[PATCH] libosmo-abis[master]: osmo_ortp.c: Add osmo_rtp_drop_frame API

[Pau Espin Pedrol]

Review at  https://gerrit.osmocom.org/3070

osmo_ortp.c: Add osmo_rtp_drop_frame API

Change-Id: Id6099372b6231c0a4b6ea0716f46f5daee7049e1
---
M src/trau/osmo_ortp.c
1 file changed, 21 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.osmocom.org:29418/libosmo-abis refs/changes/70/3070/1

diff --git a/src/trau/osmo_ortp.c b/src/trau/osmo_ortp.c
index b2ee040..05817aa 100644
--- a/src/trau/osmo_ortp.c
+++ b/src/trau/osmo_ortp.c
@@ -436,6 +436,27 @@
 		return osmo_rtp_socket_fdreg(rs);
 }
 
+/*! \brief Increment seq and timestamp on a RTP socket without sending any packet
+ *  \param[in] rs OsmoRTP socket
+ *  \param[in] duration duration in number of RTP clock ticks
+ *
+ * Useful if willing to update the RTP internal clock at a given time but no
+ * audio content is available. From receiver point of view, it will look like
+ * a packet was lost somewhere in the path.
+ *
+ *  \returns 0 on success, <0 in case of error.
+ */
+int osmo_rtp_drop_frame(struct osmo_rtp_socket *rs, unsigned int duration)
+{
+	if (rs->flags & OSMO_RTP_F_DISABLED)
+		return 0;
+
+	uint16_t next_seq = rtp_session_get_seq_number(rs->sess) + 1;
+	rtp_session_set_seq_number(rs->sess, next_seq);
+	rs->tx_timestamp += duration;
+	return 0;
+}
+
 /*! \brief Send one RTP frame via a RTP socket
  *  \param[in] rs OsmoRTP socket
  *  \param[in] payload pointer to buffer with RTP payload data

-- 
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Gerrit-MessageType: newchange
Gerrit-Change-Id: Id6099372b6231c0a4b6ea0716f46f5daee7049e1
Gerrit-PatchSet: 1
Gerrit-Project: libosmo-abis
Gerrit-Branch: master
Gerrit-Owner: Pau Espin Pedrol <pespin at sysmocom.de>