mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
238 lines
8.0 KiB
C
238 lines
8.0 KiB
C
/*
|
|
* This file is part of Libav.
|
|
*
|
|
* Libav 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.
|
|
*
|
|
* Libav 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 Libav; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
#include "libavresample/avresample.h"
|
|
#include "libavutil/audio_fifo.h"
|
|
#include "libavutil/mathematics.h"
|
|
#include "libavutil/opt.h"
|
|
#include "libavutil/samplefmt.h"
|
|
|
|
#include "audio.h"
|
|
#include "avfilter.h"
|
|
|
|
typedef struct ASyncContext {
|
|
const AVClass *class;
|
|
|
|
AVAudioResampleContext *avr;
|
|
int64_t pts; ///< timestamp in samples of the first sample in fifo
|
|
int min_delta; ///< pad/trim min threshold in samples
|
|
|
|
/* options */
|
|
int resample;
|
|
float min_delta_sec;
|
|
int max_comp;
|
|
} ASyncContext;
|
|
|
|
#define OFFSET(x) offsetof(ASyncContext, x)
|
|
#define A AV_OPT_FLAG_AUDIO_PARAM
|
|
static const AVOption options[] = {
|
|
{ "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { 0 }, 0, 1, A },
|
|
{ "min_delta", "Minimum difference between timestamps and audio data "
|
|
"(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { 0.1 }, 0, INT_MAX, A },
|
|
{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { 500 }, 0, INT_MAX, A },
|
|
{ NULL },
|
|
};
|
|
|
|
static const AVClass async_class = {
|
|
.class_name = "asyncts filter",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
static int init(AVFilterContext *ctx, const char *args, void *opaque)
|
|
{
|
|
ASyncContext *s = ctx->priv;
|
|
int ret;
|
|
|
|
s->class = &async_class;
|
|
av_opt_set_defaults(s);
|
|
|
|
if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
|
|
av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
|
|
return ret;
|
|
}
|
|
av_opt_free(s);
|
|
|
|
s->pts = AV_NOPTS_VALUE;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void uninit(AVFilterContext *ctx)
|
|
{
|
|
ASyncContext *s = ctx->priv;
|
|
|
|
if (s->avr) {
|
|
avresample_close(s->avr);
|
|
avresample_free(&s->avr);
|
|
}
|
|
}
|
|
|
|
static int config_props(AVFilterLink *link)
|
|
{
|
|
ASyncContext *s = link->src->priv;
|
|
int ret;
|
|
|
|
s->min_delta = s->min_delta_sec * link->sample_rate;
|
|
link->time_base = (AVRational){1, link->sample_rate};
|
|
|
|
s->avr = avresample_alloc_context();
|
|
if (!s->avr)
|
|
return AVERROR(ENOMEM);
|
|
|
|
av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0);
|
|
av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);
|
|
av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0);
|
|
av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0);
|
|
av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0);
|
|
av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0);
|
|
|
|
if (s->resample)
|
|
av_opt_set_int(s->avr, "force_resampling", 1, 0);
|
|
|
|
if ((ret = avresample_open(s->avr)) < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int request_frame(AVFilterLink *link)
|
|
{
|
|
AVFilterContext *ctx = link->src;
|
|
ASyncContext *s = ctx->priv;
|
|
int ret = avfilter_request_frame(ctx->inputs[0]);
|
|
int nb_samples;
|
|
|
|
/* flush the fifo */
|
|
if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
|
|
AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
|
|
nb_samples);
|
|
if (!buf)
|
|
return AVERROR(ENOMEM);
|
|
avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
|
|
nb_samples, NULL, 0, 0);
|
|
buf->pts = s->pts;
|
|
ff_filter_samples(link, buf);
|
|
return 0;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
|
|
{
|
|
avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
|
|
buf->linesize[0], buf->audio->nb_samples);
|
|
avfilter_unref_buffer(buf);
|
|
}
|
|
|
|
/* get amount of data currently buffered, in samples */
|
|
static int64_t get_delay(ASyncContext *s)
|
|
{
|
|
return avresample_available(s->avr) + avresample_get_delay(s->avr);
|
|
}
|
|
|
|
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
ASyncContext *s = ctx->priv;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
|
|
int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
|
|
av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
|
|
int out_size;
|
|
int64_t delta;
|
|
|
|
/* buffer data until we get the first timestamp */
|
|
if (s->pts == AV_NOPTS_VALUE) {
|
|
if (pts != AV_NOPTS_VALUE) {
|
|
s->pts = pts - get_delay(s);
|
|
}
|
|
write_to_fifo(s, buf);
|
|
return;
|
|
}
|
|
|
|
/* now wait for the next timestamp */
|
|
if (pts == AV_NOPTS_VALUE) {
|
|
write_to_fifo(s, buf);
|
|
return;
|
|
}
|
|
|
|
/* when we have two timestamps, compute how many samples would we have
|
|
* to add/remove to get proper sync between data and timestamps */
|
|
delta = pts - s->pts - get_delay(s);
|
|
out_size = avresample_available(s->avr);
|
|
|
|
if (labs(delta) > s->min_delta) {
|
|
av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
|
|
out_size += delta;
|
|
} else if (s->resample) {
|
|
int comp = av_clip(delta, -s->max_comp, s->max_comp);
|
|
av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
|
|
avresample_set_compensation(s->avr, delta, inlink->sample_rate);
|
|
}
|
|
|
|
if (out_size > 0) {
|
|
AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
|
|
out_size);
|
|
if (!buf_out)
|
|
return;
|
|
|
|
avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
|
|
buf_out->pts = s->pts;
|
|
|
|
if (delta > 0) {
|
|
av_samples_set_silence(buf_out->extended_data, out_size - delta,
|
|
delta, nb_channels, buf->format);
|
|
}
|
|
ff_filter_samples(outlink, buf_out);
|
|
} else {
|
|
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
|
|
"whole buffer.\n");
|
|
}
|
|
|
|
/* drain any remaining buffered data */
|
|
avresample_read(s->avr, NULL, avresample_available(s->avr));
|
|
|
|
s->pts = pts - avresample_get_delay(s->avr);
|
|
avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
|
|
buf->linesize[0], buf->audio->nb_samples);
|
|
avfilter_unref_buffer(buf);
|
|
}
|
|
|
|
AVFilter avfilter_af_asyncts = {
|
|
.name = "asyncts",
|
|
.description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),
|
|
|
|
.init = init,
|
|
.uninit = uninit,
|
|
|
|
.priv_size = sizeof(ASyncContext),
|
|
|
|
.inputs = (const AVFilterPad[]) {{ .name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_samples = filter_samples },
|
|
{ NULL }},
|
|
.outputs = (const AVFilterPad[]) {{ .name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_props,
|
|
.request_frame = request_frame },
|
|
{ NULL }},
|
|
};
|