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dexter gerrit-no-reply at lists.osmocom.orgReview at https://gerrit.osmocom.org/7394 cosmetic: rename .._codec_reset() to .._codec_init() The function mgcp_rtp_codec_reset() is soley called from mgcp_rtp_conn_init(), lets change the prefix here to _init too. - rename mgcp_rtp_conn_reset() to mgcp_rtp_conn_init() Change-Id: I246aabc896089c1f2b3d0409ec3422a85e43575c --- M src/libosmo-mgcp/mgcp_conn.c 1 file changed, 3 insertions(+), 3 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmo-mgw refs/changes/94/7394/1 diff --git a/src/libosmo-mgcp/mgcp_conn.c b/src/libosmo-mgcp/mgcp_conn.c index 9491213..2f691f0 100644 --- a/src/libosmo-mgcp/mgcp_conn.c +++ b/src/libosmo-mgcp/mgcp_conn.c @@ -68,7 +68,7 @@ } /* Reset codec state and free memory */ -static void mgcp_rtp_codec_reset(struct mgcp_rtp_codec *codec) +static void mgcp_rtp_codec_init(struct mgcp_rtp_codec *codec) { codec->payload_type = -1; codec->subtype_name = NULL; @@ -106,8 +106,8 @@ end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS; end->output_enabled = 0; - mgcp_rtp_codec_reset(&end->codec); - mgcp_rtp_codec_reset(&end->alt_codec); + mgcp_rtp_codec_init(&end->codec); + mgcp_rtp_codec_init(&end->alt_codec); } /*! allocate a new connection list entry. -- To view, visit https://gerrit.osmocom.org/7394 To unsubscribe, visit https://gerrit.osmocom.org/settings Gerrit-MessageType: newchange Gerrit-Change-Id: I246aabc896089c1f2b3d0409ec3422a85e43575c Gerrit-PatchSet: 1 Gerrit-Project: osmo-mgw Gerrit-Branch: master Gerrit-Owner: dexter <pmaier at sysmocom.de>