This patch adds the fields channels, subtype_name, and audio_name to the struct. The field audio_name contains the full string that has been used for the last part of a SDP a=rtpmap line. The others contain decoded parts of that string. If no a=rtpmap line has been given (e.g. because dynamic payload types are not used), values are assigned when the payload type matches one of the predefined ones (GSM, G729, PCMA). Sponsored-by: On-Waves ehf --- openbsc/include/openbsc/mgcp_internal.h | 3 ++ openbsc/src/libmgcp/mgcp_protocol.c | 82 +++++++++++++++++++++++-------- 2 files changed, 64 insertions(+), 21 deletions(-) diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h index 56c280d..72ac8e9 100644 --- a/openbsc/include/openbsc/mgcp_internal.h +++ b/openbsc/include/openbsc/mgcp_internal.h @@ -81,11 +81,14 @@ struct mgcp_rtp_end { /* per endpoint data */ int payload_type; uint32_t rate; + int channels; uint32_t frame_duration_num; uint32_t frame_duration_den; int frames_per_packet; uint32_t packet_duration_ms; char *fmtp_extra; + char *audio_name; + char *subtype_name; int output_enabled; /* RTP patching */ diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c index 0f8614c..761a35c 100644 --- a/openbsc/src/libmgcp/mgcp_protocol.c +++ b/openbsc/src/libmgcp/mgcp_protocol.c @@ -28,6 +28,7 @@ #include <time.h> #include <limits.h> #include <unistd.h> +#include <errno.h> #include <osmocom/core/msgb.h> #include <osmocom/core/talloc.h> @@ -77,6 +78,7 @@ char *strline_r(char *str, char **saveptr) #define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000 #define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20 #define DEFAULT_RTP_AUDIO_DEFAULT_RATE 8000 +#define DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS 1 static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end); @@ -231,6 +233,8 @@ static struct msgb *create_response_with_sdp(struct mgcp_endpoint *endp, { const char *addr = endp->cfg->local_ip; const char *fmtp_extra = endp->bts_end.fmtp_extra; + const char *audio_name = endp->bts_end.audio_name; + int payload_type = endp->bts_end.payload_type; char sdp_record[4096]; int len; @@ -244,11 +248,12 @@ static struct msgb *create_response_with_sdp(struct mgcp_endpoint *endp, "c=IN IP4 %s\r\n" "t=0 0\r\n" "m=audio %d RTP/AVP %d\r\n" - "a=rtpmap:%d %s\r\n" + "a=rtpmap:%d%s%s\r\n" "%s%s", endp->ci, endp->ci, addr, addr, - endp->net_end.local_port, endp->bts_end.payload_type, - endp->bts_end.payload_type, endp->tcfg->audio_name, + endp->net_end.local_port, payload_type, + payload_type, + audio_name ? " " : "", audio_name ? audio_name : "", fmtp_extra ? fmtp_extra : "", fmtp_extra ? "\r\n" : ""); if (len < 0 || len >= sizeof(sdp_record)) @@ -514,6 +519,47 @@ static int parse_conn_mode(const char *msg, struct mgcp_endpoint *endp) return ret; } +static int set_audio_info(struct mgcp_rtp_end *rtp, + int payload_type, const char *audio_name) +{ + int rate = 8000; + int channels = 1; + char audio_codec[64]; + + talloc_free(rtp->subtype_name); + rtp->subtype_name = NULL; + talloc_free(rtp->audio_name); + rtp->audio_name = NULL; + + rtp->payload_type = payload_type; + + if (!audio_name) { + switch (payload_type) { + case 3: audio_name = "GSM/8000/1"; break; + case 8: audio_name = "PCMA/8000/1"; break; + case 18: audio_name = "G729/8000/1"; break; + default: + rtp->rate = 8000; + rtp->channels = 1; + return 0; + } + } + + if (sscanf(audio_name, "%63[^/]/%d/%d", + audio_codec, &rate, &channels) < 2) + return -EINVAL; + + rtp->rate = rate; + rtp->channels = channels; + rtp->subtype_name = talloc_strdup(NULL, audio_codec); + rtp->audio_name = talloc_strdup(NULL, audio_name); + if (channels != 1) + LOGP(DMGCP, LOGL_NOTICE, + "Channels != 1 in SDP: '%s'\n", audio_name); + + return 0; +} + static int allocate_port(struct mgcp_endpoint *endp, struct mgcp_rtp_end *end, struct mgcp_port_range *range, int (*alloc)(struct mgcp_endpoint *endp, int port)) @@ -600,29 +646,18 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p) break; case 'a': { int payload; - int rate; - int channels = 1; int ptime, ptime2 = 0; char audio_name[64]; - char audio_codec[64]; if (audio_payload == -1) break; - if (sscanf(line, "a=rtpmap:%d %64s", + if (sscanf(line, "a=rtpmap:%d %63s", &payload, audio_name) == 2) { if (payload != audio_payload) break; - if (sscanf(audio_name, "%[^/]/%d/%d", - audio_codec, &rate, &channels) < 2) - break; - - rtp->rate = rate; - if (channels != 1) - LOGP(DMGCP, LOGL_NOTICE, - "Channels != 1 in SDP: '%s' on 0x%x\n", - line, ENDPOINT_NUMBER(p->endp)); + set_audio_info(rtp, payload, audio_name); } else if (sscanf(line, "a=ptime:%d-%d", &ptime, &ptime2) >= 1) { if (ptime2 > 0 && ptime2 != ptime) @@ -645,8 +680,8 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p) &port, &audio_payload) == 2) { rtp->rtp_port = htons(port); rtp->rtcp_port = htons(port + 1); - rtp->payload_type = audio_payload; found_media = 1; + set_audio_info(rtp, audio_payload, NULL); } break; } @@ -826,7 +861,7 @@ mgcp_header_done: endp->allocated = 1; /* set up RTP media parameters */ - endp->bts_end.payload_type = tcfg->audio_payload; + set_audio_info(&endp->bts_end, tcfg->audio_payload, tcfg->audio_name); endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints, tcfg->audio_fmtp_extra); if (have_sdp) { @@ -1252,6 +1287,7 @@ static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end) end->frames_per_packet = 0; /* unknown */ end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS; end->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE; + end->channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS; end->output_enabled = 0; } @@ -1329,6 +1365,9 @@ static void send_msg(struct mgcp_endpoint *endp, int endpoint, int port, { char buf[2096]; int len; + const char *fmtp_extra = endp->bts_end.fmtp_extra; + const char *audio_name = endp->bts_end.audio_name; + int payload_type = endp->bts_end.payload_type; /* hardcoded to AMR right now, we do not know the real type at this point */ len = snprintf(buf, sizeof(buf), @@ -1338,10 +1377,11 @@ static void send_msg(struct mgcp_endpoint *endp, int endpoint, int port, "\r\n" "c=IN IP4 %s\r\n" "m=audio %d RTP/AVP %d\r\n" - "a=rtpmap:%d %s\r\n", + "a=rtpmap:%d%s%s\r\n", msg, endpoint, mode, endp->cfg->source_addr, - port, endp->tcfg->audio_payload, - endp->tcfg->audio_payload, endp->tcfg->audio_name); + port, payload_type, + payload_type, + audio_name ? " " : "", audio_name ? audio_name : ""); if (len < 0) return; -- 1.7.9.5

This patch adds the get_net_downlink_format_cb() callback to provide payload_type, subtype_name, and fmtp_extra suitable for use in a MGCP response sent to the network. Per default, the BTS side values are returned since these must be honoured by the net peer when sending audio to the media gateway (unless transcoding is done). Sponsored-by: On-Waves ehf --- openbsc/include/openbsc/mgcp.h | 8 ++++++++ openbsc/include/openbsc/mgcp_internal.h | 5 +++++ openbsc/src/libmgcp/mgcp_network.c | 13 +++++++++++++ openbsc/src/libmgcp/mgcp_protocol.c | 20 ++++++++++++++------ 4 files changed, 40 insertions(+), 6 deletions(-) diff --git a/openbsc/include/openbsc/mgcp.h b/openbsc/include/openbsc/mgcp.h index 6ba0769..b595aba 100644 --- a/openbsc/include/openbsc/mgcp.h +++ b/openbsc/include/openbsc/mgcp.h @@ -92,6 +92,12 @@ typedef int (*mgcp_processing)(struct mgcp_rtp_end *dst_end, typedef int (*mgcp_processing_setup)(struct mgcp_endpoint *endp, struct mgcp_rtp_end *dst_end, struct mgcp_rtp_end *src_end); + +typedef void (*mgcp_get_format)(struct mgcp_endpoint *endp, + int *payload_type, + const char**subtype_name, + const char**fmtp_extra); + #define PORT_ALLOC_STATIC 0 #define PORT_ALLOC_DYNAMIC 1 @@ -166,6 +172,8 @@ struct mgcp_config { mgcp_processing rtp_processing_cb; mgcp_processing_setup setup_rtp_processing_cb; + mgcp_get_format get_net_downlink_format_cb; + struct osmo_wqueue gw_fd; struct mgcp_port_range bts_ports; diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h index 72ac8e9..e74b9fa 100644 --- a/openbsc/include/openbsc/mgcp_internal.h +++ b/openbsc/include/openbsc/mgcp_internal.h @@ -209,4 +209,9 @@ int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp, struct mgcp_rtp_end *dst_end, struct mgcp_rtp_end *src_end); +void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp, + int *payload_type, + const char**subtype_name, + const char**fmtp_extra); + #endif diff --git a/openbsc/src/libmgcp/mgcp_network.c b/openbsc/src/libmgcp/mgcp_network.c index 4d1ad35..dcbb97a 100644 --- a/openbsc/src/libmgcp/mgcp_network.c +++ b/openbsc/src/libmgcp/mgcp_network.c @@ -361,6 +361,19 @@ int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp, return 0; } +void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp, + int *payload_type, + const char**audio_name, + const char**fmtp_extra) +{ + /* Use the BTS side parameters when passing the SDP data (for + * downlink) to the net peer. + */ + *payload_type = endp->bts_end.payload_type; + *audio_name = endp->bts_end.audio_name; + *fmtp_extra = endp->bts_end.fmtp_extra; +} + /** * The RFC 3550 Appendix A assumes there are multiple sources but * some of the supported endpoints (e.g. the nanoBTS) can only handle diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c index 761a35c..b23a56a 100644 --- a/openbsc/src/libmgcp/mgcp_protocol.c +++ b/openbsc/src/libmgcp/mgcp_protocol.c @@ -232,12 +232,15 @@ static struct msgb *create_response_with_sdp(struct mgcp_endpoint *endp, const char *msg, const char *trans_id) { const char *addr = endp->cfg->local_ip; - const char *fmtp_extra = endp->bts_end.fmtp_extra; - const char *audio_name = endp->bts_end.audio_name; - int payload_type = endp->bts_end.payload_type; + const char *fmtp_extra; + const char *audio_name; + int payload_type; char sdp_record[4096]; int len; + endp->cfg->get_net_downlink_format_cb(endp, &payload_type, + &audio_name, &fmtp_extra); + if (!addr) addr = endp->cfg->source_addr; @@ -1213,6 +1216,8 @@ struct mgcp_config *mgcp_config_alloc(void) cfg->rtp_processing_cb = &mgcp_rtp_processing_default; cfg->setup_rtp_processing_cb = &mgcp_setup_rtp_processing_default; + cfg->get_net_downlink_format_cb = &mgcp_get_net_downlink_format_default; + /* default trunk handling */ cfg->trunk.cfg = cfg; cfg->trunk.trunk_nr = 0; @@ -1365,9 +1370,12 @@ static void send_msg(struct mgcp_endpoint *endp, int endpoint, int port, { char buf[2096]; int len; - const char *fmtp_extra = endp->bts_end.fmtp_extra; - const char *audio_name = endp->bts_end.audio_name; - int payload_type = endp->bts_end.payload_type; + const char *fmtp_extra; + const char *audio_name; + int payload_type; + + endp->cfg->get_net_downlink_format_cb(endp, &payload_type, + &audio_name, &fmtp_extra); /* hardcoded to AMR right now, we do not know the real type at this point */ len = snprintf(buf, sizeof(buf), -- 1.7.9.5

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 c6b2508..ea4bd74 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

This modification allows it to set the number of samples per packet that is written to the output. Sponsored-by: On-Waves ehf --- openbsc/contrib/testconv/testconv_main.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/openbsc/contrib/testconv/testconv_main.c b/openbsc/contrib/testconv/testconv_main.c index aee7304..e74c686 100644 --- a/openbsc/contrib/testconv/testconv_main.c +++ b/openbsc/contrib/testconv/testconv_main.c @@ -46,6 +46,10 @@ int main(int argc, char **argv) struct mgcp_endpoint endp = {0}; struct mgcp_process_rtp_state *state; int in_size; + int in_samples = 160; + int out_samples = 0; + uint32_t ts = 0; + uint16_t seq = 0; osmo_init_logging(&log_info); @@ -58,12 +62,20 @@ int main(int argc, char **argv) src_end = &endp.net_end; if (argc <= 2) - errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16}"); + errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16} [SPP]"); 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]); + if (argc > 3) + out_samples = atoi(argv[3]); + + if (out_samples) { + dst_end->frame_duration_den = dst_end->rate; + dst_end->frame_duration_num = out_samples; + dst_end->frames_per_packet = 1; + } rc = mgcp_transcoding_setup(&endp, dst_end, src_end); if (rc < 0) @@ -72,7 +84,7 @@ int main(int argc, char **argv) state = dst_end->rtp_process_data; OSMO_ASSERT(state != NULL); - in_size = mgcp_transcoding_get_frame_size(state, 160, 0); + in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0); OSMO_ASSERT(sizeof(buf) >= in_size + 12); buf[1] = src_end->payload_type; @@ -87,13 +99,19 @@ int main(int argc, char **argv) if (cc != in_size) err(1, "read"); + *(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, dst_end, - buf, &len, sizeof(buf)); + buf, &len, sizeof(buf)); if (cont == -EAGAIN) { fprintf(stderr, "Got EAGAIN\n"); break; -- 1.7.9.5

This patch moves the files relevant to transcoding from src/osmo-bsc_mgcp to src/libmgcp and src/include/openbsc. Makefiles and include directives are being updated accordingly. Sponsored-by: On-Waves ehf --- openbsc/contrib/testconv/Makefile | 3 +- openbsc/contrib/testconv/testconv_main.c | 2 +- openbsc/include/openbsc/Makefile.am | 3 +- openbsc/include/openbsc/mgcp_transcode.h | 36 ++ openbsc/src/libmgcp/Makefile.am | 12 +- openbsc/src/libmgcp/g711common.h | 187 ++++++++++ openbsc/src/libmgcp/mgcp_transcode.c | 553 ++++++++++++++++++++++++++++ openbsc/src/osmo-bsc_mgcp/Makefile.am | 9 +- openbsc/src/osmo-bsc_mgcp/g711common.h | 187 ---------- openbsc/src/osmo-bsc_mgcp/mgcp_main.c | 7 +- openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c | 553 ---------------------------- openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h | 36 -- openbsc/tests/mgcp/Makefile.am | 2 +- openbsc/tests/mgcp/mgcp_transcoding_test.c | 2 +- 14 files changed, 795 insertions(+), 797 deletions(-) create mode 100644 openbsc/include/openbsc/mgcp_transcode.h create mode 100644 openbsc/src/libmgcp/g711common.h create mode 100644 openbsc/src/libmgcp/mgcp_transcode.c delete mode 100644 openbsc/src/osmo-bsc_mgcp/g711common.h delete mode 100644 openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c delete mode 100644 openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h diff --git a/openbsc/contrib/testconv/Makefile b/openbsc/contrib/testconv/Makefile index 90adecc..bb856f7 100644 --- a/openbsc/contrib/testconv/Makefile +++ b/openbsc/contrib/testconv/Makefile @@ -1,5 +1,5 @@ -OBJS = testconv_main.o mgcp_transcode.o +OBJS = testconv_main.o CC = gcc CFLAGS = -O0 -ggdb -Wall @@ -11,7 +11,6 @@ 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 index e74c686..89dce1a 100644 --- a/openbsc/contrib/testconv/testconv_main.c +++ b/openbsc/contrib/testconv/testconv_main.c @@ -17,7 +17,7 @@ #error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)" #endif -#include "src/osmo-bsc_mgcp/mgcp_transcode.h" +#include "openbsc/mgcp_transcode.h" static int audio_name_to_type(const char *name) { diff --git a/openbsc/include/openbsc/Makefile.am b/openbsc/include/openbsc/Makefile.am index 8f7c1c4..6b08d07 100644 --- a/openbsc/include/openbsc/Makefile.am +++ b/openbsc/include/openbsc/Makefile.am @@ -13,7 +13,8 @@ noinst_HEADERS = abis_nm.h abis_rsl.h db.h gsm_04_08.h gsm_data.h \ osmo_bsc_rf.h osmo_bsc.h network_listen.h bsc_nat_sccp.h \ osmo_msc_data.h osmo_bsc_grace.h sms_queue.h abis_om2000.h \ bss.h gsm_data_shared.h control_cmd.h ipaccess.h mncc_int.h \ - arfcn_range_encode.h nat_rewrite_trie.h bsc_nat_callstats.h + arfcn_range_encode.h nat_rewrite_trie.h bsc_nat_callstats.h \ + mgcp_transcode.h openbsc_HEADERS = gsm_04_08.h meas_rep.h bsc_api.h openbscdir = $(includedir)/openbsc diff --git a/openbsc/include/openbsc/mgcp_transcode.h b/openbsc/include/openbsc/mgcp_transcode.h new file mode 100644 index 0000000..0961634 --- /dev/null +++ b/openbsc/include/openbsc/mgcp_transcode.h @@ -0,0 +1,36 @@ +/* + * (C) 2014 by On-Waves + * All Rights Reserved + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU Affero General Public License as published by + * the Free Software Foundation; either version 3 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Affero