2 * Copyright (c) 2013 Paul B Mahol
4 * This file is part of FFmpeg.
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.
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.
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
21 #include "libavutil/avstring.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
28 typedef struct ChanDelay {
35 typedef struct AudioDelayContext {
44 void (*delay_channel)(ChanDelay *d, int nb_samples,
45 const uint8_t *src, uint8_t *dst);
48 #define OFFSET(x) offsetof(AudioDelayContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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 },
56 AVFILTER_DEFINE_CLASS(adelay);
58 static int query_formats(AVFilterContext *ctx)
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,
69 layouts = ff_all_channel_layouts();
71 return AVERROR(ENOMEM);
72 ret = ff_set_common_channel_layouts(ctx, layouts);
76 formats = ff_make_format_list(sample_fmts);
78 return AVERROR(ENOMEM);
79 ret = ff_set_common_formats(ctx, formats);
83 formats = ff_all_samplerates();
85 return AVERROR(ENOMEM);
86 return ff_set_common_samplerates(ctx, formats);
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) \
93 const type *src = (type *)ssrc; \
94 type *dst = (type *)ddst; \
95 type *samples = (type *)d->samples; \
97 while (nb_samples) { \
98 if (d->delay_index < d->delay) { \
99 const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
101 memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
102 memset(dst, fill, len * sizeof(type)); \
103 d->delay_index += len; \
108 *dst = samples[d->index]; \
109 samples[d->index] = *src; \
113 d->index = d->index >= d->delay ? 0 : d->index; \
118 DELAY(u8, uint8_t, 0x80)
119 DELAY(s16, int16_t, 0)
120 DELAY(s32, int32_t, 0)
122 DELAY(dbl, double, 0)
124 static int config_input(AVFilterLink *inlink)
126 AVFilterContext *ctx = inlink->dst;
127 AudioDelayContext *s = ctx->priv;
128 char *p, *arg, *saveptr = NULL;
131 s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
133 return AVERROR(ENOMEM);
134 s->nb_delays = inlink->channels;
135 s->block_align = av_get_bytes_per_sample(inlink->format);
138 for (i = 0; i < s->nb_delays; i++) {
139 ChanDelay *d = &s->chandelay[i];
142 if (!(arg = av_strtok(p, "|", &saveptr)))
146 sscanf(arg, "%f", &delay);
148 d->delay = delay * inlink->sample_rate / 1000.0;
150 av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
151 return AVERROR(EINVAL);
155 for (i = 0; i < s->nb_delays; i++) {
156 ChanDelay *d = &s->chandelay[i];
161 d->samples = av_malloc_array(d->delay, s->block_align);
163 return AVERROR(ENOMEM);
165 s->max_delay = FFMAX(s->max_delay, d->delay);
169 av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n");
170 return AVERROR(EINVAL);
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;
184 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
186 AVFilterContext *ctx = inlink->dst;
187 AudioDelayContext *s = ctx->priv;
191 if (ctx->is_disabled || !s->delays)
192 return ff_filter_frame(ctx->outputs[0], frame);
194 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
196 return AVERROR(ENOMEM);
197 av_frame_copy_props(out_frame, frame);
199 for (i = 0; i < s->nb_delays; i++) {
200 ChanDelay *d = &s->chandelay[i];
201 const uint8_t *src = frame->extended_data[i];
202 uint8_t *dst = out_frame->extended_data[i];
205 memcpy(dst, src, frame->nb_samples * s->block_align);
207 s->delay_channel(d, frame->nb_samples, src, dst);
210 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
211 av_frame_free(&frame);
212 return ff_filter_frame(ctx->outputs[0], out_frame);
215 static int request_frame(AVFilterLink *outlink)
217 AVFilterContext *ctx = outlink->src;
218 AudioDelayContext *s = ctx->priv;
221 ret = ff_request_frame(ctx->inputs[0]);
222 if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
223 int nb_samples = FFMIN(s->max_delay, 2048);
226 frame = ff_get_audio_buffer(outlink, nb_samples);
228 return AVERROR(ENOMEM);
229 s->max_delay -= nb_samples;
231 av_samples_set_silence(frame->extended_data, 0,
236 frame->pts = s->next_pts;
237 if (s->next_pts != AV_NOPTS_VALUE)
238 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
240 ret = filter_frame(ctx->inputs[0], frame);
246 static av_cold void uninit(AVFilterContext *ctx)
248 AudioDelayContext *s = ctx->priv;
251 for (i = 0; i < s->nb_delays; i++)
252 av_freep(&s->chandelay[i].samples);
253 av_freep(&s->chandelay);
256 static const AVFilterPad adelay_inputs[] = {
259 .type = AVMEDIA_TYPE_AUDIO,
260 .config_props = config_input,
261 .filter_frame = filter_frame,
266 static const AVFilterPad adelay_outputs[] = {
269 .request_frame = request_frame,
270 .type = AVMEDIA_TYPE_AUDIO,
275 AVFilter ff_af_adelay = {
277 .description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
278 .query_formats = query_formats,
279 .priv_size = sizeof(AudioDelayContext),
280 .priv_class = &adelay_class,
282 .inputs = adelay_inputs,
283 .outputs = adelay_outputs,
284 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,