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Merge branch 'zecke/features/dynamic-codec-switch'

This commit is contained in:
Holger Hans Peter Freyther 2014-09-02 09:28:44 +02:00
parent c20a661272 4680121fe6
commit e3283ec3eb
12 changed files with 384 additions and 112 deletions
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10
openbsc/contrib/testconv/testconv_main.c
View File

@ -64,16 +64,16 @@ int main(int argc, char **argv)
if (argc <= 2)
errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16} [SPP]");
if ((src_end->payload_type = audio_name_to_type(argv[1])) == -1)
if ((src_end->codec.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)
if ((dst_end->codec.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->codec.frame_duration_den = dst_end->codec.rate;
dst_end->codec.frame_duration_num = out_samples;
dst_end->frames_per_packet = 1;
}
@ -87,7 +87,7 @@ int main(int argc, char **argv)
in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
OSMO_ASSERT(sizeof(buf) >= in_size + 12);
buf[1] = src_end->payload_type;
buf[1] = src_end->codec.payload_type;
*(uint16_t*)(buf+2) = htons(1);
*(uint32_t*)(buf+4) = htonl(0);
*(uint32_t*)(buf+8) = htonl(0xaabbccdd);

22
openbsc/include/openbsc/mgcp_internal.h
View File

@ -67,6 +67,17 @@ struct mgcp_rtp_state {
struct mgcp_rtp_stream_state out_stream;
};
struct mgcp_rtp_codec {
uint32_t rate;
int channels;
uint32_t frame_duration_num;
uint32_t frame_duration_den;
int payload_type;
char *audio_name;
char *subtype_name;
};
struct mgcp_rtp_end {
/* statistics */
unsigned int packets;
@ -77,17 +88,14 @@ struct mgcp_rtp_end {
/* in network byte order */
int rtp_port, rtcp_port;
/* audio codec information */
struct mgcp_rtp_codec codec;
struct mgcp_rtp_codec alt_codec; /* TODO/XXX: make it generic */
/* 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;
int force_output_ptime;

1
openbsc/include/openbsc/rtp.h
View File

@ -35,4 +35,5 @@ struct rtp_hdr {
uint16_t sequence;
uint32_t timestamp;
uint32_t ssrc;
uint8_t data[0];
} __attribute__((packed));

12
openbsc/src/libmgcp/mgcp_network.c
View File

@ -239,7 +239,7 @@ static int adjust_rtp_timestamp_offset(struct mgcp_endpoint *endp,
inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
endp->conn_mode);
} else {
tsdelta = rtp_end->rate * 20 / 1000;
tsdelta = rtp_end->codec.rate * 20 / 1000;
LOGP(DMGCP, LOGL_NOTICE,
"Fixed packet duration and last timestamp delta "
"are not available on 0x%x, "
@ -326,8 +326,8 @@ void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
/* 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;
*payload_type = endp->bts_end.codec.payload_type;
*audio_name = endp->bts_end.codec.audio_name;
*fmtp_extra = endp->bts_end.fmtp_extra;
}
@ -348,7 +348,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
uint16_t seq, udelta;
uint32_t timestamp, ssrc;
struct rtp_hdr *rtp_hdr;
int payload = rtp_end->payload_type;
int payload = rtp_end->codec.payload_type;
if (len < sizeof(*rtp_hdr))
return;
@ -356,7 +356,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
rtp_hdr = (struct rtp_hdr *) data;
seq = ntohs(rtp_hdr->sequence);
timestamp = ntohl(rtp_hdr->timestamp);
arrival_time = get_current_ts(rtp_end->rate);
arrival_time = get_current_ts(rtp_end->codec.rate);
ssrc = ntohl(rtp_hdr->ssrc);
transit = arrival_time - timestamp;
@ -380,7 +380,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
endp->conn_mode);
if (state->packet_duration == 0) {
state->packet_duration = rtp_end->rate * 20 / 1000;
state->packet_duration = rtp_end->codec.rate * 20 / 1000;
LOGP(DMGCP, LOGL_NOTICE,
"Fixed packet duration is not available on 0x%x, "
"using fixed 20ms instead: %d from %s:%d in %d\n",

