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More webrtc API fixup

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This commit is contained in:
Sebastian Dröge 2018-03-15 17:34:40 +02:00
parent 3bf11dd4b4
commit 217a8671a5
2 changed files with 2 additions and 83 deletions
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78
gir-files/GstWebRTC-1.0.gir
View file

@ -455,34 +455,6 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
</return-value>
</constructor>
<method name="get_parameters"
c:identifier="gst_webrtc_rtp_receiver_get_parameters">
<return-value transfer-ownership="full">
<type name="Gst.Structure" c:type="GstStructure*"/>
</return-value>
<parameters>
<instance-parameter name="receiver" transfer-ownership="none">
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
</instance-parameter>
<parameter name="kind" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</parameter>
</parameters>
</method>
<method name="set_parameters"
c:identifier="gst_webrtc_rtp_receiver_set_parameters">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="receiver" transfer-ownership="none">
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
</instance-parameter>
<parameter name="parameters" transfer-ownership="none">
<type name="Gst.Structure" c:type="GstStructure*"/>
</parameter>
</parameters>
</method>
<method name="set_rtcp_transport"
c:identifier="gst_webrtc_rtp_receiver_set_rtcp_transport">
<return-value transfer-ownership="none">
@ -545,48 +517,11 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
glib:type-name="GstWebRTCRTPSender"
glib:get-type="gst_webrtc_rtp_sender_get_type"
glib:type-struct="WebRTCRTPSenderClass">
<constructor name="new"
c:identifier="gst_webrtc_rtp_sender_new"
introspectable="0">
<constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
<return-value transfer-ownership="none">
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
</return-value>
<parameters>
<parameter name="send_encodings" transfer-ownership="none">
<array name="GLib.Array" c:type="GArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
</parameter>
</parameters>
</constructor>
<method name="get_parameters"
c:identifier="gst_webrtc_rtp_sender_get_parameters">
<return-value transfer-ownership="full">
<type name="Gst.Structure" c:type="GstStructure*"/>
</return-value>
<parameters>
<instance-parameter name="sender" transfer-ownership="none">
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
</instance-parameter>
<parameter name="kind" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</parameter>
</parameters>
</method>
<method name="set_parameters"
c:identifier="gst_webrtc_rtp_sender_set_parameters">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="sender" transfer-ownership="none">
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
</instance-parameter>
<parameter name="parameters" transfer-ownership="none">
<type name="Gst.Structure" c:type="GstStructure*"/>
</parameter>
</parameters>
</method>
<method name="set_rtcp_transport"
c:identifier="gst_webrtc_rtp_sender_set_rtcp_transport">
<return-value transfer-ownership="none">
@ -655,17 +590,6 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
glib:type-name="GstWebRTCRTPTransceiver"
glib:get-type="gst_webrtc_rtp_transceiver_get_type"
glib:type-struct="WebRTCRTPTransceiverClass">
<method name="stop" c:identifier="gst_webrtc_rtp_transceiver_stop">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="transceiver" transfer-ownership="none">
<type name="WebRTCRTPTransceiver"
c:type="GstWebRTCRTPTransceiver*"/>
</instance-parameter>
</parameters>
</method>
<property name="mlineindex"
writable="1"
construct-only="1"

7
gstreamer-webrtc-sys/src/lib.rs
View file

@ -362,8 +362,6 @@ extern "C" {
//=========================================================================
pub fn gst_webrtc_rtp_receiver_get_type() -> GType;
pub fn gst_webrtc_rtp_receiver_new() -> *mut GstWebRTCRTPReceiver;
pub fn gst_webrtc_rtp_receiver_get_parameters(receiver: *mut GstWebRTCRTPReceiver, kind: *mut c_char) -> *mut gst::GstStructure;
pub fn gst_webrtc_rtp_receiver_set_parameters(receiver: *mut GstWebRTCRTPReceiver, parameters: *mut gst::GstStructure) -> gboolean;
pub fn gst_webrtc_rtp_receiver_set_rtcp_transport(receiver: *mut GstWebRTCRTPReceiver, transport: *mut GstWebRTCDTLSTransport);
pub fn gst_webrtc_rtp_receiver_set_transport(receiver: *mut GstWebRTCRTPReceiver, transport: *mut GstWebRTCDTLSTransport);
@ -371,9 +369,7 @@ extern "C" {
// GstWebRTCRTPSender
//=========================================================================
pub fn gst_webrtc_rtp_sender_get_type() -> GType;
pub fn gst_webrtc_rtp_sender_new(send_encodings: *mut glib::GArray) -> *mut GstWebRTCRTPSender;
pub fn gst_webrtc_rtp_sender_get_parameters(sender: *mut GstWebRTCRTPSender, kind: *mut c_char) -> *mut gst::GstStructure;
pub fn gst_webrtc_rtp_sender_set_parameters(sender: *mut GstWebRTCRTPSender, parameters: *mut gst::GstStructure) -> gboolean;
pub fn gst_webrtc_rtp_sender_new() -> *mut GstWebRTCRTPSender;
pub fn gst_webrtc_rtp_sender_set_rtcp_transport(sender: *mut GstWebRTCRTPSender, transport: *mut GstWebRTCDTLSTransport);
pub fn gst_webrtc_rtp_sender_set_transport(sender: *mut GstWebRTCRTPSender, transport: *mut GstWebRTCDTLSTransport);
@ -381,7 +377,6 @@ extern "C" {
// GstWebRTCRTPTransceiver
//=========================================================================
pub fn gst_webrtc_rtp_transceiver_get_type() -> GType;
pub fn gst_webrtc_rtp_transceiver_stop(transceiver: *mut GstWebRTCRTPTransceiver);
//=========================================================================
// Other functions