interaudiosink: Ensure adapters don't store buffers with audio meta

The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5416>

diff --git a/subprojects/gst-integration-testsuites/testsuites/validate.testslist b/subprojects/gst-integration-testsuites/testsuites/validate.testslist
index c5e54380295aea34f11f2b25b16bf842dcb564fb..02f22cca5ced829e25d88279ab638f5fbe5d8a05 100644 (file)
--- a/subprojects/gst-integration-testsuites/testsuites/validate.testslist
+++ b/subprojects/gst-integration-testsuites/testsuites/validate.testslist
@@ -936,6 +936,7 @@ validate.test.h264.parse.trickmode_predicted.seek_trickmode_predicted
 validate.test.h264.x264enc_youtube_bitrate.fullhd_lowframerate
 validate.test.h265parse.alternate
 validate.test.h265parse.trickmode_predicted.seek_trickmode_predicted
+validate.test.inter.audio-planar
 validate.test.interlace.interlace_deinterlace
 validate.test.interlace.interlace_deinterlace_alternate
 validate.test.matroska.demux_flush_within_cluster.default

diff --git a/subprojects/gst-integration-testsuites/testsuites/validate/inter/audio-planar.validatetest b/subprojects/gst-integration-testsuites/testsuites/validate/inter/audio-planar.validatetest
new file mode 100644 (file)
index 0000000..42e2c48
--- /dev/null
+++ b/subprojects/gst-integration-testsuites/testsuites/validate/inter/audio-planar.validatetest
@@ -0,0 +1,4 @@
+meta,
+  args = {
+       "audiotestsrc num-buffers=10 ! audio/x-raw,format=S16LE,channels=2,layout=non-interleaved ! interaudiosink interaudiosrc num-buffers=10 ! audioconvert ! audio/x-raw,layout=interleaved,format=S16LE,channels=2 ! queue ! fakeaudiosink",
+  }

diff --git a/subprojects/gst-plugins-bad/gst/inter/gstinteraudiosink.c b/subprojects/gst-plugins-bad/gst/inter/gstinteraudiosink.c
index a33fe8194f7317774328adaa5a26e9494e789455..6c654e889e05fed8c23565efec775fa206a3fa0e 100644 (file)
--- a/subprojects/gst-plugins-bad/gst/inter/gstinteraudiosink.c
+++ b/subprojects/gst-plugins-bad/gst/inter/gstinteraudiosink.c
@@ -293,6 +293,7 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
   guint n, bpf;
   guint64 period_time, buffer_time;
   guint64 period_samples, buffer_samples;
+  GstBuffer *tmp;
 
   GST_DEBUG_OBJECT (interaudiosink, "render %" G_GSIZE_FORMAT,
       gst_buffer_get_size (buffer));
@@ -330,16 +331,26 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
 
   n = gst_adapter_available (interaudiosink->input_adapter);
   if (period_samples * bpf > gst_buffer_get_size (buffer) + n) {
-    gst_adapter_push (interaudiosink->input_adapter, gst_buffer_ref (buffer));
+    GstAudioMeta *audio_meta = NULL;
+
+    tmp = gst_buffer_copy_deep (buffer);
+    audio_meta = gst_buffer_get_audio_meta (tmp);
+    if (audio_meta != NULL)
+      gst_buffer_remove_meta (tmp, GST_META_CAST (audio_meta));
+
+    gst_adapter_push (interaudiosink->input_adapter, tmp);
   } else {
-    GstBuffer *tmp;
+    GstAudioMeta *audio_meta = NULL;
 
     if (n > 0) {
       tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
       gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
     }
-    gst_adapter_push (interaudiosink->surface->audio_adapter,
-        gst_buffer_ref (buffer));
+    tmp = gst_buffer_copy_deep (buffer);
+    audio_meta = gst_buffer_get_audio_meta (tmp);
+    if (audio_meta != NULL)
+      gst_buffer_remove_meta (tmp, GST_META_CAST (audio_meta));
+    gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
   }
   g_mutex_unlock (&interaudiosink->surface->mutex);