From: Ali Sabil <ali.sabil@gmail.com>
Date: Tue, 3 Mar 2009 15:26:03 +0000 (+0100)
Subject: gstreamer-rtp-0.10: regenerated with gstreamer-0.10.21
X-Git-Tag: 0.6.0~68
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=e4318bd6bd2ed863f2d63fedd72350e29b201392;p=thirdparty%2Fvala.git

gstreamer-rtp-0.10: regenerated with gstreamer-0.10.21
---

diff --git a/vapi/gstreamer-rtp-0.10.vapi b/vapi/gstreamer-rtp-0.10.vapi
index c7006ec65..cd8476f8b 100644
--- a/vapi/gstreamer-rtp-0.10.vapi
+++ b/vapi/gstreamer-rtp-0.10.vapi
@@ -14,6 +14,7 @@ namespace Gst {
 		public void set_frame_options (int frame_duration, int frame_size);
 		public void set_sample_based ();
 		public void set_sample_options (int sample_size);
+		public void set_samplebits_options (int sample_size);
 	}
 	[CCode (cheader_filename = "gst/rtp/gstbasertpdepayload.h")]
 	public class BaseRTPDepayload : Gst.Element {
@@ -29,6 +30,8 @@ namespace Gst {
 		[NoWrapper]
 		public virtual Gst.FlowReturn add_to_queue (Gst.Buffer @in);
 		[NoWrapper]
+		public virtual bool packet_lost (Gst.Event event);
+		[NoWrapper]
 		public virtual unowned Gst.Buffer process (Gst.Buffer @in);
 		public Gst.FlowReturn push (Gst.Buffer out_buf);
 		public Gst.FlowReturn push_ts (uint32 timestamp, Gst.Buffer out_buf);
@@ -366,6 +369,8 @@ namespace Gst {
 	public static void rtp_buffer_set_csrc (Gst.Buffer buffer, uchar idx, uint32 csrc);
 	[CCode (cheader_filename = "gst/rtp/gstrtpbuffer.h")]
 	public static void rtp_buffer_set_extension (Gst.Buffer buffer, bool extension);
+	[CCode (cheader_filename = "gst/gst.h")]
+	public static bool rtp_buffer_set_extension_data (Gst.Buffer buffer, uint16 bits, uint16 length);
 	[CCode (cheader_filename = "gst/rtp/gstrtpbuffer.h")]
 	public static void rtp_buffer_set_marker (Gst.Buffer buffer, bool marker);
 	[CCode (cheader_filename = "gst/rtp/gstrtpbuffer.h")]
diff --git a/vapi/packages/gstreamer-rtp-0.10/gstreamer-rtp-0.10.gi b/vapi/packages/gstreamer-rtp-0.10/gstreamer-rtp-0.10.gi
index e04169fa6..d6d8d9a90 100644
--- a/vapi/packages/gstreamer-rtp-0.10/gstreamer-rtp-0.10.gi
+++ b/vapi/packages/gstreamer-rtp-0.10/gstreamer-rtp-0.10.gi
@@ -284,6 +284,14 @@
 				<parameter name="extension" type="gboolean"/>
 			</parameters>
 		</function>
+		<function name="rtp_buffer_set_extension_data" symbol="gst_rtp_buffer_set_extension_data">
+			<return-type type="gboolean"/>
+			<parameters>
+				<parameter name="buffer" type="GstBuffer*"/>
+				<parameter name="bits" type="guint16"/>
+				<parameter name="length" type="guint16"/>
+			</parameters>
+		</function>
 		<function name="rtp_buffer_set_marker" symbol="gst_rtp_buffer_set_marker">
 			<return-type type="void"/>
 			<parameters>
@@ -705,6 +713,13 @@
 					<parameter name="sample_size" type="gint"/>
 				</parameters>
 			</method>
+			<method name="set_samplebits_options" symbol="gst_base_rtp_audio_payload_set_samplebits_options">
+				<return-type type="void"/>
+				<parameters>
+					<parameter name="basertpaudiopayload" type="GstBaseRTPAudioPayload*"/>
+					<parameter name="sample_size" type="gint"/>
+				</parameters>
+			</method>
 			<field name="base_ts" type="GstClockTime"/>
 			<field name="frame_size" type="gint"/>
 			<field name="frame_duration" type="gint"/>
@@ -734,6 +749,13 @@
 					<parameter name="in" type="GstBuffer*"/>
 				</parameters>
 			</vfunc>
+			<vfunc name="packet_lost">
+				<return-type type="gboolean"/>
+				<parameters>
+					<parameter name="filter" type="GstBaseRTPDepayload*"/>
+					<parameter name="event" type="GstEvent*"/>
+				</parameters>
+			</vfunc>
 			<vfunc name="process">
 				<return-type type="GstBuffer*"/>
 				<parameters>