[farsight2/master] Re-indent FsRtpSession according to new rules
Olivier Crête
olivier.crete at collabora.co.uk
Tue Dec 23 15:22:58 PST 2008
---
gst/fsrtpconference/fs-rtp-session.c | 219 ++++++++++++++++++++++------------
1 files changed, 143 insertions(+), 76 deletions(-)
diff --git a/gst/fsrtpconference/fs-rtp-session.c b/gst/fsrtpconference/fs-rtp-session.c
index b5556bf..ac451e3 100644
--- a/gst/fsrtpconference/fs-rtp-session.c
+++ b/gst/fsrtpconference/fs-rtp-session.c
@@ -385,10 +385,9 @@ fs_rtp_session_dispose (GObject *object)
GList *item = NULL;
GstBin *conferencebin = NULL;
- if (self->priv->disposed) {
+ if (self->priv->disposed)
/* If dispose did already run, return. */
return;
- }
conferencebin = GST_BIN (self->priv->conference);
@@ -443,7 +442,8 @@ fs_rtp_session_dispose (GObject *object)
/* Now they should all be stopped, we can remove them in peace */
- if (self->priv->media_sink_pad) {
+ if (self->priv->media_sink_pad)
+ {
gst_pad_set_active (self->priv->media_sink_pad, FALSE);
gst_element_remove_pad (GST_ELEMENT (self->priv->conference),
self->priv->media_sink_pad);
@@ -451,7 +451,8 @@ fs_rtp_session_dispose (GObject *object)
}
- if (self->priv->rtpbin_send_rtcp_src) {
+ if (self->priv->rtpbin_send_rtcp_src)
+ {
gst_pad_set_active (self->priv->rtpbin_send_rtcp_src, FALSE);
gst_element_release_request_pad (self->priv->conference->gstrtpbin,
self->priv->rtpbin_send_rtcp_src);
@@ -459,7 +460,8 @@ fs_rtp_session_dispose (GObject *object)
self->priv->rtpbin_send_rtcp_src = NULL;
}
- if (self->priv->rtpbin_send_rtp_sink) {
+ if (self->priv->rtpbin_send_rtp_sink)
+ {
gst_pad_set_active (self->priv->rtpbin_send_rtp_sink, FALSE);
gst_element_release_request_pad (self->priv->conference->gstrtpbin,
self->priv->rtpbin_send_rtp_sink);
@@ -467,7 +469,8 @@ fs_rtp_session_dispose (GObject *object)
self->priv->rtpbin_send_rtp_sink = NULL;
}
- if (self->priv->rtpbin_recv_rtp_sink) {
+ if (self->priv->rtpbin_recv_rtp_sink)
+ {
gst_pad_set_active (self->priv->rtpbin_recv_rtp_sink, FALSE);
gst_element_release_request_pad (self->priv->conference->gstrtpbin,
self->priv->rtpbin_recv_rtp_sink);
@@ -475,7 +478,8 @@ fs_rtp_session_dispose (GObject *object)
self->priv->rtpbin_recv_rtp_sink = NULL;
}
- if (self->priv->rtpbin_recv_rtcp_sink) {
+ if (self->priv->rtpbin_recv_rtcp_sink)
+ {
gst_pad_set_active (self->priv->rtpbin_recv_rtcp_sink, FALSE);
gst_element_release_request_pad (self->priv->conference->gstrtpbin,
self->priv->rtpbin_recv_rtcp_sink);
@@ -485,7 +489,8 @@ fs_rtp_session_dispose (GObject *object)
- if (self->priv->transmitters) {
+ if (self->priv->transmitters)
+ {
g_hash_table_foreach_remove (self->priv->transmitters, _remove_transmitter,
self);
@@ -493,14 +498,16 @@ fs_rtp_session_dispose (GObject *object)
self->priv->transmitters = NULL;
}
- if (self->priv->free_substreams) {
+ if (self->priv->free_substreams)
+ {
g_list_foreach (self->priv->free_substreams, (GFunc) g_object_unref, NULL);
g_list_free (self->priv->free_substreams);
self->priv->free_substreams = NULL;
}
- if (self->priv->blueprints) {
+ if (self->priv->blueprints)
+ {
fs_rtp_blueprints_unref (self->priv->media_type);
self->priv->blueprints = NULL;
}
@@ -559,7 +566,8 @@ fs_rtp_session_get_property (GObject *object,
{
FsRtpSession *self = FS_RTP_SESSION (object);
- switch (prop_id) {
+ switch (prop_id)
+ {
case PROP_MEDIA_TYPE:
g_value_set_enum (value, self->priv->media_type);
break;
@@ -644,7 +652,8 @@ fs_rtp_session_set_property (GObject *object,
{
FsRtpSession *self = FS_RTP_SESSION (object);
- switch (prop_id) {
+ switch (prop_id)
+ {
case PROP_MEDIA_TYPE:
self->priv->media_type = g_value_get_enum (value);
break;
@@ -683,7 +692,8 @@ fs_rtp_session_constructed (GObject *object)
GstPadLinkReturn ret;
gchar *tmp;
- if (self->id == 0) {
+ if (self->id == 0)
+ {
g_error ("You can no instantiate this element directly, you MUST"
" call fs_rtp_session_new ()");
return;
@@ -692,7 +702,8 @@ fs_rtp_session_constructed (GObject *object)
self->priv->blueprints = fs_rtp_blueprints_get (self->priv->media_type,
&self->priv->construction_error);
- if (!self->priv->blueprints) {
+ if (!self->priv->blueprints)
+ {
if (!self->priv->construction_error)
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_INTERNAL,
