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avfilter/af_astats: add slice threading support

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This commit is contained in:
Paul B Mahol 2020-04-16 17:55:11 +02:00
parent 87b4fb65a8
commit 715da29501
1 changed files with 29 additions and 14 deletions
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43
libavfilter/af_astats.c
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@ -539,7 +539,7 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
} }
#define UPDATE_STATS_P(type, update_func, update_float, channel_func) \ #define UPDATE_STATS_P(type, update_func, update_float, channel_func) \
for (int c = 0; c < channels; c++) { \ for (int c = start; c < end; c++) { \
ChannelStats *p = &s->chstats[c]; \ ChannelStats *p = &s->chstats[c]; \
const type *src = (const type *)data[c]; \ const type *src = (const type *)data[c]; \
const type * const srcend = src + samples; \ const type * const srcend = src + samples; \
@ -551,7 +551,7 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
} }
#define UPDATE_STATS_I(type, update_func, update_float, channel_func) \ #define UPDATE_STATS_I(type, update_func, update_float, channel_func) \
for (int c = 0; c < channels; c++) { \ for (int c = start; c < end; c++) { \
ChannelStats *p = &s->chstats[c]; \ ChannelStats *p = &s->chstats[c]; \
const type *src = (const type *)data[0]; \ const type *src = (const type *)data[0]; \
const type * const srcend = src + samples * channels; \ const type * const srcend = src + samples * channels; \
@ -569,21 +569,16 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
UPDATE_STATS_##planar(type, update_minmax(s, p, sample), , p->nmin = p->min normalizer_suffix; p->nmax = p->max normalizer_suffix;); \ UPDATE_STATS_##planar(type, update_minmax(s, p, sample), , p->nmin = p->min normalizer_suffix; p->nmax = p->max normalizer_suffix;); \
} }
static int filter_frame(AVFilterLink *inlink, AVFrame *buf) static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{ {
AudioStatsContext *s = inlink->dst->priv; AudioStatsContext *s = ctx->priv;
AVDictionary **metadata = &buf->metadata; AVFilterLink *inlink = ctx->inputs[0];
AVFrame *buf = arg;
const uint8_t * const * const data = (const uint8_t * const *)buf->extended_data;
const int channels = s->nb_channels; const int channels = s->nb_channels;
const int samples = buf->nb_samples; const int samples = buf->nb_samples;
const uint8_t * const * const data = (const uint8_t * const *)buf->extended_data; const int start = (buf->channels * jobnr) / nb_jobs;
const int end = (buf->channels * (jobnr+1)) / nb_jobs;
if (s->reset_count > 0) {
if (s->nb_frames >= s->reset_count) {
reset_stats(s);
s->nb_frames = 0;
}
s->nb_frames++;
}
switch (inlink->format) { switch (inlink->format) {
case AV_SAMPLE_FMT_DBLP: case AV_SAMPLE_FMT_DBLP:
@ -618,6 +613,25 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
break; break;
} }
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
AudioStatsContext *s = ctx->priv;
AVDictionary **metadata = &buf->metadata;
if (s->reset_count > 0) {
if (s->nb_frames >= s->reset_count) {
reset_stats(s);
s->nb_frames = 0;
}
s->nb_frames++;
}
ctx->internal->execute(ctx, filter_channel, buf, NULL, FFMIN(inlink->channels, ff_filter_get_nb_threads(ctx)));
if (s->metadata) if (s->metadata)
set_metadata(s, metadata); set_metadata(s, metadata);
@ -812,4 +826,5 @@ AVFilter ff_af_astats = {
.uninit = uninit, .uninit = uninit,
.inputs = astats_inputs, .inputs = astats_inputs,
.outputs = astats_outputs, .outputs = astats_outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS,
}; };