mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-02 03:06:28 +02:00
790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
483 lines
19 KiB
C
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/mem.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,
|
|
};
|