General Public License for more details. + * + * You should have received a copy of the GNU Affero General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ +#ifndef OPENBSC_MGCP_TRANSCODE_H +#define OPENBSC_MGCP_TRANSCODE_H + +int mgcp_transcoding_setup(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + struct mgcp_rtp_end *src_end); + +void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp, + int *payload_type, + const char**audio_name, + const char**fmtp_extra); + +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 */ diff --git a/openbsc/src/libmgcp/Makefile.am b/openbsc/src/libmgcp/Makefile.am index 72f625d..bd02e61 100644 --- a/openbsc/src/libmgcp/Makefile.am +++ b/openbsc/src/libmgcp/Makefile.am @@ -1,7 +1,15 @@ AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir) -AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOVTY_CFLAGS) $(COVERAGE_CFLAGS) -AM_LDFLAGS = $(LIBOSMOCORE_LIBS) $(LIBOSMOGSM_LIBS) $(COVERAGE_LDFLAGS) +AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOVTY_CFLAGS) $(COVERAGE_CFLAGS) \ + $(LIBBCG729_CFLAGS) +AM_LDFLAGS = $(LIBOSMOCORE_LIBS) $(LIBOSMOGSM_LIBS) $(COVERAGE_LDFLAGS) \ + $(LIBBCG729_LIBS) noinst_LIBRARIES = libmgcp.a +noinst_HEADERS = g711common.h + libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c + +if BUILD_MGCP_TRANSCODING + libmgcp_a_SOURCES += mgcp_transcode.c +endif diff --git a/openbsc/src/libmgcp/g711common.h b/openbsc/src/libmgcp/g711common.h new file mode 100644 index 0000000..cb35fc6 --- /dev/null +++ b/openbsc/src/libmgcp/g711common.h @@ -0,0 +1,187 @@ +/* + * PCM - A-Law conversion + * Copyright (c) 2000 by Abramo Bagnara &lt;abramo(a)alsa-project.org&gt; + * + * Wrapper for linphone Codec class by Simon Morlat &lt;simon.morlat(a)linphone.org&gt; + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +static inline int val_seg(int val) +{ + int r = 0; + val >>= 7; /*7 = 4 + 3*/ + if (val & 0xf0) { + val >>= 4; + r += 4; + } + if (val & 0x0c) { + val >>= 2; + r += 2; + } + if (val & 0x02) + r += 1; + return r; +} + +/* + * s16_to_alaw() - Convert a 16-bit linear PCM value to 8-bit A-law + * + * s16_to_alaw() accepts an 16-bit integer and encodes it as A-law data. + * + * Linear Input Code Compressed Code + * ------------------------ --------------- + * 0000000wxyza 000wxyz + * 0000001wxyza 001wxyz + * 000001wxyzab 010wxyz + * 00001wxyzabc 011wxyz + * 0001wxyzabcd 100wxyz + * 001wxyzabcde 101wxyz + * 01wxyzabcdef 110wxyz + * 1wxyzabcdefg 111wxyz + * + * For further information see John C. Bellamy's Digital Telephony, 1982, + * John Wiley & Sons, pps 98-111 and 472-476. + * G711 is designed for 13 bits input signal, this function add extra shifting to take this into account. + */ + +static inline unsigned char s16_to_alaw(int pcm_val) +{ + int mask; + int seg; + unsigned char aval; + + if (pcm_val >= 0) { + mask = 0xD5; + } else { + mask = 0x55; + pcm_val = -pcm_val; + if (pcm_val > 0x7fff) + pcm_val = 0x7fff; + } + + if (pcm_val < 256) /*256 = 32 << 3*/ + aval = pcm_val >> 4; /*4 = 1 + 3*/ + else { + /* Convert the scaled magnitude to segment number. */ + seg = val_seg(pcm_val); + aval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f); + } + return aval ^ mask; +} + +/* + * alaw_to_s16() - Convert an A-law value to 16-bit linear PCM + * + */ +static inline int alaw_to_s16(unsigned char a_val) +{ + int t; + int seg; + + a_val ^= 0x55; + t = a_val & 0x7f; + if (t < 16) + t = (t << 4) + 8; + else { + seg = (t >> 4) & 0x07; + t = ((t & 0x0f) << 4) + 0x108; + t <<= seg -1; + } + return ((a_val & 0x80) ? t : -t); +} +/* + * s16_to_ulaw() - Convert a linear PCM value to u-law + * + * In order to simplify the encoding process, the original linear magnitude + * is biased by adding 33 which shifts the encoding range from (0 - 8158) to + * (33 - 8191). The result can be seen in the following encoding table: + * + * Biased Linear Input Code Compressed Code + * ------------------------ --------------- + * 00000001wxyza 000wxyz + * 0000001wxyzab 001wxyz + * 000001wxyzabc 010wxyz + * 00001wxyzabcd 011wxyz + * 0001wxyzabcde 100wxyz + * 001wxyzabcdef 101wxyz + * 01wxyzabcdefg 110wxyz + * 1wxyzabcdefgh 111wxyz + * + * Each biased linear code has a leading 1 which identifies the segment + * number. The value of the segment number is equal to 7 minus the number + * of leading 0's. The quantization interval is directly available as the + * four bits wxyz. * The trailing bits (a - h) are ignored. + * + * Ordinarily the complement of the resulting code word is used for + * transmission, and so the code word is complemented before it is returned. + * + * For further information see John C. Bellamy's Digital Telephony, 1982, + * John Wiley & Sons, pps 98-111 and 472-476. + */ + +static inline unsigned char s16_to_ulaw(int pcm_val) /* 2's complement (16-bit range) */ +{ + int mask; + int seg; + unsigned char uval; + + if (pcm_val < 0) { + pcm_val = 0x84 - pcm_val; + mask = 0x7f; + } else { + pcm_val += 0x84; + mask = 0xff; + } + if (pcm_val > 0x7fff) + pcm_val = 0x7fff; + + /* Convert the scaled magnitude to segment number. */ + seg = val_seg(pcm_val); + + /* + * Combine the sign, segment, quantization bits; + * and complement the code word. + */ + uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f); + return uval ^ mask; +} + +/* + * ulaw_to_s16() - Convert a u-law value to 16-bit linear PCM + * + * First, a biased linear code is derived from the code word. An unbiased + * output can then be obtained by subtracting 33 from the biased code. + * + * Note that this function expects to be passed the complement of the + * original code word. This is in keeping with ISDN conventions. + */ +static inline int ulaw_to_s16(unsigned char u_val) +{ + int t; + + /* Complement to obtain normal u-law value. */ + u_val = ~u_val; + + /* + * Extract and bias the quantization bits. Then + * shift up by the segment number and subtract out the bias. + */ + t = ((u_val & 0x0f) << 3) + 0x84; + t <<= (u_val & 0x70) >> 4; + + return ((u_val & 0x80) ? (0x84 - t) : (t - 0x84)); +} diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c new file mode 100644 index 0000000..edd3178 --- /dev/null +++ b/openbsc/src/libmgcp/mgcp_transcode.c @@ -0,0 +1,553 @@ +/* + * (C) 2014 by On-Waves + * All Rights Reserved + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU Affero General Public License as published by + * the Free Software Foundation; either version 3 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Affero General Public License for more details. + * + * You should have received a copy of the GNU Affero General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <stdlib.h> +#include <string.h> +#include <errno.h> + + +#include "../../bscconfig.h" + +#include "g711common.h" +#include <gsm.h> +#ifdef HAVE_BCG729 +#include <bcg729/decoder.h> +#include <bcg729/encoder.h> +#endif + +#include <openbsc/debug.h> +#include <openbsc/mgcp.h> +#include <openbsc/mgcp_internal.h> + +#include <osmocom/core/talloc.h> + +enum audio_format { + AF_INVALID, + AF_S16, + AF_L16, + AF_GSM, + AF_G729, + AF_PCMA +}; + +struct mgcp_process_rtp_state { + /* decoding */ + enum audio_format src_fmt; + union { + gsm gsm_handle; +#ifdef HAVE_BCG729 + bcg729DecoderChannelContextStruct *g729_dec; +#endif + } src; + size_t src_frame_size; + size_t src_samples_per_frame; + + /* processing */ + + /* encoding */ + enum audio_format dst_fmt; + union { + gsm gsm_handle; +#ifdef HAVE_BCG729 + bcg729EncoderChannelContextStruct *g729_enc; +#endif + } dst; + size_t dst_frame_size; + size_t dst_samples_per_frame; + int dst_packet_duration; + + int is_running; + uint16_t next_seq; + uint32_t next_time; + int16_t samples[10*160]; + size_t sample_cnt; + size_t sample_offs; +}; + +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) { + if (!strcmp("GSM", rtp_end->subtype_name)) + return AF_GSM; + if (!strcmp("PCMA", rtp_end->subtype_name)) + return AF_PCMA; +#ifdef HAVE_BCG729 + if (!strcmp("G729", rtp_end->subtype_name)) + return AF_G729; +#endif + if (!strcmp("L16", rtp_end->subtype_name)) + return AF_L16; + } + + switch (rtp_end->payload_type) { + case 3 /* GSM */: + return AF_GSM; + case 8 /* PCMA */: + return AF_PCMA; +#ifdef HAVE_BCG729 + case 18 /* G.729 */: + return AF_G729; +#endif + case 11 /* L16 */: + return AF_L16; + default: + return AF_INVALID; + } +} + +static void l16_encode(short *sample, unsigned char *buf, size_t n) +{ + for (; n > 0; --n, ++sample, buf += 2) { + buf[0] = sample[0] >> 8; + buf[1] = sample[0] & 0xff; + } +} + +static void l16_decode(unsigned char *buf, short *sample, size_t n) +{ + for (; n > 0; --n, ++sample, buf += 2) + sample[0] = ((short)buf[0] << 8) | buf[1]; +} + +static void alaw_encode(short *sample, unsigned char *buf, size_t n) +{ + for (; n > 0; --n) + *(buf++) = s16_to_alaw(*(sample++)); +} + +static void alaw_decode(unsigned char *buf, short *sample, size_t n) +{ + for (; n > 0; --n) + *(sample++) = alaw_to_s16(*(buf++)); +} + +static int processing_state_destructor(struct mgcp_process_rtp_state *state) +{ + switch (state->src_fmt) { + case AF_GSM: + if (state->dst.gsm_handle) + gsm_destroy(state->src.gsm_handle); + break; +#ifdef HAVE_BCG729 + case AF_G729: + if (state->src.g729_dec) + closeBcg729DecoderChannel(state->src.g729_dec); + break; +#endif + default: + break; + } + switch (state->dst_fmt) { + case AF_GSM: + if (state->dst.gsm_handle) + gsm_destroy(state->dst.gsm_handle); + break; +#ifdef HAVE_BCG729 + case AF_G729: + if (state->dst.g729_enc) + closeBcg729EncoderChannel(state->dst.g729_enc); + break; +#endif + default: + break; + } + return 0; +} + +int mgcp_transcoding_setup(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + struct mgcp_rtp_end *src_end) +{ + struct mgcp_process_rtp_state *state = dst_end->rtp_process_data; + enum audio_format src_fmt, dst_fmt; + + /* cleanup first */ + if (state) { + talloc_free(state); + dst_end->rtp_process_data = NULL; + } + + if (!src_end) + return 0; + + src_fmt = get_audio_format(src_end); + dst_fmt = get_audio_format(dst_end); + + LOGP(DMGCP, LOGL_ERROR, + "Checking transcoding: %s (%d) -> %s (%d)\n", + src_end->subtype_name, src_end->payload_type, + dst_end->subtype_name, dst_end->payload_type); + + if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) { + if (!src_end->subtype_name || !dst_end->subtype_name) + /* Not enough info, do nothing */ + return 0; + + if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0) + /* Nothing to do */ + return 0; + + LOGP(DMGCP, LOGL_ERROR, + "Cannot transcode: %s codec not supported (%s -> %s).\n", + src_fmt != AF_INVALID ? "destination" : "source", + src_end->audio_name, dst_end->audio_name); + return -EINVAL; + } + + if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) { + LOGP(DMGCP, LOGL_ERROR, + "Cannot transcode: rate conversion (%d -> %d) not supported.\n", + src_end->rate, dst_end->rate); + return -EINVAL; + } + + state = talloc_zero(endp->tcfg->cfg, struct mgcp_process_rtp_state); + talloc_set_destructor(state, processing_state_destructor); + dst_end->rtp_process_data = state; + + state->src_fmt = src_fmt; + + switch (state->src_fmt) { + case AF_L16: + case AF_S16: + state->src_frame_size = 80 * sizeof(short); + state->src_samples_per_frame = 80; + break; + case AF_GSM: + state->src_frame_size = sizeof(gsm_frame); + state->src_samples_per_frame = 160; + state->src.gsm_handle = gsm_create(); + if (!state->src.gsm_handle) { + LOGP(DMGCP, LOGL_ERROR, + "Failed to initialize GSM decoder.\n"); + return -EINVAL; + } + break; +#ifdef HAVE_BCG729 + case AF_G729: + state->src_frame_size = 10; + state->src_samples_per_frame = 80; + state->src.g729_dec = initBcg729DecoderChannel(); + if (!state->src.g729_dec) { + LOGP(DMGCP, LOGL_ERROR, + "Failed to initialize G.729 decoder.\n"); + return -EINVAL; + } + break; +#endif + case AF_PCMA: + state->src_frame_size = 80; + state->src_samples_per_frame = 80; + break; + default: + break; + } + + state->dst_fmt = dst_fmt; + + switch (state->dst_fmt) { + case AF_L16: + case AF_S16: + state->dst_frame_size = 80*sizeof(short); + state->dst_samples_per_frame = 80; + break; + case AF_GSM: + state->dst_frame_size = sizeof(gsm_frame); + state->dst_samples_per_frame = 160; + state->dst.gsm_handle = gsm_create(); + if (!state->dst.gsm_handle) { + LOGP(DMGCP, LOGL_ERROR, + "Failed to initialize GSM encoder.\n"); + return -EINVAL; + } + break; +#ifdef HAVE_BCG729 + case AF_G729: + state->dst_frame_size = 10; + state->dst_samples_per_frame = 80; + state->dst.g729_enc = initBcg729EncoderChannel(); + if (!state->dst.g729_enc) { + LOGP(DMGCP, LOGL_ERROR, + "Failed to initialize G.729 decoder.\n"); + return -EINVAL; + } + break; +#endif + case AF_PCMA: + state->dst_frame_size = 80; + state->dst_samples_per_frame = 80; + break; + default: + break; + } + + if (dst_end->force_output_ptime) + state->dst_packet_duration = mgcp_rtp_packet_duration(endp, dst_end); + + LOGP(DMGCP, LOGL_INFO, + "Initialized RTP processing on: 0x%x " + "conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n", + ENDPOINT_NUMBER(endp), + src_fmt, src_end->payload_type, src_end->rate, src_end->fmtp_extra, + dst_fmt, dst_end->payload_type, dst_end->rate, dst_end->fmtp_extra); + + return 0; +} + +void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp, + int *payload_type, + const char**audio_name, + const char**fmtp_extra) +{ + struct mgcp_process_rtp_state *state = endp->net_end.rtp_process_data; + if (!state || endp->net_end.payload_type < 0) { + *payload_type = endp->bts_end.payload_type; + *audio_name = endp->bts_end.audio_name; + *fmtp_extra = endp->bts_end.fmtp_extra; + return; + } + + *payload_type = endp->net_end.payload_type; + *fmtp_extra = endp->net_end.fmtp_extra; + *audio_name = endp->net_end.audio_name; +} + +static int decode_audio(struct mgcp_process_rtp_state *state, + uint8_t **src, size_t *nbytes) +{ + while (*nbytes >= state->src_frame_size) { + if (state->sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(state->samples)) { + LOGP(DMGCP, LOGL_ERROR, + "Sample buffer too small: %d > %d.\n", + state->sample_cnt + state->src_samples_per_frame, + ARRAY_SIZE(state->samples)); + return -ENOSPC; + } + switch (state->src_fmt) { + case AF_GSM: + if (gsm_decode(state->src.gsm_handle, + (gsm_byte *)*src, state->samples + state->sample_cnt) < 0) { + LOGP(DMGCP, LOGL_ERROR, + "Failed to decode GSM.\n"); + return -EINVAL; + } + break; +#ifdef HAVE_BCG729 + case AF_G729: + bcg729Decoder(state->src.g729_dec, *src, 0, state->samples + state->sample_cnt); + break; +#endif + case AF_PCMA: + alaw_decode(*src, state->samples + state->sample_cnt, + state->src_samples_per_frame); + break; + case AF_S16: + memmove(state->samples + state->sample_cnt, *src, + state->src_frame_size); + break; + case AF_L16: + l16_decode(*src, state->samples + state->sample_cnt, + state->src_samples_per_frame); + break; + default: + break; + } + *src += state->src_frame_size; + *nbytes -= state->src_frame_size; + state->sample_cnt += state->src_samples_per_frame; + } + return 0; +} + +static int encode_audio(struct mgcp_process_rtp_state *state, + uint8_t *dst, size_t buf_size, size_t max_samples) +{ + int nbytes = 0; + size_t nsamples = 0; + /* Encode samples into dst */ + while (nsamples + state->dst_samples_per_frame <= max_samples) { + if (nbytes + state->dst_frame_size > buf_size) { + if (nbytes > 0) + break; + + /* Not even one frame fits into the buffer */ + LOGP(DMGCP, LOGL_INFO, + "Encoding (RTP) buffer too small: %d > %d.