@@ -716,14 +727,16 @@ fs_rtp_session_constructed (GObject *object)
valve = gst_element_factory_make ("fsvalve", tmp);
g_free (tmp);
- if (!valve) {
+ if (!valve)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the fsvalve element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), valve)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), valve))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the valve element to the FsRtpConference");
@@ -753,14 +766,16 @@ fs_rtp_session_constructed (GObject *object)
tee = gst_element_factory_make ("tee", tmp);
g_free (tmp);
- if (!tee) {
+ if (!tee)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the tee element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), tee)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), tee))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the tee element to the FsRtpConference");
@@ -789,14 +804,16 @@ fs_rtp_session_constructed (GObject *object)
funnel = gst_element_factory_make ("fsfunnel", tmp);
g_free (tmp);
- if (!funnel) {
+ if (!funnel)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtp funnel element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), funnel)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), funnel))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtp funnel element to the FsRtpConference");
@@ -816,7 +833,8 @@ fs_rtp_session_constructed (GObject *object)
ret = gst_pad_link (funnel_src_pad, self->priv->rtpbin_recv_rtp_sink);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not link pad %s (%p) with pad %s (%p)",
@@ -839,14 +857,16 @@ fs_rtp_session_constructed (GObject *object)
funnel = gst_element_factory_make ("fsfunnel", tmp);
g_free (tmp);
- if (!funnel) {
+ if (!funnel)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtcp funnel element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), funnel)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), funnel))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtcp funnel element to the FsRtcpConference");
@@ -866,7 +886,8 @@ fs_rtp_session_constructed (GObject *object)
ret = gst_pad_link (funnel_src_pad, self->priv->rtpbin_recv_rtcp_sink);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not link pad %s (%p) with pad %s (%p)",
@@ -888,14 +909,16 @@ fs_rtp_session_constructed (GObject *object)
muxer = gst_element_factory_make ("rtpdtmfmux", tmp);
g_free (tmp);
- if (!muxer) {
+ if (!muxer)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtp muxer element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), muxer)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), muxer))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtp muxer element to the FsRtpConference");
@@ -915,7 +938,8 @@ fs_rtp_session_constructed (GObject *object)
ret = gst_pad_link (muxer_src_pad, self->priv->rtpbin_send_rtp_sink);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not link pad %s (%p) with pad %s (%p)",
@@ -938,14 +962,16 @@ fs_rtp_session_constructed (GObject *object)
tee = gst_element_factory_make ("tee", tmp);
g_free (tmp);
- if (!tee) {
+ if (!tee)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtp tee element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), tee)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), tee))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtp tee element to the FsRtpConference");
@@ -960,7 +986,8 @@ fs_rtp_session_constructed (GObject *object)
tmp = g_strdup_printf ("send_rtp_src_%u", self->id);
if (!gst_element_link_pads (
self->priv->conference->gstrtpbin, tmp,
- self->priv->transmitter_rtp_tee, "sink")) {
+ self->priv->transmitter_rtp_tee, "sink"))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not link rtpbin %s pad to tee sink", tmp);
@@ -975,14 +1002,16 @@ fs_rtp_session_constructed (GObject *object)
fakesink = gst_element_factory_make ("fakesink", tmp);
g_free (tmp);
- if (!fakesink) {
+ if (!fakesink)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtp fakesink element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), fakesink)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), fakesink))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtp fakesink element to the FsRtpConference");
