/*
- *
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
AVAudioResampleContext *avr;
AVDictionary *options;
+ int resampling;
int64_t next_pts;
+ int64_t next_in_pts;
/* set by filter_frame() to signal an output frame to request_frame() */
int got_output;
char buf1[64], buf2[64];
int ret;
+ int64_t resampling_forced;
+
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
return AVERROR(ENOMEM);
if (s->options) {
+ int ret;
AVDictionaryEntry *e = NULL;
while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
- av_opt_set_dict(s->avr, &s->options);
+ ret = av_opt_set_dict(s->avr, &s->options);
+ if (ret < 0)
+ return ret;
}
av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
- outlink->time_base = (AVRational){ 1, outlink->sample_rate };
- s->next_pts = AV_NOPTS_VALUE;
+ av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
+ s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);
+
+ if (s->resampling) {
+ outlink->time_base = (AVRational){ 1, outlink->sample_rate };
+ s->next_pts = AV_NOPTS_VALUE;
+ s->next_in_pts = AV_NOPTS_VALUE;
+ } else
+ outlink->time_base = inlink->time_base;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
return (ret == 0) ? AVERROR_EOF : ret;
}
+ frame->nb_samples = ret;
frame->pts = s->next_pts;
return ff_filter_frame(outlink, frame);
}
av_assert0(!avresample_available(s->avr));
- if (s->next_pts == AV_NOPTS_VALUE) {
+ if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {
if (in->pts == AV_NOPTS_VALUE) {
av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
"assuming 0.\n");
goto fail;
}
- out->sample_rate = outlink->sample_rate;
- if (in->pts != AV_NOPTS_VALUE) {
- out->pts = av_rescale_q(in->pts, inlink->time_base,
- outlink->time_base) -
- av_rescale(delay, outlink->sample_rate,
- inlink->sample_rate);
+ if (s->resampling) {
+ out->sample_rate = outlink->sample_rate;
+ /* Only convert in->pts if there is a discontinuous jump.
+ This ensures that out->pts tracks the number of samples actually
+ output by the resampler in the absence of such a jump.
+ Otherwise, the rounding in av_rescale_q() and av_rescale()
+ causes off-by-1 errors. */
+ if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
+ out->pts = av_rescale_q(in->pts, inlink->time_base,
+ outlink->time_base) -
+ av_rescale(delay, outlink->sample_rate,
+ inlink->sample_rate);
+ } else
+ out->pts = s->next_pts;
+
+ s->next_pts = out->pts + out->nb_samples;
+ s->next_in_pts = in->pts + in->nb_samples;
} else
- out->pts = s->next_pts;
-
- s->next_pts = out->pts + out->nb_samples;
+ out->pts = in->pts;
ret = ff_filter_frame(outlink, out);
s->got_output = 1;