aboutsummaryrefslogtreecommitdiffstats
path: root/openbsc/tests/mgcp/mgcp_transcoding_test.c
diff options
context:
space:
mode:
Diffstat (limited to 'openbsc/tests/mgcp/mgcp_transcoding_test.c')
-rw-r--r--openbsc/tests/mgcp/mgcp_transcoding_test.c377
1 files changed, 377 insertions, 0 deletions
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c
new file mode 100644
index 000000000..e5da13856
--- /dev/null
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c
@@ -0,0 +1,377 @@
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdio.h>
+#include <string.h>
+#include <err.h>
+
+#include <osmocom/core/talloc.h>
+#include <osmocom/core/application.h>
+
+#include <openbsc/debug.h>
+#include <openbsc/gsm_data.h>
+#include <openbsc/mgcp.h>
+#include <openbsc/mgcp_internal.h>
+
+#include "bscconfig.h"
+#ifndef BUILD_MGCP_TRANSCODING
+#error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)"
+#endif
+
+#include "src/osmo-bsc_mgcp/mgcp_transcode.h"
+
+uint8_t *audio_frame_l16[] = {
+};
+
+struct rtp_packets {
+ float t;
+ int len;
+ char *data;
+};
+
+struct rtp_packets audio_packets_l16[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 332,
+ "\x80\x0B\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ "\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+ },
+};
+
+struct rtp_packets audio_packets_gsm[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 45,
+ "\x80\x03\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\xD4\x7C\xE3\xE9\x62\x50\x39\xF0\xF8\xB4\x68\xEA\x6C\x0E\x81\x1B"
+ "\x56\x2A\xD5\xBC\x69\x9C\xD1\xF0\x66\x7A\xEC\x49\x7A\x33\x3D\x0A"
+ "\xDE"
+ },
+};
+
+struct rtp_packets audio_packets_gsm_invalid_size[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 41,
+ "\x80\x03\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\xD4\x7C\xE3\xE9\x62\x50\x39\xF0\xF8\xB4\x68\xEA\x6C\x0E\x81\x1B"
+ "\x56\x2A\xD5\xBC\x69\x9C\xD1\xF0\x66\x7A\xEC\x49\x7A\x33\x3D\x0A"
+ "\xDE"
+ },
+};
+
+struct rtp_packets audio_packets_gsm_invalid_data[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 45,
+ "\x80\x03\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE"
+ "\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE"
+ "\xEE"
+ },
+};
+
+struct rtp_packets audio_packets_gsm_invalid_ptype[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 45,
+ "\x80\x08\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\xD4\x7C\xE3\xE9\x62\x50\x39\xF0\xF8\xB4\x68\xEA\x6C\x0E\x81\x1B"
+ "\x56\x2A\xD5\xBC\x69\x9C\xD1\xF0\x66\x7A\xEC\x49\x7A\x33\x3D\x0A"
+ "\xDE"
+ },
+};
+
+struct rtp_packets audio_packets_g729[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 32,
+ "\x80\x12\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\xAF\xC2\x81\x40\x00\xFA\xCE\xA4\x21\x7C\xC5\xC3\x4F\xA5\x98\xF5"
+ "\xB2\x95\xC4\xAD"
+ },
+};
+
+struct rtp_packets audio_packets_pcma[] = {
+ /* RTP: SeqNo=1, TS=160 */
+ {0.020000, 172,
+ "\x80\x08\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+ "\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"
+ "\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"
+ "\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"
+ "\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"
+ "\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"
+ },
+};
+
+
+
+static int audio_name_to_type(const char *name)
+{
+ if (!strcasecmp(name, "gsm"))
+ return 3;
+#ifdef HAVE_BCG729
+ else if (!strcasecmp(name, "g729"))
+ return 18;
+#endif
+ else if (!strcasecmp(name, "pcma"))
+ return 8;
+ else if (!strcasecmp(name, "l16"))
+ return 11;
+ return -1;
+}
+
+int mgcp_get_trans_frame_size(void *state_, int nsamples, int dst);
+
+static int transcode_test(const char *srcfmt, const char *dstfmt,
+ uint8_t *src_pkts, size_t src_pkt_size)
+{
+ char buf[4096] = {0x80, 0};
+ int rc;
+ 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;
+ int in_size;
+ int in_samples = 160;
+ int len, cont;
+
+ printf("== Transcoding test ==\n");
+ printf("converting %s -> %s\n", srcfmt, dstfmt);
+
+ tcfg.endpoints = &endp;
+ tcfg.number_endpoints = 1;
+ endp.tcfg = &tcfg;
+ mgcp_free_endp(&endp);
+
+ dst_end = &endp.bts_end;
+ src_end = &endp.net_end;
+
+ src_end->payload_type = audio_name_to_type(srcfmt);
+ dst_end->payload_type = audio_name_to_type(dstfmt);
+
+ 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;
+ OSMO_ASSERT(state != NULL);
+
+ in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
+ OSMO_ASSERT(sizeof(buf) >= in_size + 12);
+
+ memcpy(buf, src_pkts, src_pkt_size);
+
+ len = src_pkt_size;
+
+ cont = mgcp_transcoding_process_rtp(&endp, dst_end,
