[FFmpeg-cvslog] af_amerge: use the buferqueue API.
Nicolas George
git at videolan.org
Sun Jun 3 10:20:08 CEST 2012
ffmpeg | branch: master | Nicolas George <nicolas.george at normalesup.org> | Thu May 31 21:47:10 2012 +0200| [7f17f4f1a7ff1904ba431c62a6a576dc768203aa] | committer: Nicolas George
af_amerge: use the buferqueue API.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7f17f4f1a7ff1904ba431c62a6a576dc768203aa
---
libavfilter/af_amerge.c | 86 +++++++++++++++++++----------------------------
1 file changed, 34 insertions(+), 52 deletions(-)
diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c
index 27a35a8..971f7bc 100644
--- a/libavfilter/af_amerge.c
+++ b/libavfilter/af_amerge.c
@@ -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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