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

This makes synchronization simpler for filters with multiple inputs.

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;
 };
 
 /**

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;
 }