neels has submitted this change and it was merged. (
https://gerrit.osmocom.org/c/osmo-mgw/+/15132 )
Change subject: mgcp_codec: split codec_free() off of codec_init()
......................................................................
mgcp_codec: split codec_free() off of codec_init()
Both are used only in the same .c file, so make them static.
Move codec_set() guts into codec_add(): codec_set is only called by codec_add.
If codec_set were left separate, it'd look like the codec_init() is a bug and
lacks a codec_free() first. When looking at the entire context in codec_add(),
it becomes obvious that codec_init() should be called, not codec_free(),
because it is populating a previously unused entry.
Preparation to fix a memleak in a conn's codec list.
Change-Id: I120cab0a352a1e7b31c8f9c720c47b2c291311d7
---
M src/libosmo-mgcp/mgcp_codec.c
1 file changed, 35 insertions(+), 39 deletions(-)
Approvals:
neels: Looks good to me, approved
pespin: Looks good to me, but someone else must approve
Jenkins Builder: Verified
diff --git a/src/libosmo-mgcp/mgcp_codec.c b/src/libosmo-mgcp/mgcp_codec.c
index d5b99e8..30c185c 100644
--- a/src/libosmo-mgcp/mgcp_codec.c
+++ b/src/libosmo-mgcp/mgcp_codec.c
@@ -76,18 +76,24 @@
}
/* Initalize or reset codec information with default data. */
-void codec_init(struct mgcp_rtp_codec *codec)
+static void codec_init(struct mgcp_rtp_codec *codec)
+{
+ *codec = (struct mgcp_rtp_codec){
+ .payload_type = -1,
+ .frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM,
+ .frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN,
+ .rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE,
+ .channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS,
+ };
+}
+
+static void codec_free(struct mgcp_rtp_codec *codec)
{
if (codec->subtype_name)
talloc_free(codec->subtype_name);
if (codec->audio_name)
talloc_free(codec->audio_name);
- memset(codec, 0, sizeof(*codec));
- codec->payload_type = -1;
- codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
- codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
- codec->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
- codec->channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
+ *codec = (struct mgcp_rtp_codec){};
}
/*! Initalize or reset codec information with default data.
@@ -99,13 +105,30 @@
conn->end.codec = NULL;
}
-/* Set members of struct mgcp_rtp_codec, extrapolate in missing information.
Param audio_name is expected in uppercase. */
-static int codec_set(void *ctx, struct mgcp_rtp_codec *codec, int
payload_type, const char *audio_name,
- unsigned int pt_offset, const struct mgcp_codec_param
*param)
+/*! Add codec configuration depending on payload type and/or codec name. This
+ * function uses the input parameters to extrapolate the full codec
information.
+ * \param[out] codec configuration (caller provided memory).
+ * \param[out] conn related rtp-connection.
+ * \param[in] payload_type codec type id (e.g. 3 for GSM, -1 when undefined).
+ * \param[in] audio_name audio codec name, in uppercase (e.g. "GSM/8000/1").
+ * \param[in] param optional codec parameters (set to NULL when unused).
+ * \returns 0 on success, -EINVAL on failure. */
+int mgcp_codec_add(struct mgcp_conn_rtp *conn, int payload_type, const char
*audio_name, const struct mgcp_codec_param *param)
{
int rate;
int channels;
char audio_codec[64];
+ struct mgcp_rtp_codec *codec;
+ unsigned int pt_offset = conn->end.codecs_assigned;
+ void *ctx = conn->conn;
+
+ /* The amount of codecs we can store is limited, make sure we do not
+ * overrun this limit. */
+ if (conn->end.codecs_assigned >= MGCP_MAX_CODECS)
+ return -EINVAL;
+
+ /* First unused entry */
+ codec = &conn->end.codecs[conn->end.codecs_assigned];
/* Initalize the codec struct with some default data to begin with */
codec_init(codec);
@@ -226,41 +249,14 @@
} else
codec->param_present = false;
+ conn->end.codecs_assigned++;
return 0;
error:
/* Make sure we leave a clean codec entry on error. */
- codec_init(codec);
- memset(codec, 0, sizeof(*codec));
+ codec_free(codec);
return -EINVAL;
}
-/*! Add codec configuration depending on payload type and/or codec name. This
- * function uses the input parameters to extrapolate the full codec
information.
- * \param[out] codec configuration (caller provided memory).
- * \param[out] conn related rtp-connection.
- * \param[in] payload_type codec type id (e.g. 3 for GSM, -1 when undefined).
- * \param[in] audio_name audio codec name, in uppercase (e.g. "GSM/8000/1").
- * \param[in] param optional codec parameters (set to NULL when unused).
- * \returns 0 on success, -EINVAL on failure. */
-int mgcp_codec_add(struct mgcp_conn_rtp *conn, int payload_type, const char
*audio_name, const struct mgcp_codec_param *param)
-{
- int rc;
-
- /* The amount of codecs we can store is limited, make sure we do not
- * overrun this limit. */
- if (conn->end.codecs_assigned >= MGCP_MAX_CODECS)
- return -EINVAL;
-
- rc = codec_set(conn->conn,
&conn->end.codecs[conn->end.codecs_assigned], payload_type, audio_name,
- conn->end.codecs_assigned, param);
- if (rc != 0)
- return -EINVAL;
-
- conn->end.codecs_assigned++;
-
- return 0;
-}
-
/* Check if the given codec is applicable on the specified endpoint
* Helper function for mgcp_codec_decide() */
static bool is_codec_compatible(const struct mgcp_endpoint *endp, const struct
mgcp_rtp_codec *codec)
--
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Gerrit-Project: osmo-mgw
Gerrit-Branch: master
Gerrit-Change-Id: I120cab0a352a1e7b31c8f9c720c47b2c291311d7
Gerrit-Change-Number: 15132
Gerrit-PatchSet: 3
Gerrit-Owner: neels <nhofm...@sysmocom.de>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: laforge <lafo...@gnumonks.org>
Gerrit-Reviewer: neels <nhofm...@sysmocom.de>
Gerrit-Reviewer: pespin <pes...@sysmocom.de>
Gerrit-MessageType: merged
