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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 02:01:12 +00:00
gst: Run everything through gst-indent again
This commit is contained in:
parent
48ad01beba
commit
ef878da703
5 changed files with 28 additions and 19 deletions
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@ -2063,7 +2063,8 @@ rtsp_ctrl_timeout_remove (GstRTSPClientPrivate * priv)
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}
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static gchar *
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stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStream * stream)
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stream_make_keymgmt (GstRTSPClient * client, const gchar * location,
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GstRTSPStream * stream)
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{
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gchar *base64, *result = NULL;
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GstMIKEYMessage *mikey_msg;
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@ -2074,7 +2075,8 @@ stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStrea
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GstBuffer *key;
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GType ciphertype, authtype;
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GEnumClass *cipher_enum, *auth_enum;
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GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value, *srtp_auth_value;
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GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value,
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*srtp_auth_value;
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rtcp_encoder = gst_rtsp_stream_get_srtp_encoder (stream);
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@ -2090,8 +2092,9 @@ stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStrea
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/* We need to bring the encoder to READY so that it generates its key */
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gst_element_set_state (rtcp_encoder, GST_STATE_READY);
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g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth", &srtcp_auth,
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"rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key", &key, NULL);
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g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth",
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&srtcp_auth, "rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key",
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&key, NULL);
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g_object_unref (rtcp_encoder);
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srtcp_cipher_value = g_enum_get_value (cipher_enum, srtcp_cipher);
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@ -2845,11 +2848,14 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
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n_streams = gst_rtsp_media_n_streams (media);
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for (i = 0; i < n_streams; i++) {
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GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
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gchar *location = g_strdup_printf ("rtsp://%s%s:8554/stream=%d", priv->server_ip, path, i);
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gchar *location =
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g_strdup_printf ("rtsp://%s%s:8554/stream=%d", priv->server_ip, path,
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i);
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gchar *keymgmt = stream_make_keymgmt (client, location, stream);
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if (keymgmt)
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gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT, keymgmt);
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gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT,
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keymgmt);
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g_free (location);
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}
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@ -1098,7 +1098,7 @@ gst_rtsp_media_factory_get_retransmission_time (GstRTSPMediaFactory * factory)
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*/
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void
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gst_rtsp_media_factory_set_do_retransmission (GstRTSPMediaFactory * factory,
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gboolean do_retransmission)
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gboolean do_retransmission)
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{
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GstRTSPMediaFactoryPrivate *priv;
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@ -228,6 +228,7 @@ static gboolean wait_preroll (GstRTSPMedia * media);
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static GstElement *find_payload_element (GstElement * payloader);
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static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
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static gboolean check_complete (GstRTSPMedia * media);
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#define C_ENUM(v) ((gint) v)
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@ -603,8 +604,7 @@ do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
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{
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gint64 tmp;
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if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream))
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{
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if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
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GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
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return;
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}
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@ -1465,7 +1465,8 @@ gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
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* Since: 1.16
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*/
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void
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gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media, gboolean do_retransmission)
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gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
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gboolean do_retransmission)
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{
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GstRTSPMediaPrivate *priv;
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@ -3143,7 +3144,8 @@ start_prepare (GstRTSPMedia * media)
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}
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if (priv->rtpbin)
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g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission, NULL);
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g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
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NULL);
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for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
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GstElement *elem = walk->data;
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@ -2332,7 +2332,7 @@ gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
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}
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static void
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add_rtx_pt (gpointer key, GstCaps *caps, GstStructure *pt_map)
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add_rtx_pt (gpointer key, GstCaps * caps, GstStructure * pt_map)
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{
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guint pt = GPOINTER_TO_INT (key);
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const GstStructure *s = gst_caps_get_structure (caps, 0);
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@ -134,9 +134,11 @@ get_server_uri (gint port, const gchar * mount_point)
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}
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static GstRTSPFilterResult
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check_transport (GstRTSPStream *stream, GstRTSPStreamTransport *strans, gpointer user_data)
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check_transport (GstRTSPStream * stream, GstRTSPStreamTransport * strans,
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gpointer user_data)
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{
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const GstRTSPTransport *trans = gst_rtsp_stream_transport_get_transport (strans);
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const GstRTSPTransport *trans =
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gst_rtsp_stream_transport_get_transport (strans);
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server_send_rtcp_port = trans->client_port.max;
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@ -149,7 +151,8 @@ new_state_cb (GstRTSPMedia * media, gint state, gpointer user_data)
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if (state == GST_STATE_PLAYING) {
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GstRTSPStream *stream = gst_rtsp_media_get_stream (media, 0);
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gst_rtsp_stream_transport_filter (stream, (GstRTSPStreamTransportFilterFunc) check_transport, user_data);
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gst_rtsp_stream_transport_filter (stream,
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(GstRTSPStreamTransportFilterFunc) check_transport, user_data);
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}
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}
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@ -160,8 +163,7 @@ media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
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GstElement **p_sink = user_data;
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GstElement *bin;
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g_signal_connect (media, "new-state",
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G_CALLBACK (new_state_cb), user_data);
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g_signal_connect (media, "new-state", G_CALLBACK (new_state_cb), user_data);
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bin = gst_rtsp_media_get_element (media);
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*p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
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@ -180,8 +182,7 @@ GST_START_TEST (test_record)
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gint i;
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mfactory =
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start_record_server
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("( rtppcmadepay name=depay0 ! appsink name=sink )");
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start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
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g_signal_connect (mfactory, "media-constructed",
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G_CALLBACK (media_constructed_cb), &server_sink);
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