@@ -191,6 +191,13 @@ Null audio sink, do absolutely nothing with the input audio. It is

 mainly useful as a template and to be employed in analysis / debugging

 tools.

+@section abuffersink

+This sink is intended for programmatic use. Frames that arrive on this sink can

+be retrieved by the calling program using the interface defined in

+@file{libavfilter/buffersink.h}.

+

+This filter accepts no parameters.

+

 @c man end AUDIO SINKS

 @chapter Video Filters

@@ -104,6 +104,10 @@ void avfilter_register_all(void)

         avfilter_register(&avfilter_vsink_buffer);

     }

     {

+        extern AVFilter avfilter_asink_abuffer;

+        avfilter_register(&avfilter_asink_abuffer);

+    }

+    {

         extern AVFilter avfilter_vf_scale;

         avfilter_register(&avfilter_vf_scale);

     }

@@ -23,13 +23,20 @@

  * buffer sink

  */

+#include "libavutil/audio_fifo.h"

+#include "libavutil/audioconvert.h"

 #include "libavutil/fifo.h"

+#include "libavutil/mathematics.h"

+#include "audio.h"

 #include "avfilter.h"

 #include "buffersink.h"

 typedef struct {

-    AVFifoBuffer *fifo;          ///< FIFO buffer of video frame references

+    AVFifoBuffer *fifo;          ///< FIFO buffer of frame references

+

+    AVAudioFifo  *audio_fifo;    ///< FIFO for audio samples

+    int64_t next_pts;            ///< interpolating audio pts

 } BufferSinkContext;

 #define FIFO_INIT_SIZE 8

@@ -44,6 +51,9 @@ static av_cold void uninit(AVFilterContext *ctx)

         avfilter_unref_buffer(buf);

     }

     av_fifo_free(sink->fifo);

+

+    if (sink->audio_fifo)

+        av_audio_fifo_free(sink->audio_fifo);

 }

 static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)

@@ -58,9 +68,8 @@ static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)

     return 0;

 }

-static void end_frame(AVFilterLink *link)

+static void write_buf(AVFilterContext *ctx, AVFilterBufferRef *buf)

 {

-    AVFilterContext   *ctx = link->dst;

     BufferSinkContext *sink = ctx->priv;

     if (av_fifo_space(sink->fifo) < sizeof(AVFilterBufferRef *) &&

@@ -69,10 +78,20 @@ static void end_frame(AVFilterLink *link)

             return;

     }

-    av_fifo_generic_write(sink->fifo, &link->cur_buf, sizeof(link->cur_buf), NULL);

+    av_fifo_generic_write(sink->fifo, &buf, sizeof(buf), NULL);

+}

+

+static void end_frame(AVFilterLink *link)

+{

+    write_buf(link->dst, link->cur_buf);

     link->cur_buf = NULL;

 }

+static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)

+{

+    write_buf(link->dst, buf);

+}

+

 int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)

 {

     BufferSinkContext *sink = ctx->priv;

@@ -98,6 +117,66 @@ int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)

     return 0;

 }

+static int read_from_fifo(AVFilterContext *ctx, AVFilterBufferRef **pbuf,

+                          int nb_samples)

+{

+    BufferSinkContext *s = ctx->priv;

+    AVFilterLink   *link = ctx->inputs[0];

+    AVFilterBufferRef *buf;

+

+    if (!(buf = ff_get_audio_buffer(link, AV_PERM_WRITE, nb_samples)))

+        return AVERROR(ENOMEM);

+    av_audio_fifo_read(s->audio_fifo, (void**)buf->extended_data, nb_samples);

+

+    buf->pts = s->next_pts;

+    s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate},

+                                link->time_base);

+

+    *pbuf = buf;

+    return 0;

+

+}

+

+int av_buffersink_read_samples(AVFilterContext *ctx, AVFilterBufferRef **pbuf,

+                               int nb_samples)

+{

+    BufferSinkContext *s = ctx->priv;

+    AVFilterLink   *link = ctx->inputs[0];

+    int ret = 0;

+

+    if (!s->audio_fifo) {

+        int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);

+        if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples)))

+            return AVERROR(ENOMEM);

+    }

+

+    while (ret >= 0) {

+        AVFilterBufferRef *buf;

+

+        if (av_audio_fifo_size(s->audio_fifo) >= nb_samples)

+            return read_from_fifo(ctx, pbuf, nb_samples);

+

+        ret = av_buffersink_read(ctx, &buf);

+        if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo))

+            return read_from_fifo(ctx, pbuf, av_audio_fifo_size(s->audio_fifo));

+        else if (ret < 0)

+            return ret;

+

+        if (buf->pts != AV_NOPTS_VALUE) {

+            s->next_pts = buf->pts -

+                          av_rescale_q(av_audio_fifo_size(s->audio_fifo),

+                                       (AVRational){ 1, link->sample_rate },

+                                       link->time_base);

+        }

+

+        ret = av_audio_fifo_write(s->audio_fifo, (void**)buf->extended_data,

+                                  buf->audio->nb_samples);

+        avfilter_unref_buffer(buf);

+    }

+

+    return ret;

+}

+

 AVFilter avfilter_vsink_buffer = {

     .name      = "buffersink",

     .description = NULL_IF_CONFIG_SMALL("Buffer video frames, and make them available to the end of the filter graph."),

@@ -112,3 +191,18 @@ AVFilter avfilter_vsink_buffer = {

                                   { .name = NULL }},

     .outputs   = (AVFilterPad[]) {{ .name = NULL }},

 };

+

+AVFilter avfilter_asink_abuffer = {

+    .name      = "abuffersink",

+    .description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them available to the end of the filter graph."),

+    .priv_size = sizeof(BufferSinkContext),

+    .init      = init,

+    .uninit    = uninit,

+

+    .inputs    = (AVFilterPad[]) {{ .name           = "default",

+                                    .type           = AVMEDIA_TYPE_AUDIO,

+                                    .filter_samples = filter_samples,

+                                    .min_perms      = AV_PERM_READ, },

+                                  { .name = NULL }},

+    .outputs   = (AVFilterPad[]) {{ .name = NULL }},

+};

@@ -29,7 +29,7 @@

 /**

  * Get a buffer with filtered data from sink and put it in buf.

  *

- * @param sink pointer to a context of a buffersink AVFilter.

+ * @param sink pointer to a context of a buffersink or abuffersink AVFilter.

  * @param buf pointer to the buffer will be written here if buf is non-NULL. buf

  *            must be freed by the caller using avfilter_unref_buffer().

  *            Buf may also be NULL to query whether a buffer is ready to be

@@ -40,4 +40,23 @@

  */

 int av_buffersink_read(AVFilterContext *sink, AVFilterBufferRef **buf);

+/**

+ * Same as av_buffersink_read, but with the ability to specify the number of

+ * samples read. This function is less efficient than av_buffersink_read(),

+ * because it copies the data around.

+ *

+ * @param sink pointer to a context of the abuffersink AVFilter.

+ * @param buf pointer to the buffer will be written here if buf is non-NULL. buf

+ *            must be freed by the caller using avfilter_unref_buffer(). buf

+ *            will contain exactly nb_samples audio samples, except at the end

+ *            of stream, when it can contain less than nb_samples.

+ *            Buf may also be NULL to query whether a buffer is ready to be

+ *            output.

+ *

+ * @warning do not mix this function with av_buffersink_read(). Use only one or

+ * the other with a single sink, not both.

+ */

+int av_buffersink_read_samples(AVFilterContext *ctx, AVFilterBufferRef **buf,

+                               int nb_samples);

+

 #endif /* AVFILTER_BUFFERSINK_H */