     = sizeof(AudioDelayContext),
277     .priv_class    = &adelay_class,
278     .uninit        = uninit,
279     .inputs        = adelay_inputs,
280     .outputs       = adelay_outputs,
281     .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
282 };