From 136a319e910eec81ea9ff8f8a34c324557109d03 Mon Sep 17 00:00:00 2001 From: Jacob Erlbeck Date: Thu, 13 Mar 2014 14:33:37 +0100 Subject: [PATCH] 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 --- 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 000000000..90adeccca --- /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 000000000..c2785f23e --- /dev/null +++ b/openbsc/contrib/testconv/testconv_main.c @@ -0,0 +1,91 @@ +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include +#include + +#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 cadc8769e..91c0c383c 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 2dfb06abf..0961634da 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 */