| 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_free(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 | }; |