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asterisk
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Version 0.3.0 from FTP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Mark Spencer 23 years ago
parent c78714436e
commit e403f86b4b
3 changed files with 186 additions and 19 deletions
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23
frame.c
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@ -11,6 +11,7 @@
* the GNU General Public License
*/
#include <asterisk/lock.h>
#include <asterisk/frame.h>
#include <asterisk/logger.h>
#include <asterisk/options.h>
@ -34,23 +35,27 @@ struct ast_smoother {
int size;
int format;
int readdata;
float timeperbyte;
float samplesperbyte;
struct ast_frame f;
char data[SMOOTHER_SIZE];
char framedata[SMOOTHER_SIZE + AST_FRIENDLY_OFFSET];
int len;
};
void ast_smoother_reset(struct ast_smoother *s, int size)
{
memset(s, 0, sizeof(struct ast_smoother));
s->size = size;
}
struct ast_smoother *ast_smoother_new(int size)
{
struct ast_smoother *s;
if (size < 1)
return NULL;
s = malloc(sizeof(struct ast_smoother));
if (s) {
memset(s, 0, sizeof(struct ast_smoother));
s->size = size;
}
if (s)
ast_smoother_reset(s, size);
return s;
}
@ -62,7 +67,7 @@ int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f)
}
if (!s->format) {
s->format = f->subclass;
s->timeperbyte = (float)f->timelen / (float)f->datalen;
s->samplesperbyte = (float)f->samples / (float)f->datalen;
} else if (s->format != f->subclass) {
ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
return -1;
@ -88,7 +93,7 @@ struct ast_frame *ast_smoother_read(struct ast_smoother *s)
s->f.data = s->framedata + AST_FRIENDLY_OFFSET;
s->f.offset = AST_FRIENDLY_OFFSET;
s->f.datalen = s->size;
s->f.timelen = s->size * s->timeperbyte;
s->f.samples = s->size * s->samplesperbyte;
/* Fill Data */
memcpy(s->f.data, s->data, s->size);
s->len -= s->size;
@ -169,7 +174,7 @@ struct ast_frame *ast_frisolate(struct ast_frame *fr)
out->frametype = fr->frametype;
out->subclass = fr->subclass;
out->datalen = 0;
out->timelen = fr->timelen;
out->samples = fr->samples;
out->offset = 0;
out->src = NULL;
out->data = NULL;
@ -299,6 +304,8 @@ int ast_getformatbyname(char *name)
return AST_FORMAT_LPC10;
else if (!strcasecmp(name, "adpcm"))
return AST_FORMAT_ADPCM;
else if (!strcasecmp(name, "g729"))
return AST_FORMAT_G729A;
else if (!strcasecmp(name, "speex"))
return AST_FORMAT_SPEEX;
else if (!strcasecmp(name, "all"))

12
include/asterisk/frame.h
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@ -33,8 +33,8 @@ struct ast_frame {
int subclass;
/*! Length of data */
int datalen;
/*! Amount of time associated with this frame */
int timelen;
/*! Number of 8khz samples in this frame */
int samples;
/*! Was the data malloc'd? i.e. should we free it when we discard the frame? */
int mallocd;
/*! How far into "data" the data really starts */
@ -165,6 +165,10 @@ struct ast_frame_chain {
#define AST_CONTROL_WINK 10
/*! Set a low-level option */
#define AST_CONTROL_OPTION 11
/*! Key Radio */
#define AST_CONTROL_RADIO_KEY 12
/*! Un-Key Radio */
#define AST_CONTROL_RADIO_UNKEY 13
/* Option identifiers and flags */
#define AST_OPTION_FLAG_REQUEST 0
@ -181,6 +185,9 @@ struct ast_frame_chain {
/* Put a compatible channel into TDD (TTY for the hearing-impared) mode */
#define AST_OPTION_TDD 2
/* Relax the parameters for DTMF reception (mainly for radio use) */
#define AST_OPTION_RELAXDTMF 3
struct ast_option_header {
/* Always keep in network byte order */
#if __BYTE_ORDER == __BIG_ENDIAN
