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FFmpeg/libavfilter/af_adelay.c
Andreas Rheinhardt 50ea7389ec avfilter: Deduplicate default audio inputs/outputs
Lots of audio filters use very simple inputs or outputs:
An array with a single AVFilterPad whose name is "default"
and whose type is AVMEDIA_TYPE_AUDIO; everything else is unset.

Given that we never use pointer equality for inputs or outputs*,
we can simply use a single AVFilterPad instead of dozens; this
even saves .data.rel.ro (4784B here) as well as relocations.

*: In fact, several filters (like the filters in af_biquads.c)
already use the same inputs; furthermore, ff_filter_alloc()
duplicates the input and output pads so that we do not even
work with the pads directly.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-08-07 09:21:13 +02:00

483 lines
19 KiB
C

/*
* Copyright (c) 2013 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#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 {
int64_t delay;
size_t delay_index;
size_t index;
unsigned int samples_size;
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 offset;
int64_t next_pts;
int eof;
AVFrame *input;
void (*delay_channel)(ChanDelay *d, int nb_samples,
const uint8_t *src, uint8_t *dst);
int (*resize_channel_samples)(ChanDelay *d, int64_t new_delay);
} 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 | AV_OPT_FLAG_RUNTIME_PARAM },
{ "all", "use last available delay for remained channels", OFFSET(all), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(adelay);
#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)
#define CHANGE_DELAY(name, type, fill) \
static int resize_samples_## name ##p(ChanDelay *d, int64_t new_delay) \
{ \
type *samples; \
\
if (new_delay == d->delay) { \
return 0; \
} \
\
if (new_delay == 0) { \
av_freep(&d->samples); \
d->samples_size = 0; \
d->delay = 0; \
d->index = 0; \
d->delay_index = 0; \
return 0; \
} \
\
samples = (type *) av_fast_realloc(d->samples, &d->samples_size, new_delay * sizeof(type)); \
if (!samples) { \
return AVERROR(ENOMEM); \
} \
\
if (new_delay < d->delay) { \
if (d->index > new_delay) { \
d->index -= new_delay; \
memmove(samples, &samples[new_delay], d->index * sizeof(type)); \
d->delay_index = new_delay; \
} else if (d->delay_index > d->index) { \
memmove(&samples[d->index], &samples[d->index+(d->delay-new_delay)], \
(new_delay - d->index) * sizeof(type)); \
d->delay_index -= d->delay - new_delay; \
} \
} else { \
size_t block_size; \
if (d->delay_index >= d->delay) { \
block_size = (d->delay - d->index) * sizeof(type); \
memmove(&samples[d->index+(new_delay - d->delay)], &samples[d->index], block_size); \
d->delay_index = new_delay; \
} else { \
d->delay_index += new_delay - d->delay; \
} \
block_size = (new_delay - d->delay) * sizeof(type); \
memset(&samples[d->index], fill, block_size); \
} \
d->delay = new_delay; \
d->samples = (void *) samples; \
return 0; \
}
CHANGE_DELAY(u8, uint8_t, 0x80)
CHANGE_DELAY(s16, int16_t, 0)
CHANGE_DELAY(s32, int32_t, 0)
CHANGE_DELAY(flt, float, 0)
CHANGE_DELAY(dbl, double, 0)
static int parse_delays(char *p, char **saveptr, int64_t *result, AVFilterContext *ctx, int sample_rate) {
float delay, div;
int ret;
char *arg;
char type = 0;
if (!(arg = av_strtok(p, "|", saveptr)))
return 1;
ret = av_sscanf(arg, "%"SCNd64"%c", result, &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);
}
*result = delay * sample_rate / div;
}
if (*result < 0) {
av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
return AVERROR(EINVAL);
}
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioDelayContext *s = ctx->priv;
char *p, *saveptr = NULL;
int i;
s->next_pts = AV_NOPTS_VALUE;
s->chandelay = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->chandelay));
if (!s->chandelay)
return AVERROR(ENOMEM);
s->nb_delays = inlink->ch_layout.nb_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];
int ret;
ret = parse_delays(p, &saveptr, &d->delay, ctx, inlink->sample_rate);
if (ret == 1)
break;
else if (ret < 0)
return ret;
p = NULL;
}
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;
