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lavfi: remove request_samples.

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Filters can use min_samples/max_samples if the number is constant
or activate and ff_inlink_consume_samples().
This commit is contained in:
Nicolas George 2020-08-12 19:17:29 +02:00
parent 4ca1fb9d2a
commit 03c8fe49ea
2 changed files with 3 additions and 104 deletions
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8
libavfilter/avfilter.h
View File

@ -490,14 +490,6 @@ struct AVFilterLink {
struct AVFilterChannelLayouts *in_channel_layouts;
struct AVFilterChannelLayouts *out_channel_layouts;
/**
* Audio only, the destination filter sets this to a non-zero value to
* request that buffers with the given number of samples should be sent to
* it.
* Last buffer before EOF will be padded with silence.
*/
int request_samples;
/** stage of the initialization of the link properties (dimensions, etc) */
enum {
AVLINK_UNINIT = 0, ///< not started

99
libavfilter/fifo.c
View File

@ -143,112 +143,19 @@ static int calc_ptr_alignment(AVFrame *frame)
return min_align;
}
static int return_audio_frame(AVFilterContext *ctx)
{
AVFilterLink *link = ctx->outputs[0];
FifoContext *s = ctx->priv;
AVFrame *head = s->root.next ? s->root.next->frame : NULL;
AVFrame *out;
int ret;
/* if head is NULL then we're flushing the remaining samples in out */
if (!head && !s->out)
return AVERROR_EOF;
if (!s->out &&
head->nb_samples >= link->request_samples &&
calc_ptr_alignment(head) >= 32) {
if (head->nb_samples == link->request_samples) {
out = head;
queue_pop(s);
} else {
out = av_frame_clone(head);
if (!out)
return AVERROR(ENOMEM);
out->nb_samples = link->request_samples;
buffer_offset(link, head, link->request_samples);
}
} else {
int nb_channels = link->channels;
if (!s->out) {
s->out = ff_get_audio_buffer(link, link->request_samples);
if (!s->out)
return AVERROR(ENOMEM);
s->out->nb_samples = 0;
s->out->pts = head->pts;
s->allocated_samples = link->request_samples;
} else if (link->request_samples != s->allocated_samples) {
av_log(ctx, AV_LOG_ERROR, "request_samples changed before the "
"buffer was returned.\n");
return AVERROR(EINVAL);
}
while (s->out->nb_samples < s->allocated_samples) {
int len;
if (!s->root.next) {
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) {
av_samples_set_silence(s->out->extended_data,
s->out->nb_samples,
s->allocated_samples -
s->out->nb_samples,
nb_channels, link->format);
s->out->nb_samples = s->allocated_samples;
break;
} else if (ret < 0)
return ret;
if (!s->root.next)
return 0;
}
head = s->root.next->frame;
len = FFMIN(s->allocated_samples - s->out->nb_samples,
head->nb_samples);
av_samples_copy(s->out->extended_data, head->extended_data,
s->out->nb_samples, 0, len, nb_channels,
link->format);
s->out->nb_samples += len;
if (len == head->nb_samples) {
av_frame_free(&head);
queue_pop(s);
} else {
buffer_offset(link, head, len);
}
}
out = s->out;
s->out = NULL;
}
return ff_filter_frame(link, out);
}
static int request_frame(AVFilterLink *outlink)
{
FifoContext *s = outlink->src->priv;
int ret = 0;
if (!s->root.next) {
if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0) {
if (ret == AVERROR_EOF && outlink->request_samples)
return return_audio_frame(outlink->src);
if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0)
return ret;
}
if (!s->root.next)
return 0;
}
if (outlink->request_samples) {
return return_audio_frame(outlink->src);
} else {
ret = ff_filter_frame(outlink, s->root.next->frame);
queue_pop(s);
}
ret = ff_filter_frame(outlink, s->root.next->frame);
queue_pop(s);
return ret;
}