// 1: output sample position
int64_t position[2];
+ // first input timestamp, all other timestamps are offset by this one
+ int64_t start_pts;
+
// sample format:
enum AVSampleFormat format;
uint64_t nsamples_out;
} ATempoContext;
+#define YAE_ATEMPO_MIN 0.5
+#define YAE_ATEMPO_MAX 100.0
+
#define OFFSET(x) offsetof(ATempoContext, x)
static const AVOption atempo_options[] = {
{ "tempo", "set tempo scale factor",
- OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0.5, 2.0,
+ OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 },
+ YAE_ATEMPO_MIN,
+ YAE_ATEMPO_MAX,
AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM },
{ NULL }
};
atempo->nfrag = 0;
atempo->state = YAE_LOAD_FRAGMENT;
+ atempo->start_pts = AV_NOPTS_VALUE;
atempo->position[0] = 0;
atempo->position[1] = 0;
return AVERROR(EINVAL);
}
- if (tempo < 0.5 || tempo > 2.0) {
- av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [0.5, 2.0] range\n",
- tempo);
+ if (tempo < YAE_ATEMPO_MIN || tempo > YAE_ATEMPO_MAX) {
+ av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [%f, %f] range\n",
+ tempo, YAE_ATEMPO_MIN, YAE_ATEMPO_MAX);
return AVERROR(EINVAL);
}
return 0;
}
- // samples are not expected to be skipped:
- av_assert0(read_size <= atempo->ring);
+ // samples are not expected to be skipped, unless tempo is greater than 2:
+ av_assert0(read_size <= atempo->ring || atempo->tempo > 2.0);
while (atempo->position[0] < stop_here && src < src_end) {
int src_samples = (src_end - src) / atempo->stride;
atempo->dst_buffer->nb_samples = n_out;
// adjust the PTS:
- atempo->dst_buffer->pts =
+ atempo->dst_buffer->pts = atempo->start_pts +
av_rescale_q(atempo->nsamples_out,
(AVRational){ 1, outlink->sample_rate },
outlink->time_base);
const uint8_t *src = src_buffer->data[0];
const uint8_t *src_end = src + n_in * atempo->stride;
+ if (atempo->start_pts == AV_NOPTS_VALUE)
+ atempo->start_pts = av_rescale_q(src_buffer->pts,
+ inlink->time_base,
+ outlink->time_base);
+
while (src < src_end) {
if (!atempo->dst_buffer) {
atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);