@@ -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;

 };

 /**

@@ -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;

 }