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source: don't update packet SSRC

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Remove the code to update the SSRC in packets, it can never be called now that
we always use a source with matching packet SSRC.
This commit is contained in:
Wim Taymans 2013-07-26 10:42:44 +02:00
parent abc90da1dc
commit 88f5a5f355
1 changed files with 1 additions and 35 deletions
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36
gst/rtpmanager/rtpsource.c
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@ -126,7 +126,7 @@ rtp_source_class_init (RTPSourceClass * klass)
* name application/x-rtp-source-stats with the following fields: * name application/x-rtp-source-stats with the following fields:
* *
* "ssrc" G_TYPE_UINT The SSRC of this source * "ssrc" G_TYPE_UINT The SSRC of this source
* "internal" G_TYPE_BOOLEAN If this source is the source of the session * "internal" G_TYPE_BOOLEAN If this source is a source of the session
* "validated" G_TYPE_BOOLEAN If the source is validated * "validated" G_TYPE_BOOLEAN If the source is validated
* "received-bye" G_TYPE_BOOLEAN If we received a BYE from this source * "received-bye" G_TYPE_BOOLEAN If we received a BYE from this source
* "is-csrc" G_TYPE_BOOLEAN If this source was found as CSRC * "is-csrc" G_TYPE_BOOLEAN If this source was found as CSRC
@ -1151,19 +1151,6 @@ rtp_source_mark_bye (RTPSource * src, const gchar * reason)
src->marked_bye = TRUE; src->marked_bye = TRUE;
} }
static gboolean
set_ssrc (GstBuffer ** buffer, guint idx, RTPSource * src)
{
GstRTPBuffer rtp = { NULL };
*buffer = gst_buffer_make_writable (*buffer);
if (gst_rtp_buffer_map (*buffer, GST_MAP_WRITE, &rtp)) {
gst_rtp_buffer_set_ssrc (&rtp, src->ssrc);
gst_rtp_buffer_unmap (&rtp);
}
return TRUE;
}
/** /**
* rtp_source_send_rtp: * rtp_source_send_rtp:
* @src: an #RTPSource * @src: an #RTPSource
@ -1189,7 +1176,6 @@ rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
GstBufferList *list = NULL; GstBufferList *list = NULL;
GstBuffer *buffer = NULL; GstBuffer *buffer = NULL;
guint packets; guint packets;
guint32 ssrc;
GstRTPBuffer rtp = { NULL }; GstRTPBuffer rtp = { NULL };
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR); g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
@ -1268,28 +1254,8 @@ rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
gst_rtp_buffer_unmap (&rtp); gst_rtp_buffer_unmap (&rtp);
goto no_callback; goto no_callback;
} }
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
gst_rtp_buffer_unmap (&rtp); gst_rtp_buffer_unmap (&rtp);
if (ssrc != src->ssrc) {
/* the SSRC of the packet is not correct, make a writable buffer and
* update the SSRC. This could involve a complete copy of the packet when
* it is not writable. Usually the payloader will use caps negotiation to
* get the correct SSRC from the session manager before pushing anything. */
/* FIXME, we don't want to warn yet because we can't inform any payloader
* of the changes SSRC yet because we don't implement pad-alloc. */
GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
src->ssrc);
if (is_list) {
list = gst_buffer_list_make_writable (list);
gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
} else {
set_ssrc (&buffer, 0, src);
}
}
GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet", GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
src->stats.packets_sent); src->stats.packets_sent);