]> git.ipfire.org Git - thirdparty/vala.git/commitdiff
gstreamer: Update from 1.17.0+ git master
authorRico Tzschichholz <ricotz@ubuntu.com>
Thu, 23 May 2019 09:48:54 +0000 (11:48 +0200)
committerRico Tzschichholz <ricotz@ubuntu.com>
Thu, 23 May 2019 09:48:54 +0000 (11:48 +0200)
vapi/gstreamer-audio-1.0.vapi
vapi/gstreamer-base-1.0.vapi
vapi/gstreamer-webrtc-1.0.vapi

index 5ff80bf239de85c5a2080284fe584893b8a1262a..b0f71c31135528a5a97184477264cff5d7f70c5d 100644 (file)
@@ -21,6 +21,8 @@ namespace Gst {
                        public uint64 discont_wait { get; set; }
                        [NoAccessorMethod]
                        public uint64 output_buffer_duration { get; set; }
+                       [NoAccessorMethod]
+                       public Gst.Fraction output_buffer_duration_fraction { owned get; set; }
                }
                [CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_aggregator_convert_pad_get_type ()")]
                [GIR (name = "AudioAggregatorConvertPad")]
index 67fce6d75d780b1268a965e065d93ec6f52e098d..fdeeb0ea05b05f10b6cb47c742509e8200eab711 100644 (file)
@@ -593,6 +593,7 @@ namespace Gst {
                        [Version (since = "1.16")]
                        public Gst.ClockTime get_processing_deadline ();
                        public Gst.ClockTime get_render_delay ();
+                       [Version (since = "1.18")]
                        public Gst.Structure get_stats ();
                        public bool get_sync ();
                        public uint64 get_throttle_time ();
@@ -661,6 +662,8 @@ namespace Gst {
                        [NoAccessorMethod]
                        public bool qos { get; set; }
                        public uint64 render_delay { get; set; }
+                       [Version (since = "1.18")]
+                       public Gst.Structure stats { owned get; }
                        public bool sync { get; set; }
                        public uint64 throttle_time { get; set; }
                        public int64 ts_offset { get; set; }
index ab79bde18f3828118057b6c3b39b048abf47eda7..7583d184261a6084b260e3efcc24c853d24b9b59 100644 (file)
@@ -104,6 +104,7 @@ namespace Gst {
                public void free ();
        }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_BUNDLE_POLICY_", type_id = "gst_webrtc_bundle_policy_get_type ()")]
+       [Version (since = "1.16")]
        public enum WebRTCBundlePolicy {
                NONE,
                BALANCED,
@@ -126,6 +127,7 @@ namespace Gst {
                CONNECTED
        }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DATA_CHANNEL_STATE_", type_id = "gst_webrtc_data_channel_state_get_type ()")]
+       [Version (since = "1.16")]
        public enum WebRTCDataChannelState {
                NEW,
                CONNECTING,
@@ -134,6 +136,7 @@ namespace Gst {
                CLOSED
        }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_FEC_TYPE_", type_id = "gst_webrtc_fec_type_get_type ()")]
+       [Version (since = "1.14.1")]
        public enum WebRTCFECType {
                NONE,
                ULP_RED
@@ -165,6 +168,7 @@ namespace Gst {
                CONTROLLING
        }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_TRANSPORT_POLICY_", type_id = "gst_webrtc_ice_transport_policy_get_type ()")]
+       [Version (since = "1.16")]
        public enum WebRTCICETransportPolicy {
                ALL,
                RELAY
@@ -179,6 +183,7 @@ namespace Gst {
                CLOSED
        }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PRIORITY_TYPE_", type_id = "gst_webrtc_priority_type_get_type ()")]
+       [Version (since = "1.16")]
        public enum WebRTCPriorityType {
                VERY_LOW,
                LOW,
@@ -194,6 +199,7 @@ namespace Gst {
                SENDRECV
        }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SCTP_TRANSPORT_STATE_", type_id = "gst_webrtc_sctp_transport_state_get_type ()")]
+       [Version (since = "1.16")]
        public enum WebRTCSCTPTransportState {
                NEW,
                CONNECTING,