\n", + nbytes + state->dst_frame_size, buf_size); + return -ENOSPC; + } + switch (state->dst_fmt) { + case AF_GSM: + gsm_encode(state->dst.gsm_handle, + state->samples + state->sample_offs, dst); + break; +#ifdef HAVE_BCG729 + case AF_G729: + bcg729Encoder(state->dst.g729_enc, + state->samples + state->sample_offs, dst); + break; +#endif + case AF_PCMA: + alaw_encode(state->samples + state->sample_offs, dst, + state->src_samples_per_frame); + break; + case AF_S16: + memmove(dst, state->samples + state->sample_offs, + state->dst_frame_size); + break; + case AF_L16: + l16_encode(state->samples + state->sample_offs, dst, + state->src_samples_per_frame); + break; + default: + break; + } + dst += state->dst_frame_size; + nbytes += state->dst_frame_size; + state->sample_offs += state->dst_samples_per_frame; + nsamples += state->dst_samples_per_frame; + } + state->sample_cnt -= nsamples; + return nbytes; +} + +int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + char *data, int *len, int buf_size) +{ + struct mgcp_process_rtp_state *state = dst_end->rtp_process_data; + size_t rtp_hdr_size = 12; + char *payload_data = data + rtp_hdr_size; + int payload_len = *len - rtp_hdr_size; + // size_t sample_idx; + uint8_t *src = (uint8_t *)payload_data; + uint8_t *dst = (uint8_t *)payload_data; + size_t nbytes = payload_len; + // size_t frame_remainder; + size_t nsamples; + size_t max_samples; + uint32_t ts_no; + int rc; + + if (!state) + return 0; + + if (state->src_fmt == state->dst_fmt) { + if (!state->dst_packet_duration) + return 0; + + /* TODO: repackage without transcoding */ + } + + /* If the remaining samples do not fit into a fixed ptime, + * a) discard them, if the next packet is much later + * b) add silence and * send it, if the current packet is not + * yet too late + * c) append the sample data, if the timestamp matches exactly + */ + + /* TODO: check payload type (-> G.711 comfort noise) */ + + if (payload_len > 0) { + ts_no = ntohl(*(uint32_t*)(data+4)); + if (!state->is_running) + state->next_seq = ntohs(*(uint32_t*)(data+4)); + + state->is_running = 1; + + if (state->sample_cnt > 0) { + int32_t delta = ts_no - state->next_time; + /* TODO: check sequence? reordering? packet loss? */ + + if (delta > state->sample_cnt) + /* There is a time gap between the last packet + * and the current one. Just discard the + * partial data that is left in the buffer. + * TODO: This can be improved by adding silence + * instead if the delta is small enough. + */ + state->sample_cnt = 0; + else if (delta < 0) { + LOGP(DMGCP, LOGL_NOTICE, + "RTP time jumps backwards, delta = %d, " + "discarding buffered samples\n", + delta); + state->sample_cnt = 0; + state->sample_offs = 0; + return -EAGAIN; + } + + /* Make sure the samples start without offset */ + fprintf(stderr, "Moving %d samples to buffer start (offset %d)\n", state->sample_cnt, state->sample_offs); + if (state->sample_offs && state->sample_cnt) + memmove(&state->samples[0], + &state->samples[state->sample_offs], + state->sample_cnt * sizeof(state->samples[0])); + } + + state->sample_offs = 0; + + /* Append decoded audio to samples */ + decode_audio(state, &src, &nbytes); + + if (nbytes > 0) + LOGP(DMGCP, LOGL_NOTICE, + "Skipped audio frame in RTP packet: %d octets\n", + nbytes); + } else + ts_no = state->next_time; + + if (state->sample_cnt < state->dst_packet_duration) + return -EAGAIN; + + max_samples = + state->dst_packet_duration ? + state->dst_packet_duration : state->sample_cnt; + + nsamples = state->sample_cnt; + + rc = encode_audio(state, dst, buf_size, max_samples); + if (rc <= 0) + return rc; + + nsamples -= state->sample_cnt; + fprintf(stderr, "Wrote %d samples to buffer (offset %d)\n", nsamples, state->sample_offs); + + *len = rtp_hdr_size + rc; + *(uint16_t*)(data+2) = htonl(state->next_seq); + *(uint32_t*)(data+4) = htonl(ts_no); + + state->next_seq += 1; + state->next_time = ts_no + nsamples; + + return nsamples ? rtp_hdr_size : 0; +} diff --git a/openbsc/src/osmo-bsc_mgcp/Makefile.am b/openbsc/src/osmo-bsc_mgcp/Makefile.am index a620e7a..da02380 100644 --- a/openbsc/src/osmo-bsc_mgcp/Makefile.am +++ b/openbsc/src/osmo-bsc_mgcp/Makefile.am @@ -1,16 +1,11 @@ AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir) AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOGSM_CFLAGS) \ - $(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS) \ - $(LIBBCG729_CFLAGS) + $(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS) bin_PROGRAMS = osmo-bsc_mgcp osmo_bsc_mgcp_SOURCES = mgcp_main.c -if BUILD_MGCP_TRANSCODING - osmo_bsc_mgcp_SOURCES += mgcp_transcode.c -endif + osmo_bsc_mgcp_LDADD = $(top_builddir)/src/libcommon/libcommon.a \ $(top_builddir)/src/libmgcp/libmgcp.a -lrt \ $(LIBOSMOVTY_LIBS) $(LIBOSMOCORE_LIBS) $(LIBBCG729_LIBS) - -noinst_HEADERS = g711common.h mgcp_transcode.h diff --git a/openbsc/src/osmo-bsc_mgcp/g711common.h b/openbsc/src/osmo-bsc_mgcp/g711common.h deleted file mode 100644 index cb35fc6..0000000 --- a/openbsc/src/osmo-bsc_mgcp/g711common.h +++ /dev/null @@ -1,187 +0,0 @@ -/* - * PCM - A-Law conversion - * Copyright (c) 2000 by Abramo Bagnara &lt;abramo(a)alsa-project.org&gt; - * - * Wrapper for linphone Codec class by Simon Morlat &lt;simon.morlat(a)linphone.org&gt; - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -static inline int val_seg(int val) -{ - int r = 0; - val >>= 7; /*7 = 4 + 3*/ - if (val & 0xf0) { - val >>= 4; - r += 4; - } - if (val & 0x0c) { - val >>= 2; - r += 2; - } - if (val & 0x02) - r += 1; - return r; -} - -/* - * s16_to_alaw() - Convert a 16-bit linear PCM value to 8-bit A-law - * - * s16_to_alaw() accepts an 16-bit integer and encodes it as A-law data. - * - * Linear Input Code Compressed Code - * ------------------------ --------------- - * 0000000wxyza 000wxyz - * 0000001wxyza 001wxyz - * 000001wxyzab 010wxyz - * 00001wxyzabc 011wxyz - * 0001wxyzabcd 100wxyz - * 001wxyzabcde 101wxyz - * 01wxyzabcdef 110wxyz - * 1wxyzabcdefg 111wxyz - * - * For further information see John C. Bellamy's Digital Telephony, 1982, - * John Wiley & Sons, pps 98-111 and 472-476. - * G711 is designed for 13 bits input signal, this function add extra shifting to take this into account. - */ - -static inline unsigned char s16_to_alaw(int pcm_val) -{ - int mask; - int seg; - unsigned char aval; - - if (pcm_val >= 0) { - mask = 0xD5; - } else { - mask = 0x55; - pcm_val = -pcm_val; - if (pcm_val > 0x7fff) - pcm_val = 0x7fff; - } - - if (pcm_val < 256) /*256 = 32 << 3*/ - aval = pcm_val >> 4; /*4 = 1 + 3*/ - else { - /* Convert the scaled magnitude to segment number. */ - seg = val_seg(pcm_val); - aval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f); - } - return aval ^ mask; -} - -/* - * alaw_to_s16() - Convert an A-law value to 16-bit linear PCM - * - */ -static inline int alaw_to_s16(unsigned char a_val) -{ - int t; - int seg; - - a_val ^= 0x55; - t = a_val & 0x7f; - if (t < 16) - t = (t << 4) + 8; - else { - seg = (t >> 4) & 0x07; - t = ((t & 0x0f) << 4) + 0x108; - t <<= seg -1; - } - return ((a_val & 0x80) ? t : -t); -} -/* - * s16_to_ulaw() - Convert a linear PCM value to u-law - * - * In order to simplify the encoding process, the original linear magnitude - * is biased by adding 33 which shifts the encoding range from (0 - 8158) to - * (33 - 8191). The result can be seen in the following encoding table: - * - * Biased Linear Input Code Compressed Code - * ------------------------ --------------- - * 00000001wxyza 000wxyz - * 0000001wxyzab 001wxyz - * 000001wxyzabc 010wxyz - * 00001wxyzabcd 011wxyz - * 0001wxyzabcde 100wxyz - * 001wxyzabcdef 101wxyz - * 01wxyzabcdefg 110wxyz - * 1wxyzabcdefgh 111wxyz - * - * Each biased linear code has a leading 1 which identifies the segment - * number. The value of the segment number is equal to 7 minus the number - * of leading 0's. The quantization interval is directly available as the - * four bits wxyz. * The trailing bits (a - h) are ignored. - * - * Ordinarily the complement of the resulting code word is used for - * transmission, and so the code word is complemented before it is returned. - * - * For further information see John C. Bellamy's Digital Telephony, 1982, - * John Wiley & Sons, pps 98-111 and 472-476. - */ - -static inline unsigned char s16_to_ulaw(int pcm_val) /* 2's complement (16-bit range) */ -{ - int mask; - int seg; - unsigned char uval; - - if (pcm_val < 0) { - pcm_val = 0x84 - pcm_val; - mask = 0x7f; - } else { - pcm_val += 0x84; - mask = 0xff; - } - if (pcm_val > 0x7fff) - pcm_val = 0x7fff; - - /* Convert the scaled magnitude to segment number. */ - seg = val_seg(pcm_val); - - /* - * Combine