Commit d7fead80 authored by Paul B Mahol's avatar Paul B Mahol

avfilter/af_dynaudnorm: switch to activate

parent a40bcb5c
......@@ -34,6 +34,7 @@
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
typedef struct cqueue {
......@@ -67,6 +68,8 @@ typedef struct DynamicAudioNormalizerContext {
int channels;
int delay;
int eof;
int64_t pts;
cqueue **gain_history_original;
cqueue **gain_history_minimum;
......@@ -292,10 +295,7 @@ static int config_input(AVFilterLink *inlink)
uninit(ctx);
s->frame_len =
inlink->min_samples =
inlink->max_samples =
inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec);
s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
......@@ -661,7 +661,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFilterContext *ctx = inlink->dst;
DynamicAudioNormalizerContext *s = ctx->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
int ret = 0;
int ret = 1;
if (!cqueue_empty(s->gain_history_smoothed[0])) {
AVFrame *out = ff_bufqueue_get(&s->queue);
......@@ -670,6 +670,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
ret = ff_filter_frame(outlink, out);
}
av_frame_make_writable(in);
analyze_frame(s, in);
ff_bufqueue_add(ctx, &s->queue, in);
......@@ -701,34 +702,75 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
return filter_frame(inlink, out);
}
static int request_frame(AVFilterLink *outlink)
static int flush(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
DynamicAudioNormalizerContext *s = ctx->priv;
int ret = 0;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay) {
if (!cqueue_empty(s->gain_history_smoothed[0])) {
ret = flush_buffer(s, ctx->inputs[0], outlink);
} else if (s->queue.available) {
AVFrame *out = ff_bufqueue_get(&s->queue);
s->pts = out->pts;
ret = ff_filter_frame(outlink, out);
}
}
return ret;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
DynamicAudioNormalizerContext *s = ctx->priv;
AVFrame *in = NULL;
int ret = 0, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (!s->eof) {
ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &in);
if (ret < 0)
return ret;
if (ret > 0) {
ret = filter_frame(inlink, in);
if (ret <= 0)
return ret;
}
if (ff_inlink_queued_samples(inlink) >= s->frame_len) {
ff_filter_set_ready(ctx, 10);
return 0;
}
}
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF)
s->eof = 1;
}
if (s->eof && s->delay > 0)
return flush(outlink);
if (s->eof && s->delay <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
if (!s->eof)
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
.needs_writable = 1,
},
{ NULL }
};
......@@ -737,7 +779,6 @@ static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
},
{ NULL }
};
......@@ -749,6 +790,7 @@ AVFilter ff_af_dynaudnorm = {
.priv_size = sizeof(DynamicAudioNormalizerContext),
.init = init,
.uninit = uninit,
.activate = activate,
.inputs = avfilter_af_dynaudnorm_inputs,
.outputs = avfilter_af_dynaudnorm_outputs,
.priv_class = &dynaudnorm_class,
......
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