107
openbsc/src/libmgcp/mgcp_protocol.c
View File

@ -598,20 +598,20 @@ static int parse_conn_mode(const char *msg, struct mgcp_endpoint *endp)
return ret;
}
static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
static int set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
int payload_type, const char *audio_name)
{
int rate = rtp->rate;
int channels = rtp->channels;
int rate = codec->rate;
int channels = codec->channels;
char audio_codec[64];
talloc_free(rtp->subtype_name);
rtp->subtype_name = NULL;
talloc_free(rtp->audio_name);
rtp->audio_name = NULL;
talloc_free(codec->subtype_name);
codec->subtype_name = NULL;
talloc_free(codec->audio_name);
codec->audio_name = NULL;
if (payload_type != PTYPE_UNDEFINED)
rtp->payload_type = payload_type;
codec->payload_type = payload_type;
if (!audio_name) {
switch (payload_type) {
@ -630,17 +630,17 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
audio_codec, &rate, &channels) < 1)
return -EINVAL;
rtp->rate = rate;
rtp->channels = channels;
rtp->subtype_name = talloc_strdup(ctx, audio_codec);
rtp->audio_name = talloc_strdup(ctx, audio_name);
codec->rate = rate;
codec->channels = channels;
codec->subtype_name = talloc_strdup(ctx, audio_codec);
codec->audio_name = talloc_strdup(ctx, audio_name);
if (!strcmp(audio_codec, "G729")) {
rtp->frame_duration_num = 10;
rtp->frame_duration_den = 1000;
codec->frame_duration_num = 10;
codec->frame_duration_den = 1000;
} else {
rtp->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
rtp->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
}
if (payload_type < 0) {
@ -654,7 +654,7 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
payload_type = 18;
}
rtp->payload_type = payload_type;
codec->payload_type = payload_type;
}
if (channels != 1)
@ -738,7 +738,9 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
{
char *line;
int found_media = 0;
/* TODO/XXX make it more generic */
int audio_payload = -1;
int audio_payload_alt = -1;
for_each_line(line, p->save) {
switch (line[0]) {
@ -758,10 +760,12 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
if (sscanf(line, "a=rtpmap:%d %63s",
&payload, audio_name) == 2) {
if (payload != audio_payload)
break;
set_audio_info(p->cfg, rtp, payload, audio_name);
if (payload == audio_payload)
set_audio_info(p->cfg, &rtp->codec,
payload, audio_name);
else if (payload == audio_payload_alt)
set_audio_info(p->cfg, &rtp->alt_codec,
payload, audio_name);
} else if (sscanf(line, "a=ptime:%d-%d",
&ptime, &ptime2) >= 1) {
if (ptime2 > 0 && ptime2 != ptime)
@ -769,23 +773,29 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
else
rtp->packet_duration_ms = ptime;
} else if (sscanf(line, "a=maxptime:%d", &ptime2) == 1) {
if (ptime2 * rtp->frame_duration_den >
rtp->frame_duration_num * 1500)
/* TODO/XXX: Store this per codec and derive it on use */
if (ptime2 * rtp->codec.frame_duration_den >
rtp->codec.frame_duration_num * 1500)
/* more than 1 frame */
rtp->packet_duration_ms = 0;
}
break;
}
case 'm': {
int port;
int port, rc;
audio_payload = -1;
audio_payload_alt = -1;
if (sscanf(line, "m=audio %d RTP/AVP %d",
&port, &audio_payload) == 2) {
rc = sscanf(line, "m=audio %d RTP/AVP %d %d",
&port, &audio_payload, &audio_payload_alt);
if (rc >= 2) {
rtp->rtp_port = htons(port);
rtp->rtcp_port = htons(port + 1);
found_media = 1;
set_audio_info(p->cfg, rtp, audio_payload, NULL);
set_audio_info(p->cfg, &rtp->codec, audio_payload, NULL);
if (rc == 3)
