gst: Run everything through gst-indent again

This commit is contained in:
Sebastian Dröge 2018-04-04 10:06:06 +03:00
parent 48ad01beba
commit ef878da703
5 changed files with 28 additions and 19 deletions

View file

@ -2063,7 +2063,8 @@ rtsp_ctrl_timeout_remove (GstRTSPClientPrivate * priv)
}
static gchar *
stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStream * stream)
stream_make_keymgmt (GstRTSPClient * client, const gchar * location,
GstRTSPStream * stream)
{
gchar *base64, *result = NULL;
GstMIKEYMessage *mikey_msg;
@ -2074,7 +2075,8 @@ stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStrea
GstBuffer *key;
GType ciphertype, authtype;
GEnumClass *cipher_enum, *auth_enum;
GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value, *srtp_auth_value;
GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value,
*srtp_auth_value;
rtcp_encoder = gst_rtsp_stream_get_srtp_encoder (stream);
@ -2090,8 +2092,9 @@ stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStrea
/* We need to bring the encoder to READY so that it generates its key */
gst_element_set_state (rtcp_encoder, GST_STATE_READY);
g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth", &srtcp_auth,
"rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key", &key, NULL);
g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth",
&srtcp_auth, "rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key",
&key, NULL);
g_object_unref (rtcp_encoder);
srtcp_cipher_value = g_enum_get_value (cipher_enum, srtcp_cipher);
@ -2845,11 +2848,14 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
gchar *location = g_strdup_printf ("rtsp://%s%s:8554/stream=%d", priv->server_ip, path, i);
gchar *location =
g_strdup_printf ("rtsp://%s%s:8554/stream=%d", priv->server_ip, path,
i);
gchar *keymgmt = stream_make_keymgmt (client, location, stream);
if (keymgmt)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT, keymgmt);
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT,
keymgmt);
g_free (location);
}

View file

@ -1098,7 +1098,7 @@ gst_rtsp_media_factory_get_retransmission_time (GstRTSPMediaFactory * factory)
*/
void
gst_rtsp_media_factory_set_do_retransmission (GstRTSPMediaFactory * factory,
gboolean do_retransmission)
gboolean do_retransmission)
{
GstRTSPMediaFactoryPrivate *priv;

View file

@ -228,6 +228,7 @@ static gboolean wait_preroll (GstRTSPMedia * media);
static GstElement *find_payload_element (GstElement * payloader);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
static gboolean check_complete (GstRTSPMedia * media);
#define C_ENUM(v) ((gint) v)
@ -603,8 +604,7 @@ do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
{
gint64 tmp;
if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream))
{
if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
return;
}
@ -1465,7 +1465,8 @@ gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
* Since: 1.16
*/
void
gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media, gboolean do_retransmission)
gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
gboolean do_retransmission)
{
GstRTSPMediaPrivate *priv;
@ -3143,7 +3144,8 @@ start_prepare (GstRTSPMedia * media)
}
if (priv->rtpbin)
g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission, NULL);
g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
NULL);
for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
GstElement *elem = walk->data;

View file

@ -2332,7 +2332,7 @@ gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
}
static void
add_rtx_pt (gpointer key, GstCaps *caps, GstStructure *pt_map)
add_rtx_pt (gpointer key, GstCaps * caps, GstStructure * pt_map)
{
guint pt = GPOINTER_TO_INT (key);
const GstStructure *s = gst_caps_get_structure (caps, 0);

View file

@ -134,9 +134,11 @@ get_server_uri (gint port, const gchar * mount_point)
}
static GstRTSPFilterResult
check_transport (GstRTSPStream *stream, GstRTSPStreamTransport *strans, gpointer user_data)
check_transport (GstRTSPStream * stream, GstRTSPStreamTransport * strans,
gpointer user_data)
{
const GstRTSPTransport *trans = gst_rtsp_stream_transport_get_transport (strans);
const GstRTSPTransport *trans =
gst_rtsp_stream_transport_get_transport (strans);
server_send_rtcp_port = trans->client_port.max;
@ -149,7 +151,8 @@ new_state_cb (GstRTSPMedia * media, gint state, gpointer user_data)
if (state == GST_STATE_PLAYING) {
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, 0);
gst_rtsp_stream_transport_filter (stream, (GstRTSPStreamTransportFilterFunc) check_transport, user_data);
gst_rtsp_stream_transport_filter (stream,
(GstRTSPStreamTransportFilterFunc) check_transport, user_data);
}
}
@ -160,8 +163,7 @@ media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
GstElement **p_sink = user_data;
GstElement *bin;
g_signal_connect (media, "new-state",
G_CALLBACK (new_state_cb), user_data);
g_signal_connect (media, "new-state", G_CALLBACK (new_state_cb), user_data);
bin = gst_rtsp_media_get_element (media);
*p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
@ -180,8 +182,7 @@ GST_START_TEST (test_record)
gint i;
mfactory =
start_record_server
("( rtppcmadepay name=depay0 ! appsink name=sink )");
start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
g_signal_connect (mfactory, "media-constructed",
G_CALLBACK (media_constructed_cb), &server_sink);