mirror of
https://gitlab.gnome.org/GNOME/calls.git
synced 2025-01-06 11:35:32 +00:00
sip: codestyle changes
Shuffle the code around and make use of docstrings to conform to the newly introduced coding style as described in `HACKING.md` This commit also introduces docstrings describing each source file.
This commit is contained in:
Evangelos Ribeiro Tzaras
parent
765cd2ebb9
commit
7ed1ee2502
5 changed files with 135 additions and 82 deletions
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@ -24,17 +24,34 @@
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#define G_LOG_DOMAIN "CallsSipCall"
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#include "calls-sip-call.h"
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#include "calls-call.h"
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#include "calls-message-source.h"
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#include "calls-sip-call.h"
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#include "calls-sip-media-manager.h"
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#include "calls-sip-media-pipeline.h"
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#include "calls-sip-util.h"
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#include "calls-call.h"
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#include <glib/gi18n.h>
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#include <sofia-sip/nua.h>
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/**
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* SECTION:sip-call
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* @short_description: A #CallsCall for the SIP protocol
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* @Title: CallsSipCall
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*
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* #CallsSipCall derives from #CallsCall. Apart from allowing call control
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* like answering and hanging up it also coordinates with #CallsSipMediaManager
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* to prepare and control appropriate #CallsSipMediaPipeline objects.
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*/
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enum {
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PROP_0,
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PROP_CALL_HANDLE,
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PROP_LAST_PROP
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};
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static GParamSpec *props[PROP_LAST_PROP];
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struct _CallsSipCall
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{
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@ -62,14 +79,6 @@ G_DEFINE_TYPE_WITH_CODE (CallsSipCall, calls_sip_call, CALLS_TYPE_CALL,
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G_IMPLEMENT_INTERFACE (CALLS_TYPE_MESSAGE_SOURCE,
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calls_sip_call_message_source_interface_init))
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enum {
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PROP_0,
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PROP_CALL_HANDLE,
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PROP_LAST_PROP
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};
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static GParamSpec *props[PROP_LAST_PROP];
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static gboolean
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try_setting_up_media_pipeline (CallsSipCall *self)
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{
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@ -115,6 +124,7 @@ calls_sip_call_get_number (CallsCall *call)
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return self->number;
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}
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static CallsCallState
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calls_sip_call_get_state (CallsCall *call)
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{
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@ -123,6 +133,7 @@ calls_sip_call_get_state (CallsCall *call)
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return self->state;
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}
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static gboolean
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calls_sip_call_get_inbound (CallsCall *call)
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{
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@ -131,6 +142,7 @@ calls_sip_call_get_inbound (CallsCall *call)
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return self->inbound;
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}
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static void
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calls_sip_call_answer (CallsCall *call)
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{
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@ -169,6 +181,7 @@ calls_sip_call_answer (CallsCall *call)
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calls_sip_call_set_state (self, CALLS_CALL_STATE_ACTIVE);
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}
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static void
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calls_sip_call_hang_up (CallsCall *call)
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{
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@ -205,6 +218,7 @@ calls_sip_call_hang_up (CallsCall *call)
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}
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}
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static void
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calls_sip_call_set_property (GObject *object,
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guint property_id,
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@ -287,6 +301,7 @@ calls_sip_call_class_init (CallsSipCallClass *klass)
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g_object_class_install_property (object_class, PROP_CALL_HANDLE, props[PROP_CALL_HANDLE]);
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}
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static void
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calls_sip_call_message_source_interface_init (CallsMessageSourceInterface *iface)
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{
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@ -24,12 +24,21 @@
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#define G_LOG_DOMAIN "CallsSipMediaManager"
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#include "calls-sip-media-pipeline.h"
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#include "gst-rfc3551.h"
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#include "calls-sip-media-manager.h"
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#include "gst-rfc3551.h"
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#include <gst/gst.h>
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/**
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* SECTION:sip-media-manager
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* @short_description: The media manager singleton
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* @Title: CallsSipMediaManager
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*
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* #CallsSipMediaManager is mainly responsible for generating appropriate
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* SDP messages for the set of supported codecs. In the future it
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* shall also manage the #CallsSipMediaPipeline objects that are in use.
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*/
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typedef struct _CallsSipMediaManager
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{
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GObject parent;
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@ -29,6 +29,35 @@
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#include <gst/gst.h>
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#include <gio/gio.h>
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/**
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* SECTION:sip-media-pipeline
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* @short_description:
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* @Title:
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*
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* #CallsSipMediaPipeline is responsible for building Gstreamer pipelines.
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* Usually a sender and receiver pipeline is employed.
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*
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* The sender pipeline records audio and uses RTP to send it out over the network
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* to the specified host.
