mirror of
https://github.com/FFmpeg/FFmpeg.git
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334 lines
11 KiB
C
334 lines
11 KiB
C
/*
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* Copyright (c) 2013 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avstring.h"
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#include "libavutil/eval.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "internal.h"
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typedef struct ChanDelay {
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int64_t delay;
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size_t delay_index;
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size_t index;
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uint8_t *samples;
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} ChanDelay;
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typedef struct AudioDelayContext {
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const AVClass *class;
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int all;
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char *delays;
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ChanDelay *chandelay;
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int nb_delays;
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int block_align;
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int64_t padding;
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int64_t max_delay;
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int64_t next_pts;
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int eof;
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void (*delay_channel)(ChanDelay *d, int nb_samples,
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const uint8_t *src, uint8_t *dst);
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} AudioDelayContext;
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#define OFFSET(x) offsetof(AudioDelayContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption adelay_options[] = {
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{ "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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{ "all", "use last available delay for remained channels", OFFSET(all), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(adelay);
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#define DELAY(name, type, fill) \
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static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
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const uint8_t *ssrc, uint8_t *ddst) \
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{ \
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const type *src = (type *)ssrc; \
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type *dst = (type *)ddst; \
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type *samples = (type *)d->samples; \
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\
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while (nb_samples) { \
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if (d->delay_index < d->delay) { \
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const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
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\
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memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
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memset(dst, fill, len * sizeof(type)); \
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d->delay_index += len; \
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src += len; \
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dst += len; \
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nb_samples -= len; \
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} else { \
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*dst = samples[d->index]; \
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samples[d->index] = *src; \
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nb_samples--; \
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d->index++; \
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src++, dst++; \
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d->index = d->index >= d->delay ? 0 : d->index; \
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} \
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} \
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}
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DELAY(u8, uint8_t, 0x80)
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DELAY(s16, int16_t, 0)
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DELAY(s32, int32_t, 0)
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DELAY(flt, float, 0)
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DELAY(dbl, double, 0)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDelayContext *s = ctx->priv;
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char *p, *arg, *saveptr = NULL;
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int i;
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s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
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if (!s->chandelay)
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return AVERROR(ENOMEM);
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s->nb_delays = inlink->channels;
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s->block_align = av_get_bytes_per_sample(inlink->format);
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p = s->delays;
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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float delay, div;
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char type = 0;
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int ret;
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if (!(arg = av_strtok(p, "|", &saveptr)))
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break;
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p = NULL;
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ret = av_sscanf(arg, "%"SCNd64"%c", &d->delay, &type);
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if (ret != 2 || type != 'S') {
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div = type == 's' ? 1.0 : 1000.0;
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if (av_sscanf(arg, "%f", &delay) != 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid syntax for delay.\n");
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return AVERROR(EINVAL);
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}
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d->delay = delay * inlink->sample_rate / div;
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}
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if (d->delay < 0) {
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av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
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return AVERROR(EINVAL);
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}
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}
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if (s->all && i) {
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for (int j = i; j < s->nb_delays; j++)
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s->chandelay[j].delay = s->chandelay[i-1].delay;
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}
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s->padding = s->chandelay[0].delay;
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for (i = 1; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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s->padding = FFMIN(s->padding, d->delay);
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}
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if (s->padding) {
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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d->delay -= s->padding;
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}
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}
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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if (!d->delay)
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continue;
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if (d->delay > SIZE_MAX) {
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av_log(ctx, AV_LOG_ERROR, "Requested delay is too big.\n");
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return AVERROR(EINVAL);
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}
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d->samples = av_malloc_array(d->delay, s->block_align);
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if (!d->samples)
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return AVERROR(ENOMEM);
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s->max_delay = FFMAX(s->max_delay, d->delay);
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}
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switch (inlink->format) {
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case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
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case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
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case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
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case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
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case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioDelayContext *s = ctx->priv;
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AVFrame *out_frame;
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int i;
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if (ctx->is_disabled || !s->delays)
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return ff_filter_frame(outlink, frame);
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out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
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if (!out_frame) {
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av_frame_free(&frame);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out_frame, frame);
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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const uint8_t *src = frame->extended_data[i];
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uint8_t *dst = out_frame->extended_data[i];
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if (!d->delay)
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memcpy(dst, src, frame->nb_samples * s->block_align);
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else
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s->delay_channel(d, frame->nb_samples, src, dst);
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}
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out_frame->pts = s->next_pts;
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s->next_pts += av_rescale_q(frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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av_frame_free(&frame);
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return ff_filter_frame(outlink, out_frame);
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}
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static int activate(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AudioDelayContext *s = ctx->priv;
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AVFrame *frame = NULL;
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int ret, status;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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if (s->padding) {
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int nb_samples = FFMIN(s->padding, 2048);
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frame = ff_get_audio_buffer(outlink, nb_samples);
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if (!frame)
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return AVERROR(ENOMEM);
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s->padding -= nb_samples;
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av_samples_set_silence(frame->extended_data, 0,
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frame->nb_samples,
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outlink->channels,
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frame->format);
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frame->pts = s->next_pts;
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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return ff_filter_frame(outlink, frame);
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}
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ret = ff_inlink_consume_frame(inlink, &frame);
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if (ret < 0)
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return ret;
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if (ret > 0)
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return filter_frame(inlink, frame);
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if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
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if (status == AVERROR_EOF)
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s->eof = 1;
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}
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if (s->eof && s->max_delay) {
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int nb_samples = FFMIN(s->max_delay, 2048);
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frame = ff_get_audio_buffer(outlink, nb_samples);
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if (!frame)
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return AVERROR(ENOMEM);
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s->max_delay -= nb_samples;
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av_samples_set_silence(frame->extended_data, 0,
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frame->nb_samples,
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outlink->channels,
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frame->format);
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frame->pts = s->next_pts;
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return filter_frame(inlink, frame);
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}
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if (s->eof && s->max_delay == 0) {
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ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
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return 0;
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}
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if (!s->eof)
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FF_FILTER_FORWARD_WANTED(outlink, inlink);
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return FFERROR_NOT_READY;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioDelayContext *s = ctx->priv;
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if (s->chandelay) {
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for (int i = 0; i < s->nb_delays; i++)
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av_freep(&s->chandelay[i].samples);
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}
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av_freep(&s->chandelay);
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}
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static const AVFilterPad adelay_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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},
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};
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static const AVFilterPad adelay_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_af_adelay = {
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.name = "adelay",
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.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
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.priv_size = sizeof(AudioDelayContext),
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.priv_class = &adelay_class,
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.activate = activate,
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.uninit = uninit,
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FILTER_INPUTS(adelay_inputs),
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FILTER_OUTPUTS(adelay_outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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};
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