From: Rico Tzschichholz Date: Thu, 23 May 2019 09:48:54 +0000 (+0200) Subject: gstreamer: Update from 1.17.0+ git master X-Git-Tag: 0.45.2~21 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=44f94a26333d7c30288f8544f2eaebe9d8c51080;p=thirdparty%2Fvala.git gstreamer: Update from 1.17.0+ git master --- diff --git a/vapi/gstreamer-audio-1.0.vapi b/vapi/gstreamer-audio-1.0.vapi index 5ff80bf23..b0f71c311 100644 --- a/vapi/gstreamer-audio-1.0.vapi +++ b/vapi/gstreamer-audio-1.0.vapi @@ -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")] diff --git a/vapi/gstreamer-base-1.0.vapi b/vapi/gstreamer-base-1.0.vapi index 67fce6d75..fdeeb0ea0 100644 --- a/vapi/gstreamer-base-1.0.vapi +++ b/vapi/gstreamer-base-1.0.vapi @@ -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; } diff --git a/vapi/gstreamer-webrtc-1.0.vapi b/vapi/gstreamer-webrtc-1.0.vapi index ab79bde18..7583d1842 100644 --- a/vapi/gstreamer-webrtc-1.0.vapi +++ b/vapi/gstreamer-webrtc-1.0.vapi @@ -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,