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af_amerge: use the buferqueue API.

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This commit is contained in:
Nicolas George 2012-05-31 21:47:10 +02:00
parent 2f2d47ab63
commit 7f17f4f1a7
1 changed files with 34 additions and 52 deletions
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86
libavfilter/af_amerge.c
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@ -26,28 +26,27 @@
#include "libswresample/swresample.h" // only for SWR_CH_MAX #include "libswresample/swresample.h" // only for SWR_CH_MAX
#include "avfilter.h" #include "avfilter.h"
#include "audio.h" #include "audio.h"
#include "bufferqueue.h"
#include "internal.h" #include "internal.h"
#define QUEUE_SIZE 16
typedef struct { typedef struct {
int nb_in_ch[2]; /**< number of channels for each input */ int nb_in_ch[2]; /**< number of channels for each input */
int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */ int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
int bps; int bps;
struct amerge_queue { struct amerge_input {
AVFilterBufferRef *buf[QUEUE_SIZE]; struct FFBufQueue queue;
int nb_buf, nb_samples, pos; int nb_samples;
} queue[2]; int pos;
} in[2];
} AMergeContext; } AMergeContext;
static av_cold void uninit(AVFilterContext *ctx) static av_cold void uninit(AVFilterContext *ctx)
{ {
AMergeContext *am = ctx->priv; AMergeContext *am = ctx->priv;
int i, j; int i;
for (i = 0; i < 2; i++) for (i = 0; i < 2; i++)
for (j = 0; j < am->queue[i].nb_buf; j++) ff_bufqueue_discard_all(&am->in[i].queue);
avfilter_unref_buffer(am->queue[i].buf[j]);
} }
static int query_formats(AVFilterContext *ctx) static int query_formats(AVFilterContext *ctx)
@ -144,7 +143,7 @@ static int request_frame(AVFilterLink *outlink)
int i, ret; int i, ret;
for (i = 0; i < 2; i++) 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) if ((ret = avfilter_request_frame(ctx->inputs[i])) < 0)
return ret; return ret;
return 0; return 0;
@ -189,47 +188,38 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
AMergeContext *am = ctx->priv; AMergeContext *am = ctx->priv;
AVFilterLink *const outlink = ctx->outputs[0]; AVFilterLink *const outlink = ctx->outputs[0];
int input_number = inlink == ctx->inputs[1]; int input_number = inlink == ctx->inputs[1];
struct amerge_queue *inq = &am->queue[input_number];
int nb_samples, ns, i; int nb_samples, ns, i;
AVFilterBufferRef *outbuf, **inbuf[2]; AVFilterBufferRef *outbuf, *inbuf[2];
uint8_t *ins[2], *outs; uint8_t *ins[2], *outs;
if (inq->nb_buf == QUEUE_SIZE) { ff_bufqueue_add(ctx, &am->in[input_number].queue, insamples);
av_log(ctx, AV_LOG_ERROR, "Packet queue overflow; dropped\n"); am->in[input_number].nb_samples += insamples->audio->nb_samples;
avfilter_unref_buffer(insamples); if (!am->in[!input_number].nb_samples)
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)
return; return;
nb_samples = FFMIN(am->queue[0].nb_samples, nb_samples = FFMIN(am->in[0].nb_samples,
am->queue[1].nb_samples); am->in[1].nb_samples);
outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
nb_samples);
outs = outbuf->data[0]; outs = outbuf->data[0];
for (i = 0; i < 2; i++) { for (i = 0; i < 2; i++) {
inbuf[i] = am->queue[i].buf; inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
ins[i] = (*inbuf[i])->data[0] + ins[i] = inbuf[i]->data[0] +
am->queue[i].pos * am->nb_in_ch[i] * am->bps; am->in[i].pos * am->nb_in_ch[i] * am->bps;
} }
outbuf->pts = (*inbuf[0])->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE : outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
(*inbuf[0])->pts + inbuf[0]->pts +
av_rescale_q(am->queue[0].pos, av_rescale_q(am->in[0].pos,
(AVRational){ 1, ctx->inputs[0]->sample_rate }, (AVRational){ 1, ctx->inputs[0]->sample_rate },
ctx->outputs[0]->time_base); 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->nb_samples = nb_samples;
outbuf->audio->channel_layout = outlink->channel_layout; outbuf->audio->channel_layout = outlink->channel_layout;
while (nb_samples) { while (nb_samples) {
ns = nb_samples; ns = nb_samples;
for (i = 0; i < 2; i++) 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, /* Unroll the most common sample formats: speed +~350% for the loop,
+~13% overall (including two common decoders) */ +~13% overall (including two common decoders) */
switch (am->bps) { switch (am->bps) {
@ -249,25 +239,17 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
nb_samples -= ns; nb_samples -= ns;
for (i = 0; i < 2; i++) { for (i = 0; i < 2; i++) {
am->queue[i].nb_samples -= ns; am->in[i].nb_samples -= ns;
am->queue[i].pos += ns; am->in[i].pos += ns;
if (am->queue[i].pos == (*inbuf[i])->audio->nb_samples) { if (am->in[i].pos == inbuf[i]->audio->nb_samples) {
am->queue[i].pos = 0; am->in[i].pos = 0;
avfilter_unref_buffer(*inbuf[i]); avfilter_unref_buffer(inbuf[i]);
*inbuf[i] = NULL; ff_bufqueue_get(&am->in[i].queue);
inbuf[i]++; inbuf[i] = ff_bufqueue_peek(&am->in[i].queue, 0);
ins[i] = *inbuf[i] ? (*inbuf[i])->data[0] : NULL; 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); ff_filter_samples(ctx->outputs[0], outbuf);
} }
@ -283,11 +265,11 @@ AVFilter avfilter_af_amerge = {
{ .name = "in1", { .name = "in1",
.type = AVMEDIA_TYPE_AUDIO, .type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples, .filter_samples = filter_samples,
.min_perms = AV_PERM_READ, }, .min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = "in2", { .name = "in2",
.type = AVMEDIA_TYPE_AUDIO, .type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples, .filter_samples = filter_samples,
.min_perms = AV_PERM_READ, }, .min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = NULL } { .name = NULL }
}, },
.outputs = (const AVFilterPad[]) { .outputs = (const AVFilterPad[]) {