#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"
float min_delta_sec;
int max_comp;
- /* set by filter_samples() to signal an output frame to request_frame() */
+ /* set by filter_frame() 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, { 0 }, 0, 1, A },
+ { "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, { 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 },
+ "(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 },
};
}
av_opt_free(s);
- s->pts = AV_NOPTS_VALUE;
-
return 0;
}
nb_samples);
if (!buf)
return AVERROR(ENOMEM);
- avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
- nb_samples, NULL, 0, 0);
+ ret = avresample_convert(s->avr, buf->extended_data,
+ buf->linesize[0], nb_samples, NULL, 0, 0);
+ if (ret <= 0) {
+ avfilter_unref_bufferp(&buf);
+ return (ret < 0) ? ret : AVERROR_EOF;
+ }
+
buf->pts = s->pts;
- ff_filter_samples(link, buf);
- return 0;
+ return ff_filter_frame(link, buf);
}
return ret;
}
-static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
+static int 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);
+ int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
+ buf->linesize[0], buf->audio->nb_samples);
avfilter_unref_buffer(buf);
+ return ret;
}
/* get amount of data currently buffered, in samples */
return avresample_available(s->avr) + avresample_get_delay(s->avr);
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ASyncContext *s = ctx->priv;
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;
+ int out_size, ret;
int64_t delta;
/* buffer data until we get the first timestamp */
if (pts != AV_NOPTS_VALUE) {
s->pts = pts - get_delay(s);
}
- write_to_fifo(s, buf);
- return;
+ return write_to_fifo(s, buf);
}
/* now wait for the next timestamp */
if (pts == AV_NOPTS_VALUE) {
- write_to_fifo(s, buf);
- return;
+ return write_to_fifo(s, buf);
}
/* when we have two timestamps, compute how many samples would we have
if (labs(delta) > s->min_delta) {
av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
- out_size += 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);
if (out_size > 0) {
AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
out_size);
- if (!buf_out)
- return;
+ if (!buf_out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
- avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
+ avresample_read(s->avr, 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);
+ ret = ff_filter_frame(outlink, buf_out);
+ if (ret < 0)
+ goto fail;
s->got_output = 1;
} else {
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
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);
+ ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
+ buf->linesize[0], buf->audio->nb_samples);
+
+fail:
avfilter_unref_buffer(buf);
+
+ return ret;
}
+static const AVFilterPad avfilter_af_asyncts_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_asyncts_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_props,
+ .request_frame = request_frame
+ },
+ { NULL }
+};
+
AVFilter avfilter_af_asyncts = {
.name = "asyncts",
.description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),
.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 }},
+ .inputs = avfilter_af_asyncts_inputs,
+ .outputs = avfilter_af_asyncts_outputs,
};
projects / ffmpeg / blobdiff