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gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not provided with caps but with a signal.

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Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(create_stream), (payload_type_change), (new_ssrc_pad_found):
Also handle lip-sync when the clock-rate is not provided with caps but
with a signal.
This commit is contained in:
Olivier Crete 2008-01-25 16:58:00 +00:00 committed by Wim Taymans
parent 717657e696
commit b65d97e2e1
2 changed files with 56 additions and 3 deletions
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9
ChangeLog
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@ -1,3 +1,12 @@
2008-01-25 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(create_stream), (payload_type_change), (new_ssrc_pad_found):
Also handle lip-sync when the clock-rate is not provided with caps but
with a signal.
2008-01-25 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: Olivier Crete <tester@tester.ca>

50
gst/rtpmanager/gstrtpbin.c
View file

@ -291,6 +291,7 @@ struct _GstRtpBinStream
GstElement *demux;
gulong demux_newpad_sig;
gulong demux_ptreq_sig;
gulong demux_pt_change_sig;
/* the internal pad we use to get RTCP sync messages */
GstPad *sync_pad;
@ -308,6 +309,7 @@ struct _GstRtpBinStream
gint clock_rate;
gint64 ts_offset;
gint64 prev_ts_offset;
gint last_pt;
};
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
@ -721,8 +723,30 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
/* we can only continue if we know the local clock-base and clock-rate */
if (stream->clock_base == -1)
goto no_clock_base;
if (stream->clock_rate <= 0)
goto no_clock_rate;
if (stream->clock_rate <= 0) {
gint pt = -1;
GstCaps *caps = NULL;
GstStructure *s = NULL;
GST_RTP_SESSION_LOCK (stream->session);
pt = stream->last_pt;
GST_RTP_SESSION_UNLOCK (stream->session);
if (pt < 0)
goto no_clock_rate;
caps = get_pt_map (stream->session, pt);
if (!caps)
goto no_clock_rate;
s = gst_caps_get_structure (caps, 0);
gst_structure_get_int (s, "clock-rate", &stream->clock_rate);
gst_caps_unref (caps);
if (stream->clock_rate <= 0)
goto no_clock_rate;
}
/* map last RTP time to local timeline using our clock-base */
stream->local_rtp = stream->last_extrtptime - stream->clock_base;
@ -939,6 +963,7 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
stream->buffer = buffer;
stream->demux = demux;
stream->last_extrtptime = -1;
stream->last_pt = -1;
stream->have_sync = FALSE;
session->streams = g_slist_prepend (session->streams, stream);
@ -1671,6 +1696,15 @@ caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
GST_RTP_SESSION_UNLOCK (session);
}
/* Stores the last payload type received on a particular stream */
static void
payload_type_change (GstElement * element, guint pt, GstRtpBinStream * stream)
{
GST_RTP_SESSION_LOCK (stream->session);
stream->last_pt = pt;
GST_RTP_SESSION_UNLOCK (stream->session);
}
/* a new pad (SSRC) was created in @session */
static void
new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
@ -1699,9 +1733,14 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "clock-rate", &stream->clock_rate))
if (!gst_structure_get_int (s, "clock-rate", &stream->clock_rate)) {
stream->clock_rate = -1;
GST_WARNING_OBJECT (session->bin,
"Caps have no clock rate %s from pad %s:%s",
gst_caps_to_string (caps), GST_DEBUG_PAD_NAME (pad));
}
if (gst_structure_get_uint (s, "clock-base", &val))
stream->clock_base = val;
else
@ -1734,6 +1773,11 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
* depayloaders. */
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
"request-pt-map", (GCallback) pt_map_requested, session);
/* connect to the payload-type-change signal so that we can know which
* PT is the current PT so that the jitterbuffer can be matched to the right
* offset. */
stream->demux_pt_change_sig = g_signal_connect (stream->demux,
"payload-type-change", (GCallback) payload_type_change, stream);
GST_RTP_SESSION_UNLOCK (session);