// 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,
- AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM },
+ 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 | AV_OPT_FLAG_RUNTIME_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 0;
}
-static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)
+static int yae_update(AVFilterContext *ctx)
{
const AudioFragment *prev;
ATempoContext *atempo = ctx->priv;
- char *tail = NULL;
- double tempo = av_strtod(arg_tempo, &tail);
-
- if (tail && *tail) {
- av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);
- 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);
- return AVERROR(EINVAL);
- }
prev = yae_prev_frag(atempo);
atempo->origin[0] = prev->position[0] + atempo->window / 2;
atempo->origin[1] = prev->position[1] + atempo->window / 2;
- atempo->tempo = tempo;
return 0;
}
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);
int res_len,
int flags)
{
- return !strcmp(cmd, "tempo") ? yae_set_tempo(ctx, arg) : AVERROR(ENOSYS);
+ int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
+
+ if (ret < 0)
+ return ret;
+
+ return yae_update(ctx);
}
static const AVFilterPad atempo_inputs[] = {
{ NULL }
};
-AVFilter ff_af_atempo = {
+const AVFilter ff_af_atempo = {
.name = "atempo",
.description = NULL_IF_CONFIG_SMALL("Adjust audio tempo."),
.init = init,