}
s->offset = av_rescale_q(s->padding,
av_make_q(1, inlink->sample_rate),
inlink->time_base);
}
for (i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
if (!d->delay)
continue;
if (d->delay > SIZE_MAX) {
av_log(ctx, AV_LOG_ERROR, "Requested delay is too big.\n");
return AVERROR(EINVAL);
}
d->samples = av_malloc_array(d->delay, s->block_align);
if (!d->samples)
return AVERROR(ENOMEM);
d->samples_size = d->delay * s->block_align;
s->max_delay = FFMAX(s->max_delay, d->delay);
}
switch (inlink->format) {
case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ;
s->resize_channel_samples = resize_samples_u8p; break;
case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p;
s->resize_channel_samples = resize_samples_s16p; break;
case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p;
s->resize_channel_samples = resize_samples_s32p; break;
case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp;
s->resize_channel_samples = resize_samples_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp;
s->resize_channel_samples = resize_samples_dblp; break;
}
return 0;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret = AVERROR(ENOSYS);
AVFilterLink *inlink = ctx->inputs[0];
AudioDelayContext *s = ctx->priv;
if (!strcmp(cmd, "delays")) {
int64_t delay;
char *p, *saveptr = NULL;
int64_t all_delay = -1;
int64_t max_delay = 0;
char *args_cpy = av_strdup(args);
if (args_cpy == NULL) {
return AVERROR(ENOMEM);
}
ret = 0;
p = args_cpy;
if (!strncmp(args, "all:", 4)) {
p = &args_cpy[4];
ret = parse_delays(p, &saveptr, &all_delay, ctx, inlink->sample_rate);
if (ret == 1)
ret = AVERROR(EINVAL);
else if (ret == 0)
delay = all_delay;
}
if (!ret) {
for (int i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
if (all_delay < 0) {
ret = parse_delays(p, &saveptr, &delay, ctx, inlink->sample_rate);
if (ret != 0) {
ret = 0;
break;
}
p = NULL;
}
ret = s->resize_channel_samples(d, delay);
if (ret)
break;
max_delay = FFMAX(max_delay, d->delay);
}
s->max_delay = FFMAX(s->max_delay, max_delay);
}
av_freep(&args_cpy);
}
return ret;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioDelayContext *s = ctx->priv;
AVFrame *out_frame;
int i;
if (ctx->is_disabled || !s->delays) {
s->input = NULL;
return ff_filter_frame(outlink, frame);
}
s->next_pts = av_rescale_q(frame->pts, inlink->time_base, outlink->time_base);
out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
if (!out_frame) {
s->input = NULL;
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->offset;
out_frame->duration = av_rescale_q(out_frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
s->next_pts += out_frame->duration;
av_frame_free(&frame);
s->input = NULL;
return ff_filter_frame(outlink, 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->input) {
ret = ff_inlink_consume_frame(inlink, &s->input);
if (ret < 0)
return ret;
}
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF)
s->eof = 1;
}
if (s->next_pts == AV_NOPTS_VALUE && pts != AV_NOPTS_VALUE)
s->next_pts = av_rescale_q(pts, inlink->time_base, outlink->time_base);
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->ch_layout.nb_channels,
frame->format);
frame->duration = av_rescale_q(frame->nb_samples,
(AVRational){1, outlink->sample_rate},
outlink->time_base);
frame->pts = s->next_pts;
s->next_pts += frame->duration;
return ff_filter_frame(outlink, frame);
}
if (s->input)
return filter_frame(inlink, s->input);
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->ch_layout.nb_channels,
frame->format);
frame->duration = av_rescale_q(frame->nb_samples,
(AVRational){1, outlink->sample_rate},
outlink->time_base);
frame->pts = s->next_pts;
s->next_pts += frame->duration;
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,
},
};
const AVFilter ff_af_adelay = {
.name = "adelay",
.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
.priv_size = sizeof(AudioDelayContext),
.priv_class = &adelay_class,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(adelay_inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.process_command = process_command,
};