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rtpjitterbuffer: Lock clock_rate variable

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The priv->clock_rate variable could become -1 between when its checked to not
be -1 and when its used, causing an assertion. Fixed by taking the mutex
earlier in the chain() function.

Fixes #593955
This commit is contained in:
Olivier Crête 2009-09-02 15:21:02 -04:00 committed by Wim Taymans
parent 3fcde4486d
commit f542f710cf
1 changed files with 31 additions and 18 deletions
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49
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -609,7 +609,10 @@ gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
priv = jitterbuffer->priv;
/* this will trigger a new pt-map request signal, FIXME, do something better. */
JBUF_LOCK (priv);
priv->clock_rate = -1;
JBUF_UNLOCK (priv);
}
static GstCaps *
@ -647,6 +650,10 @@ gst_rtp_jitter_buffer_getcaps (GstPad * pad)
return caps;
}
/*
* Must be called with JBUF_LOCK held
*/
static gboolean
gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
GstCaps * caps)
@ -737,7 +744,9 @@ gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps)
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
JBUF_UNLOCK (priv);
/* set same caps on srcpad on success */
if (res)
@ -1032,6 +1041,10 @@ gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event)
return TRUE;
}
/*
* Must be called with JBUF_LOCK held
*/
static gboolean
gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
guint8 pt)
@ -1094,6 +1107,23 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
pt = gst_rtp_buffer_get_payload_type (buffer);
/* take the timestamp of the buffer. This is the time when the packet was
* received and is used to calculate jitter and clock skew. We will adjust
* this timestamp with the smoothed value after processing it in the
* jitterbuffer. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* bring to running time */
timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
seqnum = gst_rtp_buffer_get_seq (buffer);
GST_DEBUG_OBJECT (jitterbuffer,
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
JBUF_LOCK_CHECK (priv, out_flushing);
if (G_UNLIKELY (priv->last_pt != pt)) {
GstCaps *caps;
@ -1117,22 +1147,6 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
goto no_clock_rate;
}
/* take the timestamp of the buffer. This is the time when the packet was
* received and is used to calculate jitter and clock skew. We will adjust
* this timestamp with the smoothed value after processing it in the
* jitterbuffer. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* bring to running time */
timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
seqnum = gst_rtp_buffer_get_seq (buffer);
GST_DEBUG_OBJECT (jitterbuffer,
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
JBUF_LOCK_CHECK (priv, out_flushing);
/* don't accept more data on EOS */
if (G_UNLIKELY (priv->eos))
goto have_eos;
@ -1252,8 +1266,7 @@ no_clock_rate:
GST_WARNING_OBJECT (jitterbuffer,
"No clock-rate in caps!, dropping buffer");
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
goto finished;
}
out_flushing:
{