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-rw-r--r--tests/mgcp/mgcp_transcoding_test.c654
1 files changed, 654 insertions, 0 deletions
diff --git a/tests/mgcp/mgcp_transcoding_test.c b/tests/mgcp/mgcp_transcoding_test.c
new file mode 100644
index 000000000..c5c0a0bab
--- /dev/null
+++ b/tests/mgcp/mgcp_transcoding_test.c
@@ -0,0 +1,654 @@
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdio.h>
+#include <string.h>
+#include <err.h>
+#include <stdint.h>
+
+#include <osmocom/core/talloc.h>
+#include <osmocom/core/application.h>
+
+#include <osmocom/netif/rtp.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 "openbsc/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"
+ },
+ /* 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"
+ },
+};
+
+
+
+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 given_configured_endpoint(int in_samples, int out_samples,
+ const char *srcfmt, const char *dstfmt,
+ void **out_ctx, struct mgcp_endpoint **out_endp)
+{
+ int rc;
+ struct mgcp_rtp_end *dst_end;
+ struct mgcp_rtp_end *src_end;
+ struct mgcp_config *cfg;
+ struct mgcp_trunk_config *tcfg;
+ struct mgcp_endpoint *endp;
+
+ cfg = mgcp_config_alloc();
+ tcfg = talloc_zero(cfg, struct mgcp_trunk_config);
+ endp = talloc_zero(tcfg, struct mgcp_endpoint);
+
+ cfg->setup_rtp_processing_cb = mgcp_transcoding_setup;
+ cfg->rtp_processing_cb = mgcp_transcoding_process_rtp;
+ cfg->get_net_downlink_format_cb = mgcp_transcoding_net_downlink_format;
+
+ tcfg->endpoints = endp;
+ tcfg->number_endpoints = 1;
+ tcfg->cfg = cfg;
+ endp->tcfg = tcfg;
+ endp->cfg = cfg;
+ mgcp_initialize_endp(endp);
+
+ dst_end = &endp->bts_end;
+ dst_end->codec.payload_type = audio_name_to_type(dstfmt);
+
+ src_end = &endp->net_end;
+ src_end->codec.payload_type = audio_name_to_type(srcfmt);
+
+ if (out_samples) {
+ dst_end->codec.frame_duration_den = dst_end->codec.rate;
+ dst_end->codec.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) {
+ printf("setup failed: %s", strerror(-rc));
+ abort();
+ }
+
+ *out_ctx = cfg;
+ *out_endp = endp;
+ return 0;
+}
+
+
+static int transcode_test(const char *srcfmt, const char *dstfmt,
+ uint8_t *src_pkts, size_t src_pkt_size)
+{
+ char buf[4096] = {0x80, 0};
+ void *ctx;
+
+ struct mgcp_rtp_end *dst_end;
+ struct mgcp_process_rtp_state *state;
+ struct mgcp_endpoint *endp;
+ int in_size;
+ const int in_samples = 160;
+ int len, cont;
+
+ printf("== Transcoding test ==\n");
+ printf("converting %s -> %s\n", srcfmt, dstfmt);
+
+ given_configured_endpoint(in_samples, 0, srcfmt, dstfmt, &ctx, &endp);
+
+ dst_end = &endp->bts_end;
+ 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) {
+ printf("Nothing encoded due: %s\n", strerror(-cont));
+ talloc_free(ctx);
+ return -1;
+ }
+
+ 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);
+ }
+ printf("counted: %d\n", cont);
+ talloc_free(ctx);
+ return 0;
+}
+
+static void test_rtp_seq_state(void)
+{
+ char buf[4096];
+ int len;
+ int cont;
+ void *ctx;
+ struct mgcp_endpoint *endp;
+ struct mgcp_process_rtp_state *state;
+ struct rtp_hdr *hdr;
+ uint32_t ts_no;
+ uint16_t seq_no;
+
+ given_configured_endpoint(160, 0, "pcma", "l16", &ctx, &endp);
+ state = endp->bts_end.rtp_process_data;
+ OSMO_ASSERT(!state->is_running);
+ OSMO_ASSERT(state->next_seq == 0);
+ OSMO_ASSERT(state->next_time == 0);
+
+ /* initialize packet */
+ len = audio_packets_pcma[0].len;
+ memcpy(buf, audio_packets_pcma[0].data, len);
+ cont = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, len);
+ OSMO_ASSERT(cont >= 0);
+ OSMO_ASSERT(state->is_running);
+ OSMO_ASSERT(state->next_seq == 2);
+ OSMO_ASSERT(state->next_time == 240);
+
+ /* verify that the right timestamp was written */
+ OSMO_ASSERT(len == audio_packets_pcma[0].len);
+ hdr = (struct rtp_hdr *) &buf[0];
+
+ memcpy(&ts_no, &hdr->timestamp, sizeof(ts_no));
+ OSMO_ASSERT(htonl(ts_no) == 160);
+ memcpy(&seq_no, &hdr->sequence, sizeof(seq_no));
+ OSMO_ASSERT(htons(seq_no) == 1);
