diff options
| author | Pau Espin Pedrol <pespin@sysmocom.de> | 2017-06-28 15:31:46 +0200 |
|---|---|---|
| committer | Pau Espin Pedrol <pespin@sysmocom.de> | 2017-06-29 17:27:25 +0200 |
| commit | 524923a96a9892445234248623d8be642eb5229d (patch) | |
| tree | e7bda5b6b72743306bfd6d8107b7df19ea39d413 | |
| parent | c633cfbb9115485f3f54efe49480f4e2cec1cfa4 (diff) | |
osmo_ortp: Add osmo_rtp_drop_frame API
This API will be used by osmo-bts to maintain the sampling clock of the
RTP socket whenever a frame should be sent but it is not available, for
instance due to low quality of signal.
Change-Id: Id6099372b6231c0a4b6ea0716f46f5daee7049e1
| -rw-r--r-- | include/osmocom/trau/osmo_ortp.h | 1 | ||||
| -rw-r--r-- | src/trau/osmo_ortp.c | 22 |
2 files changed, 23 insertions, 0 deletions
diff --git a/include/osmocom/trau/osmo_ortp.h b/include/osmocom/trau/osmo_ortp.h index 9eb176f..578b1cb 100644 --- a/include/osmocom/trau/osmo_ortp.h +++ b/include/osmocom/trau/osmo_ortp.h @@ -76,6 +76,7 @@ int osmo_rtp_socket_bind(struct osmo_rtp_socket *rs, const char *ip, int port); int osmo_rtp_socket_connect(struct osmo_rtp_socket *rs, const char *ip, uint16_t port); int osmo_rtp_socket_set_pt(struct osmo_rtp_socket *rs, int payload_type); int osmo_rtp_socket_free(struct osmo_rtp_socket *rs); +int osmo_rtp_skipped_frame(struct osmo_rtp_socket *rs, unsigned int duration); int osmo_rtp_send_frame(struct osmo_rtp_socket *rs, const uint8_t *payload, unsigned int payload_len, unsigned int duration); int osmo_rtp_send_frame_ext(struct osmo_rtp_socket *rs, const uint8_t *payload, diff --git a/src/trau/osmo_ortp.c b/src/trau/osmo_ortp.c index b2ee040..c9480ff 100644 --- a/src/trau/osmo_ortp.c +++ b/src/trau/osmo_ortp.c @@ -436,6 +436,28 @@ int osmo_rtp_socket_connect(struct osmo_rtp_socket *rs, const char *ip, uint16_t return osmo_rtp_socket_fdreg(rs); } +/*! \brief Increment 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 to keep the RTP internal clock up to date if an RTP frame should be + * send at a given time but no audio content is available. When next packet is + * sent, the receiver will see a different increase on the sequence number and + * the timestamp, and it should then take it as a synchronization point. For + * that same reason, it is advisable to enable the marker bit on the next RTP + * packet to be sent after calling this function. + * + * \returns 0 on success, <0 in case of error. + */ +int osmo_rtp_skipped_frame(struct osmo_rtp_socket *rs, unsigned int duration) +{ + if (rs->flags & OSMO_RTP_F_DISABLED) + return 0; + + 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 |