mgcp: Add CLI tool to test audio conversion
This tool uses mgcp_transcode.c to convert audio data from stdin to stdout. Sponsored-by: On-Waves ehf
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@ -0,0 +1,17 @@
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OBJS = testconv_main.o mgcp_transcode.o
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CC = gcc
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CFLAGS = -O0 -ggdb -Wall
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LDFLAGS =
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CPPFLAGS = -I../.. -I../../include $(shell pkg-config --cflags libosmocore) $(shell pkg-config --cflags libbcg729)
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LIBS = ../../src/libmgcp/libmgcp.a ../../src/libcommon/libcommon.a $(shell pkg-config --libs libosmocore) $(shell pkg-config --libs libbcg729) -lgsm -lrt
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testconv: $(OBJS)
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$(CC) -o $@ $^ $(LDFLAGS) $(LIBS)
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testconv_main.o: testconv_main.c
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mgcp_transcode.o: ../../src/osmo-bsc_mgcp/mgcp_transcode.c
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$(OBJS):
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$(CC) $(CFLAGS) $(CPPFLAGS) -c -o $@ $<
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#include <stdlib.h>
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#include <unistd.h>
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#include <stdio.h>
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#include <string.h>
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#include <err.h>
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#include <osmocom/core/talloc.h>
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#include <osmocom/core/application.h>
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#include <openbsc/debug.h>
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#include <openbsc/gsm_data.h>
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#include <openbsc/mgcp.h>
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#include <openbsc/mgcp_internal.h>
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#include "bscconfig.h"
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#ifndef BUILD_MGCP_TRANSCODING
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#error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)"
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#endif
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#include "src/osmo-bsc_mgcp/mgcp_transcode.h"
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static int audio_name_to_type(const char *name)
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{
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if (!strcasecmp(name, "gsm"))
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return 3;
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#ifdef HAVE_BCG729
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else if (!strcasecmp(name, "g729"))
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return 18;
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#endif
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else if (!strcasecmp(name, "pcma"))
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return 8;
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else if (!strcasecmp(name, "l16"))
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return 11;
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return -1;
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}
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int mgcp_get_trans_frame_size(void *state_, int nsamples, int dst);
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int main(int argc, char **argv)
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{
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char buf[4096] = {0};
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int cc, rc;
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struct mgcp_rtp_end dst_end = {0};
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struct mgcp_rtp_end src_end = {0};
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struct mgcp_trunk_config tcfg = {{0}};
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struct mgcp_endpoint endp = {0};
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struct mgcp_process_rtp_state *state;
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int in_size;
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osmo_init_logging(&log_info);
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tcfg.endpoints = &endp;
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tcfg.number_endpoints = 1;
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endp.tcfg = &tcfg;
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if (argc <= 2)
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errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16}");
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if ((src_end.payload_type = audio_name_to_type(argv[1])) == -1)
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errx(1, "invalid input format '%s'", argv[1]);
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if ((dst_end.payload_type = audio_name_to_type(argv[2])) == -1)
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errx(1, "invalid output format '%s'", argv[2]);
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rc = mgcp_transcoding_setup(&endp, &dst_end, &src_end);
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if (rc < 0)
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errx(1, "setup failed: %s", strerror(-rc));
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state = dst_end.rtp_process_data;
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OSMO_ASSERT(state != NULL);
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in_size = mgcp_transcoding_get_frame_size(state, 160, 0);
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OSMO_ASSERT(sizeof(buf) >= in_size + 12);
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while ((cc = read(0, buf + 12, in_size))) {
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if (cc != in_size)
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err(1, "read");
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cc += 12; /* include RTP header */
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rc = mgcp_transcoding_process_rtp(&endp, &dst_end,
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buf, &cc, sizeof(buf));
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if (rc < 0)
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errx(1, "processing failed: %s", strerror(-rc));
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cc -= 12; /* ignore RTP header */
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if (write(1, buf + 12, cc) != cc)
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err(1, "write");
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}
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return 0;
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}
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@ -72,6 +72,19 @@ struct mgcp_process_rtp_state {
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size_t dst_samples_per_frame;
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};
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int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
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{
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struct mgcp_process_rtp_state *state = state_;
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if (dst)
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return (nsamples >= 0 ?
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nsamples / state->dst_samples_per_frame :
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1) * state->dst_frame_size;
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else
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return (nsamples >= 0 ?
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nsamples / state->src_samples_per_frame :
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1) * state->src_frame_size;
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}
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static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
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{
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if (rtp_end->subtype_name) {
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@ -31,4 +31,6 @@ void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
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int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
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struct mgcp_rtp_end *dst_end,
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char *data, int *len, int buf_size);
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int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst);
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#endif /* OPENBSC_MGCP_TRANSCODE_H */
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