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af_amerge: use the buferqueue API.

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pull/59/head
Nicolas George 13 years ago
parent 2f2d47ab63
commit 7f17f4f1a7
1 changed files with 34 additions and 52 deletions
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  1. 86
      libavfilter/af_amerge.c

86
libavfilter/af_amerge.c
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@ -26,28 +26,27 @@
#include "libswresample/swresample.h" // only for SWR_CH_MAX
#include "avfilter.h"
#include "audio.h"
#include "bufferqueue.h"
#include "internal.h"
#define QUEUE_SIZE 16
typedef struct {
int nb_in_ch[2]; /**< number of channels for each input */
int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
int bps;
struct amerge_queue {
AVFilterBufferRef *buf[QUEUE_SIZE];
int nb_buf, nb_samples, pos;
} queue[2];
struct amerge_input {
struct FFBufQueue queue;
int nb_samples;
int pos;
} in[2];
} AMergeContext;
static av_cold void uninit(AVFilterContext *ctx)
{
AMergeContext *am = ctx->priv;
int i, j;
int i;
for (i = 0; i < 2; i++)
for (j = 0; j < am->queue[i].nb_buf; j++)
avfilter_unref_buffer(am->queue[i].buf[j]);
ff_bufqueue_discard_all(&am->in[i].queue);
}
static int query_formats(AVFilterContext *ctx)
@ -144,7 +143,7 @@ static int request_frame(AVFilterLink *outlink)
int i, ret;
for (i = 0; i < 2; i++)
if (!am->queue[i].nb_samples)
if (!am->in[i].nb_samples)
if ((ret = avfilter_request_frame(ctx->inputs[i])) < 0)
return ret;
return 0;
@ -189,47 +188,38 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
AMergeContext *am = ctx->priv;
AVFilterLink *const outlink = ctx->outputs[0];
int input_number = inlink == ctx->inputs[1];
struct amerge_queue *inq = &am->queue[input_number];
int nb_samples, ns, i;
AVFilterBufferRef *outbuf, **inbuf[2];
AVFilterBufferRef *outbuf, *inbuf[2];
uint8_t *ins[2], *outs;
if (inq->nb_buf == QUEUE_SIZE) {
av_log(ctx, AV_LOG_ERROR, "Packet queue overflow; dropped\n");
avfilter_unref_buffer(insamples);
return;
}
inq->buf[inq->nb_buf++] = avfilter_ref_buffer(insamples, AV_PERM_READ |
AV_PERM_PRESERVE);
inq->nb_samples += insamples->audio->nb_samples;
avfilter_unref_buffer(insamples);
if (!am->queue[!input_number].nb_samples)
ff_bufqueue_add(ctx, &am->in[input_number].queue, insamples);
am->in[input_number].nb_samples += insamples->audio->nb_samples;
if (!am->in[!input_number].nb_samples)
return;
nb_samples = FFMIN(am->queue[0].nb_samples,
am->queue[1].nb_samples);
outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE,
nb_samples);
nb_samples = FFMIN(am->in[0].nb_samples,
am->in[1].nb_samples);
outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
outs = outbuf->data[0];
for (i = 0; i < 2; i++) {
inbuf[i] = am->queue[i].buf;
ins[i] = (*inbuf[i])->data[0] +
am->queue[i].pos * am->nb_in_ch[i] * am->bps;
inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
ins[i] = inbuf[i]->data[0] +
am->in[i].pos * am->nb_in_ch[i] * am->bps;
}
outbuf->pts = (*inbuf[0])->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
(*inbuf[0])->pts +
av_rescale_q(am->queue[0].pos,
outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
inbuf[0]->pts +
av_rescale_q(am->in[0].pos,
(AVRational){ 1, ctx->inputs[0]->sample_rate },
ctx->outputs[0]->time_base);
avfilter_copy_buffer_ref_props(outbuf, *inbuf[0]);
avfilter_copy_buffer_ref_props(outbuf, inbuf[0]);
outbuf->audio->nb_samples = nb_samples;
outbuf->audio->channel_layout = outlink->channel_layout;
while (nb_samples) {
ns = nb_samples;
for (i = 0; i < 2; i++)
ns = FFMIN(ns, (*inbuf[i])->audio->nb_samples - am->queue[i].pos);
ns = FFMIN(ns, inbuf[i]->audio->nb_samples - am->in[i].pos);
/* Unroll the most common sample formats: speed +~350% for the loop,
+~13% overall (including two common decoders) */
switch (am->bps) {
@ -249,25 +239,17 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
nb_samples -= ns;
for (i = 0; i < 2; i++) {
am->queue[i].nb_samples -= ns;
am->queue[i].pos += ns;
if (am->queue[i].pos == (*inbuf[i])->audio->nb_samples) {
am->queue[i].pos = 0;
avfilter_unref_buffer(*inbuf[i]);
*inbuf[i] = NULL;
inbuf[i]++;
ins[i] = *inbuf[i] ? (*inbuf[i])->data[0] : NULL;
am->in[i].nb_samples -= ns;
am->in[i].pos += ns;
if (am->in[i].pos == inbuf[i]->audio->nb_samples) {
am->in[i].pos = 0;
avfilter_unref_buffer(inbuf[i]);
ff_bufqueue_get(&am->in[i].queue);
inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL;
}
}
}
for (i = 0; i < 2; i++) {
int nbufused = inbuf[i] - am->queue[i].buf;
if (nbufused) {
am->queue[i].nb_buf -= nbufused;
memmove(am->queue[i].buf, inbuf[i],
am->queue[i].nb_buf * sizeof(**inbuf));
}
}
ff_filter_samples(ctx->outputs[0], outbuf);
}
@ -283,11 +265,11 @@ AVFilter avfilter_af_amerge = {
{ .name = "in1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = "in2",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {

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