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sip: codestyle changes

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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 2021-04-11 00:25:13 +02:00
parent 765cd2ebb9
commit 7ed1ee2502
5 changed files with 135 additions and 82 deletions
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35
plugins/sip/calls-sip-call.c
View file

@ -24,17 +24,34 @@
#define G_LOG_DOMAIN "CallsSipCall"
#include "calls-sip-call.h"
#include "calls-call.h"
#include "calls-message-source.h"
#include "calls-sip-call.h"
#include "calls-sip-media-manager.h"
#include "calls-sip-media-pipeline.h"
#include "calls-sip-util.h"
#include "calls-call.h"
#include <glib/gi18n.h>
#include <sofia-sip/nua.h>
/**
* SECTION:sip-call
* @short_description: A #CallsCall for the SIP protocol
* @Title: CallsSipCall
*
* #CallsSipCall derives from #CallsCall. Apart from allowing call control
* like answering and hanging up it also coordinates with #CallsSipMediaManager
* to prepare and control appropriate #CallsSipMediaPipeline objects.
*/
enum {
PROP_0,
PROP_CALL_HANDLE,
PROP_LAST_PROP
};
static GParamSpec *props[PROP_LAST_PROP];
struct _CallsSipCall
{
@ -62,14 +79,6 @@ G_DEFINE_TYPE_WITH_CODE (CallsSipCall, calls_sip_call, CALLS_TYPE_CALL,
G_IMPLEMENT_INTERFACE (CALLS_TYPE_MESSAGE_SOURCE,
calls_sip_call_message_source_interface_init))
enum {
PROP_0,
PROP_CALL_HANDLE,
PROP_LAST_PROP
};
static GParamSpec *props[PROP_LAST_PROP];
static gboolean
try_setting_up_media_pipeline (CallsSipCall *self)
{
@ -115,6 +124,7 @@ calls_sip_call_get_number (CallsCall *call)
return self->number;
}
static CallsCallState
calls_sip_call_get_state (CallsCall *call)
{
@ -123,6 +133,7 @@ calls_sip_call_get_state (CallsCall *call)
return self->state;
}
static gboolean
calls_sip_call_get_inbound (CallsCall *call)
{
@ -131,6 +142,7 @@ calls_sip_call_get_inbound (CallsCall *call)
return self->inbound;
}
static void
calls_sip_call_answer (CallsCall *call)
{
@ -169,6 +181,7 @@ calls_sip_call_answer (CallsCall *call)
calls_sip_call_set_state (self, CALLS_CALL_STATE_ACTIVE);
}
static void
calls_sip_call_hang_up (CallsCall *call)
{
@ -205,6 +218,7 @@ calls_sip_call_hang_up (CallsCall *call)
}
}
static void
calls_sip_call_set_property (GObject *object,
guint property_id,
@ -287,6 +301,7 @@ calls_sip_call_class_init (CallsSipCallClass *klass)
g_object_class_install_property (object_class, PROP_CALL_HANDLE, props[PROP_CALL_HANDLE]);
}
static void
calls_sip_call_message_source_interface_init (CallsMessageSourceInterface *iface)
{

13
plugins/sip/calls-sip-media-manager.c
View file

@ -24,12 +24,21 @@
#define G_LOG_DOMAIN "CallsSipMediaManager"
#include "calls-sip-media-pipeline.h"
#include "gst-rfc3551.h"
#include "calls-sip-media-manager.h"
#include "gst-rfc3551.h"
#include <gst/gst.h>
/**
* SECTION:sip-media-manager
* @short_description: The media manager singleton
* @Title: CallsSipMediaManager
*
* #CallsSipMediaManager is mainly responsible for generating appropriate
* SDP messages for the set of supported codecs. In the future it
* shall also manage the #CallsSipMediaPipeline objects that are in use.
*/
typedef struct _CallsSipMediaManager
{
GObject parent;

