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rtspsrc: Handle TCP as lower transport with RTSP 2.0

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Meaning that the interleave fields have to be updated as
if streams setup was working when using pipelined setup
request. Otherwise there is a mismatch between the server
channel count and our own.

This also makes RTSP 2.0 over HTTP working.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
This commit is contained in:
Thibault Saunier 2017-07-13 14:46:55 -04:00
parent 8121752887
commit ffcd173c09
1 changed files with 11 additions and 6 deletions
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17
gst/rtsp/gstrtspsrc.c
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@ -6082,10 +6082,14 @@ gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
g_string_append_printf (str, "%d", src->free_channel); g_string_append_printf (str, "%d", src->free_channel);
else if (next[3] == '2') else if (next[3] == '2')
g_string_append_printf (str, "%d", src->free_channel + 1); g_string_append_printf (str, "%d", src->free_channel + 1);
} }
p = next + 4; p = next + 4;
} }
if (src->version >= GST_RTSP_VERSION_2_0)
src->free_channel += 2;
/* append final part */ /* append final part */
g_string_append (str, p); g_string_append (str, p);
@ -6202,10 +6206,12 @@ gst_rtsp_src_setup_stream_from_response (GstRTSPSrc * src,
if (protocols) if (protocols)
*protocols = GST_RTSP_LOWER_TRANS_TCP; *protocols = GST_RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE; src->interleaved = TRUE;
/* update free channels */ if (src->version < GST_RTSP_VERSION_2_0) {
src->free_channel = MAX (transport.interleaved.min, src->free_channel); /* update free channels */
src->free_channel = MAX (transport.interleaved.max, src->free_channel); src->free_channel = MAX (transport.interleaved.min, src->free_channel);
src->free_channel++; src->free_channel = MAX (transport.interleaved.max, src->free_channel);
src->free_channel++;
}
break; break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST: case GST_RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */ /* only allow multicast for other streams */
@ -6331,7 +6337,7 @@ gst_rtspsrc_setup_streams_end (GstRTSPSrc * src, gboolean async)
gst_rtsp_src_receive_response (src, conninfo, &response, NULL); gst_rtsp_src_receive_response (src, conninfo, &response, NULL);
gst_rtsp_src_setup_stream_from_response (src, stream, gst_rtsp_src_setup_stream_from_response (src, stream,
&response, NULL, 1, NULL, NULL); &response, NULL, 0, NULL, NULL);
} }
return GST_RTSP_OK; return GST_RTSP_OK;
@ -6505,7 +6511,6 @@ gst_rtspsrc_setup_streams_start (GstRTSPSrc * src, gboolean async)
} }
GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports)); GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
/* create SETUP request */ /* create SETUP request */
res = res =
gst_rtspsrc_init_request (src, &request, GST_RTSP_SETUP, gst_rtspsrc_init_request (src, &request, GST_RTSP_SETUP,