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* The receiver pipeline receives RTP from the network and plays the audio
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* on the system.
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*
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* Both pipelines are using RTCP.
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*/
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enum {
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PROP_0,
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PROP_CODEC,
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PROP_REMOTE,
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PROP_LPORT_RTP,
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PROP_RPORT_RTP,
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PROP_LPORT_RTCP,
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PROP_RPORT_RTCP,
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PROP_DEBUG,
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PROP_LAST_PROP,
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};
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static GParamSpec *props[PROP_LAST_PROP];
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struct _CallsSipMediaPipeline {
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GObject parent;
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@ -78,24 +107,10 @@ static void initable_iface_init (GInitableIface *iface);
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G_DEFINE_TYPE_WITH_CODE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT,
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G_IMPLEMENT_INTERFACE (G_TYPE_INITABLE, initable_iface_init));
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enum {
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PROP_0,
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PROP_CODEC,
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PROP_REMOTE,
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PROP_LPORT_RTP,
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PROP_RPORT_RTP,
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PROP_LPORT_RTCP,
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PROP_RPORT_RTCP,
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PROP_DEBUG,
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PROP_LAST_PROP,
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};
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static GParamSpec *props[PROP_LAST_PROP];
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/* rtpbin adds a pad once the payload is verified */
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static void
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pad_added_cb (GstElement *rtpbin,
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GstPad *srcpad,
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on_pad_added (GstElement *rtpbin,
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GstPad *srcpad,
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GstElement *depayloader)
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{
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GstPad *sinkpad;
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@ -113,9 +128,9 @@ pad_added_cb (GstElement *rtpbin,
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static gboolean
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bus_cb (GstBus *bus,
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GstMessage *message,
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gpointer data)
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on_bus_message (GstBus *bus,
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GstMessage *message,
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gpointer data)
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{
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CallsSipMediaPipeline *pipeline = CALLS_SIP_MEDIA_PIPELINE (data);
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@ -426,8 +441,8 @@ initable_init (GInitable *initable,
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/* get the busses and establish watches */
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self->bus_send = gst_pipeline_get_bus (GST_PIPELINE (self->send_pipeline));
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self->bus_recv = gst_pipeline_get_bus (GST_PIPELINE (self->recv_pipeline));
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self->bus_watch_send = gst_bus_add_watch (self->bus_send, bus_cb, self);
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self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, bus_cb, self);
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self->bus_watch_send = gst_bus_add_watch (self->bus_send, on_bus_message, self);
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self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, on_bus_message, self);
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gst_bin_add_many (GST_BIN (self->recv_pipeline), self->depayloader, self->decoder,
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self->audiosink, NULL);
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@ -568,7 +583,7 @@ initable_init (GInitable *initable,
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gst_object_unref (sinkpad);
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/* need to link RTP pad to the depayloader */
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g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (pad_added_cb), self->depayloader);
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g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
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/* out/send direction */
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@ -24,24 +24,53 @@
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#define G_LOG_DOMAIN "CallsSipOrigin"
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#include "config.h"
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#include "calls-sip-origin.h"
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#include "calls-message-source.h"
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#include "calls-origin.h"
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#include "calls-sip-call.h"
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#include "calls-sip-util.h"
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#include "calls-sip-enums.h"
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#include "calls-sip-origin.h"
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#include "calls-sip-util.h"
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#include "calls-sip-media-manager.h"
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#include "config.h"
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#include <glib/gi18n.h>
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#include <glib-object.h>
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#include <sofia-sip/nua.h>
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#include <sofia-sip/su_tag.h>
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#include <sofia-sip/su_tag_io.h>
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#include <sofia-sip/sip_util.h>
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#include <sofia-sip/sdp.h>
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/**
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* SECTION:sip-origin
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* @short_description: A #CallsOrigin for the SIP protocol
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* @Title: CallsSipOrigin
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*
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* #CallsSipOrigin implements the #CallsOriginInterface and is mainly
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* responsible for managing the sofia-sip callbacks, keeping track of #CallsSipCall
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* objects and coordinating with #CallsSipMediaManager.
