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rtspsrc: strip trailing spaces

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This commit is contained in:
Stefan Kost 2011-02-07 17:08:47 +02:00
parent 5e071d51f2
commit 6f6b2a7efc
1 changed files with 9 additions and 9 deletions
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18
gst/rtsp/gstrtspsrc.c
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@ -56,14 +56,14 @@
* For each stream listed in the SDP a new rtp_stream%d pad will be created * For each stream listed in the SDP a new rtp_stream%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type * with caps derived from the SDP media description. This is a caps of mime type
* "application/x-rtp" that can be connected to any available RTP depayloader * "application/x-rtp" that can be connected to any available RTP depayloader
* element. * element.
* *
* rtspsrc will internally instantiate an RTP session manager element * rtspsrc will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal, * that will handle the RTCP messages to and from the server, jitter removal,
* packet reordering along with providing a clock for the pipeline. * packet reordering along with providing a clock for the pipeline.
* This feature is implemented using the gstrtpbin element. * This feature is implemented using the gstrtpbin element.
* *
* rtspsrc acts like a live source and will therefore only generate data in the * rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state. * PLAYING state.
* *
* <refsect2> * <refsect2>
@ -1309,7 +1309,7 @@ gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
/* /*
* Mapping of caps to and from SDP fields: * Mapping of caps to and from SDP fields:
* *
* m=<media> <UDP port> RTP/AVP <payload> * m=<media> <UDP port> RTP/AVP <payload>
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>] * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
* a=fmtp:<payload> <param>[=<value>];... * a=fmtp:<payload> <param>[=<value>];...
*/ */
@ -4133,11 +4133,11 @@ gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
g_slist_free (list); g_slist_free (list);
} }
/* Parse a WWW-Authenticate Response header and determine the /* Parse a WWW-Authenticate Response header and determine the
* available authentication methods * available authentication methods
* *
* This code should also cope with the fact that each WWW-Authenticate * This code should also cope with the fact that each WWW-Authenticate
* header can contain multiple challenge methods + tokens * header can contain multiple challenge methods + tokens
* *
* At the moment, for Basic auth, we just do a minimal check and don't * At the moment, for Basic auth, we just do a minimal check and don't
* even parse out the realm */ * even parse out the realm */
@ -4166,8 +4166,8 @@ gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
* gst_rtspsrc_setup_auth: * gst_rtspsrc_setup_auth:
* @src: the rtsp source * @src: the rtsp source
* *
* Configure a username and password and auth method on the * Configure a username and password and auth method on the
* connection object based on a response we received from the * connection object based on a response we received from the
* peer. * peer.
* *
* Currently, this requires that a username and password were supplied * Currently, this requires that a username and password were supplied
@ -5672,7 +5672,7 @@ not_supported:
* *
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...] * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
* *
* rtptime corresponds to the timestamp for the NPT time given in the header * rtptime corresponds to the timestamp for the NPT time given in the header
* seqbase corresponds to the next sequence number we received. This number * seqbase corresponds to the next sequence number we received. This number
* indicates the first seqnum after the seek and should be used to discard * indicates the first seqnum after the seek and should be used to discard
* packets that are from before the seek. * packets that are from before the seek.