set_audio_info(p->cfg, &rtp->alt_codec,
audio_payload_alt, NULL);
}
break;
}
@ -814,8 +824,8 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
LOGP(DMGCP, LOGL_NOTICE,
"Got media info via SDP: port %d, payload %d (%s), "
"duration %d, addr %s\n",
ntohs(rtp->rtp_port), rtp->payload_type,
rtp->subtype_name ? rtp->subtype_name : "unknown",
ntohs(rtp->rtp_port), rtp->codec.payload_type,
rtp->codec.subtype_name ? rtp->codec.subtype_name : "unknown",
rtp->packet_duration_ms, inet_ntoa(rtp->addr));
return found_media;
@ -872,13 +882,13 @@ uint32_t mgcp_rtp_packet_duration(struct mgcp_endpoint *endp,
/* Get the number of frames per channel and packet */
if (rtp->frames_per_packet)
f = rtp->frames_per_packet;
else if (rtp->packet_duration_ms && rtp->frame_duration_num) {
int den = 1000 * rtp->frame_duration_num;
f = (rtp->packet_duration_ms * rtp->frame_duration_den + den/2)
else if (rtp->packet_duration_ms && rtp->codec.frame_duration_num) {
int den = 1000 * rtp->codec.frame_duration_num;
f = (rtp->packet_duration_ms * rtp->codec.frame_duration_den + den/2)
/ den;
}
return rtp->rate * f * rtp->frame_duration_num / rtp->frame_duration_den;
return rtp->codec.rate * f * rtp->codec.frame_duration_num / rtp->codec.frame_duration_den;
}
static int mgcp_parse_osmux_cid(const char *line)
@ -1025,13 +1035,13 @@ mgcp_header_done:
endp->allocated = 1;
/* set up RTP media parameters */
set_audio_info(p->cfg, &endp->bts_end, tcfg->audio_payload, tcfg->audio_name);
set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints,
tcfg->audio_fmtp_extra);
if (have_sdp)
parse_sdp_data(&endp->net_end, p);
else if (endp->local_options.codec)
set_audio_info(p->cfg, &endp->net_end,
set_audio_info(p->cfg, &endp->net_end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
if (p->cfg->bts_force_ptime) {
@ -1147,7 +1157,7 @@ static struct msgb *handle_modify_con(struct mgcp_parse_data *p)
local_options);
if (!have_sdp && endp->local_options.codec)
set_audio_info(p->cfg, &endp->net_end,
set_audio_info(p->cfg, &endp->net_end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
setup_rtp_processing(endp);
@ -1449,6 +1459,19 @@ struct mgcp_trunk_config *mgcp_trunk_num(struct mgcp_config *cfg, int index)
return NULL;
}
static void mgcp_rtp_codec_reset(struct mgcp_rtp_codec *codec)
{
codec->payload_type = -1;
talloc_free(codec->subtype_name);
codec->subtype_name = NULL;
talloc_free(codec->audio_name);
codec->audio_name = NULL;
codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
codec->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
codec->channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
}
static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end)
{
if (end->local_alloc == PORT_ALLOC_DYNAMIC) {
@ -1461,25 +1484,19 @@ static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end)
end->dropped_packets = 0;
memset(&end->addr, 0, sizeof(end->addr));
end->rtp_port = end->rtcp_port = 0;
end->payload_type = -1;
end->local_alloc = -1;
talloc_free(end->fmtp_extra);
end->fmtp_extra = NULL;
talloc_free(end->subtype_name);
end->subtype_name = NULL;
talloc_free(end->audio_name);
end->audio_name = 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;
end->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
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;
mgcp_rtp_codec_reset(&end->codec);
mgcp_rtp_codec_reset(&end->alt_codec);
}
static void mgcp_rtp_end_init(struct mgcp_rtp_end *end)

