[PATCH 1/2 v2][spice-gtk] audio: new-sample callback must return GST_FLOW_OK

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Changes since v1
* Use g_return_val_if_fail instead of g_return_if_fail

---
 gtk/spice-gstaudio.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/gtk/spice-gstaudio.c b/gtk/spice-gstaudio.c
index 6173783..5f9abb2 100644
--- a/gtk/spice-gstaudio.c
+++ b/gtk/spice-gstaudio.c
@@ -120,13 +120,13 @@ static void spice_gstaudio_class_init(SpiceGstaudioClass *klass)
     g_type_class_add_private(klass, sizeof(SpiceGstaudioPrivate));
 }
 
-static void record_new_buffer(GstAppSink *appsink, gpointer data)
+static GstFlowReturn record_new_buffer(GstAppSink *appsink, gpointer data)
 {
     SpiceGstaudio *gstaudio = data;
     SpiceGstaudioPrivate *p = gstaudio->priv;
     GstMessage *msg;
 
-    g_return_if_fail(p != NULL);
+    g_return_val_if_fail(p != NULL, GST_FLOW_ERROR);
 
 #ifdef WITH_GST1AUDIO
     msg = gst_message_new_application(GST_OBJECT(p->record.pipe),
@@ -135,6 +135,7 @@ static void record_new_buffer(GstAppSink *appsink, gpointer data)
     msg = gst_message_new_application(GST_OBJECT(p->record.pipe), NULL);
 #endif
     gst_element_post_message(p->record.pipe, msg);
+    return GST_FLOW_OK;
 }
 
 static void record_stop(SpiceGstaudio *gstaudio)
-- 
2.1.0

_______________________________________________
Spice-devel mailing list
Spice-devel@xxxxxxxxxxxxxxxxxxxxx
http://lists.freedesktop.org/mailman/listinfo/spice-devel





[Index of Archives]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [ECOS]     [Asterisk Internet PBX]     [Linux API]     [Monitors]