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*/
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enum {
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PROP_0,
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PROP_NAME,
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PROP_ACC_USER,
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PROP_ACC_PASSWORD,
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PROP_ACC_HOST,
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PROP_ACC_PORT,
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PROP_ACC_PROTOCOL,
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PROP_ACC_DIRECT,
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PROP_ACC_AUTO_CONNECT,
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PROP_SIP_CONTEXT,
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PROP_SIP_LOCAL_PORT,
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PROP_ACC_STATE,
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PROP_CALLS,
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PROP_COUNTRY_CODE,
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PROP_LAST_PROP,
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};
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static GParamSpec *props[PROP_LAST_PROP];
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struct _CallsSipOrigin
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{
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@ -90,26 +119,6 @@ G_DEFINE_TYPE_WITH_CODE (CallsSipOrigin, calls_sip_origin, G_TYPE_OBJECT,
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G_IMPLEMENT_INTERFACE (CALLS_TYPE_ORIGIN,
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calls_sip_origin_origin_interface_init))
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enum {
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PROP_0,
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PROP_NAME,
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PROP_ACC_USER,
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PROP_ACC_PASSWORD,
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PROP_ACC_HOST,
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PROP_ACC_PORT,
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PROP_ACC_PROTOCOL,
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PROP_ACC_DIRECT,
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PROP_ACC_AUTO_CONNECT,
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PROP_SIP_CONTEXT,
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PROP_SIP_LOCAL_PORT,
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PROP_ACC_STATE,
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PROP_CALLS,
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PROP_COUNTRY_CODE,
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PROP_LAST_PROP,
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};
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static GParamSpec *props[PROP_LAST_PROP];
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static void
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remove_call (CallsSipOrigin *self,
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CallsCall *call,
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@ -166,10 +175,10 @@ remove_calls (CallsSipOrigin *self,
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static void
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on_call_state_changed_cb (CallsSipOrigin *self,
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CallsCallState new_state,
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CallsCallState old_state,
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CallsCall *call)
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on_call_state_changed (CallsSipOrigin *self,
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CallsCallState new_state,
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CallsCallState old_state,
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CallsCall *call)
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{
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g_assert (CALLS_IS_SIP_ORIGIN (self));
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g_assert (CALLS_IS_CALL (call));
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@ -211,7 +220,7 @@ add_call (CallsSipOrigin *self,
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g_signal_emit_by_name (CALLS_ORIGIN (self), "call-added", call);
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g_signal_connect_swapped (call, "state-changed",
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G_CALLBACK (on_call_state_changed_cb),
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G_CALLBACK (on_call_state_changed),
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self);
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if (!inbound) {
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@ -1166,7 +1175,7 @@ calls_sip_origin_new (const gchar *name,
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/* calls_sip_origin_go_online:
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* @self: A #CallsSipOrigin
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* @enabled: Whether to go online or offline
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* @enabled: %TRUE to go online, %FALSE to go offline
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*/
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void
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calls_sip_origin_go_online (CallsSipOrigin *self,
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@ -24,23 +24,35 @@
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#define G_LOG_DOMAIN "CallsSipProvider"
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#define SU_ROOT_MAGIC_T CallsSipProvider
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#include "calls-sip-provider.h"
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#define SIP_ACCOUNT_FILE "sip-account.cfg"
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#include "calls-message-source.h"
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#include "calls-provider.h"
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#include "calls-sip-origin.h"
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#include "calls-sip-util.h"
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#include "calls-sip-enums.h"
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#include "calls-sip-origin.h"
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#include "calls-sip-provider.h"
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#include "calls-sip-util.h"
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#include "config.h"
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#include <libpeas/peas.h>
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#include <sofia-sip/nua.h>
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#include <sofia-sip/su_glib.h>
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/**
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* SECTION:sip-provider
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* @short_description: A #CallsProvider for the SIP protocol
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* @Title: CallsSipProvider
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*
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* #CallsSipProvider is derived from #CallsProvider and is responsible
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* for setting up the sofia-sip stack.
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*/
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#define SIP_ACCOUNT_FILE "sip-account.cfg"
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enum {
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PROP_0,
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PROP_SIP_STATE,
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PROP_LAST_PROP,
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};
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static GParamSpec *props[PROP_LAST_PROP];
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struct _CallsSipProvider
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{
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@ -54,13 +66,6 @@ struct _CallsSipProvider
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gchar *filename;
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};
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enum {
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PROP_0,
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PROP_SIP_STATE,
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PROP_LAST_PROP,
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};
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static GParamSpec *props[PROP_LAST_PROP];
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static void calls_sip_provider_message_source_interface_init (CallsMessageSourceInterface *iface);
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@ -339,6 +344,12 @@ calls_sip_provider_dispose (GObject *object)
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}
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static void
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calls_sip_provider_class_finalize (CallsSipProviderClass *klass)
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{
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}
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static void
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calls_sip_provider_class_init (CallsSipProviderClass *klass)
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{
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@ -430,12 +441,6 @@ calls_sip_provider_new ()
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}
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static void
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calls_sip_provider_class_finalize (CallsSipProviderClass *klass)
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{
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}
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G_MODULE_EXPORT void
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peas_register_types (PeasObjectModule *module)
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{
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