Merge commit 'zecke/fixes/mgcp-transcoding'
Include the last bits of fixes for the transcoding code. These fixes
and the appropriate mgcp config are known to work.
diff --git a/openbsc/include/openbsc/mgcp.h b/openbsc/include/openbsc/mgcp.h
index d4d6140..1790f84 100644
--- a/openbsc/include/openbsc/mgcp.h
+++ b/openbsc/include/openbsc/mgcp.h
@@ -87,6 +87,12 @@
typedef int (*mgcp_reset)(struct mgcp_trunk_config *cfg);
typedef int (*mgcp_rqnt)(struct mgcp_endpoint *endp, char tone);
+/**
+ * Return:
+ * < 0 in case no audio was processed
+ * >= 0 in case audio was processed. The remaining payload
+ * length will be returned.
+ */
typedef int (*mgcp_processing)(struct mgcp_endpoint *endp,
struct mgcp_rtp_end *dst_end,
char *data, int *len, int buf_size);
diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
index 296020c..4d4cec8 100644
--- a/openbsc/src/libmgcp/mgcp_transcode.c
+++ b/openbsc/src/libmgcp/mgcp_transcode.c
@@ -428,10 +428,12 @@
if (payload_len > 0) {
ts_no = ntohl(*(uint32_t*)(data+4));
- if (!state->is_running)
+ if (!state->is_running) {
state->next_seq = ntohs(*(uint16_t*)(data+2));
+ state->next_time = ts_no;
+ state->is_running = 1;
+ }
- state->is_running = 1;
if (state->sample_cnt > 0) {
int32_t delta = ts_no - state->next_time;
@@ -448,6 +450,7 @@
"0x%x dropping sample buffer due delta=%d sample_cnt=%d\n",
ENDPOINT_NUMBER(endp), delta, state->sample_cnt);
state->sample_cnt = 0;
+ state->next_time = ts_no;
} else if (delta < 0) {
LOGP(DMGCP, LOGL_NOTICE,
"RTP time jumps backwards, delta = %d, "
@@ -488,7 +491,14 @@
nsamples = state->sample_cnt;
rc = encode_audio(state, dst, buf_size, max_samples);
- if (rc <= 0)
+ /*
+ * There were no samples to encode?
+ * TODO: how does this work for comfort noise?
+ */
+ if (rc == 0)
+ return -ENOMSG;
+ /* Any other error during the encoding */
+ if (rc < 0)
return rc;
nsamples -= state->sample_cnt;
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c
index 404268a..2e857a1 100644
--- a/openbsc/tests/mgcp/mgcp_transcoding_test.c
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c
@@ -120,6 +120,24 @@
"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
},
+ /* RTP: SeqNo=26527, TS=232640 */
+ {0.020000, 92,
+ "\x80\x08\x67\x9f\x00\x03\x8c\xc0\x04\xaa\x67\x9f\xd5\xd5\xd5\xd5"
+ "\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5"
+ "\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5"
+ "\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5\xd5"
+ "\xd5\xd5\xd5\xd5\xd5\xd5\x55\x55\xd5\xd5\x55\x55\xd5\xd5\x55\x55"
+ "\xd5\xd5\xd5\x55\x55\xd5\xd5\xd5\x55\x55\xd5\xd5"
+ },
+ /* RTP: SeqNo=26528, TS=232720 */
+ {0.020000, 92,
+ "\x80\x08\x67\xa0\x00\x03\x8d\x10\x04\xaa\x67\x9f\x55\xd5\xd5\x55"
+ "\xd5\x55\xd5\xd5\xd5\x55\xd5\x55\xd5\xd5\x55\xd5\x55\xd5\x55\xd5"
+ "\x55\x55\xd5\x55\xd5\xd5\x55\x55\x55\x55\x55\xd5\xd5\x55\xd5\xd5"
+ "\xd5\x55\xd5\xd5\xd5\x55\x54\x55\xd5\xd5\x55\xd5\xd5\xd5\xd5\x55"
+ "\x54\x55\xd5\x55\xd5\x55\x55\x55\x55\x55\xd5\xd5\xd5\xd5\xd5\xd4"
+ "\xd5\x54\x55\xd5\xd4\xd5\x54\xd5\x55\xd5\xd5\xd5"
+ },
};
@@ -221,8 +239,9 @@
cont = mgcp_transcoding_process_rtp(endp, dst_end,
buf, &len, sizeof(buf));
if (cont < 0) {
- printf("processing failed: %s", strerror(-cont));
- abort();
+ printf("Nothing encoded due: %s\n", strerror(-cont));
+ talloc_free(ctx);
+ return -1;
}
if (len < 24) {
@@ -296,6 +315,124 @@
talloc_free(ctx);
}
+static void test_transcode_result(void)
+{
+ char buf[4096];
+ int len, res;
+ void *ctx;
+ struct mgcp_endpoint *endp;
+ struct mgcp_process_rtp_state *state;
+
+ {
+ /* from GSM to PCMA and same ptime */
+ given_configured_endpoint(160, 0, "gsm", "pcma", &ctx, &endp);
+ state = endp->bts_end.rtp_process_data;
+
+ /* result */
+ len = audio_packets_gsm[0].len;
+ memcpy(buf, audio_packets_gsm[0].data, len);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res == sizeof(struct rtp_hdr));
+ OSMO_ASSERT(state->sample_cnt == 0);
+
+ len = res;
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res == -ENOMSG);
+
+ talloc_free(ctx);
+ }
+
+ {
+ /* from GSM to PCMA and same ptime */
+ given_configured_endpoint(160, 160, "gsm", "pcma", &ctx, &endp);
+ state = endp->bts_end.rtp_process_data;
+
+ /* result */
+ len = audio_packets_gsm[0].len;
+ memcpy(buf, audio_packets_gsm[0].data, len);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res == sizeof(struct rtp_hdr));
