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