119
openbsc/src/libmgcp/mgcp_transcode.c
View File

@ -28,6 +28,7 @@
#include <openbsc/mgcp.h>
#include <openbsc/mgcp_internal.h>
#include <openbsc/mgcp_transcode.h>
#include <openbsc/rtp.h>
#include <osmocom/core/talloc.h>
@ -44,22 +45,22 @@ int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
1) * state->src_frame_size;
}
static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
static enum audio_format get_audio_format(const struct mgcp_rtp_codec *codec)
{
if (rtp_end->subtype_name) {
if (!strcmp("GSM", rtp_end->subtype_name))
if (codec->subtype_name) {
if (!strcmp("GSM", codec->subtype_name))
return AF_GSM;
if (!strcmp("PCMA", rtp_end->subtype_name))
if (!strcmp("PCMA", codec->subtype_name))
return AF_PCMA;
#ifdef HAVE_BCG729
if (!strcmp("G729", rtp_end->subtype_name))
if (!strcmp("G729", codec->subtype_name))
return AF_G729;
#endif
if (!strcmp("L16", rtp_end->subtype_name))
if (!strcmp("L16", codec->subtype_name))
return AF_L16;
}
switch (rtp_end->payload_type) {
switch (codec->payload_type) {
case 3 /* GSM */:
return AF_GSM;
case 8 /* PCMA */:
@ -140,6 +141,8 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
{
struct mgcp_process_rtp_state *state;
enum audio_format src_fmt, dst_fmt;
const struct mgcp_rtp_codec *src_codec = &src_end->codec;
const struct mgcp_rtp_codec *dst_codec = &dst_end->codec;
/* cleanup first */
if (dst_end->rtp_process_data) {
@ -150,34 +153,34 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
if (!src_end)
return 0;
src_fmt = get_audio_format(src_end);
dst_fmt = get_audio_format(dst_end);
src_fmt = get_audio_format(src_codec);
dst_fmt = get_audio_format(dst_codec);
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);
src_codec->subtype_name, src_codec->payload_type,
dst_codec->subtype_name, dst_codec->payload_type);
if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
if (!src_end->subtype_name || !dst_end->subtype_name)
if (!src_codec->subtype_name || !dst_codec->subtype_name)
/* Not enough info, do nothing */
return 0;
if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
if (strcmp(src_codec->subtype_name, dst_codec->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);
src_codec->audio_name, dst_codec->audio_name);
return -EINVAL;
}
if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
if (src_codec->rate && dst_codec->rate && src_codec->rate != dst_codec->rate) {
LOGP(DMGCP, LOGL_ERROR,
"Cannot transcode: rate conversion (%d -> %d) not supported.\n",
src_end->rate, dst_end->rate);
src_codec->rate, dst_codec->rate);
return -EINVAL;
}
@ -268,8 +271,8 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
"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);
src_fmt, src_codec->payload_type, src_codec->rate, src_end->fmtp_extra,
dst_fmt, dst_codec->payload_type, dst_codec->rate, dst_end->fmtp_extra);
return 0;
}
@ -280,16 +283,19 @@ void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
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;
struct mgcp_rtp_codec *net_codec = &endp->net_end.codec;
struct mgcp_rtp_codec *bts_codec = &endp->bts_end.codec;
if (!state || net_codec->payload_type < 0) {
*payload_type = bts_codec->payload_type;
*audio_name = bts_codec->audio_name;
*fmtp_extra = endp->bts_end.fmtp_extra;
return;
}
*payload_type = endp->net_end.payload_type;
*payload_type = net_codec->payload_type;
*audio_name = net_codec->audio_name;
*fmtp_extra = endp->net_end.fmtp_extra;
*audio_name = endp->net_end.audio_name;
}
static int decode_audio(struct mgcp_process_rtp_state *state,
@ -391,13 +397,65 @@ static int encode_audio(struct mgcp_process_rtp_state *state,
return nbytes;
}
static struct mgcp_rtp_end *source_for_dest(struct mgcp_endpoint *endp,
struct mgcp_rtp_end *dst_end)
{
if (&endp->bts_end == dst_end)
return &endp->net_end;
else if (&endp->net_end == dst_end)
return &endp->bts_end;
OSMO_ASSERT(0);
}
/*
* With some modems we get offered multiple codecs
* and we have selected one of them. It might not
* be the right one and we need to detect this with
* the first audio packets. One difficulty is that
* we patch the rtp payload type in place, so we
* need to discuss this.
*/
struct mgcp_process_rtp_state *check_transcode_state(
struct mgcp_endpoint *endp,
struct mgcp_rtp_end *dst_end,
struct rtp_hdr *rtp_hdr)
{
struct mgcp_rtp_end *src_end;
/* Only deal with messages from net to bts */
if (&endp->bts_end != dst_end)
goto done;
src_end = source_for_dest(endp, dst_end);
/* Already patched */
if (rtp_hdr->payload_type == dst_end->codec.payload_type)
goto done;
/* The payload we expect */
if (rtp_hdr->payload_type == src_end->codec.payload_type)
goto done;
/* The matching alternate payload type? Then switch */
if (rtp_hdr->payload_type == src_end->alt_codec.payload_type) {
struct mgcp_config *cfg = endp->cfg;
struct mgcp_rtp_codec tmp_codec = src_end->alt_codec;
src_end->alt_codec = src_end->codec;
src_end->codec = tmp_codec;
cfg->setup_rtp_processing_cb(endp, &endp->net_end, &endp->bts_end);
cfg->setup_rtp_processing_cb(endp, &endp->bts_end, &endp->net_end);
}
done:
return dst_end->rtp_process_data;
}
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;
struct mgcp_process_rtp_state *state;
const size_t rtp_hdr_size = sizeof(struct rtp_hdr);
struct rtp_hdr *rtp_hdr = (struct rtp_hdr *) data;
char *payload_data = (char *) &rtp_hdr->data[0];
int payload_len = *len - rtp_hdr_size;
uint8_t *src = (uint8_t *)payload_data;
uint8_t *dst = (uint8_t *)payload_data;
@ -407,6 +465,7 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
uint32_t ts_no;
int rc;
state = check_transcode_state(endp, dst_end, rtp_hdr);
if (!state)
return 0;
@ -427,9 +486,9 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
/* TODO: check payload type (-> G.711 comfort noise) */
if (payload_len > 0) {
ts_no = ntohl(*(uint32_t*)(data+4));
ts_no = ntohl(rtp_hdr->timestamp);
if (!state->is_running) {
state->next_seq = ntohs(*(uint16_t*)(data+2));
state->next_seq = ntohs(rtp_hdr->sequence);
state->next_time = ts_no;
state->is_running = 1;
}
@ -504,8 +563,8 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
nsamples -= state->sample_cnt;
*len = rtp_hdr_size + rc;
*(uint16_t*)(data+2) = htons(state->next_seq);
*(uint32_t*)(data+4) = htonl(ts_no);
rtp_hdr->sequence = htons(state->next_seq);
rtp_hdr->timestamp = htonl(ts_no);
state->next_seq += 1;
state->next_time = ts_no + nsamples;

