diff options
author | Jacob Erlbeck <jerlbeck@sysmocom.de> | 2014-04-08 16:10:04 +0200 |
---|---|---|
committer | Jacob Erlbeck <jerlbeck@sysmocom.de> | 2014-06-05 14:08:53 +0200 |
commit | 84a45cbf8384be753e2b83414dddc95ad63f4f2b (patch) | |
tree | 12c7ba15d70483a095b3d40fec2fb4d1c3e33566 /openbsc/tests/mgcp/mgcp_transcoding_test.c | |
parent | 07886d9b0abe81259433ab39a8a8fc86e1c197a3 (diff) |
mgcp/test: Add test cases for transcoding and repacking
This patch adds test cases for transcoding and repacking.
Sponsored-by: On-Waves ehf
Diffstat (limited to 'openbsc/tests/mgcp/mgcp_transcoding_test.c')
-rw-r--r-- | openbsc/tests/mgcp/mgcp_transcoding_test.c | 377 |
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; +} + |