@@ -1004,7 +1033,8 @@ fs_rtp_session_constructed (GObject *object)
gst_object_unref (pad2);
gst_object_unref (pad1);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not link the rtp tee to its fakesink");
@@ -1017,14 +1047,16 @@ fs_rtp_session_constructed (GObject *object)
tee = gst_element_factory_make ("tee", tmp);
g_free (tmp);
- if (!tee) {
+ if (!tee)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtcp tee element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), tee)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), tee))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtcp tee element to the FsRtpConference");
@@ -1074,14 +1106,16 @@ fs_rtp_session_constructed (GObject *object)
fakesink = gst_element_factory_make ("fakesink", tmp);
g_free (tmp);
- if (!fakesink) {
+ if (!fakesink)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtcp fakesink element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), fakesink)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), fakesink))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtcp fakesink element to the FsRtcpConference");
@@ -1103,7 +1137,8 @@ fs_rtp_session_constructed (GObject *object)
gst_object_unref (pad2);
gst_object_unref (pad1);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not link the rtcp tee to its fakesink");
@@ -1116,14 +1151,16 @@ fs_rtp_session_constructed (GObject *object)
capsfilter = gst_element_factory_make ("capsfilter", tmp);
g_free (tmp);
- if (!capsfilter) {
+ if (!capsfilter)
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not create the rtp capsfilter element");
return;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), capsfilter)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), capsfilter))
+ {
self->priv->construction_error = g_error_new (FS_ERROR,
FS_ERROR_CONSTRUCTION,
"Could not add the rtp capsfilter element to the FsRtpConference");
@@ -1190,7 +1227,8 @@ fs_rtp_session_new_stream (FsSession *session,
FsStream *new_stream = NULL;
FsStreamTransmitter *st;
- if (!FS_IS_RTP_PARTICIPANT (participant)) {
+ if (!FS_IS_RTP_PARTICIPANT (participant))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_INVALID_ARGUMENTS,
"You have to provide a participant of type RTP");
return NULL;
@@ -1386,7 +1424,8 @@ fs_rtp_session_new (FsMediaType media_type, FsRtpConference *conference,
"id", id,
NULL);
- if (session->priv->construction_error) {
+ if (session->priv->construction_error)
+ {
g_propagate_error (error, session->priv->construction_error);
g_object_unref (session);
return NULL;
@@ -1435,7 +1474,8 @@ _get_request_pad_and_link (GstElement *tee_funnel, const gchar *tee_funnel_name,
requestpad = gst_element_get_request_pad (tee_funnel, requestpad_name);
- if (!requestpad) {
+ if (!requestpad)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Can not get the %s pad from the transmitter %s element",
requestpad_name, tee_funnel_name);
@@ -1452,7 +1492,8 @@ _get_request_pad_and_link (GstElement *tee_funnel, const gchar *tee_funnel_name,
gst_object_unref (requestpad);
gst_object_unref (transpad);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Can not link the %s to the transmitter %s", tee_funnel_name,
(direction == GST_PAD_SINK) ? "sink" : "src");
@@ -1487,7 +1528,8 @@ fs_rtp_session_get_new_stream_transmitter (FsRtpSession *self,
transmitter = g_hash_table_lookup (self->priv->transmitters,
transmitter_name);
- if (transmitter) {
+ if (transmitter)
+ {
return fs_transmitter_new_stream_transmitter (transmitter, participant,
n_parameters, parameters, error);
}
@@ -1498,14 +1540,16 @@ fs_rtp_session_get_new_stream_transmitter (FsRtpSession *self,
g_object_get (transmitter, "gst-sink", &sink, "gst-src", &src, NULL);
- if (!gst_bin_add (GST_BIN (self->priv->conference), sink)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), sink))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not add the transmitter sink for %s to the conference",
transmitter_name);
goto error;
}
- if (!gst_bin_add (GST_BIN (self->priv->conference), src)) {
+ if (!gst_bin_add (GST_BIN (self->priv->conference), src))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not add the transmitter src for %s to the conference",
