This tool uses mgcp_transcode.c to convert audio data from stdin to
stdout.
Sponsored-by: On-Waves ehf
---
openbsc/contrib/testconv/Makefile | 17 ++++++
openbsc/contrib/testconv/testconv_main.c | 91 ++++++++++++++++++++++++++++
openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c | 13 ++++
openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h | 2 +
4 files changed, 123 insertions(+)
create mode 100644 openbsc/contrib/testconv/Makefile
create mode 100644 openbsc/contrib/testconv/testconv_main.c
diff --git a/openbsc/contrib/testconv/Makefile
b/openbsc/contrib/testconv/Makefile
new file mode 100644
index 0000000..90adecc
--- /dev/null
+++ b/openbsc/contrib/testconv/Makefile
@@ -0,0 +1,17 @@
+
+OBJS = testconv_main.o mgcp_transcode.o
+
+CC = gcc
+CFLAGS = -O0 -ggdb -Wall
+LDFLAGS =
+CPPFLAGS = -I../.. -I../../include $(shell pkg-config --cflags libosmocore)
$(shell pkg-config --cflags libbcg729)
+LIBS = ../../src/libmgcp/libmgcp.a ../../src/libcommon/libcommon.a $(shell
pkg-config --libs libosmocore) $(shell pkg-config --libs libbcg729) -lgsm -lrt
+
+testconv: $(OBJS)
+ $(CC) -o $@ $^ $(LDFLAGS) $(LIBS)
+
+testconv_main.o: testconv_main.c
+mgcp_transcode.o: ../../src/osmo-bsc_mgcp/mgcp_transcode.c
+
+$(OBJS):
+ $(CC) $(CFLAGS) $(CPPFLAGS) -c -o $@ $<
diff --git a/openbsc/contrib/testconv/testconv_main.c
b/openbsc/contrib/testconv/testconv_main.c
new file mode 100644
index 0000000..c2785f2
--- /dev/null
+++ b/openbsc/contrib/testconv/testconv_main.c
@@ -0,0 +1,91 @@
+#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"
+
+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);
+
+int main(int argc, char **argv)
+{
+ char buf[4096] = {0};
+ int cc, rc;
+ struct mgcp_rtp_end dst_end = {0};
+ struct mgcp_rtp_end src_end = {0};
+ struct mgcp_trunk_config tcfg = {{0}};
+ struct mgcp_endpoint endp = {0};
+ struct mgcp_process_rtp_state *state;
+ int in_size;
+
+ osmo_init_logging(&log_info);
+
+ tcfg.endpoints = &endp;
+ tcfg.number_endpoints = 1;
+ endp.tcfg = &tcfg;
+
+ if (argc <= 2)
+ errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16}");
+
+ if ((src_end.payload_type = audio_name_to_type(argv[1])) == -1)
+ errx(1, "invalid input format '%s'", argv[1]);
+ if ((dst_end.payload_type = audio_name_to_type(argv[2])) == -1)
+ errx(1, "invalid output format '%s'", argv[2]);
+
+ 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, 160, 0);
+ OSMO_ASSERT(sizeof(buf) >= in_size + 12);
+
+ while ((cc = read(0, buf + 12, in_size))) {
+ if (cc != in_size)
+ err(1, "read");
+
+ cc += 12; /* include RTP header */
+
+ rc = mgcp_transcoding_process_rtp(&endp, &dst_end,
+ buf, &cc, sizeof(buf));
+ if (rc < 0)
+ errx(1, "processing failed: %s", strerror(-rc));
+
+ cc -= 12; /* ignore RTP header */
+ if (write(1, buf + 12, cc) != cc)
+ err(1, "write");
+ }
+ return 0;
+}
+
diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
index 7247c88..67e7e52 100644
--- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
+++ b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
@@ -72,6 +72,19 @@ struct mgcp_process_rtp_state {
size_t dst_samples_per_frame;
};
+int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
+{
+ struct mgcp_process_rtp_state *state = state_;
+ if (dst)
+ return (nsamples >= 0 ?
+ nsamples / state->dst_samples_per_frame :
+ 1) * state->dst_frame_size;
+ else
+ return (nsamples >= 0 ?
+ nsamples / state->src_samples_per_frame :
+ 1) * state->src_frame_size;
+}
+
static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
{
if (rtp_end->subtype_name) {
diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h
b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h
index 2dfb06a..0961634 100644
--- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h
+++ b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h
@@ -31,4 +31,6 @@ void mgcp_transcoding_net_downlink_format(struct
mgcp_endpoint *endp,
int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
struct mgcp_rtp_end *dst_end,
char *data, int *len, int buf_size);
+
+int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst);
#endif /* OPENBSC_MGCP_TRANSCODE_H */
--
1.7.9.5
