transaction: move cc.codecs.result -> cc.local

Prepare for CSD where this will be used too.

Related: OS#4394
Change-Id: Iaf954be0455625faa06a64c19905b79b7045f8e4

7 files changed, 41 insertions, 46 deletions

diff --git a/include/osmocom/msc/codec_filter.h b/include/osmocom/msc/codec_filter.h
index c0d8f329c..da4a67e04 100644
--- a/include/osmocom/msc/codec_filter.h
+++ b/include/osmocom/msc/codec_filter.h
@@ -42,20 +42,16 @@ struct codec_filter {
 
 	/* After a channel was assigned, this reflects the chosen codec. */
 	struct sdp_audio_codec assignment;
-
-	/* Resulting choice of supported codecs, usually the intersection of the above,
-	 * and the local RTP address to be sent to the remote call leg.
-	 * The RTP address:port in result.rtp is not modified by codec_filter_run() -- set it once. */
-	struct sdp_msg result;
 };
 
 void codec_filter_set_ran(struct codec_filter *codec_filter, enum osmo_rat_type ran_type);
 void codec_filter_set_bss(struct codec_filter *codec_filter,
 			  const struct gsm0808_speech_codec_list *codec_list_bss_supported);
-void codec_filter_set_local_rtp(struct codec_filter *codec_filter, const struct osmo_sockaddr_str *rtp);
-int codec_filter_run(struct codec_filter *codec_filter, const struct sdp_msg *remote);
+int codec_filter_run(struct codec_filter *codec_filter, struct sdp_msg *result, const struct sdp_msg *remote);
 
 int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter,
+			    const struct sdp_msg *result, const struct sdp_msg *remote);
+char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *result,
 			    const struct sdp_msg *remote);
-char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *remote);
-const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *remote);
+const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *result,
+				const struct sdp_msg *remote);
diff --git a/include/osmocom/msc/transaction.h b/include/osmocom/msc/transaction.h
index bfb32ad4f..36aab7874 100644
--- a/include/osmocom/msc/transaction.h
+++ b/include/osmocom/msc/transaction.h
@@ -109,6 +109,9 @@ struct gsm_trans {
 			struct sdp_msg remote;
 			/* Track codec choices from BSS and remote call leg */
 			struct codec_filter codecs;
+			/* Resulting choice from codecs/bearer services and the
+			 * local RTP address to be sent to the remote call leg. */
+			struct sdp_msg local;
 		} cc;
 		struct {
 			struct gsm411_smc_inst smc_inst;
diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c
index 38a12464a..a9d93a70d 100644
--- a/src/libmsc/codec_filter.c
+++ b/src/libmsc/codec_filter.c
@@ -84,19 +84,11 @@ void codec_filter_set_bss(struct codec_filter *codec_filter,
 		sdp_audio_codecs_from_speech_codec_list(&codec_filter->bss, codec_list_bss_supported);
 }
 
-void codec_filter_set_local_rtp(struct codec_filter *codec_filter, const struct osmo_sockaddr_str *rtp)
-{
-	if (!rtp)
-		codec_filter->result.rtp = (struct osmo_sockaddr_str){0};
-	else
-		codec_filter->result.rtp = *rtp;
-}
-
 /* Render intersections of all known audio codec constraints to reach a resulting choice of favorite audio codec, plus
  * possible set of alternative audio codecs, in codec_filter->result. (The result.rtp address remains unchanged.) */
-int codec_filter_run(struct codec_filter *codec_filter, const struct sdp_msg *remote)
+int codec_filter_run(struct codec_filter *codec_filter, struct sdp_msg *result, const struct sdp_msg *remote)
 {
-	struct sdp_audio_codecs *r = &codec_filter->result.audio_codecs;
+	struct sdp_audio_codecs *r = &result->audio_codecs;
 	struct sdp_audio_codec *a = &codec_filter->assignment;
 	*r = codec_filter->ran;
 	if (codec_filter->ms.count)
@@ -150,10 +142,10 @@ int codec_filter_run(struct codec_filter *codec_filter, const struct sdp_msg *re
 }
 
