This source file includes following definitions.
- query_formats
- DELAY
- filter_frame
- activate
- uninit
#include "libavutil/avstring.h"
#include "libavutil/eval.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "internal.h"
typedef struct ChanDelay {
int delay;
unsigned delay_index;
unsigned index;
uint8_t *samples;
} ChanDelay;
typedef struct AudioDelayContext {
const AVClass *class;
int all;
char *delays;
ChanDelay *chandelay;
int nb_delays;
int block_align;
int64_t padding;
int64_t max_delay;
int64_t next_pts;
int eof;
void (*delay_channel)(ChanDelay *d, int nb_samples,
const uint8_t *src, uint8_t *dst);
} AudioDelayContext;
#define OFFSET(x) offsetof(AudioDelayContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption adelay_options[] = {
{ "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
{ "all", "use last available delay for remained channels", OFFSET(all), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(adelay);
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts;
AVFilterFormats *formats;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
#define DELAY(name, type, fill) \
static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
const uint8_t *ssrc, uint8_t *ddst) \
{ \
const type *src = (type *)ssrc; \
type *dst = (type *)ddst; \
type *samples = (type *)d->samples; \
\
while (nb_samples) { \
if (d->delay_index < d->delay) { \
const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
\
memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
memset(dst, fill, len * sizeof(type)); \
d->delay_index += len; \
src += len; \
dst += len; \
nb_samples -= len; \
} else { \
*dst = samples[d->index]; \
samples[d->index] = *src; \
nb_samples--; \
d->index++; \
src++, dst++; \
d->index = d->index >= d->delay ? 0 : d->index; \
} \
} \
}
DELAY(u8, uint8_t, 0x80)
DELAY(s16, int16_t, 0)
DELAY(s32, int32_t, 0)
DELAY(flt, float, 0)
DELAY(dbl, double, 0)
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioDelayContext *s = ctx->priv;
char *p, *arg, *saveptr = NULL;
int i;
s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
if (!s->chandelay)
return AVERROR(ENOMEM);
s->nb_delays = inlink->channels;
s->block_align = av_get_bytes_per_sample(inlink->format);
p = s->delays;
for (i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
float delay, div;
char type = 0;
int ret;
if (!(arg = av_strtok(p, "|", &saveptr)))
break;
p = NULL;
ret = av_sscanf(arg, "%d%c", &d->delay, &type);
if (ret != 2 || type != 'S') {
div = type == 's' ? 1.0 : 1000.0;
if (av_sscanf(arg, "%f", &delay) != 1) {
av_log(ctx, AV_LOG_ERROR, "Invalid syntax for delay.\n");
return AVERROR(EINVAL);
}
d->delay = delay * inlink->sample_rate / div;
}
if (d->delay < 0) {
av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
return AVERROR(EINVAL);
}
}
if (s->all && i) {
for (int j = i; j < s->nb_delays; j++)
s->chandelay[j].delay = s->chandelay[i-1].delay;
}
s->padding = s->chandelay[0].delay;
for (i = 1; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
s->padding = FFMIN(s->padding, d->delay);
}
if (s->padding) {
for (i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
d->delay -= s->padding;
}
}
for (i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
if (!d->delay)
continue;
d->samples = av_malloc_array(d->delay, s->block_align);
if (!d->samples)
return AVERROR(ENOMEM);
s->max_delay = FFMAX(s->max_delay, d->delay);
}
switch (inlink->format) {
case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AudioDelayContext *s = ctx->priv;
AVFrame *out_frame;
int i;
if (ctx->is_disabled || !s->delays)
return ff_filter_frame(ctx->outputs[0], frame);
out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
if (!out_frame) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out_frame, frame);
for (i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
const uint8_t *src = frame->extended_data[i];
uint8_t *dst = out_frame->extended_data[i];
if (!d->delay)
memcpy(dst, src, frame->nb_samples * s->block_align);
else
s->delay_channel(d, frame->nb_samples, src, dst);
}
out_frame->pts = s->next_pts;
s->next_pts += av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
av_frame_free(&frame);
return ff_filter_frame(ctx->outputs[0], out_frame);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioDelayContext *s = ctx->priv;
AVFrame *frame = NULL;
int ret, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (s->padding) {
int nb_samples = FFMIN(s->padding, 2048);
frame = ff_get_audio_buffer(outlink, nb_samples);
if (!frame)
return AVERROR(ENOMEM);
s->padding -= nb_samples;
av_samples_set_silence(frame->extended_data, 0,
frame->nb_samples,
outlink->channels,
frame->format);
frame->pts = s->next_pts;
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
return ff_filter_frame(outlink, frame);
}
ret = ff_inlink_consume_frame(inlink, &frame);
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, frame);
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF)
s->eof = 1;
}
if (s->eof && s->max_delay) {
int nb_samples = FFMIN(s->max_delay, 2048);
frame = ff_get_audio_buffer(outlink, nb_samples);
if (!frame)
return AVERROR(ENOMEM);
s->max_delay -= nb_samples;
av_samples_set_silence(frame->extended_data, 0,
frame->nb_samples,
outlink->channels,
frame->format);
frame->pts = s->next_pts;
return filter_frame(inlink, frame);
}
if (s->eof && s->max_delay == 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
return 0;
}
if (!s->eof)
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioDelayContext *s = ctx->priv;
if (s->chandelay) {
for (int i = 0; i < s->nb_delays; i++)
av_freep(&s->chandelay[i].samples);
}
av_freep(&s->chandelay);
}
static const AVFilterPad adelay_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
{ NULL }
};
static const AVFilterPad adelay_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_adelay = {
.name = "adelay",
.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
.query_formats = query_formats,
.priv_size = sizeof(AudioDelayContext),
.priv_class = &adelay_class,
.activate = activate,
.uninit = uninit,
.inputs = adelay_inputs,
.outputs = adelay_outputs,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};