+ /* Check the right sequence number is written */
+ state->next_seq = 1234;
+ len = audio_packets_pcma[0].len;
+ memcpy(buf, audio_packets_pcma[0].data, len);
+ cont = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, len);
+ OSMO_ASSERT(cont >= 0);
+ OSMO_ASSERT(len == audio_packets_pcma[0].len);
+ hdr = (struct rtp_hdr *) &buf[0];
+
+ memcpy(&seq_no, &hdr->sequence, sizeof(seq_no));
+ OSMO_ASSERT(htons(seq_no) == 1234);
+
+ 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 void test_transcode_change(void)
+{
+ char buf[4096] = {0x80, 0};
+ void *ctx;
+
+ struct mgcp_endpoint *endp;
+ struct mgcp_process_rtp_state *state;
+ struct rtp_hdr *hdr;
+
+ int len, res;
+
+ {
+ /* from GSM to PCMA and same ptime */
+ printf("Testing Initial L16->GSM, PCMA->GSM\n");
+ given_configured_endpoint(160, 0, "l16", "gsm", &ctx, &endp);
+ endp->net_end.alt_codec = endp->net_end.codec;
+ endp->net_end.alt_codec.payload_type = audio_name_to_type("pcma");
+ state = endp->bts_end.rtp_process_data;
+
+ /* initial transcoding work */
+ OSMO_ASSERT(state->src_fmt == AF_L16);
+ OSMO_ASSERT(state->dst_fmt == AF_GSM);
+ OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 8);
+ OSMO_ASSERT(endp->net_end.codec.payload_type == 11);
+
+ /* result */
+ len = audio_packets_pcma[0].len;
+ memcpy(buf, audio_packets_pcma[0].data, len);
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ state = endp->bts_end.rtp_process_data;
+ OSMO_ASSERT(res == sizeof(struct rtp_hdr));
+ OSMO_ASSERT(state->sample_cnt == 0);
+ OSMO_ASSERT(state->src_fmt == AF_PCMA);
+ OSMO_ASSERT(state->dst_fmt == AF_GSM);
+ OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 11);
+ OSMO_ASSERT(endp->net_end.codec.payload_type == 8);
+
+ len = res;
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(res == -ENOMSG);
+ OSMO_ASSERT(state == endp->bts_end.rtp_process_data);
+
+
+ /* now check that comfort noise doesn't change anything */
+ len = audio_packets_pcma[1].len;
+ memcpy(buf, audio_packets_pcma[1].data, len);
+ hdr = (struct rtp_hdr *) buf;
+ hdr->payload_type = 12;
+ res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+ OSMO_ASSERT(state == endp->bts_end.rtp_process_data);
+ OSMO_ASSERT(state->sample_cnt == 80);
+ OSMO_ASSERT(state->src_fmt == AF_PCMA);
+ OSMO_ASSERT(state->dst_fmt == AF_GSM);
+ OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 11);
+ OSMO_ASSERT(endp->net_end.codec.payload_type == 8);
+
+ talloc_free(ctx);
+ }
+}
+
+static int test_repacking(int in_samples, int out_samples, int no_transcode)
+{
+ char buf[4096] = {0x80, 0};
+ int cc;
+ struct mgcp_endpoint *endp;
+ void *ctx;
+
+ 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";
+
+ printf("== Transcoding test ==\n");
+ printf("converting %s -> %s\n", srcfmt, dstfmt);
+
+ given_configured_endpoint(in_samples, out_samples, srcfmt, dstfmt, &ctx, &endp);
+
+ state = endp->bts_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] = endp->net_end.codec.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, &endp->bts_end,
+ buf, &len, sizeof(buf));
+ if (cont == -EAGAIN) {
+ fprintf(stderr, "Got EAGAIN\n");
+ break;
+ }
+
+ if (cont < 0) {
+ printf("processing failed: %s", strerror(-cont));
+ abort();
+ }
+
+ len -= 12; /* ignore RTP header */
+
+ printf("got %d %s output frames (%d octets) count=%d\n",
+ len / out_size, dstfmt, len, cont);
+
+ len = cont;
+ } while (len > 0);
+ }
+
+ talloc_free(ctx);
+ return 0;
+}
+
+int main(int argc, char **argv)
+{
+ int rc;
+ 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");
+ 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");
+ 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");
+ 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");
+
+ 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);
+ test_rtp_seq_state();
+ test_transcode_result();
+ test_transcode_change();
+
+ return 0;
+}
+
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-26 04:29:09 +0000