[FFmpeg-cvslog] lavfi: allow audio filters to request a given number of samples.
Anton Khirnov
git at videolan.org
Sun Jun 24 02:20:45 CEST 2012
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Sun May 27 14:18:49 2012 +0200| [f75be9856a99739b2c22ed73a3c51df0f54a5ce9] | committer: Anton Khirnov
lavfi: allow audio filters to request a given number of samples.
This makes synchronization simpler for filters with multiple inputs.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f75be9856a99739b2c22ed73a3c51df0f54a5ce9
---
libavfilter/avfilter.h | 9 +++
libavfilter/fifo.c | 159 +++++++++++++++++++++++++++++++++++++++++++++---
2 files changed, 160 insertions(+), 8 deletions(-)
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
index c92f7e1..f09f086 100644
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -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;
};
/**
diff --git a/libavfilter/fifo.c b/libavfilter/fifo.c
index 3fa4faa..6d28757 100644
--- a/libavfilter/fifo.c
+++ b/libavfilter/fifo.c
@@ -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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