+ buf, &len, sizeof(buf));
+ if (cont < 0)
+ errx(1, "processing failed: %s", strerror(-cont));
+
+ if (len < 24) {
+ printf("encoded: %s\n", osmo_hexdump((unsigned char *)buf, len));
+ } else {
+ const char *str = osmo_hexdump((unsigned char *)buf, len);
+ int i = 0;
+ const int prefix = 4;
+ const int cutlen = 48;
+ int nchars = 0;
+
+ printf("encoded:\n");
+ do {
+ nchars = printf("%*s%-.*s", prefix, "", cutlen, str + i);
+ i += nchars - prefix;
+ printf("\n");
+ } while (nchars - prefix >= cutlen);
+ }
+ return 0;
+}
+
+static int test_repacking(int in_samples, int out_samples, int no_transcode)
+{
+ char buf[4096] = {0x80, 0};
+ int cc, rc;
+ struct mgcp_rtp_end *dst_end;
+ struct mgcp_rtp_end *src_end;
+ struct mgcp_config *cfg;
+ struct mgcp_trunk_config tcfg = {{0}};
+ struct mgcp_endpoint endp = {0};
+ struct mgcp_process_rtp_state *state;
+ int in_cnt;
+ int out_size;
+ int in_size;
+ uint32_t ts = 0;
+ uint16_t seq = 0;
+ const char *srcfmt = "pcma";
+ const char *dstfmt = no_transcode ? "pcma" : "l16";
+
+ cfg = mgcp_config_alloc();
+
+ tcfg.endpoints = &endp;
+ tcfg.number_endpoints = 1;
+ tcfg.cfg = cfg;
+ endp.tcfg = &tcfg;
+ endp.cfg = cfg;
+ mgcp_free_endp(&endp);
+
+ dst_end = &endp.bts_end;
+ src_end = &endp.net_end;
+
+ printf("== Transcoding test ==\n");
+ printf("converting %s -> %s\n", srcfmt, dstfmt);
+
+ src_end->payload_type = audio_name_to_type(srcfmt);
+ dst_end->payload_type = audio_name_to_type(dstfmt);
+
+ if (out_samples) {
+ dst_end->frame_duration_den = dst_end->rate;
+ dst_end->frame_duration_num = out_samples;
+ dst_end->frames_per_packet = 1;
+ dst_end->force_output_ptime = 1;
+ }
+
+ 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;
+ OSMO_ASSERT(state != NULL);
+
+ in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
+ OSMO_ASSERT(sizeof(buf) >= in_size + 12);
+
+ out_size = mgcp_transcoding_get_frame_size(state, -1, 1);
+ OSMO_ASSERT(sizeof(buf) >= out_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);
+
+ for (in_cnt = 0; in_cnt < 16; in_cnt++) {
+ int cont;
+ int len;
+
+ /* fake PCMA data */
+ printf("generating %d %s input samples\n", in_samples, srcfmt);
+ for (cc = 0; cc < in_samples; cc++)
+ buf[12+cc] = cc;
+
+ *(uint16_t*)(buf+2) = htonl(seq);
+ *(uint32_t*)(buf+4) = htonl(ts);
+
+ seq += 1;
+ ts += in_samples;
+
+ cc += 12; /* include RTP header */
+
+ 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 */
+
+ printf("got %d %s output frames (%d octets)\n",
+ len / out_size, dstfmt, len);
+
+ len = cont;
+ } while (len > 0);
+ }
+ return 0;
+}
+
+int main(int argc, char **argv)
+{
+ osmo_init_logging(&log_info);
+
+ printf("=== Transcoding Good Cases ===\n");
+
+ transcode_test("l16", "l16",
+ (uint8_t *)audio_packets_l16[0].data,
+ audio_packets_l16[0].len);
+ transcode_test("l16", "gsm",
+ (uint8_t *)audio_packets_l16[0].data,
+ audio_packets_l16[0].len);
+ transcode_test("l16", "pcma",
+ (uint8_t *)audio_packets_l16[0].data,
+ audio_packets_l16[0].len);
+ transcode_test("gsm", "l16",
+ (uint8_t *)audio_packets_gsm[0].data,
+ audio_packets_gsm[0].len);
+ transcode_test("gsm", "gsm",
+ (uint8_t *)audio_packets_gsm[0].data,
+ audio_packets_gsm[0].len);
+ transcode_test("gsm", "pcma",
+ (uint8_t *)audio_packets_gsm[0].data,
+ audio_packets_gsm[0].len);
+ transcode_test("pcma", "l16",
+ (uint8_t *)audio_packets_pcma[0].data,
+ audio_packets_pcma[0].len);
+ transcode_test("pcma", "gsm",
+ (uint8_t *)audio_packets_pcma[0].data,
+ audio_packets_pcma[0].len);
+ transcode_test("pcma", "pcma",
+ (uint8_t *)audio_packets_pcma[0].data,
+ audio_packets_pcma[0].len);
+
+ printf("=== Transcoding Bad Cases ===\n");
+
+ printf("Invalid size:\n");
+ transcode_test("gsm", "pcma",
+ (uint8_t *)audio_packets_gsm_invalid_size[0].data,
+ audio_packets_gsm_invalid_size[0].len);
+
+ printf("Invalid data:\n");
+ transcode_test("gsm", "pcma",
+ (uint8_t *)audio_packets_gsm_invalid_data[0].data,
+ audio_packets_gsm_invalid_data[0].len);
+
+ printf("Invalid payload type:\n");
+ transcode_test("gsm", "pcma",
+ (uint8_t *)audio_packets_gsm_invalid_ptype[0].data,
+ audio_packets_gsm_invalid_ptype[0].len);
+
+ printf("=== Repacking ===\n");
+
+ test_repacking(160, 160, 0);
+ test_repacking(160, 160, 1);
+ test_repacking(160, 80, 0);
+ test_repacking(160, 80, 1);
+ test_repacking(160, 320, 0);
+ test_repacking(160, 320, 1);
+ test_repacking(160, 240, 0);
+ test_repacking(160, 240, 1);
+ test_repacking(160, 100, 0);
+ test_repacking(160, 100, 1);
+
+ return 0;
+}
+
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-13 23:56:22 +0000