+ OSMO_ASSERT(state->sample_cnt == 0);
+
+ len = res;
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res == -EAGAIN);
+
+ talloc_free(ctx);
+ }
+
+ {
+ /* from PCMA to GSM and wrong different ptime */
+ given_configured_endpoint(80, 160, "pcma", "gsm", &ctx, &endp);
+ state = endp->bts_end.rtp_process_data;
+
+ /* Add the first sample */
+ len = audio_packets_pcma[1].len;
+ memcpy(buf, audio_packets_pcma[1].data, len);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(state->sample_cnt == 80);
+ OSMO_ASSERT(state->next_time == 232640);
+ OSMO_ASSERT(res < 0);
+
+ /* Add the second sample and it should be consumable */
+ len = audio_packets_pcma[2].len;
+ memcpy(buf, audio_packets_pcma[2].data, len);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(state->sample_cnt == 0);
+ OSMO_ASSERT(state->next_time == 232640 + 80 + 160);
+ OSMO_ASSERT(res == sizeof(struct rtp_hdr));
+
+ talloc_free(ctx);
+ }
+
+ {
+ /* from PCMA to GSM with a big time jump */
+ struct rtp_hdr *hdr;
+ uint32_t ts;
+
+ given_configured_endpoint(80, 160, "pcma", "gsm", &ctx, &endp);
+ state = endp->bts_end.rtp_process_data;
+
+ /* Add the first sample */
+ len = audio_packets_pcma[1].len;
+ memcpy(buf, audio_packets_pcma[1].data, len);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(state->sample_cnt == 80);
+ OSMO_ASSERT(state->next_time == 232640);
+ OSMO_ASSERT(state->next_seq == 26527);
+ OSMO_ASSERT(res < 0);
+
+ /* Add a skip to the packet to force a 'resync' */
+ len = audio_packets_pcma[2].len;
+ memcpy(buf, audio_packets_pcma[2].data, len);
+ hdr = (struct rtp_hdr *) &buf[0];
+ /* jump the time and add alignment error */
+ ts = ntohl(hdr->timestamp) + 123 * 80 + 2;
+ hdr->timestamp = htonl(ts);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res < 0);
+ OSMO_ASSERT(state->sample_cnt == 80);
+ OSMO_ASSERT(state->next_time == ts);
+ OSMO_ASSERT(state->next_seq == 26527);
+ /* TODO: this can create alignment errors */
+
+
+ /* Now attempt to consume 160 samples */
+ len = audio_packets_pcma[2].len;
+ memcpy(buf, audio_packets_pcma[2].data, len);
+ hdr = (struct rtp_hdr *) &buf[0];
+ ts += 80;
+ hdr->timestamp = htonl(ts);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res == 12);
+ OSMO_ASSERT(state->sample_cnt == 0);
+ OSMO_ASSERT(state->next_time == ts + 160);
+ OSMO_ASSERT(state->next_seq == 26528);
+
+ talloc_free(ctx);
+ }
+}
+
static int test_repacking(int in_samples, int out_samples, int no_transcode)
{
char buf[4096] = {0x80, 0};
@@ -378,6 +515,7 @@
int main(int argc, char **argv)
{
+ int rc;
osmo_init_logging(&log_info);
printf("=== Transcoding Good Cases ===\n");
@@ -413,19 +551,22 @@
printf("=== Transcoding Bad Cases ===\n");
printf("Invalid size:\n");
- transcode_test("gsm", "pcma",
+ rc = transcode_test("gsm", "pcma",
(uint8_t *)audio_packets_gsm_invalid_size[0].data,
audio_packets_gsm_invalid_size[0].len);
+ OSMO_ASSERT(rc < 0);
printf("Invalid data:\n");
- transcode_test("gsm", "pcma",
+ rc = transcode_test("gsm", "pcma",
(uint8_t *)audio_packets_gsm_invalid_data[0].data,
audio_packets_gsm_invalid_data[0].len);
+ OSMO_ASSERT(rc < 0);
printf("Invalid payload type:\n");
- transcode_test("gsm", "pcma",
+ rc = transcode_test("gsm", "pcma",
(uint8_t *)audio_packets_gsm_invalid_ptype[0].data,
audio_packets_gsm_invalid_ptype[0].len);
+ OSMO_ASSERT(rc == 0);
printf("=== Repacking ===\n");
@@ -440,6 +581,7 @@
test_repacking(160, 100, 0);
test_repacking(160, 100, 1);
test_rtp_seq_state();
+ test_transcode_result();
return 0;
}
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.ok b/openbsc/tests/mgcp/mgcp_transcoding_test.ok
index e06b0e1..7c1c8ce 100644
--- a/openbsc/tests/mgcp/mgcp_transcoding_test.ok
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.ok
@@ -144,19 +144,11 @@
Invalid size:
== Transcoding test ==
converting gsm -> pcma
-encoded:
- 80 03 00 01 00 00 00 a0 11 22 33 44 d4 7c e3 e9
- 62 50 39 f0 f8 b4 68 ea 6c 0e 81 1b 56 2a d5 bc
- 69 9c d1 f0 66 7a ec 49 7a
-counted: 0
+Nothing encoded due: No message of desired type
Invalid data:
== Transcoding test ==
converting gsm -> pcma
-encoded:
- 80 03 00 01 00 00 00 a0 11 22 33 44 ee ee ee ee
- ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee
- ee ee ee ee ee ee ee ee ee ee ee ee ee
-counted: 0
+Nothing encoded due: No message of desired type
Invalid payload type:
== Transcoding test ==
converting gsm -> pcma