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_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
index 88de7d2..e961ba6 100644
--- a/openbsc/src/libmgcp/mgcp_transcode.c
+++ b/openbsc/src/libmgcp/mgcp_transcode.c
@@ -45,22 +45,22 @@
1) * state->src_frame_size;
}
-static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
+static enum audio_format get_audio_format(const struct mgcp_rtp_codec *codec)
{
- if (rtp_end->subtype_name) {
- if (!strcmp("GSM", rtp_end->subtype_name))
+ if (codec->subtype_name) {
+ if (!strcmp("GSM", codec->subtype_name))
return AF_GSM;
- if (!strcmp("PCMA", rtp_end->subtype_name))
+ if (!strcmp("PCMA", codec->subtype_name))
return AF_PCMA;
#ifdef HAVE_BCG729
- if (!strcmp("G729", rtp_end->subtype_name))
+ if (!strcmp("G729", codec->subtype_name))
return AF_G729;
#endif
- if (!strcmp("L16", rtp_end->subtype_name))
+ if (!strcmp("L16", codec->subtype_name))
return AF_L16;
}
- switch (rtp_end->payload_type) {
+ switch (codec->payload_type) {
case 3 /* GSM */:
return AF_GSM;
case 8 /* PCMA */:
@@ -141,6 +141,8 @@
{
struct mgcp_process_rtp_state *state;
enum audio_format src_fmt, dst_fmt;
+ const struct mgcp_rtp_codec *src_codec = &src_end->codec;
+ const struct mgcp_rtp_codec *dst_codec = &dst_end->codec;
/* cleanup first */
if (dst_end->rtp_process_data) {
@@ -151,34 +153,34 @@
if (!src_end)
return 0;
- src_fmt = get_audio_format(src_end);
- dst_fmt = get_audio_format(dst_end);
+ src_fmt = get_audio_format(src_codec);
+ dst_fmt = get_audio_format(dst_codec);
LOGP(DMGCP, LOGL_ERROR,
"Checking transcoding: %s (%d) -> %s (%d)\n",
- src_end->subtype_name, src_end->payload_type,
- dst_end->subtype_name, dst_end->payload_type);
+ src_codec->subtype_name, src_codec->payload_type,
+ dst_codec->subtype_name, dst_codec->payload_type);
if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
- if (!src_end->subtype_name || !dst_end->subtype_name)
+ if (!src_codec->subtype_name || !dst_codec->subtype_name)
/* Not enough info, do nothing */
return 0;
- if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
+ if (strcmp(src_codec->subtype_name, dst_codec->subtype_name) == 0)
/* Nothing to do */
return 0;
LOGP(DMGCP, LOGL_ERROR,
"Cannot transcode: %s codec not supported (%s -> %s).\n",
src_fmt != AF_INVALID ? "destination" : "source",
- src_end->audio_name, dst_end->audio_name);
+ src_codec->audio_name, dst_codec->audio_name);
return -EINVAL;
}
- if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
+ if (src_codec->rate && dst_codec->rate && src_codec->rate != dst_codec->rate) {
LOGP(DMGCP, LOGL_ERROR,
"Cannot transcode: rate conversion (%d -> %d) not supported.\n",
- src_end->rate, dst_end->rate);
+ src_codec->rate, dst_codec->rate);
return -EINVAL;
}
@@ -269,8 +271,8 @@
"Initialized RTP processing on: 0x%x "
"conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n",
ENDPOINT_NUMBER(endp),
- src_fmt, src_end->payload_type, src_end->rate, src_end->fmtp_extra,
- dst_fmt, dst_end->payload_type, dst_end->rate, dst_end->fmtp_extra);
+ src_fmt, src_codec->payload_type, src_codec->rate, src_end->fmtp_extra,
+ dst_fmt, dst_codec->payload_type, dst_codec->rate, dst_end->fmtp_extra);
return 0;
}
@@ -281,16 +283,19 @@
const char**fmtp_extra)
{
struct mgcp_process_rtp_state *state = endp->net_end.rtp_process_data;
- if (!state || endp->net_end.payload_type < 0) {
- *payload_type = endp->bts_end.payload_type;
- *audio_name = endp->bts_end.audio_name;
+ struct mgcp_rtp_codec *net_codec = &endp->net_end.codec;
+ struct mgcp_rtp_codec *bts_codec = &endp->bts_end.codec;
+
+ if (!state || net_codec->payload_type < 0) {
+ *payload_type = bts_codec->payload_type;
+ *audio_name = bts_codec->audio_name;
*fmtp_extra = endp->bts_end.fmtp_extra;
return;
}
- *payload_type = endp->net_end.payload_type;
+ *payload_type = net_codec->payload_type;
+ *audio_name = net_codec->audio_name;
*fmtp_extra = endp->net_end.fmtp_extra;
- *audio_name = endp->net_end.audio_name;
}
static int decode_audio(struct mgcp_process_rtp_state *state,