avfilter/af_amerge: port to activate API

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2018-05-04 11:48:35 +02:00
parent d05c3b9cee
commit ac86011b1f
1 changed files with 69 additions and 77 deletions

View File

@ -31,8 +31,8 @@
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "filters.h"
#include "audio.h"
#include "bufferqueue.h"
#include "internal.h"
#define SWR_CH_MAX 64
@ -43,10 +43,7 @@ typedef struct AMergeContext {
int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
int bps;
struct amerge_input {
struct FFBufQueue queue;
int nb_ch; /**< number of channels for the input */
int nb_samples;
int pos;
} *in;
} AMergeContext;
@ -67,8 +64,6 @@ static av_cold void uninit(AVFilterContext *ctx)
int i;
for (i = 0; i < s->nb_inputs; i++) {
if (s->in)
ff_bufqueue_discard_all(&s->in[i].queue);
if (ctx->input_pads)
av_freep(&ctx->input_pads[i].name);
}
@ -183,21 +178,6 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AMergeContext *s = ctx->priv;
int i, ret;
for (i = 0; i < s->nb_inputs; i++)
if (!s->in[i].nb_samples ||
/* detect EOF immediately */
(ctx->inputs[i]->status_in && !ctx->inputs[i]->status_out))
if ((ret = ff_request_frame(ctx->inputs[i])) < 0)
return ret;
return 0;
}
/**
* Copy samples from several input streams to one output stream.
* @param nb_inputs number of inputs
@ -235,88 +215,101 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
}
}
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
static void free_frames(int nb_inputs, AVFrame **input_frames)
{
AVFilterContext *ctx = inlink->dst;
AMergeContext *s = ctx->priv;
AVFilterLink *const outlink = ctx->outputs[0];
int input_number;
int nb_samples, ns, i;
AVFrame *outbuf, *inbuf[SWR_CH_MAX];
uint8_t *ins[SWR_CH_MAX], *outs;
int i;
for (i = 0; i < nb_inputs; i++)
av_frame_free(&input_frames[i]);
}
for (input_number = 0; input_number < s->nb_inputs; input_number++)
if (inlink == ctx->inputs[input_number])
break;
av_assert1(input_number < s->nb_inputs);
if (ff_bufqueue_is_full(&s->in[input_number].queue)) {
av_frame_free(&insamples);
return AVERROR(ENOMEM);
static int try_push_frame(AVFilterContext *ctx, int nb_samples)
{
AMergeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int i, ret;
AVFrame *outbuf, *inbuf[SWR_CH_MAX] = { NULL };
uint8_t *outs, *ins[SWR_CH_MAX];
for (i = 0; i < ctx->nb_inputs; i++) {
ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &inbuf[i]);
if (ret < 0) {
free_frames(i, inbuf);
return ret;
}
ins[i] = inbuf[i]->data[0];
}
ff_bufqueue_add(ctx, &s->in[input_number].queue, av_frame_clone(insamples));
s->in[input_number].nb_samples += insamples->nb_samples;
av_frame_free(&insamples);
nb_samples = s->in[0].nb_samples;
for (i = 1; i < s->nb_inputs; i++)
nb_samples = FFMIN(nb_samples, s->in[i].nb_samples);
if (!nb_samples)
return 0;
outbuf = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
if (!outbuf)
if (!outbuf) {
free_frames(s->nb_inputs, inbuf);
return AVERROR(ENOMEM);
outs = outbuf->data[0];
for (i = 0; i < s->nb_inputs; i++) {
inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
ins[i] = inbuf[i]->data[0] +
s->in[i].pos * s->in[i].nb_ch * s->bps;
}
av_frame_copy_props(outbuf, inbuf[0]);
outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
inbuf[0]->pts +
av_rescale_q(s->in[0].pos,
av_make_q(1, ctx->inputs[0]->sample_rate),
ctx->outputs[0]->time_base);
outs = outbuf->data[0];
outbuf->pts = inbuf[0]->pts;
outbuf->nb_samples = nb_samples;
outbuf->channel_layout = outlink->channel_layout;
outbuf->channels = outlink->channels;
while (nb_samples) {
ns = nb_samples;
for (i = 0; i < s->nb_inputs; i++)
ns = FFMIN(ns, inbuf[i]->nb_samples - s->in[i].pos);
/* Unroll the most common sample formats: speed +~350% for the loop,
+~13% overall (including two common decoders) */
switch (s->bps) {
case 1:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 1);
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 1);
break;
case 2:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 2);
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 2);
break;
case 4:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 4);
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 4);
break;
default:
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, s->bps);
copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, s->bps);
break;
}
nb_samples -= ns;
for (i = 0; i < s->nb_inputs; i++) {
s->in[i].nb_samples -= ns;
s->in[i].pos += ns;
if (s->in[i].pos == inbuf[i]->nb_samples) {
s->in[i].pos = 0;
av_frame_free(&inbuf[i]);
ff_bufqueue_get(&s->in[i].queue);
inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0);
ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL;
}
nb_samples = 0;
}
free_frames(s->nb_inputs, inbuf);
return ff_filter_frame(ctx->outputs[0], outbuf);
}
static int activate(AVFilterContext *ctx)
{
int i, status;
int ret, nb_samples;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
nb_samples = ff_framequeue_queued_samples(&ctx->inputs[0]->fifo);
for (i = 1; i < ctx->nb_inputs && nb_samples > 0; i++) {
nb_samples = FFMIN(ff_framequeue_queued_samples(&ctx->inputs[i]->fifo), nb_samples);
}
if (nb_samples) {
ret = try_push_frame(ctx, nb_samples);
if (ret < 0)
return ret;
}
for (i = 0; i < ctx->nb_inputs; i++) {
if (ff_framequeue_queued_samples(&ctx->inputs[i]->fifo))
continue;
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
return 0;
} else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
ff_inlink_request_frame(ctx->inputs[i]);
return 0;
}
}
return ff_filter_frame(ctx->outputs[0], outbuf);
return 0;
}
static av_cold int init(AVFilterContext *ctx)
@ -332,7 +325,6 @@ static av_cold int init(AVFilterContext *ctx)
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
};
if (!name)
return AVERROR(ENOMEM);
@ -349,7 +341,6 @@ static const AVFilterPad amerge_outputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame,
},
{ NULL }
};
@ -362,6 +353,7 @@ AVFilter ff_af_amerge = {
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.activate = activate,
.inputs = NULL,
.outputs = amerge_outputs,
.priv_class = &amerge_class,