[FFmpeg-cvslog] avfilter/af_afade: stop using ff_outlink_get_status on inputs
Paul B Mahol
git at videolan.org
Sat Jun 17 23:37:46 EEST 2023
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Jun 17 22:32:11 2023 +0200| [c44fe101603dbe1e788079f4a798eeca13dbedcd] | committer: Paul B Mahol
avfilter/af_afade: stop using ff_outlink_get_status on inputs
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c44fe101603dbe1e788079f4a798eeca13dbedcd
---
libavfilter/af_afade.c | 42 +++++++++++++++++++++++++-----------------
1 file changed, 25 insertions(+), 17 deletions(-)
diff --git a/libavfilter/af_afade.c b/libavfilter/af_afade.c
index 2f45b12904..226b63c275 100644
--- a/libavfilter/af_afade.c
+++ b/libavfilter/af_afade.c
@@ -42,8 +42,8 @@ typedef struct AudioFadeContext {
double silence;
double unity;
int overlap;
- int cf0_eof;
- int crossfade_is_over;
+ int status[2];
+ int passthrough;
int64_t pts;
void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
@@ -521,6 +521,13 @@ CROSSFADE(flt, float)
CROSSFADE(s16, int16_t)
CROSSFADE(s32, int32_t)
+static int check_input(AVFilterLink *inlink)
+{
+ const int queued_samples = ff_inlink_queued_samples(inlink);
+
+ return ff_inlink_check_available_samples(inlink, queued_samples + 1) == 1;
+}
+
static int activate(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;
@@ -531,7 +538,7 @@ static int activate(AVFilterContext *ctx)
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
- if (s->crossfade_is_over) {
+ if (s->passthrough && s->status[0]) {
ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
if (ret > 0) {
in->pts = s->pts;
@@ -541,10 +548,10 @@ static int activate(AVFilterContext *ctx)
} else if (ret < 0) {
return ret;
} else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
- ff_outlink_set_status(ctx->outputs[0], status, pts);
+ ff_outlink_set_status(outlink, status, pts);
return 0;
} else if (!ret) {
- if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+ if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}
@@ -554,6 +561,7 @@ static int activate(AVFilterContext *ctx)
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;
@@ -561,7 +569,7 @@ static int activate(AVFilterContext *ctx)
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
- } else if (s->cf0_eof && nb_samples >= s->nb_samples &&
+ } 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);
@@ -587,7 +595,7 @@ static int activate(AVFilterContext *ctx)
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
- s->crossfade_is_over = 1;
+ s->passthrough = 1;
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
@@ -627,19 +635,20 @@ static int activate(AVFilterContext *ctx)
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
- s->crossfade_is_over = 1;
+ s->passthrough = 1;
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
- } else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
- if (!s->cf0_eof && ff_outlink_get_status(ctx->inputs[0])) {
- s->cf0_eof = 1;
- }
- if (ff_outlink_get_status(ctx->inputs[1])) {
- ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, AV_NOPTS_VALUE);
+ } else if (ff_outlink_frame_wanted(outlink)) {
+ if (!s->status[0] && check_input(ctx->inputs[0]))
+ s->status[0] = AVERROR_EOF;
+ s->passthrough = !s->status[0];
+ if (check_input(ctx->inputs[1])) {
+ s->status[1] = AVERROR_EOF;
+ ff_outlink_set_status(outlink, AVERROR_EOF, AV_NOPTS_VALUE);
return 0;
}
- if (!s->cf0_eof)
+ if (!s->status[0])
ff_inlink_request_frame(ctx->inputs[0]);
else
ff_inlink_request_frame(ctx->inputs[1]);
@@ -677,8 +686,7 @@ static AVFrame *get_audio_buffer(AVFilterLink *inlink, int nb_samples)
AVFilterContext *ctx = inlink->dst;
AudioFadeContext *s = ctx->priv;
- return (s->crossfade_is_over ||
- (ff_inlink_queued_samples(inlink) > s->nb_samples)) ?
+ return s->passthrough ?
ff_null_get_audio_buffer (inlink, nb_samples) :
ff_default_get_audio_buffer(inlink, nb_samples);
}
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