8
openbsc/src/libmgcp/mgcp_vty.c
View File

@ -147,6 +147,8 @@ static int config_write_mgcp(struct vty *vty)
static void dump_rtp_end(const char *end_name, struct vty *vty,
struct mgcp_rtp_state *state, struct mgcp_rtp_end *end)
{
struct mgcp_rtp_codec *codec = &end->codec;
vty_out(vty,
" %s%s"
" Timestamp Errs: %d->%d%s"
@ -160,10 +162,10 @@ static void dump_rtp_end(const char *end_name, struct vty *vty,
state->in_stream.err_ts_counter,
state->out_stream.err_ts_counter, VTY_NEWLINE,
end->dropped_packets, VTY_NEWLINE,
end->payload_type, end->rate, end->channels, VTY_NEWLINE,
end->frame_duration_num, end->frame_duration_den, VTY_NEWLINE,
codec->payload_type, codec->rate, codec->channels, VTY_NEWLINE,
codec->frame_duration_num, codec->frame_duration_den, VTY_NEWLINE,
end->frames_per_packet, end->packet_duration_ms, VTY_NEWLINE,
end->fmtp_extra, end->audio_name, end->subtype_name, VTY_NEWLINE,
end->fmtp_extra, codec->audio_name, codec->subtype_name, VTY_NEWLINE,
end->output_enabled, end->force_output_ptime, VTY_NEWLINE);
}

4
openbsc/src/osmo-bsc_nat/bsc_mgcp_utils.c
View File

@ -552,7 +552,7 @@ static int bsc_mgcp_policy_cb(struct mgcp_trunk_config *tcfg, int endpoint, int
bsc_msg = bsc_mgcp_rewrite((char *) nat->mgcp_msg, nat->mgcp_length,
sccp->bsc_endp, nat->mgcp_cfg->source_addr,
mgcp_endp->bts_end.local_port, osmux_cid,
&mgcp_endp->net_end.payload_type);
&mgcp_endp->net_end.codec.payload_type);
if (!bsc_msg) {
LOGP(DMGCP, LOGL_ERROR, "Failed to patch the msg.\n");
return MGCP_POLICY_CONT;
@ -746,7 +746,7 @@ void bsc_mgcp_forward(struct bsc_connection *bsc, struct msgb *msg)
output = bsc_mgcp_rewrite((char * ) msg->l2h, msgb_l2len(msg), -1,
bsc->nat->mgcp_cfg->source_addr,
endp->net_end.local_port, -1,
&endp->bts_end.payload_type);
&endp->bts_end.codec.payload_type);
if (!output) {
LOGP(DMGCP, LOGL_ERROR, "Failed to rewrite MGCP msg.\n");
return;

