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https://github.com/xenia-project/FFmpeg.git
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avfilter/af_adelay: switch to activate
Also use less memory in special case when all channels have common delay.
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6a112349c2
commit
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@ -23,6 +23,7 @@
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "internal.h"
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typedef struct ChanDelay {
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@ -38,8 +39,10 @@ typedef struct AudioDelayContext {
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ChanDelay *chandelay;
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int nb_delays;
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int block_align;
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unsigned max_delay;
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int64_t padding;
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int64_t max_delay;
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int64_t next_pts;
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int eof;
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void (*delay_channel)(ChanDelay *d, int nb_samples,
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const uint8_t *src, uint8_t *dst);
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@ -158,6 +161,21 @@ static int config_input(AVFilterLink *inlink)
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}
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}
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s->padding = s->chandelay[0].delay;
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for (i = 1; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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s->padding = FFMIN(s->padding, d->delay);
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}
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if (s->padding) {
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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d->delay -= s->padding;
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}
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}
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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@ -210,21 +228,57 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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s->delay_channel(d, frame->nb_samples, src, dst);
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}
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s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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out_frame->pts = s->next_pts;
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s->next_pts += av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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av_frame_free(&frame);
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return ff_filter_frame(ctx->outputs[0], out_frame);
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}
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static int request_frame(AVFilterLink *outlink)
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static int activate(AVFilterContext *ctx)
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{
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AVFilterContext *ctx = outlink->src;
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AudioDelayContext *s = ctx->priv;
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int ret;
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AVFrame *frame = NULL;
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int ret, status;
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int64_t pts;
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ret = ff_request_frame(ctx->inputs[0]);
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if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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if (s->padding) {
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int nb_samples = FFMIN(s->padding, 2048);
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frame = ff_get_audio_buffer(outlink, nb_samples);
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if (!frame)
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return AVERROR(ENOMEM);
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s->padding -= nb_samples;
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av_samples_set_silence(frame->extended_data, 0,
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frame->nb_samples,
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outlink->channels,
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frame->format);
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frame->pts = s->next_pts;
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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return ff_filter_frame(outlink, frame);
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}
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ret = ff_inlink_consume_frame(inlink, &frame);
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if (ret < 0)
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return ret;
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if (ret > 0)
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return filter_frame(inlink, frame);
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if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
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if (status == AVERROR_EOF)
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s->eof = 1;
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}
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if (s->eof && s->max_delay) {
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int nb_samples = FFMIN(s->max_delay, 2048);
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AVFrame *frame;
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frame = ff_get_audio_buffer(outlink, nb_samples);
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if (!frame)
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@ -237,13 +291,18 @@ static int request_frame(AVFilterLink *outlink)
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frame->format);
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frame->pts = s->next_pts;
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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ret = filter_frame(ctx->inputs[0], frame);
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return filter_frame(inlink, frame);
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}
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return ret;
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if (s->eof && s->max_delay == 0) {
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ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
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return 0;
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}
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if (!s->eof)
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FF_FILTER_FORWARD_WANTED(outlink, inlink);
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return FFERROR_NOT_READY;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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@ -262,16 +321,14 @@ static const AVFilterPad adelay_inputs[] = {
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad adelay_outputs[] = {
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{
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.name = "default",
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.request_frame = request_frame,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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@ -282,6 +339,7 @@ AVFilter ff_af_adelay = {
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.query_formats = query_formats,
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.priv_size = sizeof(AudioDelayContext),
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.priv_class = &adelay_class,
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.activate = activate,
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.uninit = uninit,
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.inputs = adelay_inputs,
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.outputs = adelay_outputs,
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