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-rw-r--r--tests/mgcp/mgcp_transcoding_test.c654
1 files changed, 0 insertions, 654 deletions
diff --git a/tests/mgcp/mgcp_transcoding_test.c b/tests/mgcp/mgcp_transcoding_test.c
deleted file mode 100644
index c5c0a0bab..000000000
--- a/tests/mgcp/mgcp_transcoding_test.c
+++ /dev/null
@@ -1,654 +0,0 @@
-#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 2024-06-03 22:12:27 +0000