 int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter *codec_filter,
-			    const struct sdp_msg *remote)
+			    const struct sdp_msg *result, const struct sdp_msg *remote)
 {
 	struct osmo_strbuf sb = { .buf = buf, .len = buflen };
-	OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, &codec_filter->result);
+	OSMO_STRBUF_APPEND(sb, sdp_msg_to_str_buf, result);
 	OSMO_STRBUF_PRINTF(sb, " (from:");
 
 	if (sdp_audio_codec_is_set(&codec_filter->assignment)) {
@@ -188,12 +180,14 @@ int codec_filter_to_str_buf(char *buf, size_t buflen, const struct codec_filter
 	return sb.chars_needed;
 }
 
-char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *remote)
+char *codec_filter_to_str_c(void *ctx, const struct codec_filter *codec_filter, const struct sdp_msg *result,
+			    const struct sdp_msg *remote)
 {
-	OSMO_NAME_C_IMPL(ctx, 128, "codec_filter_to_str_c-ERROR", codec_filter_to_str_buf, codec_filter, remote)
+	OSMO_NAME_C_IMPL(ctx, 128, "codec_filter_to_str_c-ERROR", codec_filter_to_str_buf, codec_filter, result, remote)
 }
 
-const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *remote)
+const char *codec_filter_to_str(const struct codec_filter *codec_filter, const struct sdp_msg *result,
+				const struct sdp_msg *remote)
 {
-	return codec_filter_to_str_c(OTC_SELECT, codec_filter, remote);
+	return codec_filter_to_str_c(OTC_SELECT, codec_filter, result, remote);
 }
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 70a0b7726..77090ca76 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -731,9 +731,9 @@ void gsm48_cc_rx_setup_cn_local_rtp_port_known(struct gsm_trans *trans)
 		trans_free(trans);
 		return;
 	}
-	codec_filter_set_local_rtp(&trans->cc.codecs, rtp_cn_local);
+	trans->cc.local.rtp = *rtp_cn_local;
 
-	sdp = trans->cc.codecs.result.audio_codecs.count ? &trans->cc.codecs.result : NULL;
+	sdp = trans->cc.local.audio_codecs.count ? &trans->cc.local : NULL;
 	rc = sdp_msg_to_sdp_str_buf(setup.sdp, sizeof(setup.sdp), sdp);
 	if (rc >= sizeof(setup.sdp)) {
 		LOG_TRANS(trans, LOGL_ERROR, "MNCC_SETUP_IND: SDP too long (%d > %zu bytes)\n", rc, sizeof(setup.sdp));
@@ -829,7 +829,7 @@ static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg)
 	bearer_cap = (struct gsm_mncc_bearer_cap){
 		.speech_ver = { -1 },
 	};
-	sdp_audio_codecs_to_bearer_cap(&bearer_cap, &trans->cc.codecs.result.audio_codecs);
+	sdp_audio_codecs_to_bearer_cap(&bearer_cap, &trans->cc.local.audio_codecs);
 	rc = bearer_cap_set_radio(&bearer_cap);
 	if (rc) {
 		LOG_TRANS(trans, LOGL_ERROR, "Error composing Bearer Capability for CC Setup\n");
@@ -844,7 +844,8 @@ static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg)
 	 * finding a matching codec. */
 	if (bearer_cap.speech_ver[0] == -1) {
 		LOG_TRANS(trans, LOGL_ERROR, "%s: no codec match possible: %s\n",
-			  get_mncc_name(setup->msg_type), codec_filter_to_str(&trans->cc.codecs, &trans->cc.remote));
+			  get_mncc_name(setup->msg_type),
+			  codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote));
 
 		/* incompatible codecs */
 		rc = mncc_release_ind(trans->net, trans, trans->callref,
@@ -978,7 +979,7 @@ static int gsm48_cc_mt_rtp_port_and_codec_known(struct gsm_trans *trans)
 		trans_free(trans);
 		return -EINVAL;
 	}
-	codec_filter_set_local_rtp(&trans->cc.codecs, rtp_cn_local);
+	trans->cc.local.rtp = *rtp_cn_local;
 
 	trans_cc_filter_run(trans);
 