53
plugins/sip/calls-sip-media-pipeline.c
View file

@ -29,6 +29,35 @@
#include <gst/gst.h>
#include <gio/gio.h>
/**
* SECTION:sip-media-pipeline
* @short_description:
* @Title:
*
* #CallsSipMediaPipeline is responsible for building Gstreamer pipelines.
* Usually a sender and receiver pipeline is employed.
*
* The sender pipeline records audio and uses RTP to send it out over the network
* to the specified host.
* The receiver pipeline receives RTP from the network and plays the audio
* on the system.
*
* Both pipelines are using RTCP.
*/
enum {
PROP_0,
PROP_CODEC,
PROP_REMOTE,
PROP_LPORT_RTP,
PROP_RPORT_RTP,
PROP_LPORT_RTCP,
PROP_RPORT_RTCP,
PROP_DEBUG,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
struct _CallsSipMediaPipeline {
GObject parent;
@ -78,23 +107,9 @@ static void initable_iface_init (GInitableIface *iface);
G_DEFINE_TYPE_WITH_CODE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT,
G_IMPLEMENT_INTERFACE (G_TYPE_INITABLE, initable_iface_init));
enum {
PROP_0,
PROP_CODEC,
PROP_REMOTE,
PROP_LPORT_RTP,
PROP_RPORT_RTP,
PROP_LPORT_RTCP,
PROP_RPORT_RTCP,
PROP_DEBUG,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
/* rtpbin adds a pad once the payload is verified */
static void
pad_added_cb (GstElement *rtpbin,
on_pad_added (GstElement *rtpbin,
GstPad *srcpad,
GstElement *depayloader)
{
@ -113,7 +128,7 @@ pad_added_cb (GstElement *rtpbin,
static gboolean
bus_cb (GstBus *bus,
on_bus_message (GstBus *bus,
GstMessage *message,
gpointer data)
{
@ -426,8 +441,8 @@ initable_init (GInitable *initable,
/* get the busses and establish watches */
self->bus_send = gst_pipeline_get_bus (GST_PIPELINE (self->send_pipeline));
self->bus_recv = gst_pipeline_get_bus (GST_PIPELINE (self->recv_pipeline));
self->bus_watch_send = gst_bus_add_watch (self->bus_send, bus_cb, self);
self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, bus_cb, self);
self->bus_watch_send = gst_bus_add_watch (self->bus_send, on_bus_message, self);
self->bus_watch_recv = gst_bus_add_watch (self->bus_recv, on_bus_message, self);
gst_bin_add_many (GST_BIN (self->recv_pipeline), self->depayloader, self->decoder,
self->audiosink, NULL);
@ -568,7 +583,7 @@ initable_init (GInitable *initable,
gst_object_unref (sinkpad);
/* need to link RTP pad to the depayloader */
g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (pad_added_cb), self->depayloader);
g_signal_connect (self->recv_rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
/* out/send direction */

