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
28typedef struct ChanDelay {
29    int delay;
30    unsigned delay_index;
31    unsigned index;
32    uint8_t *samples;
33} ChanDelay;
34
35typedef 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
51static 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
56AVFILTER_DEFINE_CLASS(adelay);
57
58static 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)                                           \
87static 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
115DELAY(u8,  uint8_t, 0x80)
116DELAY(s16, int16_t, 0)
117DELAY(s32, int32_t, 0)
118DELAY(flt, float,   0)
119DELAY(dbl, double,  0)
120
121static 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
181static 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
212static 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
243static 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_free(s->chandelay[i].samples);
250    av_freep(&s->chandelay);
251}
252
253static 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
263static const AVFilterPad adelay_outputs[] = {
264    {
265        .name          = "default",
266        .request_frame = request_frame,
267        .type          = AVMEDIA_TYPE_AUDIO,
268    },
269    { NULL }
270};
271
272AVFilter 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};
283