@ -280,6 +287,7 @@ struct ast_smoother;
extern struct ast_smoother *ast_smoother_new(int bytes);
extern void ast_smoother_free(struct ast_smoother *s);
extern void ast_smoother_reset(struct ast_smoother *s, int bytes);
extern int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f);
extern struct ast_frame *ast_smoother_read(struct ast_smoother *s);

170
rtp.c
Unescape View File

@ -31,6 +31,11 @@
#include <asterisk/options.h>
#include <asterisk/channel.h>
#define TYPE_SILENCE 0x2
#define TYPE_HIGH 0x0
#define TYPE_LOW 0x1
#define TYPE_MASK 0x3
static int dtmftimeout = 300; /* 300 samples */
struct ast_rtp {
@ -41,6 +46,7 @@ struct ast_rtp {
unsigned int ssrc;
unsigned int lastts;
unsigned int lastrxts;
int lasttxformat;
int dtmfcount;
struct sockaddr_in us;
struct sockaddr_in them;
@ -56,6 +62,40 @@ struct ast_rtp {
};
static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_MASK:
case TYPE_SILENCE:
return 4;
break;
case TYPE_HIGH:
return 24;
break;
case TYPE_LOW:
return 20;
break;
default:
ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
}
return -1;
}
static int g723_samples(unsigned char *buf, int maxlen)
{
int pos = 0;
int samples = 0;
int res;
while(pos < maxlen) {
res = g723_len(buf[pos]);
if (res < 0)
break;
samples += 240;
pos += res;
}
return samples;
}
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
{
rtp->data = data;
@ -72,7 +112,7 @@ static void send_dtmf(struct ast_rtp *rtp)
rtp->f.frametype = AST_FRAME_DTMF;
rtp->f.subclass = rtp->resp;
rtp->f.datalen = 0;
rtp->f.timelen = 0;
rtp->f.samples = 0;
rtp->f.mallocd = 0;
rtp->f.src = "RTP";
rtp->resp = 0;
@ -185,6 +225,9 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
} else if (payloadtype == 121) {
/* CISCO proprietary DTMF bridge */
process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
} else if (payloadtype == 100) {
/* CISCO's notso proprietary DTMF bridge */
process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
}
@ -222,22 +265,25 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
switch(rtp->f.subclass) {
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
rtp->f.timelen = rtp->f.datalen / 8;
rtp->f.samples = rtp->f.datalen;
break;
case AST_FORMAT_SLINEAR:
rtp->f.timelen = rtp->f.datalen / 16;
rtp->f.samples = rtp->f.datalen / 2;
break;
case AST_FORMAT_GSM:
rtp->f.timelen = 20 * (rtp->f.datalen / 33);
rtp->f.samples = 160 * (rtp->f.datalen / 33);
break;
case AST_FORMAT_ADPCM:
rtp->f.timelen = rtp->f.datalen / 4;
rtp->f.samples = rtp->f.datalen * 2;
break;
case AST_FORMAT_G729A:
rtp->f.timelen = rtp->f.datalen;
rtp->f.samples = rtp->f.datalen * 8;
break;
case AST_FORMAT_G723_1:
rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
break;
default:
ast_log(LOG_NOTICE, "Unable to calculate timelen for format %d\n", rtp->f.subclass);
ast_log(LOG_NOTICE, "Unable to calculate samples for format %d\n", rtp->f.subclass);
break;
}
rtp->f.src = "RTP";
@ -330,12 +376,27 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
return rtp;
}
int ast_rtp_settos(struct ast_rtp *rtp, int tos)
{
int res;
if ((res = setsockopt(rtp->s, SOL_IP, IP_TOS, &tos, sizeof(tos))))
ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
return res;
}
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
rtp->them.sin_port = them->sin_port;
rtp->them.sin_addr = them->sin_addr;
}
void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
them->sin_family = AF_INET;
them->sin_port = rtp->them.sin_port;
them->sin_addr = rtp->them.sin_addr;
}
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
