mgcp: Add packet size (ptime) conversion

The current transcoder implemenation always does a 1:1 recoding
concerning the duration of a packet. So RTP timestamps and sequence
numbers are not modified.

This is not sufficient in some cases, e.g. when the BTS does only
allow for a single fixed ptime.

This patch decouples encoding from decoding and moves the decoded
samples to the state structure so that samples can be combined or
drain according to the packaging of incoming and outgoing packets.

This patch incorporates parts of Holger's experimental fixes in
0e669e05^..9eba68f9.

Ticket: OW#1111
Sponsored-by: On-Waves ehf

1 files changed, 38 insertions, 14 deletions

diff --git a/openbsc/contrib/testconv/testconv_main.c b/openbsc/contrib/testconv/testconv_main.c
index c2785f23e..aee73048c 100644
--- a/openbsc/contrib/testconv/testconv_main.c
+++ b/openbsc/contrib/testconv/testconv_main.c
@@ -38,10 +38,10 @@ int mgcp_get_trans_frame_size(void *state_, int nsamples, int dst);
 
 int main(int argc, char **argv)
 {
-	char buf[4096] = {0};
+	char buf[4096] = {0x80, 0};
 	int cc, rc;
-	struct mgcp_rtp_end dst_end = {0};
-	struct mgcp_rtp_end src_end = {0};
+	struct mgcp_rtp_end *dst_end;
+	struct mgcp_rtp_end *src_end;
 	struct mgcp_trunk_config tcfg = {{0}};
 	struct mgcp_endpoint endp = {0};
 	struct mgcp_process_rtp_state *state;
@@ -52,39 +52,63 @@ int main(int argc, char **argv)
 	tcfg.endpoints = &endp;
 	tcfg.number_endpoints = 1;
 	endp.tcfg = &tcfg;
+	mgcp_free_endp(&endp);
+
+	dst_end = &endp.bts_end;
+	src_end = &endp.net_end;
 
 	if (argc <= 2)
 		errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16}");
 
-	if ((src_end.payload_type = audio_name_to_type(argv[1])) == -1)
+	if ((src_end->payload_type = audio_name_to_type(argv[1])) == -1)
 		errx(1, "invalid input format '%s'", argv[1]);
-	if ((dst_end.payload_type = audio_name_to_type(argv[2])) == -1)
+	if ((dst_end->payload_type = audio_name_to_type(argv[2])) == -1)
 		errx(1, "invalid output format '%s'", argv[2]);
 
-	rc = mgcp_transcoding_setup(&endp, &dst_end, &src_end);
+	rc = mgcp_transcoding_setup(&endp, dst_end, src_end);
 	if (rc < 0)
 		errx(1, "setup failed: %s", strerror(-rc));
 
-	state = dst_end.rtp_process_data;
+	state = dst_end->rtp_process_data;
 	OSMO_ASSERT(state != NULL);
 
 	in_size = mgcp_transcoding_get_frame_size(state, 160, 0);
 	OSMO_ASSERT(sizeof(buf) >= in_size + 12);
 
+	buf[1] = src_end->payload_type;
+	*(uint16_t*)(buf+2) = htons(1);
+	*(uint32_t*)(buf+4) = htonl(0);
+	*(uint32_t*)(buf+8) = htonl(0xaabbccdd);
+
 	while ((cc = read(0, buf + 12, in_size))) {
+		int cont;
+		int len;
+
 		if (cc != in_size)
 			err(1, "read");
 
 		cc += 12; /* include RTP header */
 
-		rc = mgcp_transcoding_process_rtp(&endp, &dst_end,
-						  buf, &cc, sizeof(buf));
-		if (rc < 0)
-			errx(1, "processing failed: %s", strerror(-rc));
+		len = cc;
+
+		do {
+			cont = mgcp_transcoding_process_rtp(&endp, dst_end,
+							buf, &len, sizeof(buf));
+			if (cont == -EAGAIN) {
+				fprintf(stderr, "Got EAGAIN\n");
+				break;
+			}
+
+			if (cont < 0)
+				errx(1, "processing failed: %s", strerror(-cont));
+
+			len -= 12; /* ignore RTP header */
+
+			if (write(1, buf + 12, len) != len)
+				err(1, "write");
 
-		cc -= 12; /* ignore RTP header */
-		if (write(1, buf + 12, cc) != cc)
-			err(1, "write");
+			len = cont;
+		} while (len > 0);
 	}
 	return 0;
 }