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avfilter/af_afade: factorize functions generating frames

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No change in functionality.

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint
2025-07-20 20:43:05 +02:00
parent 944329f8fd
commit 4be21b9399
1 changed files with 111 additions and 95 deletions
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206
libavfilter/af_afade.c
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@ -25,6 +25,7 @@
#include "config_components.h"
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
@ -547,117 +548,132 @@ static int check_input(AVFilterLink *inlink)
return ff_inlink_check_available_samples(inlink, queued_samples + 1) == 1;
}
static int pass_frame(AVFilterLink *inlink, AVFilterLink *outlink, int64_t *pts)
{
AVFrame *in;
int ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
av_assert1(ret);
in->pts = *pts;
*pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
}
static int pass_samples(AVFilterLink *inlink, AVFilterLink *outlink, unsigned nb_samples, int64_t *pts)
{
AVFrame *in;
int ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
if (ret < 0)
return ret;
av_assert1(ret);
in->pts = *pts;
*pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
}
static int pass_crossfade(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out, *cf[2] = { NULL };
int ret;
if (s->overlap) {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data,
s->nb_samples, out->ch_layout.nb_channels,
s->curve, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
s->passthrough = 1;
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
} else {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
outlink->ch_layout.nb_channels, -1, s->nb_samples - 1, s->nb_samples, s->curve, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
outlink->ch_layout.nb_channels, 1, 0, s->nb_samples, s->curve2, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
s->passthrough = 1;
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
}
static int activate(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *in = NULL, *out, *cf[2] = { NULL };
int ret = 0, nb_samples, status;
int64_t pts;
int ret = 0, nb_samples;
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
if (s->passthrough && s->status[0]) {
ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
if (ret > 0) {
in->pts = s->pts;
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
} else if (ret < 0) {
return ret;
} else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
ff_outlink_set_status(outlink, status, pts);
return 0;
} else if (!ret) {
if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}
}
if (ff_inlink_queued_frames(ctx->inputs[1]))
return pass_frame(ctx->inputs[1], outlink, &s->pts);
FF_FILTER_FORWARD_STATUS(ctx->inputs[1], outlink);
FF_FILTER_FORWARD_WANTED(outlink, ctx->inputs[1]);
}
nb_samples = ff_inlink_queued_samples(ctx->inputs[0]);
if (nb_samples > s->nb_samples) {
nb_samples -= s->nb_samples;
s->passthrough = 1;
ret = ff_inlink_consume_samples(ctx->inputs[0], nb_samples, nb_samples, &in);
if (ret < 0)
return ret;
in->pts = s->pts;
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
return pass_samples(ctx->inputs[0], outlink, nb_samples, &s->pts);
} else if (s->status[0] && nb_samples >= s->nb_samples &&
ff_inlink_queued_samples(ctx->inputs[1]) >= s->nb_samples) {
if (s->overlap) {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data,
s->nb_samples, out->ch_layout.nb_channels,
s->curve, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
s->passthrough = 1;
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
} else {
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
outlink->ch_layout.nb_channels, -1, s->nb_samples - 1, s->nb_samples, s->curve, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}
s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
outlink->ch_layout.nb_channels, 1, 0, s->nb_samples, s->curve2, 0., 1.);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
s->passthrough = 1;
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
return pass_crossfade(ctx);
} else if (ff_outlink_frame_wanted(outlink)) {
if (!s->status[0] && check_input(ctx->inputs[0]))
s->status[0] = AVERROR_EOF;