{
memcpy(us, &rtp->us, sizeof(rtp->us));
@ -365,6 +426,67 @@ static unsigned int calc_txstamp(struct ast_rtp *rtp)
return ms;
}
int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
{
unsigned int *rtpheader;
int hdrlen = 12;
int res;
int ms;
int pred;
int x;
char data[256];
if ((digit <= '9') && (digit >= '0'))
digit -= '0';
else if (digit == '*')
digit = 10;
else if (digit == '#')
digit = 11;
else if ((digit >= 'A') && (digit <= 'D'))
digit = digit - 'A' + 12;
else if ((digit >= 'a') && (digit <= 'd'))
digit = digit - 'a' + 12;
else {
ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
return -1;
}
/* If we have no peer, return immediately */
if (!rtp->them.sin_addr.s_addr)
return 0;
ms = calc_txstamp(rtp);
/* Default prediction */
pred = ms * 8;
/* Get a pointer to the header */
rtpheader = (unsigned int *)data;
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (101 << 16) | (rtp->seqno++));
rtpheader[1] = htonl(rtp->lastts);
rtpheader[2] = htonl(rtp->ssrc);
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
for (x=0;x<4;x++) {
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
res = sendto(rtp->s, (void *)rtpheader, hdrlen + 4, 0, &rtp->them, sizeof(rtp->them));
if (res <0)
ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
#if 0
printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
#endif
}
if (x ==0) {
/* Clear marker bit and increment seqno */
rtpheader[0] = htonl((2 << 30) | (101 << 16) | (rtp->seqno++));
/* Make duration 240 */
rtpheader[3] |= htonl((240));
/* Set the End bit for the last 3 */
rtpheader[3] |= htonl((1 << 23));
}
}
return 0;
}
static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
{
unsigned int *rtpheader;
@ -387,6 +509,12 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
case AST_FORMAT_G729A:
pred = rtp->lastts + f->datalen * 8;
break;
case AST_FORMAT_GSM:
pred = rtp->lastts + f->datalen * 20 / 33;
break;
case AST_FORMAT_G723_1:
pred = rtp->lastts + g723_samples(f->data, f->datalen);
break;
default:
ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %d\n", f->subclass);
}
@ -423,8 +551,12 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
int codec;
int hdrlen = 12;
/* Make sure we have enough space for RTP header */
/* If we have no peer, return immediately */
if (!rtp->them.sin_addr.s_addr)
return 0;
/* Make sure we have enough space for RTP header */
if (_f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "RTP can only send voice\n");
return -1;
@ -436,6 +568,15 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
return -1;
}
if (rtp->lasttxformat != _f->subclass) {
/* New format, reset the smoother */
ast_log(LOG_DEBUG, "Ooh, format changed from %d to %d\n", rtp->lasttxformat, _f->subclass);
rtp->lasttxformat = _f->subclass;
if (rtp->smoother)
ast_smoother_free(rtp->smoother);
rtp->smoother = NULL;
}
switch(_f->subclass) {
case AST_FORMAT_ULAW:
@ -465,7 +606,18 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
while((f = ast_smoother_read(rtp->smoother)))
ast_rtp_raw_write(rtp, f, codec);
break;
case AST_FORMAT_GSM:
if (!rtp->smoother) {
rtp->smoother = ast_smoother_new(33);
}
if (!rtp->smoother) {
ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n");
return -1;
}
ast_smoother_feed(rtp->smoother, _f);
while((f = ast_smoother_read(rtp->smoother)))
ast_rtp_raw_write(rtp, f, codec);
break;
default:
ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", _f->subclass);
if (_f->offset < hdrlen) {

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