the sign, segment, quantization bits; - * and complement the code word. - */ - uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f); - return uval ^ mask; -} - -/* - * ulaw_to_s16() - Convert a u-law value to 16-bit linear PCM - * - * First, a biased linear code is derived from the code word. An unbiased - * output can then be obtained by subtracting 33 from the biased code. - * - * Note that this function expects to be passed the complement of the - * original code word. This is in keeping with ISDN conventions. - */ -static inline int ulaw_to_s16(unsigned char u_val) -{ - int t; - - /* Complement to obtain normal u-law value. */ - u_val = ~u_val; - - /* - * Extract and bias the quantization bits. Then - * shift up by the segment number and subtract out the bias. - */ - t = ((u_val & 0x0f) << 3) + 0x84; - t <<= (u_val & 0x70) >> 4; - - return ((u_val & 0x80) ? (0x84 - t) : (t - 0x84)); -} diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_main.c b/openbsc/src/osmo-bsc_mgcp/mgcp_main.c index 5ac4c26..8c3808a 100644 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_main.c +++ b/openbsc/src/osmo-bsc_mgcp/mgcp_main.c @@ -31,11 +31,6 @@ #include <sys/socket.h> -#include "g711common.h" -#include <gsm.h> -#include <bcg729/decoder.h> -#include <bcg729/encoder.h> - #include <openbsc/debug.h> #include <openbsc/gsm_data.h> #include <openbsc/mgcp.h> @@ -55,7 +50,7 @@ #include "../../bscconfig.h" #ifdef BUILD_MGCP_TRANSCODING -#include "mgcp_transcode.h" +#include "openbsc/mgcp_transcode.h" #endif /* this is here for the vty... it will never be called */ diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c deleted file mode 100644 index edd3178..0000000 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c +++ /dev/null @@ -1,553 +0,0 @@ -/* - * (C) 2014 by On-Waves - * All Rights Reserved - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU Affero General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU Affero General Public License for more details. - * - * You should have received a copy of the GNU Affero General Public License - * along with this program. If not, see <http://www.gnu.org/licenses/>. - * - */ - -#include <stdlib.h> -#include <string.h> -#include <errno.h> - - -#include "../../bscconfig.h" - -#include "g711common.h" -#include <gsm.h> -#ifdef HAVE_BCG729 -#include <bcg729/decoder.h> -#include <bcg729/encoder.h> -#endif - -#include <openbsc/debug.h> -#include <openbsc/mgcp.h> -#include <openbsc/mgcp_internal.h> - -#include <osmocom/core/talloc.h> - -enum audio_format { - AF_INVALID, - AF_S16, - AF_L16, - AF_GSM, - AF_G729, - AF_PCMA -}; - -struct mgcp_process_rtp_state { - /* decoding */ - enum audio_format src_fmt; - union { - gsm gsm_handle; -#ifdef HAVE_BCG729 - bcg729DecoderChannelContextStruct *g729_dec; -#endif - } src; - size_t src_frame_size; - size_t src_samples_per_frame; - - /* processing */ - - /* encoding */ - enum audio_format dst_fmt; - union { - gsm gsm_handle; -#ifdef HAVE_BCG729 - bcg729EncoderChannelContextStruct *g729_enc; -#endif - } dst; - size_t dst_frame_size; - size_t dst_samples_per_frame; - int dst_packet_duration; - - int is_running; - uint16_t next_seq; - uint32_t next_time; - int16_t samples[10*160]; - size_t sample_cnt; - size_t sample_offs; -}; - -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) { - if (!strcmp("GSM", rtp_end->subtype_name)) - return AF_GSM; - if (!strcmp("PCMA", rtp_end->subtype_name)) - return AF_PCMA; -#ifdef HAVE_BCG729 - if (!strcmp("G729", rtp_end->subtype_name)) - return AF_G729; -#endif - if (!strcmp("L16", rtp_end->subtype_name)) - return AF_L16; - } - - switch (rtp_end->payload_type) { - case 3 /* GSM */: - return AF_GSM; - case 8 /* PCMA */: - return AF_PCMA; -#ifdef HAVE_BCG729 - case 18 /* G.729 */: - return AF_G729; -#endif - case 11 /* L16 */: - return AF_L16; - default: - return AF_INVALID; - } -} - -static void l16_encode(short *sample, unsigned char *buf, size_t n) -{ - for (; n > 0; --n, ++sample, buf += 2) { - buf[0] = sample[0] >> 8; - buf[1] = sample[0] & 0xff; - } -} - -static void l16_decode(unsigned char *buf, short *sample, size_t n) -{ - for (; n > 0; --n, ++sample, buf += 2) - sample[0] = ((short)buf[0] << 8) | buf[1]; -} - -static void alaw_encode(short *sample, unsigned char *buf, size_t n) -{ - for (; n > 0; --n) - *(buf++) = s16_to_alaw(*(sample++)); -} - -static void alaw_decode(unsigned char *buf, short *sample, size_t n) -{ - for (; n > 0; --n) - *(sample++) = alaw_to_s16(*(buf++)); -} - -static int processing_state_destructor(struct mgcp_process_rtp_state *state) -{ - switch (state->src_fmt) { - case AF_GSM: - if (state->dst.gsm_handle) - gsm_destroy(state->src.gsm_handle); - break; -#ifdef HAVE_BCG729 - case AF_G729: - if (state->src.g729_dec) - closeBcg729DecoderChannel(state->src.g729_dec); - break; -#endif - default: - break; - } - switch (state->dst_fmt) { - case AF_GSM: - if (state->dst.gsm_handle) - gsm_destroy(state->dst.gsm_handle); - break; -#ifdef HAVE_BCG729 - case AF_G729: - if (state->dst.g729_enc) - closeBcg729EncoderChannel(state->dst.g729_enc); - break; -#endif - default: - break; - } - return 0; -} - -int mgcp_transcoding_setup(struct mgcp_endpoint *endp, - struct mgcp_rtp_end *dst_end, - struct mgcp_rtp_end *src_end) -{ - struct mgcp_process_rtp_state *state = dst_end->rtp_process_data; - enum audio_format src_fmt, dst_fmt; - - /* cleanup first */ - if (state) { - talloc_free(state); - dst_end->rtp_process_data = NULL; - } - - if (!src_end) - return 0; - - src_fmt = get_audio_format(src_end); - dst_fmt = get_audio_format(dst_end); - - LOGP(DMGCP, LOGL_ERROR, - "Checking transcoding: %s (%d) -> %s (%d)\n", - src_end->subtype_name, src_end->payload_type, - dst_end->subtype_name, dst_end->payload_type); - - if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) { - if (!src_end->subtype_name || !dst_end->subtype_name) - /* Not enough info, do nothing */ - return 0; - - if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0) - /* Nothing to do */ - return 0; - - LOGP(DMGCP, LOGL_ERROR, - "Cannot transcode: %s codec not supported (%s -> %s).\n", - src_fmt != AF_INVALID ? "destination" : "source", - src_end->audio_name, dst_end->audio_name); - return -EINVAL; - } - - if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) { - LOGP(DMGCP, LOGL_ERROR, - "Cannot transcode: rate conversion (%d -> %d) not supported.\n", - src_end->rate, dst_end->rate); - return -EINVAL; - } - - state = talloc_zero(endp->tcfg->cfg, struct mgcp_process_rtp_state); - talloc_set_destructor(state, processing_state_destructor); - dst_end->rtp_process_data = state; - - state->src_fmt = src_fmt; - - switch (state->src_fmt) { - case AF_L16: - case AF_S16: - state->src_frame_size = 80 * sizeof(short); - state->src_samples_per_frame = 80; - break; - case AF_GSM: - state->src_frame_size = sizeof(gsm_frame); - state->src_samples_per_frame = 160; - state->src.gsm_handle = gsm_create(); - if (!state->src.gsm_handle) { - LOGP(DMGCP, LOGL_ERROR, - "Failed to initialize GSM decoder.\n"); - return -EINVAL; - } - break; -#ifdef HAVE_BCG729 - case AF_G729: - state->src_frame_size = 10; - state->src_samples_per_frame = 80; - state->src.g729_dec = initBcg729DecoderChannel(); - if (!state->src.g729_dec) { - LOGP(DMGCP, LOGL_ERROR, - "Failed to initialize G.729 decoder.\n"); - return -EINVAL; - } - break; -#endif - case AF_PCMA: - state->src_frame_size = 80; - state->src_samples_per_frame = 80; - break; - default: - break; - } - - state->dst_fmt = dst_fmt; - - switch (state->dst_fmt) { - case AF_L16: - case AF_S16: - state->dst_frame_size = 80*sizeof(short); - state->dst_samples_per_frame = 80; - break; - case AF_GSM: - state->dst_frame_size = sizeof(gsm_frame); - state->dst_samples_per_frame = 160; - state->dst.gsm_handle = gsm_create(); - if (!state->dst.gsm_handle) { - LOGP(DMGCP, LOGL_ERROR, - "Failed to initialize GSM encoder.\n"); - return -EINVAL; - } - break; -#ifdef HAVE_BCG729 - case AF_G729: - state->dst_frame_size = 10; - state->dst_samples_per_frame = 80; - state->dst.g729_enc = initBcg729EncoderChannel(); - if (!state->dst.g729_enc) { - LOGP(DMGCP, LOGL_ERROR, - "Failed to initialize G.729 decoder.