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lavfi: allow audio filters to request a given number of samples.

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This makes synchronization simpler for filters with multiple inputs.
pull/59/head
Anton Khirnov 13 years ago
parent 58b049f2fa
commit f75be9856a
2 changed files with 160 additions and 8 deletions
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  1. 9
      libavfilter/avfilter.h
  2. 159
      libavfilter/fifo.c

9
libavfilter/avfilter.h
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@ -595,6 +595,15 @@ struct AVFilterLink {
AVFilterFormats *out_samplerates;
struct AVFilterChannelLayouts *in_channel_layouts;
struct AVFilterChannelLayouts *out_channel_layouts;
/**
* Audio only, the destination filter sets this to a non-zero value to
* request that buffers with the given number of samples should be sent to
* it. AVFilterPad.needs_fifo must also be set on the corresponding input
* pad.
* Last buffer before EOF will be padded with silence.
*/
int request_samples;
};
/**

159
libavfilter/fifo.c
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@ -23,6 +23,11 @@
* FIFO buffering filter
*/
#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
@ -36,6 +41,13 @@ typedef struct Buf {
typedef struct {
Buf root;
Buf *last; ///< last buffered frame
/**
* When a specific number of output samples is requested, the partial
* buffer is stored here
*/
AVFilterBufferRef *buf_out;
int allocated_samples; ///< number of samples buf_out was allocated for
} FifoContext;
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
@ -57,6 +69,8 @@ static av_cold void uninit(AVFilterContext *ctx)
avfilter_unref_buffer(buf->buf);
av_free(buf);
}
avfilter_unref_buffer(fifo->buf_out);
}
static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
@ -68,14 +82,143 @@ static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
fifo->last->buf = buf;
}
static void queue_pop(FifoContext *s)
{
Buf *tmp = s->root.next->next;
if (s->last == s->root.next)
s->last = &s->root;
av_freep(&s->root.next);
s->root.next = tmp;
}
static void end_frame(AVFilterLink *inlink) { }
static void draw_slice(AVFilterLink *inlink, int y, int h, int slice_dir) { }
/**
* Move data pointers and pts offset samples forward.
*/
static void buffer_offset(AVFilterLink *link, AVFilterBufferRef *buf,
int offset)
{
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
int planar = av_sample_fmt_is_planar(link->format);
int planes = planar ? nb_channels : 1;
int block_align = av_get_bytes_per_sample(link->format) * (planar ? 1 : nb_channels);
int i;
av_assert0(buf->audio->nb_samples > offset);
for (i = 0; i < planes; i++)
buf->extended_data[i] += block_align*offset;
if (buf->data != buf->extended_data)
memcpy(buf->data, buf->extended_data,
FFMIN(planes, FF_ARRAY_ELEMS(buf->data)) * sizeof(*buf->data));
buf->linesize[0] -= block_align*offset;
buf->audio->nb_samples -= offset;
if (buf->pts != AV_NOPTS_VALUE) {
buf->pts += av_rescale_q(offset, (AVRational){1, link->sample_rate},
link->time_base);
}
}
static int calc_ptr_alignment(AVFilterBufferRef *buf)
{
int planes = av_sample_fmt_is_planar(buf->format) ?
av_get_channel_layout_nb_channels(buf->audio->channel_layout) : 1;
int min_align = 128;
int p;
for (p = 0; p < planes; p++) {
int cur_align = 128;
while ((intptr_t)buf->extended_data[p] % cur_align)
cur_align >>= 1;
if (cur_align < min_align)
min_align = cur_align;
}
return min_align;
}
static int return_audio_frame(AVFilterContext *ctx)
{
AVFilterLink *link = ctx->outputs[0];
FifoContext *s = ctx->priv;
AVFilterBufferRef *head = s->root.next->buf;
AVFilterBufferRef *buf_out;
int ret;
if (!s->buf_out &&
head->audio->nb_samples >= link->request_samples &&
calc_ptr_alignment(head) >= 32) {
if (head->audio->nb_samples == link->request_samples) {
buf_out = head;
queue_pop(s);
} else {
buf_out = avfilter_ref_buffer(head, AV_PERM_READ);
buf_out->audio->nb_samples = link->request_samples;
buffer_offset(link, head, link->request_samples);
}
} else {
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
if (!s->buf_out) {
s->buf_out = ff_get_audio_buffer(link, AV_PERM_WRITE,
link->request_samples);
if (!s->buf_out)
return AVERROR(ENOMEM);
s->buf_out->audio->nb_samples = 0;
s->buf_out->pts = head->pts;
s->allocated_samples = link->request_samples;
} else if (link->request_samples != s->allocated_samples) {
av_log(ctx, AV_LOG_ERROR, "request_samples changed before the "
"buffer was returned.\n");
return AVERROR(EINVAL);
}
while (s->buf_out->audio->nb_samples < s->allocated_samples) {
int len = FFMIN(s->allocated_samples - s->buf_out->audio->nb_samples,
head->audio->nb_samples);
av_samples_copy(s->buf_out->extended_data, head->extended_data,
s->buf_out->audio->nb_samples, 0, len, nb_channels,
link->format);
s->buf_out->audio->nb_samples += len;
if (len == head->audio->nb_samples) {
avfilter_unref_buffer(head);
queue_pop(s);
if (!s->root.next &&
(ret = ff_request_frame(ctx->inputs[0])) < 0) {
if (ret == AVERROR_EOF) {
av_samples_set_silence(s->buf_out->extended_data,
s->buf_out->audio->nb_samples,
s->allocated_samples -
s->buf_out->audio->nb_samples,
nb_channels, link->format);
s->buf_out->audio->nb_samples = s->allocated_samples;
break;
}
return ret;
}
head = s->root.next->buf;
} else {
buffer_offset(link, head, len);
}
}
buf_out = s->buf_out;
s->buf_out = NULL;
}
ff_filter_samples(link, buf_out);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
FifoContext *fifo = outlink->src->priv;
Buf *tmp;
int ret;
if (!fifo->root.next) {
@ -90,20 +233,20 @@ static int request_frame(AVFilterLink *outlink)
ff_start_frame(outlink, fifo->root.next->buf);
ff_draw_slice (outlink, 0, outlink->h, 1);
ff_end_frame (outlink);
queue_pop(fifo);
break;
case AVMEDIA_TYPE_AUDIO:
ff_filter_samples(outlink, fifo->root.next->buf);
if (outlink->request_samples) {
return return_audio_frame(outlink->src);
} else {
ff_filter_samples(outlink, fifo->root.next->buf);
queue_pop(fifo);
}
break;
default:
return AVERROR(EINVAL);
}
if (fifo->last == fifo->root.next)
fifo->last = &fifo->root;
tmp = fifo->root.next->next;
av_free(fifo->root.next);
fifo->root.next = tmp;
return 0;
}

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