@@ -989,7 +990,7 @@ static int gsm48_cc_mt_rtp_port_and_codec_known(struct gsm_trans *trans)
 	}
 
 	return mncc_recv_rtp(msc_a_net(msc_a), trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local, 0, 0,
-			     &trans->cc.codecs.result);
+			     &trans->cc.local);
 }
 
 static int gsm48_cc_tx_call_proc_and_assign(struct gsm_trans *trans, void *arg)
@@ -1060,7 +1061,7 @@ static int gsm48_cc_rx_alerting(struct gsm_trans *trans, struct msgb *msg)
 	new_cc_state(trans, GSM_CSTATE_CALL_RECEIVED);
 
 	trans_cc_filter_run(trans);
-	rc = sdp_msg_to_sdp_str_buf(alerting.sdp, sizeof(alerting.sdp), &trans->cc.codecs.result);
+	rc = sdp_msg_to_sdp_str_buf(alerting.sdp, sizeof(alerting.sdp), &trans->cc.local);
 	if (rc >= sizeof(alerting.sdp)) {
 		LOG_TRANS(trans, LOGL_ERROR, "MNCC_ALERT_IND: SDP too long (%d > %zu bytes)\n",
 			  rc, sizeof(alerting.sdp));
@@ -1206,7 +1207,7 @@ static int gsm48_cc_rx_connect(struct gsm_trans *trans, struct msgb *msg)
 	rate_ctr_inc(rate_ctr_group_get_ctr(trans->net->msc_ctrs, MSC_CTR_CALL_MT_CONNECT));
 
 	trans_cc_filter_run(trans);
-	sdp_msg_to_sdp_str_buf(connect.sdp, sizeof(connect.sdp), &trans->cc.codecs.result);
+	sdp_msg_to_sdp_str_buf(connect.sdp, sizeof(connect.sdp), &trans->cc.local);
 	return mncc_recvmsg(trans->net, trans, MNCC_SETUP_CNF, &connect);
 }
 
@@ -2044,7 +2045,7 @@ int gsm48_tch_rtp_create(struct gsm_trans *trans)
 	}
 
 	trans_cc_filter_run(trans);
-	codecs = &trans->cc.codecs.result.audio_codecs;
+	codecs = &trans->cc.local.audio_codecs;
 	if (!codecs->count) {
 		LOG_TRANS_CAT(trans, DMNCC, LOGL_ERROR,
 			      "Cannot RTP CREATE to MNCC, there is no codec available\n");
@@ -2063,7 +2064,7 @@ int gsm48_tch_rtp_create(struct gsm_trans *trans)
 	}
 
 	return mncc_recv_rtp(net, trans, trans->callref, MNCC_RTP_CREATE, rtp_cn_local,
-			     codec->payload_type, mncc_payload_msg_type, &trans->cc.codecs.result);
+			     codec->payload_type, mncc_payload_msg_type, &trans->cc.local);
 }
 
 static int tch_rtp_connect(struct gsm_network *net, const struct gsm_mncc_rtp *rtp)
diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c
index b8a5de1c8..ca3820626 100644
--- a/src/libmsc/msc_a.c
+++ b/src/libmsc/msc_a.c
@@ -640,18 +640,18 @@ static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a)
 	trans_cc_filter_run(cc_trans);
 	LOG_TRANS(cc_trans, LOGL_DEBUG, "Sending Assignment Command\n");
 
-	if (!cc_trans->cc.codecs.result.audio_codecs.count) {
+	if (!cc_trans->cc.local.audio_codecs.count) {
 		LOG_TRANS(cc_trans, LOGL_ERROR, "Assignment not possible, no matching codec: %s\n",
-			  codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.remote));
+			  codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.local, &cc_trans->cc.remote));
 		call_leg_release(msc_a->cc.call_leg);
 		return;
 	}
 