\n"); - return -EINVAL; - } - break; -#endif - case AF_PCMA: - state->dst_frame_size = 80; - state->dst_samples_per_frame = 80; - break; - default: - break; - } - - if (dst_end->force_output_ptime) - state->dst_packet_duration = mgcp_rtp_packet_duration(endp, dst_end); - - LOGP(DMGCP, LOGL_INFO, - "Initialized RTP processing on: 0x%x " - "conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n", - ENDPOINT_NUMBER(endp), - src_fmt, src_end->payload_type, src_end->rate, src_end->fmtp_extra, - dst_fmt, dst_end->payload_type, dst_end->rate, dst_end->fmtp_extra); - - return 0; -} - -void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp, - int *payload_type, - const char**audio_name, - const char**fmtp_extra) -{ - struct mgcp_process_rtp_state *state = endp->net_end.rtp_process_data; - if (!state || endp->net_end.payload_type < 0) { - *payload_type = endp->bts_end.payload_type; - *audio_name = endp->bts_end.audio_name; - *fmtp_extra = endp->bts_end.fmtp_extra; - return; - } - - *payload_type = endp->net_end.payload_type; - *fmtp_extra = endp->net_end.fmtp_extra; - *audio_name = endp->net_end.audio_name; -} - -static int decode_audio(struct mgcp_process_rtp_state *state, - uint8_t **src, size_t *nbytes) -{ - while (*nbytes >= state->src_frame_size) { - if (state->sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(state->samples)) { - LOGP(DMGCP, LOGL_ERROR, - "Sample buffer too small: %d > %d.\n", - state->sample_cnt + state->src_samples_per_frame, - ARRAY_SIZE(state->samples)); - return -ENOSPC; - } - switch (state->src_fmt) { - case AF_GSM: - if (gsm_decode(state->src.gsm_handle, - (gsm_byte *)*src, state->samples + state->sample_cnt) < 0) { - LOGP(DMGCP, LOGL_ERROR, - "Failed to decode GSM.\n"); - return -EINVAL; - } - break; -#ifdef HAVE_BCG729 - case AF_G729: - bcg729Decoder(state->src.g729_dec, *src, 0, state->samples + state->sample_cnt); - break; -#endif - case AF_PCMA: - alaw_decode(*src, state->samples + state->sample_cnt, - state->src_samples_per_frame); - break; - case AF_S16: - memmove(state->samples + state->sample_cnt, *src, - state->src_frame_size); - break; - case AF_L16: - l16_decode(*src, state->samples + state->sample_cnt, - state->src_samples_per_frame); - break; - default: - break; - } - *src += state->src_frame_size; - *nbytes -= state->src_frame_size; - state->sample_cnt += state->src_samples_per_frame; - } - return 0; -} - -static int encode_audio(struct mgcp_process_rtp_state *state, - uint8_t *dst, size_t buf_size, size_t max_samples) -{ - int nbytes = 0; - size_t nsamples = 0; - /* Encode samples into dst */ - while (nsamples + state->dst_samples_per_frame <= max_samples) { - if (nbytes + state->dst_frame_size > buf_size) { - if (nbytes > 0) - break; - - /* Not even one frame fits into the buffer */ - LOGP(DMGCP, LOGL_INFO, - "Encoding (RTP) buffer too small: %d > %d.\n", - nbytes + state->dst_frame_size, buf_size); - return -ENOSPC; - } - switch (state->dst_fmt) { - case AF_GSM: - gsm_encode(state->dst.gsm_handle, - state->samples + state->sample_offs, dst); - break; -#ifdef HAVE_BCG729 - case AF_G729: - bcg729Encoder(state->dst.g729_enc, - state->samples + state->sample_offs, dst); - break; -#endif - case AF_PCMA: - alaw_encode(state->samples + state->sample_offs, dst, - state->src_samples_per_frame); - break; - case AF_S16: - memmove(dst, state->samples + state->sample_offs, - state->dst_frame_size); - break; - case AF_L16: - l16_encode(state->samples + state->sample_offs, dst, - state->src_samples_per_frame); - break; - default: - break; - } - dst += state->dst_frame_size; - nbytes += state->dst_frame_size; - state->sample_offs += state->dst_samples_per_frame; - nsamples += state->dst_samples_per_frame; - } - state->sample_cnt -= nsamples; - return nbytes; -} - -int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp, - struct mgcp_rtp_end *dst_end, - char *data, int *len, int buf_size) -{ - struct mgcp_process_rtp_state *state = dst_end->rtp_process_data; - size_t rtp_hdr_size = 12; - char *payload_data = data + rtp_hdr_size; - int payload_len = *len - rtp_hdr_size; - // size_t sample_idx; - uint8_t *src = (uint8_t *)payload_data; - uint8_t *dst = (uint8_t *)payload_data; - size_t nbytes = payload_len; - // size_t frame_remainder; - size_t nsamples; - size_t max_samples; - uint32_t ts_no; - int rc; - - if (!state) - return 0; - - if (state->src_fmt == state->dst_fmt) { - if (!state->dst_packet_duration) - return 0; - - /* TODO: repackage without transcoding */ - } - - /* If the remaining samples do not fit into a fixed ptime, - * a) discard them, if the next packet is much later - * b) add silence and * send it, if the current packet is not - * yet too late - * c) append the sample data, if the timestamp matches exactly - */ - - /* TODO: check payload type (-> G.711 comfort noise) */ - - if (payload_len > 0) { - ts_no = ntohl(*(uint32_t*)(data+4)); - if (!state->is_running) - state->next_seq = ntohs(*(uint32_t*)(data+4)); - - state->is_running = 1; - - if (state->sample_cnt > 0) { - int32_t delta = ts_no - state->next_time; - /* TODO: check sequence? reordering? packet loss? */ - - if (delta > state->sample_cnt) - /* There is a time gap between the last packet - * and the current one. Just discard the - * partial data that is left in the buffer. - * TODO: This can be improved by adding silence - * instead if the delta is small enough. - */ - state->sample_cnt = 0; - else if (delta < 0) { - LOGP(DMGCP, LOGL_NOTICE, - "RTP time jumps backwards, delta = %d, " - "discarding buffered samples\n", - delta); - state->sample_cnt = 0; - state->sample_offs = 0; - return -EAGAIN; - } - - /* Make sure the samples start without offset */ - fprintf(stderr, "Moving %d samples to buffer start (offset %d)\n", state->sample_cnt, state->sample_offs); - if (state->sample_offs && state->sample_cnt) - memmove(&state->samples[0], - &state->samples[state->sample_offs], - state->sample_cnt * sizeof(state->samples[0])); - } - - state->sample_offs = 0; - - /* Append decoded audio to samples */ - decode_audio(state, &src, &nbytes); - - if (nbytes > 0) - LOGP(DMGCP, LOGL_NOTICE, - "Skipped audio frame in RTP packet: %d octets\n", - nbytes); - } else - ts_no = state->next_time; - - if (state->sample_cnt < state->dst_packet_duration) - return -EAGAIN; - - max_samples = - state->dst_packet_duration ? - state->dst_packet_duration : state->sample_cnt; - - nsamples = state->sample_cnt; - - rc = encode_audio(state, dst, buf_size, max_samples); - if (rc <= 0) - return rc; - - nsamples -= state->sample_cnt; - fprintf(stderr, "Wrote %d samples to buffer (offset %d)\n", nsamples, state->sample_offs); - - *len = rtp_hdr_size + rc; - *(uint16_t*)(data+2) = htonl(state->next_seq); - *(uint32_t*)(data+4) = htonl(ts_no); - - state->next_seq += 1; - state->next_time = ts_no + nsamples; - - return nsamples ? rtp_hdr_size : 0; -} diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h deleted file mode 100644 index 0961634..0000000 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h +++ /dev/null @@ -1,36 +0,0 @@ -/* - * (C) 2014 by On-Waves - * All Rights Reserved - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU Affero General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU Affero General Public License for more details. - * - * You should have received a copy of the GNU Affero General Public License - * along with this program. If not, see <http://www.gnu.org/licenses/>. - * - */ -#ifndef OPENBSC_MGCP_TRANSCODE_H -#define OPENBSC_MGCP_TRANSCODE_H - -int mgcp_transcoding_setup(struct mgcp_endpoint *endp, - struct mgcp_rtp_end *dst_end, - struct mgcp_rtp_end *src_end); - -void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp, - int *payload_type, - const char**audio_name, - const char**fmtp_extra); - -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 */ diff --git a/openbsc/tests/mgcp/Makefile.am b/openbsc/tests/mgcp/Makefile.am index 81f28ae..2bc2da6 100644 --- a/openbsc/tests/mgcp/Makefile.am +++ b/openbsc/tests/mgcp/Makefile.am @@ -18,7 +18,7 @@ mgcp_test_LDADD = $(top_builddir)/src/libbsc/libbsc.a \ $(LIBOSMOCORE_LIBS) -lrt -lm $(LIBOSMOSCCP_LIBS) $(LIBOSMOVTY_LIBS) \ $(LIBRARY_DL) -mgcp_transcoding_test_SOURCES = mgcp_transcoding_test.c $(top_builddir)/src/osmo-bsc_mgcp/mgcp_transcode.c +mgcp_transcoding_test_SOURCES = mgcp_transcoding_test.c mgcp_transcoding_test_LDADD = \ $(top_builddir)/src/libbsc/libbsc.a \ diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c index a1af157..264dbe6 100644 --- a/openbsc/tests/mgcp/mgcp_transcoding_test.c +++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c @@ -17,7 +17,7 @@ #error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)" #endif -#include "src/osmo-bsc_mgcp/mgcp_transcode.h" +#include "openbsc/mgcp_transcode.h" uint8_t *audio_frame_l16[] = { }; -- 1.7.9.5