transmitter_name);
@@ -1785,7 +1829,8 @@ fs_rtp_session_update_codecs (FsRtpSession *session,
new_negotiated_codec_associations);
/* Lets remove the codec bin for any PT that has changed type */
- for (pt = 0; pt < 128; pt++) {
+ for (pt = 0; pt < 128; pt++)
+ {
CodecAssociation *old_codec_association =
lookup_codec_association_by_pt (
old_negotiated_codec_associations, pt);
@@ -1796,14 +1841,16 @@ fs_rtp_session_update_codecs (FsRtpSession *session,
if (old_codec_association == NULL && new_codec_association == NULL)
continue;
- if (old_codec_association == NULL || new_codec_association == NULL) {
+ if (old_codec_association == NULL || new_codec_association == NULL)
+ {
fs_rtp_session_invalidate_pt (session, pt, NULL);
clear_pts = TRUE;
continue;
}
if (!fs_codec_are_equal (old_codec_association->codec,
- new_codec_association->codec)) {
+ new_codec_association->codec))
+ {
fs_rtp_session_invalidate_pt (session, pt,
new_codec_association->codec);
clear_pts = TRUE;
@@ -1904,7 +1951,8 @@ fs_rtp_session_new_recv_pad (FsRtpSession *session, GstPad *new_pad,
substream = fs_rtp_sub_stream_new (session->priv->conference, session,
new_pad, ssrc, pt, no_rtcp_timeout, &error);
- if (substream == NULL) {
+ if (substream == NULL)
+ {
if (error && error->domain == FS_ERROR)
fs_session_emit_error (FS_SESSION (session), error->code,
"Could not create a substream for the new pad", error->message);
@@ -1987,12 +2035,13 @@ fs_rtp_session_new_recv_pad (FsRtpSession *session, GstPad *new_pad,
FS_RTP_SESSION_UNLOCK (session);
- if (stream) {
- if (!fs_rtp_stream_add_substream (stream, substream, &error)) {
+ if (stream)
+ {
+ if (!fs_rtp_stream_add_substream (stream, substream, &error))
fs_session_emit_error (FS_SESSION (session), error->code,
"Could not add the output ghostpad to the new substream",
error->message);
- }
+
g_clear_error (&error);
g_object_unref (stream);
}
@@ -2018,20 +2067,23 @@ _create_ghost_pad (GstElement *current_element, const gchar *padname, GstElement
GstPad *pad = gst_element_get_static_pad (current_element, padname);
gboolean ret = FALSE;
- if (!pad) {
+ if (!pad)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not find the %s pad on the element", padname);
return FALSE;
}
ghostpad = gst_ghost_pad_new (padname, pad);
- if (!ghostpad) {
+ if (!ghostpad)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not create a ghost pad for pad %s", padname);
goto done;
}
- if (!gst_pad_set_active (ghostpad, TRUE)) {
+ if (!gst_pad_set_active (ghostpad, TRUE))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not active ghostpad %s", padname);
gst_object_unref (ghostpad);
@@ -2087,12 +2139,14 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
for (walk = g_list_first (pipeline_factory); walk; walk = g_list_next (walk))
{
- if (g_list_next (g_list_first (walk->data))) {
+ if (g_list_next (g_list_first (walk->data)))
+ {
/* We have to check some kind of configuration to see if we have a
favorite */
current_element = gst_element_factory_make ("fsselector", NULL);
- if (!current_element) {
+ if (!current_element)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not create fsselector element");
goto error;
@@ -2103,14 +2157,16 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
current_element =
gst_element_factory_create (
GST_ELEMENT_FACTORY (g_list_first (walk->data)->data), NULL);
- if (!current_element) {
+ if (!current_element)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not create element for pt %d", codec->id);
goto error;
}
}
- if (!gst_bin_add (GST_BIN (codec_bin), current_element)) {
+ if (!gst_bin_add (GST_BIN (codec_bin), current_element))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not add new element to %s codec_bin for pt %d",
direction_str, codec->id);
@@ -2145,7 +2201,8 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
/* let's link them together using the specified media_caps if any
* this will ensure that multi-codec encoders/decoders will select the
* appropriate codec based on caps negotiation */
- if (previous_element) {
+ if (previous_element)
+ {
GstPad *sinkpad;
GstPad *srcpad;
GstPadLinkReturn ret;
@@ -2155,7 +2212,8 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