140
openbsc/tests/mgcp/mgcp_test.c
View File

@ -1,6 +1,6 @@
/*
* (C) 2011-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
* (C) 2011-2012 by On-Waves
* (C) 2011-2012,2014 by Holger Hans Peter Freyther <zecke@selfish.org>
* (C) 2011-2012,2014 by On-Waves
* All Rights Reserved
*
* This program is free software; you can redistribute it and/or modify
@ -268,6 +268,61 @@ static void test_strline(void)
#define PTYPE_NONE 128
#define PTYPE_NYI PTYPE_NONE
#define CRCX_MULT_1 "CRCX 2 1@mgw MGCP 1.0\r\n" \
"M: recvonly\r\n" \
"C: 2\r\n" \
"X\r\n" \
"L: p:20\r\n" \
"\r\n" \
"v=0\r\n" \
"c=IN IP4 123.12.12.123\r\n" \
"m=audio 5904 RTP/AVP 18 97\r\n"\
"a=rtpmap:18 G729/8000\r\n" \
"a=rtpmap:97 GSM-EFR/8000\r\n" \
"a=ptime:40\r\n"
#define CRCX_MULT_2 "CRCX 2 2@mgw MGCP 1.0\r\n" \
"M: recvonly\r\n" \
"C: 2\r\n" \
"X\r\n" \
"L: p:20\r\n" \
"\r\n" \
"v=0\r\n" \
"c=IN IP4 123.12.12.123\r\n" \
"m=audio 5904 RTP/AVP 18 97 101\r\n"\
"a=rtpmap:18 G729/8000\r\n" \
"a=rtpmap:97 GSM-EFR/8000\r\n" \
"a=rtpmap:101 FOO/8000\r\n" \
"a=ptime:40\r\n"
#define CRCX_MULT_3 "CRCX 2 3@mgw MGCP 1.0\r\n" \
"M: recvonly\r\n" \
"C: 2\r\n" \
"X\r\n" \
"L: p:20\r\n" \
"\r\n" \
"v=0\r\n" \
"c=IN IP4 123.12.12.123\r\n" \
"m=audio 5904 RTP/AVP\r\n" \
"a=rtpmap:18 G729/8000\r\n" \
"a=rtpmap:97 GSM-EFR/8000\r\n" \
"a=rtpmap:101 FOO/8000\r\n" \
"a=ptime:40\r\n"
#define CRCX_MULT_4 "CRCX 2 4@mgw MGCP 1.0\r\n" \
"M: recvonly\r\n" \
"C: 2\r\n" \
"X\r\n" \
"L: p:20\r\n" \
"\r\n" \
"v=0\r\n" \
"c=IN IP4 123.12.12.123\r\n" \
"m=audio 5904 RTP/AVP 18\r\n" \
"a=rtpmap:18 G729/8000\r\n" \
"a=rtpmap:97 GSM-EFR/8000\r\n" \
"a=rtpmap:101 FOO/8000\r\n" \
"a=ptime:40\r\n"
struct mgcp_test {
const char *name;
const char *req;
@ -410,7 +465,7 @@ static void test_messages(void)
/* reset endpoints */
for (i = 0; i < cfg->trunk.number_endpoints; i++) {
endp = &cfg->trunk.endpoints[i];
endp->net_end.payload_type = PTYPE_NONE;
endp->net_end.codec.payload_type = PTYPE_NONE;
endp->net_end.packet_duration_ms = -1;
OSMO_ASSERT(endp->conn_mode == MGCP_CONN_NONE);
@ -498,18 +553,18 @@ static void test_messages(void)
fprintf(stderr, "endpoint %d: "
"payload type BTS %d (exp %d), NET %d (exp %d)\n",
last_endpoint,
endp->bts_end.payload_type, t->exp_bts_ptype,
endp->net_end.payload_type, t->exp_net_ptype);
endp->bts_end.codec.payload_type, t->exp_bts_ptype,
endp->net_end.codec.payload_type, t->exp_net_ptype);
if (t->exp_bts_ptype != PTYPE_IGNORE)
OSMO_ASSERT(endp->bts_end.payload_type ==
OSMO_ASSERT(endp->bts_end.codec.payload_type ==
t->exp_bts_ptype);
if (t->exp_net_ptype != PTYPE_IGNORE)
