mgcp: Move the "codec" params to a struct
We might be offered multiple codecs by the remote and need to
switch between them once we receive data. Do this by moving it
to a struct so we can separate between proposed and current
codec. In SDP we can have multiple codecs but a global ptime.
The current code doesn't separate that clearly instead we write
it to the main codec.
diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c
index ae275a8..d2a9f09 100644
--- a/openbsc/src/libmgcp/mgcp_protocol.c
+++ b/openbsc/src/libmgcp/mgcp_protocol.c
@@ -598,20 +598,20 @@
return ret;
}
-static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
+static int set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
int payload_type, const char *audio_name)
{
- int rate = rtp->rate;
- int channels = rtp->channels;
+ int rate = codec->rate;
+ int channels = codec->channels;
char audio_codec[64];
- talloc_free(rtp->subtype_name);
- rtp->subtype_name = NULL;
- talloc_free(rtp->audio_name);
- rtp->audio_name = NULL;
+ talloc_free(codec->subtype_name);
+ codec->subtype_name = NULL;
+ talloc_free(codec->audio_name);
+ codec->audio_name = NULL;
if (payload_type != PTYPE_UNDEFINED)
- rtp->payload_type = payload_type;
+ codec->payload_type = payload_type;
if (!audio_name) {
switch (payload_type) {
@@ -630,17 +630,17 @@
audio_codec, &rate, &channels) < 1)
return -EINVAL;
- rtp->rate = rate;
- rtp->channels = channels;
- rtp->subtype_name = talloc_strdup(ctx, audio_codec);
- rtp->audio_name = talloc_strdup(ctx, audio_name);
+ codec->rate = rate;
+ codec->channels = channels;
+ codec->subtype_name = talloc_strdup(ctx, audio_codec);
+ codec->audio_name = talloc_strdup(ctx, audio_name);
if (!strcmp(audio_codec, "G729")) {
- rtp->frame_duration_num = 10;
- rtp->frame_duration_den = 1000;
+ codec->frame_duration_num = 10;
+ codec->frame_duration_den = 1000;
} else {
- rtp->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
- rtp->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
+ codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
+ codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
}
if (payload_type < 0) {
@@ -654,7 +654,7 @@
payload_type = 18;
}
- rtp->payload_type = payload_type;
+ codec->payload_type = payload_type;
}
if (channels != 1)
@@ -761,7 +761,7 @@
if (payload != audio_payload)
break;
- set_audio_info(p->cfg, rtp, payload, audio_name);
+ set_audio_info(p->cfg, &rtp->codec, payload, audio_name);
} else if (sscanf(line, "a=ptime:%d-%d",
&ptime, &ptime2) >= 1) {
if (ptime2 > 0 && ptime2 != ptime)
@@ -769,8 +769,8 @@
else
rtp->packet_duration_ms = ptime;
} else if (sscanf(line, "a=maxptime:%d", &ptime2) == 1) {
- if (ptime2 * rtp->frame_duration_den >
- rtp->frame_duration_num * 1500)
+ if (ptime2 * rtp->codec.frame_duration_den >
+ rtp->codec.frame_duration_num * 1500)
/* more than 1 frame */
rtp->packet_duration_ms = 0;
}
@@ -785,7 +785,7 @@
rtp->rtp_port = htons(port);
rtp->rtcp_port = htons(port + 1);
found_media = 1;
- set_audio_info(p->cfg, rtp, audio_payload, NULL);
+ set_audio_info(p->cfg, &rtp->codec, audio_payload, NULL);
}
break;
}
@@ -814,8 +814,8 @@
LOGP(DMGCP, LOGL_NOTICE,
"Got media info via SDP: port %d, payload %d (%s), "
"duration %d, addr %s\n",
- ntohs(rtp->rtp_port), rtp->payload_type,
- rtp->subtype_name ? rtp->subtype_name : "unknown",
+ ntohs(rtp->rtp_port), rtp->codec.payload_type,
+ rtp->codec.subtype_name ? rtp->codec.subtype_name : "unknown",
rtp->packet_duration_ms, inet_ntoa(rtp->addr));
return found_media;
@@ -872,13 +872,13 @@
/* Get the number of frames per channel and packet */
if (rtp->frames_per_packet)
f = rtp->frames_per_packet;
- else if (rtp->packet_duration_ms && rtp->frame_duration_num) {
- int den = 1000 * rtp->frame_duration_num;
- f = (rtp->packet_duration_ms * rtp->frame_duration_den + den/2)
+ else if (rtp->packet_duration_ms && rtp->codec.frame_duration_num) {
+ int den = 1000 * rtp->codec.frame_duration_num;
+ f = (rtp->packet_duration_ms * rtp->codec.frame_duration_den + den/2)
/ den;
}
- return rtp->rate * f * rtp->frame_duration_num / rtp->frame_duration_den;
+ return rtp->codec.rate * f * rtp->codec.frame_duration_num / rtp->codec.frame_duration_den;
}
static int mgcp_parse_osmux_cid(const char *line)
@@ -1025,13 +1025,13 @@
endp->allocated = 1;
/* set up RTP media parameters */
- set_audio_info(p->cfg, &endp->bts_end, tcfg->audio_payload, tcfg->audio_name);
+ set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints,
tcfg->audio_fmtp_extra);
if (have_sdp)
parse_sdp_data(&endp->net_end, p);
else if (endp->local_options.codec)
- set_audio_info(p->cfg, &endp->net_end,
+ set_audio_info(p->cfg, &endp->net_end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
if (p->cfg->bts_force_ptime) {
@@ -1147,7 +1147,7 @@
local_options);
if (!have_sdp && endp->local_options.codec)
- set_audio_info(p->cfg, &endp->net_end,
+ set_audio_info(p->cfg, &endp->net_end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
setup_rtp_processing(endp);
@@ -1461,24 +1461,24 @@
end->dropped_packets = 0;
memset(&end->addr, 0, sizeof(end->addr));
end->rtp_port = end->rtcp_port = 0;
- end->payload_type = -1;
+ end->codec.payload_type = -1;
end->local_alloc = -1;
talloc_free(end->fmtp_extra);
end->fmtp_extra = NULL;
- talloc_free(end->subtype_name);
- end->subtype_name = NULL;
- talloc_free(end->audio_name);
- end->audio_name = NULL;
+ talloc_free(end->codec.subtype_name);
+ end->codec.subtype_name = NULL;
+ talloc_free(end->codec.audio_name);
+ end->codec.audio_name = NULL;
talloc_free(end->rtp_process_data);
end->rtp_process_data = NULL;
/* Set default values */
- end->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
- end->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
+ end->codec.frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
+ end->codec.frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
end->frames_per_packet = 0; /* unknown */
end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
- end->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
- end->channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
+ end->codec.rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
+ end->codec.channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
end->output_enabled = 0;
}