61
plugins/sip/calls-sip-origin.c
View file

@ -24,24 +24,53 @@
#define G_LOG_DOMAIN "CallsSipOrigin"
#include "config.h"
#include "calls-sip-origin.h"
#include "calls-message-source.h"
#include "calls-origin.h"
#include "calls-sip-call.h"
#include "calls-sip-util.h"
#include "calls-sip-enums.h"
#include "calls-sip-origin.h"
#include "calls-sip-util.h"
#include "calls-sip-media-manager.h"
#include "config.h"
#include <glib/gi18n.h>
#include <glib-object.h>
#include <sofia-sip/nua.h>
#include <sofia-sip/su_tag.h>
#include <sofia-sip/su_tag_io.h>
#include <sofia-sip/sip_util.h>
#include <sofia-sip/sdp.h>
/**
* SECTION:sip-origin
* @short_description: A #CallsOrigin for the SIP protocol
* @Title: CallsSipOrigin
*
* #CallsSipOrigin implements the #CallsOriginInterface and is mainly
* responsible for managing the sofia-sip callbacks, keeping track of #CallsSipCall
* objects and coordinating with #CallsSipMediaManager.
*/
enum {
PROP_0,
PROP_NAME,
PROP_ACC_USER,
PROP_ACC_PASSWORD,
PROP_ACC_HOST,
PROP_ACC_PORT,
PROP_ACC_PROTOCOL,
PROP_ACC_DIRECT,
PROP_ACC_AUTO_CONNECT,
PROP_SIP_CONTEXT,
PROP_SIP_LOCAL_PORT,
PROP_ACC_STATE,
PROP_CALLS,
PROP_COUNTRY_CODE,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
struct _CallsSipOrigin
{
@ -90,26 +119,6 @@ G_DEFINE_TYPE_WITH_CODE (CallsSipOrigin, calls_sip_origin, G_TYPE_OBJECT,
G_IMPLEMENT_INTERFACE (CALLS_TYPE_ORIGIN,
calls_sip_origin_origin_interface_init))
enum {
PROP_0,
PROP_NAME,
PROP_ACC_USER,
PROP_ACC_PASSWORD,
PROP_ACC_HOST,
PROP_ACC_PORT,
PROP_ACC_PROTOCOL,
PROP_ACC_DIRECT,
PROP_ACC_AUTO_CONNECT,
PROP_SIP_CONTEXT,
PROP_SIP_LOCAL_PORT,
PROP_ACC_STATE,
PROP_CALLS,
PROP_COUNTRY_CODE,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
static void
remove_call (CallsSipOrigin *self,
CallsCall *call,
@ -166,7 +175,7 @@ remove_calls (CallsSipOrigin *self,
static void
on_call_state_changed_cb (CallsSipOrigin *self,
on_call_state_changed (CallsSipOrigin *self,
CallsCallState new_state,
CallsCallState old_state,
CallsCall *call)
@ -211,7 +220,7 @@ add_call (CallsSipOrigin *self,
g_signal_emit_by_name (CALLS_ORIGIN (self), "call-added", call);
g_signal_connect_swapped (call, "state-changed",
G_CALLBACK (on_call_state_changed_cb),
G_CALLBACK (on_call_state_changed),
self);
if (!inbound) {
@ -1166,7 +1175,7 @@ calls_sip_origin_new (const gchar *name,
/* calls_sip_origin_go_online:
* @self: A #CallsSipOrigin
* @enabled: Whether to go online or offline
* @enabled: %TRUE to go online, %FALSE to go offline
*/
void
calls_sip_origin_go_online (CallsSipOrigin *self,

43
plugins/sip/calls-sip-provider.c
View file

@ -24,23 +24,35 @@
#define G_LOG_DOMAIN "CallsSipProvider"
#define SU_ROOT_MAGIC_T CallsSipProvider
#include "calls-sip-provider.h"
#define SIP_ACCOUNT_FILE "sip-account.cfg"
#include "calls-message-source.h"
#include "calls-provider.h"
#include "calls-sip-origin.h"
#include "calls-sip-util.h"
#include "calls-sip-enums.h"
#include "calls-sip-origin.h"
#include "calls-sip-provider.h"
#include "calls-sip-util.h"
#include "config.h"
#include <libpeas/peas.h>
#include <sofia-sip/nua.h>
#include <sofia-sip/su_glib.h>
/**
* SECTION:sip-provider
* @short_description: A #CallsProvider for the SIP protocol
* @Title: CallsSipProvider
*
* #CallsSipProvider is derived from #CallsProvider and is responsible
* for setting up the sofia-sip stack.
*/
#define SIP_ACCOUNT_FILE "sip-account.cfg"
enum {
PROP_0,
PROP_SIP_STATE,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
struct _CallsSipProvider
{
@ -54,13 +66,6 @@ struct _CallsSipProvider
gchar *filename;
};
enum {
PROP_0,
PROP_SIP_STATE,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
static void calls_sip_provider_message_source_interface_init (CallsMessageSourceInterface *iface);
@ -339,6 +344,12 @@ calls_sip_provider_dispose (GObject *object)
}
static void
calls_sip_provider_class_finalize (CallsSipProviderClass *klass)
{
}
static void
calls_sip_provider_class_init (CallsSipProviderClass *klass)
{
@ -430,12 +441,6 @@ calls_sip_provider_new ()
}
static void
calls_sip_provider_class_finalize (CallsSipProviderClass *klass)
{
}
G_MODULE_EXPORT void
peas_register_types (PeasObjectModule *module)
{