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rtspsrc: refactor payload handling

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This commit is contained in:
Wim Taymans 2014-03-04 11:34:39 +01:00
parent b4caf09011
commit dbe92c9147
1 changed files with 56 additions and 47 deletions
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103
gst/rtsp/gstrtspsrc.c
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@ -1299,55 +1299,11 @@ gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
}
}
static const gchar *
get_aggregate_control (GstRTSPSrc * src)
static void
gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
const GstSDPMedia * media, GstRTSPStream * stream)
{
const gchar *base;
if (src->control)
base = src->control;
else if (src->content_base)
base = src->content_base;
else if (src->conninfo.url_str)
base = src->conninfo.url_str;
else
base = "/";
return base;
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
{
GstRTSPStream *stream;
const gchar *control_url;
const gchar *payload;
const GstSDPMedia *media;
/* get media, should not return NULL */
media = gst_sdp_message_get_media (sdp, idx);
if (media == NULL)
return NULL;
stream = g_new0 (GstRTSPStream, 1);
stream->parent = src;
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
* the element. */
stream->last_ret = GST_FLOW_NOT_LINKED;
stream->added = FALSE;
stream->disabled = FALSE;
stream->id = src->numstreams++;
stream->eos = FALSE;
stream->discont = TRUE;
stream->seqbase = -1;
stream->timebase = -1;
/* collect bandwidth information for this steam. FIXME, configure in the RTP
* session manager to scale RTCP. */
gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
/* collect connection info */
gst_rtspsrc_collect_connections (src, sdp, media, stream);
/* we must have a payload. No payload means we cannot create caps */
/* FIXME, handle multiple formats. The problem here is that we just want to
@ -1376,6 +1332,59 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
}
}
}
}
static const gchar *
get_aggregate_control (GstRTSPSrc * src)
{
const gchar *base;
if (src->control)
base = src->control;
else if (src->content_base)
base = src->content_base;
else if (src->conninfo.url_str)
base = src->conninfo.url_str;
else
base = "/";
return base;
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
{
GstRTSPStream *stream;
const gchar *control_url;
const GstSDPMedia *media;
/* get media, should not return NULL */
media = gst_sdp_message_get_media (sdp, idx);
if (media == NULL)
return NULL;
stream = g_new0 (GstRTSPStream, 1);
stream->parent = src;
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
* the element. */
stream->last_ret = GST_FLOW_NOT_LINKED;
stream->added = FALSE;
stream->disabled = FALSE;
stream->id = src->numstreams++;
stream->eos = FALSE;
stream->discont = TRUE;
stream->seqbase = -1;
stream->timebase = -1;
/* collect bandwidth information for this steam. FIXME, configure in the RTP
* session manager to scale RTCP. */
gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
/* collect connection info */
gst_rtspsrc_collect_connections (src, sdp, media, stream);
gst_rtspsrc_collect_payloads (src, sdp, media, stream);
/* collect port number */
stream->port = gst_sdp_media_get_port (media);