#include "audio.h"
#include "avfilter.h"
+#include "filters.h"
#include "formats.h"
#include "internal.h"
smoothing, formant, opitch, channels;
int64_t nb_samples_out;
int64_t nb_samples_in;
- int flushed;
+ int64_t first_pts;
+ int nb_samples;
} RubberBandContext;
#define OFFSET(x) offsetof(RubberBandContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption rubberband_options[] = {
- { "tempo", "set tempo scale factor", OFFSET(tempo), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 100, A },
- { "pitch", "set pitch scale factor", OFFSET(pitch), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 100, A },
+ { "tempo", "set tempo scale factor", OFFSET(tempo), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 100, AT },
+ { "pitch", "set pitch scale factor", OFFSET(pitch), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 100, AT },
{ "transients", "set transients", OFFSET(transients), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, A, "transients" },
{ "crisp", 0, 0, AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsCrisp}, 0, 0, A, "transients" },
{ "mixed", 0, 0, AV_OPT_TYPE_CONST, {.i64=RubberBandOptionTransientsMixed}, 0, 0, A, "transients" },
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
- RubberBandContext *s = inlink->dst->priv;
- AVFilterLink *outlink = inlink->dst->outputs[0];
+ AVFilterContext *ctx = inlink->dst;
+ RubberBandContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
int ret = 0, nb_samples;
- rubberband_process(s->rbs, (const float *const *)in->data, in->nb_samples, 0);
+ if (s->first_pts == AV_NOPTS_VALUE)
+ s->first_pts = in->pts;
+
+ rubberband_process(s->rbs, (const float *const *)in->data, in->nb_samples, ff_outlink_get_status(inlink));
s->nb_samples_in += in->nb_samples;
nb_samples = rubberband_available(s->rbs);
av_frame_free(&in);
return AVERROR(ENOMEM);
}
- out->pts = av_rescale_q(s->nb_samples_out,
+ out->pts = s->first_pts + av_rescale_q(s->nb_samples_out,
(AVRational){ 1, outlink->sample_rate },
outlink->time_base);
nb_samples = rubberband_retrieve(s->rbs, (float *const *)out->data, nb_samples);
}
av_frame_free(&in);
- return ret;
+ if (ff_inlink_queued_samples(inlink) >= s->nb_samples)
+ ff_filter_set_ready(ctx, 100);
+ return ret < 0 ? ret : nb_samples;
}
static int config_input(AVFilterLink *inlink)
if (s->rbs)
rubberband_delete(s->rbs);
s->rbs = rubberband_new(inlink->sample_rate, inlink->channels, opts, 1. / s->tempo, s->pitch);
+ if (!s->rbs)
+ return AVERROR(ENOMEM);
- inlink->partial_buf_size =
- inlink->min_samples =
- inlink->max_samples = rubberband_get_samples_required(s->rbs);
+ s->nb_samples = rubberband_get_samples_required(s->rbs);
+ s->first_pts = AV_NOPTS_VALUE;
return 0;
}
-static int request_frame(AVFilterLink *outlink)
+static int activate(AVFilterContext *ctx)
{
- AVFilterContext *ctx = outlink->src;
- RubberBandContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
- int ret = 0;
-
- ret = ff_request_frame(ctx->inputs[0]);
-
- if (ret == AVERROR_EOF && !s->flushed) {
- if (rubberband_available(s->rbs) > 0) {
- AVFrame *out = ff_get_audio_buffer(inlink, 1);
- int nb_samples;
-
- if (!out)
- return AVERROR(ENOMEM);
-
- rubberband_process(s->rbs, (const float *const *)out->data, 1, 1);
- av_frame_free(&out);
- nb_samples = rubberband_available(s->rbs);
-
- if (nb_samples > 0) {
- out = ff_get_audio_buffer(outlink, nb_samples);
- if (!out)
- return AVERROR(ENOMEM);
- out->pts = av_rescale_q(s->nb_samples_out,
- (AVRational){ 1, outlink->sample_rate },
- outlink->time_base);
- nb_samples = rubberband_retrieve(s->rbs, (float *const *)out->data, nb_samples);
- out->nb_samples = nb_samples;
- ret = ff_filter_frame(outlink, out);
- s->nb_samples_out += nb_samples;
- }
- }
- s->flushed = 1;
- av_log(ctx, AV_LOG_DEBUG, "nb_samples_in %"PRId64" nb_samples_out %"PRId64"\n",
- s->nb_samples_in, s->nb_samples_out);
+ AVFilterLink *outlink = ctx->outputs[0];
+ RubberBandContext *s = ctx->priv;
+ AVFrame *in = NULL;
+ int ret;
+
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+ ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
+ if (ret < 0)
+ return ret;
+ if (ret > 0) {
+ ret = filter_frame(inlink, in);
+ if (ret != 0)
+ return ret;
}
- return ret;
+ FF_FILTER_FORWARD_STATUS(inlink, outlink);
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return FFERROR_NOT_READY;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
RubberBandContext *s = ctx->priv;
+ int ret;
- if (!strcmp(cmd, "tempo")) {
- double arg;
-
- sscanf(args, "%lf", &arg);
- if (arg < 0.01 || arg > 100) {
- av_log(ctx, AV_LOG_ERROR,
- "Tempo scale factor '%f' out of range\n", arg);
- return AVERROR(EINVAL);
- }
- rubberband_set_time_ratio(s->rbs, 1. / arg);
- }
-
- if (!strcmp(cmd, "pitch")) {
- double arg;
+ ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+ if (ret < 0)
+ return ret;
- sscanf(args, "%lf", &arg);
- if (arg < 0.01 || arg > 100) {
- av_log(ctx, AV_LOG_ERROR,
- "Pitch scale factor '%f' out of range\n", arg);
- return AVERROR(EINVAL);
- }
- rubberband_set_pitch_scale(s->rbs, arg);
- }
+ rubberband_set_time_ratio(s->rbs, 1. / s->tempo);
+ rubberband_set_pitch_scale(s->rbs, s->pitch);
+ s->nb_samples = rubberband_get_samples_required(s->rbs);
return 0;
}
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
- .filter_frame = filter_frame,
},
{ NULL }
};
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
- .request_frame = request_frame,
},
{ NULL }
};
-AVFilter ff_af_rubberband = {
+const AVFilter ff_af_rubberband = {
.name = "rubberband",
.description = NULL_IF_CONFIG_SMALL("Apply time-stretching and pitch-shifting."),
.query_formats = query_formats,
.priv_size = sizeof(RubberBandContext),
.priv_class = &rubberband_class,
.uninit = uninit,
+ .activate = activate,
.inputs = rubberband_inputs,
.outputs = rubberband_outputs,
.process_command = process_command,
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