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laforge gerrit-no-reply at lists.osmocom.orglaforge has submitted this change. ( https://gerrit.osmocom.org/c/osmo-mgw/+/25421 )
Change subject: libosmo-mgcp: cleanup audio codex alloc
......................................................................
libosmo-mgcp: cleanup audio codex alloc
No need to complicate audio codes with pointers, our "usual" string is
barely larger than a poointer, five times smaller than a talloc header,
and most importantly not really optional anyway...
Change-Id: Icc41643050a5e1ca3c66f307d60b6911ba1b8032
---
M include/osmocom/mgcp/mgcp_network.h
M src/libosmo-mgcp/mgcp_codec.c
M src/libosmo-mgcp/mgcp_sdp.c
3 files changed, 18 insertions(+), 22 deletions(-)
Approvals:
Jenkins Builder: Verified
laforge: Looks good to me, approved
pespin: Looks good to me, but someone else must approve
diff --git a/include/osmocom/mgcp/mgcp_network.h b/include/osmocom/mgcp/mgcp_network.h
index d6d90dd..b9cf5e3 100644
--- a/include/osmocom/mgcp/mgcp_network.h
+++ b/include/osmocom/mgcp/mgcp_network.h
@@ -81,8 +81,8 @@
uint32_t frame_duration_den;
int payload_type;
- char *audio_name;
- char *subtype_name;
+ char audio_name[64];
+ char subtype_name[64];
bool param_present;
struct mgcp_codec_param param;
diff --git a/src/libosmo-mgcp/mgcp_codec.c b/src/libosmo-mgcp/mgcp_codec.c
index f237e38..06dcedc 100644
--- a/src/libosmo-mgcp/mgcp_codec.c
+++ b/src/libosmo-mgcp/mgcp_codec.c
@@ -89,15 +89,13 @@
.frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN,
.rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE,
.channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS,
+ .subtype_name = "",
+ .audio_name = "",
};
}
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);
*codec = (struct mgcp_rtp_codec){};
}
@@ -124,10 +122,8 @@
{
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. */
@@ -160,16 +156,16 @@
if (!audio_name) {
switch (payload_type) {
case 0:
- audio_name = talloc_strdup(ctx, "PCMU/8000/1");
+ strcpy(codec->audio_name, "PCMU/8000/1");
break;
case 3:
- audio_name = talloc_strdup(ctx, "GSM/8000/1");
+ strcpy(codec->audio_name, "GSM/8000/1");
break;
case 8:
- audio_name = talloc_strdup(ctx, "PCMA/8000/1");
+ strcpy(codec->audio_name, "PCMA/8000/1");
break;
case 18:
- audio_name = talloc_strdup(ctx, "G729/8000/1");
+ strcpy(codec->audio_name, "G729/8000/1");
break;
default:
/* The given payload type is not known to us, or it
@@ -179,36 +175,36 @@
payload_type);
goto error;
}
+ } else {
+ strncpy(codec->audio_name, audio_name, sizeof(codec->audio_name));
}
/* Now we extract the codec subtype name, rate and channels. The latter
* two are optional. If they are not present we use the safe defaults
* above. */
- if (strlen(audio_name) >= sizeof(audio_codec)) {
- LOGP(DLMGCP, LOGL_ERROR, "Audio codec too long: %s\n", osmo_quote_str(audio_name, -1));
+ if (strlen(codec->audio_name) >= sizeof(codec->subtype_name)) {
+ LOGP(DLMGCP, LOGL_ERROR, "Audio codec too long: %s\n", osmo_quote_str(codec->audio_name, -1));
goto error;
}
channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
- if (sscanf(audio_name, "%63[^/]/%d/%d", audio_codec, &rate, &channels) < 1) {
- LOGP(DLMGCP, LOGL_ERROR, "Invalid audio codec: %s\n", osmo_quote_str(audio_name, -1));
+ if (sscanf(codec->audio_name, "%63[^/]/%d/%d", codec->subtype_name, &rate, &channels) < 1) {
+ LOGP(DLMGCP, LOGL_ERROR, "Invalid audio codec: %s\n", osmo_quote_str(codec->audio_name, -1));
goto error;
}
/* Note: We only accept configurations with one audio channel! */
if (channels != 1) {
LOGP(DLMGCP, LOGL_ERROR, "Cannot handle audio codec with more than one channel: %s\n",
- osmo_quote_str(audio_name, -1));
+ osmo_quote_str(codec->audio_name, -1));
goto error;
}
codec->rate = rate;
codec->channels = channels;
- codec->subtype_name = talloc_strdup(ctx, audio_codec);
- codec->audio_name = talloc_strdup(ctx, audio_name);
codec->payload_type = payload_type;
- if (!strcmp(audio_codec, "G729")) {
+ if (!strcmp(codec->subtype_name, "G729")) {
codec->frame_duration_num = 10;
codec->frame_duration_den = 1000;
} else {
@@ -287,7 +283,7 @@
/* A codec name must be set, if not, this might mean that the codec
* (payload type) that was assigned is unknown to us so we must stop
* here. */
- if (!codec->subtype_name)
+ if (!strlen(codec->subtype_name))
return false;
/* FIXME: implement meaningful checks to make sure that the given codec
diff --git a/src/libosmo-mgcp/mgcp_sdp.c b/src/libosmo-mgcp/mgcp_sdp.c
index eabaf53..077ac96 100644
--- a/src/libosmo-mgcp/mgcp_sdp.c
+++ b/src/libosmo-mgcp/mgcp_sdp.c
@@ -439,7 +439,7 @@
for (i = 0; i < codecs_used; i++) {
LOGPC(DLMGCP, LOGL_NOTICE, "%d=%s",
rtp->codecs[i].payload_type,
- rtp->codecs[i].subtype_name ? rtp-> codecs[i].subtype_name : "unknown");
+ strlen(rtp->codecs[i].subtype_name) ? rtp->codecs[i].subtype_name : "unknown");
LOGPC(DLMGCP, LOGL_NOTICE, " ");
}
LOGPC(DLMGCP, LOGL_NOTICE, "\n");
--
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Gerrit-Project: osmo-mgw
Gerrit-Branch: master
Gerrit-Change-Id: Icc41643050a5e1ca3c66f307d60b6911ba1b8032
Gerrit-Change-Number: 25421
Gerrit-PatchSet: 8
Gerrit-Owner: Hoernchen <ewild at sysmocom.de>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: laforge <laforge at osmocom.org>
Gerrit-Reviewer: pespin <pespin at sysmocom.de>
Gerrit-CC: dexter <pmaier at sysmocom.de>
Gerrit-MessageType: merged
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