lavfi: add an audio buffer sink.

4 files changed, 129 insertions, 5 deletions

diff --git a/doc/filters.texi b/doc/filters.texi
index 8d4242607a..0314bfaf20 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -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
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 25cd8222c3..c84b3f2587 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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);
     }
diff --git a/libavfilter/buffersink.c b/libavfilter/buffersink.c
index e4cbe3be42..8787268f4c 100644
--- a/libavfilter/buffersink.c
+++ b/libavfilter/buffersink.c
@@ -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 }},
+};
diff --git a/libavfilter/buffersink.h b/libavfilter/buffersink.h
index e579b9ad03..8f94a9c94d 100644
--- a/libavfilter/buffersink.h
+++ b/libavfilter/buffersink.h
@@ -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 */