This patch adds the callbacks rtp_processing_cb and setup_rtp_processing_cb to mgcp_config to support arbitrary RTP payload processing. Sponsored-by: On-Waves ehf --- openbsc/include/openbsc/mgcp.h | 10 +++++++++ openbsc/include/openbsc/mgcp_internal.h | 8 ++++++++ openbsc/src/libmgcp/mgcp_network.c | 20 +++++++++++++++--- openbsc/src/libmgcp/mgcp_protocol.c | 34 ++++++++++++++++++++++++++++++- 4 files changed, 68 insertions(+), 4 deletions(-) diff --git a/openbsc/include/openbsc/mgcp.h b/openbsc/include/openbsc/mgcp.h index 1d74078..6ba0769 100644 --- a/openbsc/include/openbsc/mgcp.h +++ b/openbsc/include/openbsc/mgcp.h @@ -65,6 +65,7 @@ static inline int rtp_calculate_port(int multiplex, int base) struct mgcp_endpoint; struct mgcp_config; struct mgcp_trunk_config; +struct mgcp_rtp_end; #define MGCP_ENDP_CRCX 1 #define MGCP_ENDP_DLCX 2 @@ -86,6 +87,11 @@ typedef int (*mgcp_policy)(struct mgcp_trunk_config *cfg, int endpoint, int stat typedef int (*mgcp_reset)(struct mgcp_trunk_config *cfg); typedef int (*mgcp_rqnt)(struct mgcp_endpoint *endp, char tone); +typedef int (*mgcp_processing)(struct mgcp_rtp_end *dst_end, + char *data, int *len, int buf_size); +typedef int (*mgcp_processing_setup)(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + struct mgcp_rtp_end *src_end); #define PORT_ALLOC_STATIC 0 #define PORT_ALLOC_DYNAMIC 1 @@ -156,6 +162,10 @@ struct mgcp_config { struct in_addr transcoder_in; int transcoder_remote_base; + /* RTP processing */ + mgcp_processing rtp_processing_cb; + mgcp_processing_setup setup_rtp_processing_cb; + struct osmo_wqueue gw_fd; struct mgcp_port_range bts_ports; diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h index 9b97165..56c280d 100644 --- a/openbsc/include/openbsc/mgcp_internal.h +++ b/openbsc/include/openbsc/mgcp_internal.h @@ -91,6 +91,7 @@ struct mgcp_rtp_end { /* RTP patching */ int force_constant_ssrc; /* -1: always, 0: don't, 1: once */ int force_aligned_timing; + void *rtp_process_data; /* * Each end has a socket... @@ -197,5 +198,12 @@ void mgcp_state_calc_loss(struct mgcp_rtp_state *s, struct mgcp_rtp_end *, uint32_t *expected, int *loss); uint32_t mgcp_state_calc_jitter(struct mgcp_rtp_state *); +/* payload processing default functions */ +int mgcp_rtp_processing_default(struct mgcp_rtp_end *dst_end, + char *data, int *len, int buf_size); + +int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + struct mgcp_rtp_end *src_end); #endif diff --git a/openbsc/src/libmgcp/mgcp_network.c b/openbsc/src/libmgcp/mgcp_network.c index 8a5656a..4d1ad35 100644 --- a/openbsc/src/libmgcp/mgcp_network.c +++ b/openbsc/src/libmgcp/mgcp_network.c @@ -77,6 +77,7 @@ struct rtp_hdr { #define RTP_SEQ_MOD (1 << 16) #define RTP_MAX_DROPOUT 3000 #define RTP_MAX_MISORDER 100 +#define RTP_BUF_SIZE 4096 enum { @@ -347,6 +348,18 @@ static int align_rtp_timestamp_offset(struct mgcp_endpoint *endp, return timestamp_error; } +int mgcp_rtp_processing_default(struct mgcp_rtp_end *dst_end, + char *data, int *len, int buf_size) +{ + return 0; +} + +int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp, + struct mgcp_rtp_end *dst_end, + struct mgcp_rtp_end *src_end) +{ + return 0; +} /** * The RFC 3550 Appendix A assumes there are multiple sources but @@ -597,6 +610,7 @@ static int mgcp_send(struct mgcp_endpoint *endp, int dest, int is_rtp, rtp_end->dropped_packets += 1; else if (is_rtp) { mgcp_patch_and_count(endp, rtp_state, rtp_end, addr, buf, rc); + endp->cfg->rtp_processing_cb(rtp_end, buf, &rc, RTP_BUF_SIZE); forward_data(rtp_end->rtp.fd, &endp->taps[tap_idx], buf, rc); return mgcp_udp_send(rtp_end->rtp.fd, &rtp_end->addr, @@ -635,7 +649,7 @@ static int receive_from(struct mgcp_endpoint *endp, int fd, struct sockaddr_in * static int rtp_data_net(struct osmo_fd *fd, unsigned int what) { - char buf[4096]; + char buf[RTP_BUF_SIZE]; struct sockaddr_in addr; struct mgcp_endpoint *endp; int rc, proto; @@ -719,7 +733,7 @@ static void discover_bts(struct mgcp_endpoint *endp, int proto, struct sockaddr_ static int rtp_data_bts(struct osmo_fd *fd, unsigned int what) { - char buf[4096]; + char buf[RTP_BUF_SIZE]; struct sockaddr_in addr; struct mgcp_endpoint *endp; int rc, proto; @@ -781,7 +795,7 @@ static int rtp_data_bts(struct osmo_fd *fd, unsigned int what) static int rtp_data_transcoder(struct mgcp_rtp_end *end, struct mgcp_endpoint *_endp, int dest, struct osmo_fd *fd) { - char buf[4096]; + char buf[RTP_BUF_SIZE]; struct sockaddr_in addr; struct mgcp_config *cfg; int rc, proto; diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c index 5c88c9d..0f8614c 100644 --- a/openbsc/src/libmgcp/mgcp_protocol.c +++ b/openbsc/src/libmgcp/mgcp_protocol.c @@ -107,6 +107,8 @@ static struct msgb *handle_noti_req(struct mgcp_parse_data *data); static void create_transcoder(struct mgcp_endpoint *endp); static void delete_transcoder(struct mgcp_endpoint *endp); +static void setup_rtp_processing(struct mgcp_endpoint *endp); + static int mgcp_analyze_header(struct mgcp_parse_data *parse, char *data); static uint32_t generate_call_id(struct mgcp_config *cfg) @@ -827,8 +829,10 @@ mgcp_header_done: endp->bts_end.payload_type = tcfg->audio_payload; endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints, tcfg->audio_fmtp_extra); - if (have_sdp) + if (have_sdp) { parse_sdp_data(&endp->net_end, p); + setup_rtp_processing(endp); + } /* policy CB */ if (p->cfg->policy_cb) { @@ -929,6 +933,8 @@ static struct msgb *handle_modify_con(struct mgcp_parse_data *p) set_local_cx_options(endp->tcfg->endpoints, &endp->local_options, local_options); + setup_rtp_processing(endp); + /* policy CB */ if (p->cfg->policy_cb) { int rc; @@ -1169,6 +1175,9 @@ struct mgcp_config *mgcp_config_alloc(void) cfg->bts_ports.base_port = RTP_PORT_DEFAULT; cfg->net_ports.base_port = RTP_PORT_NET_DEFAULT; + cfg->rtp_processing_cb = &mgcp_rtp_processing_default; + cfg->setup_rtp_processing_cb = &mgcp_setup_rtp_processing_default; + /* default trunk handling */ cfg->trunk.cfg = cfg; cfg->trunk.trunk_nr = 0; @@ -1234,6 +1243,8 @@ static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end) end->local_alloc = -1; talloc_free(end->fmtp_extra); end->fmtp_extra = NULL; + talloc_free(end->rtp_process_data); + end->rtp_process_data = NULL; /* Set default values */ end->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM; @@ -1388,6 +1399,27 @@ int mgcp_send_reset_ep(struct mgcp_endpoint *endp, int endpoint) return send_agent(endp->cfg, buf, len); } +static void setup_rtp_processing(struct mgcp_endpoint *endp) +{ + struct mgcp_config *cfg = endp->cfg; + + if (endp->type != MGCP_RTP_DEFAULT) + return; + + if (endp->conn_mode == MGCP_CONN_LOOPBACK) + return; + + if (endp->conn_mode & MGCP_CONN_SEND_ONLY) + cfg->setup_rtp_processing_cb(endp, &endp->net_end, &endp->bts_end); + else + cfg->setup_rtp_processing_cb(endp, &endp->net_end, NULL); + + if (endp->conn_mode & MGCP_CONN_RECV_ONLY) + cfg->setup_rtp_processing_cb(endp, &endp->bts_end, &endp->net_end); + else + cfg->setup_rtp_processing_cb(endp, &endp->bts_end, NULL); +} + static void create_transcoder(struct mgcp_endpoint *endp) { int port; -- 1.7.9.5