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sip: call: Defer setting up codecs for pipeline until activated

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This let's us get rid of some ugly code
This commit is contained in:
Evangelos Ribeiro Tzaras 2022-05-06 00:55:39 +02:00
parent 5a15638db3
commit 96f1cc0a30
3 changed files with 16 additions and 47 deletions
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60
plugins/sip/calls-sip-call.c
View file

@ -78,35 +78,6 @@ 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))
static gboolean
try_setting_up_media_pipeline (CallsSipCall *self)
{
g_assert (CALLS_SIP_CALL (self));
if (!self->codecs)
return FALSE;
if (calls_sip_media_pipeline_get_state (self->pipeline) ==
CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC) {
MediaCodecInfo *codec = (MediaCodecInfo *) self->codecs->data;
g_debug ("Setting codec '%s' for pipeline", codec->name);
calls_sip_media_pipeline_set_codec (self->pipeline, codec);
}
g_debug ("Setting remote ports: RTP/RTCP %u/%u",
self->rport_rtp, self->rport_rtcp);
g_object_set (self->pipeline,
"remote", self->remote,
"rport-rtp", self->rport_rtp,
"rport-rtcp", self->rport_rtcp,
NULL);
return TRUE;
}
static void
calls_sip_call_answer (CallsCall *call)
{
@ -129,8 +100,6 @@ calls_sip_call_answer (CallsCall *call)
rtp_port = calls_sip_media_pipeline_get_rtp_port (self->pipeline);
rtcp_port = calls_sip_media_pipeline_get_rtcp_port (self->pipeline);
calls_sip_call_setup_local_media_connection (self);
local_sdp = calls_sip_media_manager_get_capabilities (self->manager,
self->ip,
rtp_port,
@ -302,18 +271,6 @@ calls_sip_call_init (CallsSipCall *self)
self->manager = calls_sip_media_manager_default ();
}
/**
* calls_sip_call_setup_local_media_connection:
* @self: A #CallsSipCall
*/
void
calls_sip_call_setup_local_media_connection (CallsSipCall *self)
{
g_return_if_fail (CALLS_IS_SIP_CALL (self));
/* XXX maybe we can get rid of this completely */
try_setting_up_media_pipeline (self);
}
/**
* calls_sip_call_setup_remote_media_connection:
@ -335,7 +292,14 @@ calls_sip_call_setup_remote_media_connection (CallsSipCall *self,
self->rport_rtp = port_rtp;
self->rport_rtcp = port_rtcp;
try_setting_up_media_pipeline (self);
g_debug ("Setting remote ports: RTP/RTCP %u/%u",
self->rport_rtp, self->rport_rtcp);
g_object_set (self->pipeline,
"remote", self->remote,
"rport-rtp", self->rport_rtp,
"rport-rtcp", self->rport_rtcp,
NULL);
}
/**
@ -357,6 +321,14 @@ calls_sip_call_activate_media (CallsSipCall *self,
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self->pipeline));
if (enabled) {
if (calls_sip_media_pipeline_get_state (self->pipeline) ==
CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC) {
MediaCodecInfo *codec = (MediaCodecInfo *) self->codecs->data;
g_debug ("Setting codec '%s' for pipeline", codec->name);
calls_sip_media_pipeline_set_codec (self->pipeline, codec);
}
calls_sip_media_pipeline_start (self->pipeline);
} else {
calls_sip_media_pipeline_stop (self->pipeline);

1
plugins/sip/calls-sip-call.h
View file

@ -45,7 +45,6 @@ void calls_sip_call_setup_remote_media_connection (CallsSi
const char *remote,
guint port_rtp,
guint port_rtcp);
void calls_sip_call_setup_local_media_connection (CallsSipCall *self);
void calls_sip_call_activate_media (CallsSipCall *self,
gboolean enabled);
void calls_sip_call_set_state (CallsSipCall *self,

2
plugins/sip/calls-sip-origin.c
View file

@ -272,8 +272,6 @@ add_call (CallsSipOrigin *self,
self);
if (!inbound) {
calls_sip_call_setup_local_media_connection (sip_call);
local_sdp = calls_sip_media_manager_static_capabilities (self->media_manager,
self->own_ip,
rtp_port,