 	/* Compose 48.008 Channel Type from the current set of codecs determined from both local and remote codec
 	 * capabilities. */
-	if (sdp_audio_codecs_to_gsm0808_channel_type(&channel_type, &cc_trans->cc.codecs.result.audio_codecs)) {
+	if (sdp_audio_codecs_to_gsm0808_channel_type(&channel_type, &cc_trans->cc.local.audio_codecs)) {
 		LOG_MSC_A(msc_a, LOGL_ERROR, "Cannot compose Channel Type (Permitted Speech) from codecs: %s\n",
-			  codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.remote));
+			  codec_filter_to_str(&cc_trans->cc.codecs, &cc_trans->cc.local, &cc_trans->cc.remote));
 		trans_free(cc_trans);
 		return;
 	}
@@ -1455,7 +1455,7 @@ static void msc_a_up_call_assignment_complete(struct msc_a *msc_a, const struct
 	trans_cc_filter_run(cc_trans);
 	LOG_TRANS(cc_trans, LOGL_INFO, "Assignment Complete: RAN: %s, CN: %s\n",
 		  sdp_audio_codecs_to_str(&rtps_to_ran->codecs),
-		  sdp_audio_codecs_to_str(&cc_trans->cc.codecs.result.audio_codecs));
+		  sdp_audio_codecs_to_str(&cc_trans->cc.local.audio_codecs));
 
 	if (cc_on_assignment_done(cc_trans)) {
 		/* If an error occurred, it was logged in cc_assignment_done() */
@@ -1874,13 +1874,13 @@ static int msc_a_start_assignment(struct msc_a *msc_a, struct gsm_trans *cc_tran
 	 * issued first here will also be the first CRCX sent to the MGW. Usually both still need to be set up. */
 	if (!cn_rtp_available)
 		call_leg_ensure_ci(cl, RTP_TO_CN, cc_trans->callref, cc_trans,
-				   &cc_trans->cc.codecs.result.audio_codecs, NULL);
+				   &cc_trans->cc.local.audio_codecs, NULL);
 	if (!ran_rtp_available) {
 		struct sdp_audio_codecs *codecs;
 		if (msc_a->c.ran->force_mgw_codecs_to_ran.count)
 			codecs = &msc_a->c.ran->force_mgw_codecs_to_ran;
 		else
-			codecs = &cc_trans->cc.codecs.result.audio_codecs;
+			codecs = &cc_trans->cc.local.audio_codecs;
 		return call_leg_ensure_ci(cl, RTP_TO_RAN, cc_trans->callref, cc_trans, codecs, NULL);
 	}
 
diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c
index 746728762..2a5891aea 100644
--- a/src/libmsc/msc_ho.c
+++ b/src/libmsc/msc_ho.c
@@ -416,14 +416,14 @@ static void msc_ho_send_handover_request(struct msc_a *msc_a)
 
 	if (cc_trans) {
 		if (sdp_audio_codecs_to_gsm0808_channel_type(&channel_type,
-							     &cc_trans->cc.codecs.result.audio_codecs)) {
+							     &cc_trans->cc.local.audio_codecs)) {
 			msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE,
 				      "Failed to determine Channel Type for Handover Request message\n");
 			return;
 		}
 		ran_enc_msg.handover_request.geran.channel_type = &channel_type;
 
-		sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.codecs.result.audio_codecs);
+		sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs);
 		if (!scl.len) {
 			msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE, "Failed to compose"
 				      " Codec List (MSC Preferred) for Handover Request message\n");
diff --git a/src/libmsc/transaction_cc.c b/src/libmsc/transaction_cc.c
index 4c30f84bb..d221d7cf4 100644
--- a/src/libmsc/transaction_cc.c
+++ b/src/libmsc/transaction_cc.c
@@ -42,8 +42,9 @@ void trans_cc_filter_set_bss(struct gsm_trans *trans, struct msc_a *msc_a)
 
 void trans_cc_filter_run(struct gsm_trans *trans)
 {
-	codec_filter_run(&trans->cc.codecs, &trans->cc.remote);
-	LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n", codec_filter_to_str(&trans->cc.codecs, &trans->cc.remote));
+	codec_filter_run(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote);
+	LOG_TRANS(trans, LOGL_DEBUG, "codecs: %s\n",
+		  codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote));
 }
 
 void trans_cc_filter_set_ms_from_bc(struct gsm_trans *trans, const struct gsm_mncc_bearer_cap *bcap)