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rtspsrc: fix some comments, remove property check
Fix some comments, clarify some FIXMEs Remove the on-ntp-stop signal check now that the jitterbuffer is in -good and we know that it supports this signal.
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1 changed files with 12 additions and 9 deletions
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@ -814,7 +814,11 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
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gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
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gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
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/* we must have a payload. No payload means we cannot create caps */
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/* we must have a payload. No payload means we cannot create caps */
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/* FIXME, handle multiple formats. */
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/* FIXME, handle multiple formats. The problem here is that we just want to
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* take the first available format that we can handle but in order to do that
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* we need to scan for depayloader plugins. Scanning for payloader plugins is
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* also suboptimal because the user maybe just wants to save the raw stream
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* and then we don't care. */
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if ((payload = gst_sdp_media_get_format (media, 0))) {
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if ((payload = gst_sdp_media_get_format (media, 0))) {
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stream->pt = atoi (payload);
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stream->pt = atoi (payload);
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/* convert caps */
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/* convert caps */
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@ -1203,7 +1207,11 @@ gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
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if (valpos) {
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if (valpos) {
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/* we have a '=' and thus a value, remove the '=' with \0 */
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/* we have a '=' and thus a value, remove the '=' with \0 */
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*valpos = '\0';
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*valpos = '\0';
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/* value is everything between '=' and ';'. FIXME, strip? */
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/* value is everything between '=' and ';'. We split the pairs at ;
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* boundaries so we can take the remainder of the value. Some servers
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* put spaces around the value which we strip off here. Alternatively
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* we could strip those spaces in the depayloaders should these spaces
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* actually carry any meaning in the future. */
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val = g_strstrip (valpos + 1);
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val = g_strstrip (valpos + 1);
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} else {
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} else {
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/* simple <param>;.. is translated into <param>=1;... */
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/* simple <param>;.. is translated into <param>=1;... */
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@ -2091,13 +2099,8 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
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src);
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src);
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g_signal_connect (src->session, "on-timeout", (GCallback) on_timeout,
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g_signal_connect (src->session, "on-timeout", (GCallback) on_timeout,
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src);
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src);
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/* FIXME: remove this once the rtpjitterbuffer is in -good */
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g_signal_connect (src->session, "on-npt-stop", (GCallback) on_npt_stop,
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if (g_signal_lookup ("on-npt-stop", G_OBJECT_TYPE (src->session)) != 0) {
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src);
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g_signal_connect (src->session, "on-npt-stop", (GCallback) on_npt_stop,
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src);
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} else {
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GST_INFO_OBJECT (src, "skipping on-npt-stop handling, not implemented");
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}
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}
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}
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/* we stream directly to the manager, get some pads. Each RTSP stream goes
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/* we stream directly to the manager, get some pads. Each RTSP stream goes
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