OSMO_ASSERT(endp->net_end.payload_type ==
OSMO_ASSERT(endp->net_end.codec.payload_type ==
t->exp_net_ptype);
/* Reset them again for next test */
endp->net_end.payload_type = PTYPE_NONE;
endp->net_end.codec.payload_type = PTYPE_NONE;
}
}
@ -830,7 +885,7 @@ static void test_packet_error_detection(int patch_ssrc, int patch_ts)
mgcp_initialize_endp(&endp);
rtp->payload_type = 98;
rtp->codec.payload_type = 98;
for (i = 0; i < ARRAY_SIZE(test_rtp_packets1); ++i) {
struct rtp_packet_info *info = test_rtp_packets1 + i;
@ -877,6 +932,72 @@ static void test_packet_error_detection(int patch_ssrc, int patch_ts)
force_monotonic_time_us = -1;
}
static void test_multilple_codec(void)
{
struct mgcp_config *cfg;
struct mgcp_endpoint *endp;
struct msgb *inp, *resp;
printf("Testing multiple payload types\n");
cfg = mgcp_config_alloc();
cfg->trunk.number_endpoints = 64;
mgcp_endpoints_allocate(&cfg->trunk);
cfg->policy_cb = mgcp_test_policy_cb;
mgcp_endpoints_allocate(mgcp_trunk_alloc(cfg, 1));
/* Allocate endpoint 1@mgw with two codecs */
last_endpoint = -1;
inp = create_msg(CRCX_MULT_1);
resp = mgcp_handle_message(cfg, inp);
msgb_free(inp);
msgb_free(resp);
OSMO_ASSERT(last_endpoint == 1);
endp = &cfg->trunk.endpoints[last_endpoint];
OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 97);
/* Allocate 2@mgw with three codecs, last one ignored */
last_endpoint = -1;
inp = create_msg(CRCX_MULT_2);
resp = mgcp_handle_message(cfg, inp);
msgb_free(inp);
msgb_free(resp);
OSMO_ASSERT(last_endpoint == 2);
endp = &cfg->trunk.endpoints[last_endpoint];
OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 97);
/* Allocate 3@mgw with no codecs, check for PT == -1 */
last_endpoint = -1;
inp = create_msg(CRCX_MULT_3);
resp = mgcp_handle_message(cfg, inp);
msgb_free(inp);
msgb_free(resp);
OSMO_ASSERT(last_endpoint == 3);
endp = &cfg->trunk.endpoints[last_endpoint];
OSMO_ASSERT(endp->net_end.codec.payload_type == -1);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == -1);
/* Allocate 4@mgw with a single codec */
last_endpoint = -1;
inp = create_msg(CRCX_MULT_4);
resp = mgcp_handle_message(cfg, inp);
msgb_free(inp);
msgb_free(resp);
OSMO_ASSERT(last_endpoint == 4);
endp = &cfg->trunk.endpoints[last_endpoint];
OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == -1);
talloc_free(cfg);
}
int main(int argc, char **argv)
{
osmo_init_logging(&log_info);
@ -892,6 +1013,7 @@ int main(int argc, char **argv)
test_packet_error_detection(0, 0);
test_packet_error_detection(0, 1);
test_packet_error_detection(1, 1);
test_multilple_codec();
printf("Done\n");
return EXIT_SUCCESS;

