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/*
* 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/common.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.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;
/* set by filter_samples() to signal an output frame to request_frame() */
int got_output;
} 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, { .i64 = 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, { .dbl = 0.1 }, 0, INT_MAX, A },
{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A },
{ "first_pts", "Assume the first pts should be this value.", OFFSET(pts), AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_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)
{
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);
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 = 0;
int nb_samples;
s->got_output = 0;
while (ret >= 0 && !s->got_output)
ret = ff_request_frame(ctx->inputs[0]);
/* 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;
return ff_filter_samples(link, buf);
}
return ret;
}
static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
{
int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
buf->linesize[0], buf->audio->nb_samples);
avfilter_unref_buffer(buf);
return ret;
}
/* 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 int 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, ret;
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);
}
return write_to_fifo(s, buf);
}
/* now wait for the next timestamp */
if (pts == AV_NOPTS_VALUE) {
return write_to_fifo(s, buf);
}
/* 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 = av_clipl_int32((int64_t)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);
}
delta = 0;
}
if (out_size > 0) {
AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
out_size);
if (!buf_out) {
ret = AVERROR(ENOMEM);
goto fail;
}
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);
}
ret = ff_filter_samples(outlink, buf_out);
if (ret < 0)
goto fail;
s->got_output = 1;
} 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);
ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
buf->linesize[0], buf->audio->nb_samples);
fail:
avfilter_unref_buffer(buf);
return ret;
}
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 }},
};
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