gstreamer-webrtc: Update manual code to 2018 edition

gstreamer-webrtc/Cargo.toml

@@ -11,16 +11,14 @@ homepage = "https://gstreamer.freedesktop.org"
 documentation = "https://gstreamer.pages.freedesktop.org/gstreamer-rs/gstreamer_webrtc/"
 keywords = ["gstreamer", "multimedia", "audio", "video", "gnome"]
 build = "build.rs"
+edition = "2018"
 
 [dependencies]
 libc = "0.2"
-glib-sys = { git = "https://github.com/gtk-rs/gtk-rs" }
+ffi = { package = "gstreamer-webrtc-sys", path = "../gstreamer-webrtc/sys" }
-gobject-sys = { git = "https://github.com/gtk-rs/gtk-rs" }
-gstreamer-sys = { path = "../gstreamer/sys", features = ["v1_8"] }
-gstreamer-webrtc-sys = { path = "../gstreamer-webrtc/sys" }
 glib = { git = "https://github.com/gtk-rs/gtk-rs" }
-gstreamer = { path = "../gstreamer", features = ["v1_14"] }
+gst = { package = "gstreamer", path = "../gstreamer", features = ["v1_14"] }
-gstreamer-sdp = { path = "../gstreamer-sdp", features = ["v1_14"] }
+gst-sdp = { package = "gstreamer-sdp", path = "../gstreamer-sdp", features = ["v1_14"] }
 
 [build-dependencies]
 gstreamer-rs-lgpl-docs = { path = "../docs", optional = true }
@@ -29,9 +27,9 @@ gstreamer-rs-lgpl-docs = { path = "../docs", optional = true }
 default = []
 embed-lgpl-docs = ["gstreamer-rs-lgpl-docs"]
 purge-lgpl-docs = ["gstreamer-rs-lgpl-docs"]
-dox = ["v1_18", "gstreamer-webrtc-sys/dox", "glib/dox", "gstreamer/dox", "gstreamer-sdp/dox"]
+dox = ["v1_18", "ffi/dox", "glib/dox", "gst/dox", "gst-sdp/dox"]
-v1_16 = ["gstreamer/v1_16", "gstreamer-sdp/v1_16", "gstreamer-webrtc-sys/v1_16"]
+v1_16 = ["gst/v1_16", "gst-sdp/v1_16", "ffi/v1_16"]
-v1_18 = ["gstreamer/v1_18", "gstreamer-sdp/v1_18", "gstreamer-webrtc-sys/v1_18", "v1_16"]
+v1_18 = ["gst/v1_18", "gst-sdp/v1_18", "ffi/v1_18", "v1_16"]
 
 [package.metadata.docs.rs]
 features = ["dox", "embed-lgpl-docs"]

gstreamer-webrtc/src/auto/flags.rs

@@ -2,6 +2,6 @@
 // from gir-files (https://github.com/gtk-rs/gir-files)
 // DO NOT EDIT
 
-use gst_web_rtc_sys;
+use ffi;
 use glib::translate::*;
 

gstreamer-webrtc/src/lib.rs

@@ -8,20 +8,11 @@
 
 #![cfg_attr(feature = "dox", feature(doc_cfg))]
 
-extern crate libc;
+pub use ffi;
-
-#[macro_use]
-extern crate glib;
-extern crate glib_sys;
-extern crate gobject_sys;
-extern crate gstreamer as gst;
-extern crate gstreamer_sdp as gst_sdp;
-extern crate gstreamer_sys as gst_sys;
-extern crate gstreamer_webrtc_sys as gst_web_rtc_sys;
 
 macro_rules! assert_initialized_main_thread {
     () => {
-        if unsafe { ::gst_sys::gst_is_initialized() } != ::glib_sys::GTRUE {
+        if unsafe { gst::ffi::gst_is_initialized() } != glib::ffi::GTRUE {
             panic!("GStreamer has not been initialized. Call `gst::init` first.");
         }
     };
@@ -35,7 +26,7 @@ macro_rules! skip_assert_initialized {
 #[allow(clippy::too_many_arguments)]
 #[allow(clippy::match_same_arms)]
 mod auto;
-pub use auto::*;
+pub use crate::auto::*;
 
 #[cfg(any(feature = "v1_18", feature = "dox"))]
 #[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
@@ -48,5 +39,5 @@ pub mod prelude {
     pub use glib::prelude::*;
     pub use gst::prelude::*;
 
-    pub use auto::traits::*;
+    pub use crate::auto::traits::*;
 }

gstreamer-webrtc/src/web_rtc_data_channel.rs

@@ -6,9 +6,8 @@
 // option. This file may not be copied, modified, or distributed
 // except according to those terms.
 
+use crate::WebRTCDataChannel;
 use glib::translate::*;
-use gst_web_rtc_sys;
-use WebRTCDataChannel;
 
 use std::mem;
 
@@ -16,7 +15,7 @@ impl WebRTCDataChannel {
     pub fn on_error(&self, error: glib::Error) {
         let error = mem::ManuallyDrop::new(error);
         unsafe {
-            gst_web_rtc_sys::gst_webrtc_data_channel_on_error(
+            ffi::gst_webrtc_data_channel_on_error(
                 self.to_glib_none().0,
                 mut_override(error.to_glib_none().0),
             );

gstreamer-webrtc/src/web_rtc_session_description.rs

@@ -6,26 +6,24 @@
 // option. This file may not be copied, modified, or distributed
 // except according to those terms.
 
+use crate::WebRTCSDPType;
+use crate::WebRTCSessionDescription;
 use glib::translate::*;
-use gst_sdp;
-use gst_web_rtc_sys;
 use std::mem;
-use WebRTCSDPType;
-use WebRTCSessionDescription;
 
 impl WebRTCSessionDescription {
     pub fn new(type_: WebRTCSDPType, sdp: gst_sdp::SDPMessage) -> WebRTCSessionDescription {
         assert_initialized_main_thread!();
         unsafe {
             let mut sdp = mem::ManuallyDrop::new(sdp);
-            from_glib_full(gst_web_rtc_sys::gst_webrtc_session_description_new(
+            from_glib_full(ffi::gst_webrtc_session_description_new(
                 type_.to_glib(),
                 sdp.to_glib_none_mut().0,
             ))
         }
     }
 
-    pub fn get_type(&self) -> ::WebRTCSDPType {
+    pub fn get_type(&self) -> crate::WebRTCSDPType {
         unsafe { from_glib((*self.to_glib_none().0).type_) }
     }