else
sinkpad = gst_element_get_static_pad (current_element, "sink");
- if (!sinkpad) {
+ if (!sinkpad)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not get the sink pad one of the elements in the %s codec bin"
" for pt %d", direction_str, codec->id);
@@ -2168,7 +2226,8 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
else
srcpad = gst_element_get_static_pad (previous_element, "src");
- if (!srcpad) {
+ if (!srcpad)
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not get the src pad one of the elements in the %s codec bin"
" for pt %d", direction_str, codec->id);
@@ -2181,7 +2240,8 @@ _create_codec_bin (CodecBlueprint *blueprint, const FsCodec *codec,
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
- if (GST_PAD_LINK_FAILED (ret)) {
+ if (GST_PAD_LINK_FAILED (ret))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not link element inside the %s codec bin for pt %d",
direction_str, codec->id);
@@ -2386,7 +2446,8 @@ fs_rtp_session_select_send_codec_locked (FsRtpSession *session,
return NULL;
}
- if (session->priv->requested_send_codec) {
+ if (session->priv->requested_send_codec)
+ {
ca = lookup_codec_association_by_codec_without_config (
session->priv->codec_associations,
session->priv->requested_send_codec);
@@ -2528,7 +2589,8 @@ fs_rtp_session_add_send_codec_bin (FsRtpSession *session,
goto error;
}
- if (!gst_element_sync_state_with_parent (codecbin)) {
+ if (!gst_element_sync_state_with_parent (codecbin))
+ {
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not sync the state of the codec bin for pt %d with the state"
" of the conference", codec->id);
@@ -2654,10 +2716,12 @@ _send_src_pad_have_data_callback (GstPad *pad, GstMiniObject *miniobj,
if (codecbin)
{
- if (GST_IS_BUFFER (miniobj)) {
+ if (GST_IS_BUFFER (miniobj))
+ {
GstPad *codecbin_sink_pad = gst_pad_get_peer (pad);
- if (!gst_pad_accept_caps (codecbin_sink_pad, GST_BUFFER_CAPS (miniobj))) {
+ if (!gst_pad_accept_caps (codecbin_sink_pad, GST_BUFFER_CAPS (miniobj)))
+ {
ret = FALSE;
GST_WARNING ("Dropping buffer because its caps do not match the"
" requirements of the new send codec bin");
@@ -2712,7 +2776,8 @@ fs_rtp_session_verify_send_codec_bin_locked (FsRtpSession *self, GError **error)
goto error;
}
- if (self->priv->current_send_codec) {
+ if (self->priv->current_send_codec)
+ {
if (fs_codec_are_equal (codec, self->priv->current_send_codec))
goto done;
@@ -2828,7 +2893,8 @@ fs_rtp_session_associate_ssrc_cname (FsRtpSession *session,
g_free (localcname);
}
- if (!stream) {
+ if (!stream)
+ {
gchar *str = g_strdup_printf ("There is no particpant with cname %s for"
" ssrc %u", cname, ssrc);
fs_session_emit_error (FS_SESSION (session), FS_ERROR_UNKNOWN_CNAME,
@@ -2849,7 +2915,8 @@ fs_rtp_session_associate_ssrc_cname (FsRtpSession *session,
g_object_get (localsubstream, "ssrc", &localssrc, NULL);
GST_LOG ("Have substream with ssrc %x, looking for %x", localssrc, ssrc);
- if (ssrc == localssrc) {
+ if (ssrc == localssrc)
+ {
substream = localsubstream;
session->priv->free_substreams = g_list_delete_link (
session->priv->free_substreams, item);
@@ -2866,9 +2933,9 @@ fs_rtp_session_associate_ssrc_cname (FsRtpSession *session,
}
while (
- g_signal_handlers_disconnect_by_func (substream, "error", session) > 0) {}
+ g_signal_handlers_disconnect_by_func (substream, "error", session) > 0);
while (
- g_signal_handlers_disconnect_by_func (substream, "no-rtcp-timedout", session) > 0) {}
+ g_signal_handlers_disconnect_by_func (substream, "no-rtcp-timedout", session) > 0);
if (!fs_rtp_stream_add_substream (stream, substream, &error))
fs_session_emit_error (FS_SESSION (session), error->code,
@@ -2914,9 +2981,9 @@ _substream_no_rtcp_timedout_cb (FsRtpSubStream *substream,
substream);
while (
- g_signal_handlers_disconnect_by_func (substream, "error", session) > 0) {}
+ g_signal_handlers_disconnect_by_func (substream, "error", session) > 0);
while (
- g_signal_handlers_disconnect_by_func (substream, "no-rtcp-timedout", session) > 0) {}
+ g_signal_handlers_disconnect_by_func (substream, "no-rtcp-timedout", session) > 0);
if (!fs_rtp_stream_add_substream (
g_list_first (session->priv->streams)->data,
--
1.5.6.5
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