1
openbsc/tests/mgcp/mgcp_test.ok
View File

@ -474,4 +474,5 @@ Stats: Jitter = 0, Transit = -144000
In TS: 160320, dTS: 160, Seq: 1002
Out TS change: 160, dTS: 160, Seq change: 1, TS Err change: in +0, out +0
Stats: Jitter = 0, Transit = -144000
Testing multiple payload types
Done

71
openbsc/tests/mgcp/mgcp_transcoding_test.c
View File

@ -174,6 +174,9 @@ static int given_configured_endpoint(int in_samples, int out_samples,
tcfg = talloc_zero(cfg, struct mgcp_trunk_config);
endp = talloc_zero(tcfg, struct mgcp_endpoint);
cfg->setup_rtp_processing_cb = mgcp_transcoding_setup;
cfg->rtp_processing_cb = mgcp_transcoding_process_rtp;
cfg->get_net_downlink_format_cb = mgcp_transcoding_net_downlink_format;
tcfg->endpoints = endp;
tcfg->number_endpoints = 1;
@ -183,14 +186,14 @@ static int given_configured_endpoint(int in_samples, int out_samples,
mgcp_initialize_endp(endp);
dst_end = &endp->bts_end;
dst_end->payload_type = audio_name_to_type(dstfmt);
dst_end->codec.payload_type = audio_name_to_type(dstfmt);
src_end = &endp->net_end;
src_end->payload_type = audio_name_to_type(srcfmt);
src_end->codec.payload_type = audio_name_to_type(srcfmt);
if (out_samples) {
dst_end->frame_duration_den = dst_end->rate;
dst_end->frame_duration_num = out_samples;
dst_end->codec.frame_duration_den = dst_end->codec.rate;
dst_end->codec.frame_duration_num = out_samples;
dst_end->frames_per_packet = 1;
dst_end->force_output_ptime = 1;
}
@ -433,6 +436,63 @@ static void test_transcode_result(void)
}
}
static void test_transcode_change(void)
{
char buf[4096] = {0x80, 0};
void *ctx;
struct mgcp_endpoint *endp;
struct mgcp_process_rtp_state *state;
struct rtp_hdr *hdr;
int len, res;
{
/* from GSM to PCMA and same ptime */
printf("Testing Initial G729->GSM, PCMA->GSM\n");
given_configured_endpoint(160, 0, "g729", "gsm", &ctx, &endp);
endp->net_end.alt_codec = endp->net_end.codec;
endp->net_end.alt_codec.payload_type = audio_name_to_type("pcma");
state = endp->bts_end.rtp_process_data;
/* initial transcoding work */
OSMO_ASSERT(state->src_fmt == AF_G729);
OSMO_ASSERT(state->dst_fmt == AF_GSM);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 8);
OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
/* result */
len = audio_packets_pcma[0].len;
memcpy(buf, audio_packets_pcma[0].data, len);
res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
OSMO_ASSERT(res == sizeof(struct rtp_hdr));
OSMO_ASSERT(state->sample_cnt == 0);
OSMO_ASSERT(state->src_fmt == AF_PCMA);
OSMO_ASSERT(state->dst_fmt == AF_GSM);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 18);
OSMO_ASSERT(endp->net_end.codec.payload_type == 8);
len = res;
res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
OSMO_ASSERT(res == -ENOMSG);
/* now check that comfort noise doesn't change anything */
len = audio_packets_pcma[1].len;
memcpy(buf, audio_packets_pcma[1].data, len);
hdr = (struct rtp_hdr *) buf;
hdr->payload_type = 11;
res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
OSMO_ASSERT(state->sample_cnt == 80);
OSMO_ASSERT(state->src_fmt == AF_PCMA);
OSMO_ASSERT(state->dst_fmt == AF_GSM);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 18);
OSMO_ASSERT(endp->net_end.codec.payload_type == 8);
talloc_free(ctx);
}
}
static int test_repacking(int in_samples, int out_samples, int no_transcode)
{
char buf[4096] = {0x80, 0};
@ -463,7 +523,7 @@ static int test_repacking(int in_samples, int out_samples, int no_transcode)
out_size = mgcp_transcoding_get_frame_size(state, -1, 1);
OSMO_ASSERT(sizeof(buf) >= out_size + 12);
buf[1] = endp->net_end.payload_type;
buf[1] = endp->net_end.codec.payload_type;
*(uint16_t*)(buf+2) = htons(1);
*(uint32_t*)(buf+4) = htonl(0);
*(uint32_t*)(buf+8) = htonl(0xaabbccdd);
@ -582,6 +642,7 @@ int main(int argc, char **argv)
test_repacking(160, 100, 1);
test_rtp_seq_state();
test_transcode_result();
test_transcode_change();
return 0;
}

1
openbsc/tests/mgcp/mgcp_transcoding_test.ok
View File

@ -536,3 +536,4 @@ got 1 pcma output frames (80 octets) count=12
generating 160 pcma input samples
got 1 pcma output frames (80 octets) count=12
got 1 pcma output frames (80 octets) count=12
Testing Initial G729->GSM, PCMA->GSM