Version 0.2.0 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@503 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
markster
parent
d08e5f9e35
commit
dd6aeaf199
188
rtp.c
188
rtp.c
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@ -26,6 +26,7 @@
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#include <fcntl.h>
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#include <asterisk/rtp.h>
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#include <asterisk/frame.h>
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#include <asterisk/logger.h>
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#include <asterisk/options.h>
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#include <asterisk/channel.h>
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@ -45,6 +46,7 @@ struct ast_rtp {
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struct sockaddr_in them;
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struct timeval rxcore;
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struct timeval txcore;
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struct ast_smoother *smoother;
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int *ioid;
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unsigned short seqno;
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struct sched_context *sched;
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@ -104,6 +106,40 @@ static void process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len)
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rtp->dtmfcount = dtmftimeout;
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}
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static void process_type121(struct ast_rtp *rtp, unsigned char *data, int len)
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{
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char resp = 0;
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unsigned char b0,b1,b2,b3,b4,b5,b6,b7;
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b0=*(data+0);b1=*(data+1);b2=*(data+2);b3=*(data+3);
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b4=*(data+4);b5=*(data+5);b6=*(data+6);b7=*(data+7);
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// printf("%u %u %u %u %u %u %u %u\n",b0,b1,b2,b3,b4,b5,b6,b7);
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if (b2==32) {
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// printf("Start %d\n",b3);
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if (b4==0) {
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// printf("Detection point for DTMF %d\n",b3);
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if (b3<10) {
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resp='0'+b3;
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} else if (b3<11) {
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resp='*';
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} else if (b3<12) {
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resp='#';
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} else if (b3<16) {
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resp='A'+(b3-12);
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}
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rtp->resp=resp;
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send_dtmf(rtp);
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}
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}
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if (b2==3) {
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// printf("Stop(3) %d\n",b3);
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}
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if (b2==0) {
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// printf("Stop(0) %d\n",b3);
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}
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}
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static int rtpread(int *id, int fd, short events, void *cbdata)
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{
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struct ast_rtp *rtp = cbdata;
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@ -114,7 +150,6 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
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int payloadtype;
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int hdrlen = 12;
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unsigned int timestamp;
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unsigned int *rtpheader;
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len = sizeof(sin);
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@ -147,8 +182,12 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
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if (payloadtype == 101) {
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/* It's special -- rfc2833 process it */
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process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
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} else
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} else if (payloadtype == 121) {
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/* CISCO proprietary DTMF bridge */
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process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
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} else {
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ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
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}
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return 1;
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}
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@ -192,7 +231,10 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
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rtp->f.timelen = 20 * (rtp->f.datalen / 33);
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break;
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case AST_FORMAT_ADPCM:
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rtp->f.timelen = rtp->f.datalen / 8;
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rtp->f.timelen = rtp->f.datalen / 4;
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break;
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case AST_FORMAT_G729A:
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rtp->f.timelen = rtp->f.datalen;
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break;
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default:
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ast_log(LOG_NOTICE, "Unable to calculate timelen for format %d\n", rtp->f.subclass);
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@ -207,13 +249,14 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
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static struct {
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int rtp;
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int ast;
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char *label;
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} cmap[] = {
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{ 0, AST_FORMAT_ULAW },
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{ 3, AST_FORMAT_GSM },
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{ 4, AST_FORMAT_G723_1 },
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{ 5, AST_FORMAT_ADPCM },
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{ 8, AST_FORMAT_ALAW },
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{ 18, AST_FORMAT_G729A },
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{ 0, AST_FORMAT_ULAW, "PCMU" },
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{ 3, AST_FORMAT_GSM, "GSM" },
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{ 4, AST_FORMAT_G723_1, "G723" },
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{ 5, AST_FORMAT_ADPCM, "ADPCM" },
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{ 8, AST_FORMAT_ALAW, "PCMA" },
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{ 18, AST_FORMAT_G729A, "G729" },
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};
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int rtp2ast(int id)
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@ -236,6 +279,15 @@ int ast2rtp(int id)
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return -1;
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}
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char *ast2rtpn(int id)
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{
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int x;
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for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
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if (cmap[x].ast == id)
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return cmap[x].label;
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}
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return "";
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}
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struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
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{
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struct ast_rtp *rtp;
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@ -291,6 +343,8 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
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void ast_rtp_destroy(struct ast_rtp *rtp)
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{
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if (rtp->smoother)
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ast_smoother_free(rtp->smoother);
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if (rtp->ioid)
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ast_io_remove(rtp->io, rtp->ioid);
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if (rtp->s > -1)
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@ -307,46 +361,46 @@ static unsigned int calc_txstamp(struct ast_rtp *rtp)
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}
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gettimeofday(&now, NULL);
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ms = (now.tv_sec - rtp->txcore.tv_sec) * 1000;
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ms += (now.tv_usec - rtp->txcore.tv_usec);
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ms += (now.tv_usec - rtp->txcore.tv_usec) / 1000;
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return ms;
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}
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int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
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{
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int hdrlen = 12;
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struct ast_frame *f;
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int codec;
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int res;
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unsigned int ms;
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unsigned int *rtpheader;
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/* Make sure we have enough space for RTP header */
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if (_f->frametype != AST_FRAME_VOICE) {
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ast_log(LOG_WARNING, "RTP can only send voice\n");
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return -1;
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}
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int hdrlen = 12;
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int res;
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int ms;
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int pred;
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codec = ast2rtp(_f->subclass);
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if (codec < 0) {
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ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
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return -1;
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}
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if (_f->offset < hdrlen) {
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f = ast_frdup(_f);
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} else
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f = _f;
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ms = calc_txstamp(rtp) * 8;
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ms = calc_txstamp(rtp);
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/* Default prediction */
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pred = ms * 8;
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switch(f->subclass) {
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case AST_FORMAT_ULAW:
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case AST_FORMAT_ALAW:
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/* If we're within +/- 20ms from when where we
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predict we should be, use that */
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pred = rtp->lastts + f->datalen;
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break;
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case AST_FORMAT_G729A:
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pred = rtp->lastts + f->datalen * 8;
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break;
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default:
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ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %d\n", f->subclass);
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}
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rtp->lastts += f->datalen;
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/* Re-calculate last TS */
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rtp->lastts = ms * 8;
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/* If it's close to ou prediction, go for it */
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if (abs(rtp->lastts - pred) < 640)
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rtp->lastts = pred;
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#if 0
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else
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printf("Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
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#endif
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/* Get a pointer to the header */
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rtpheader = (unsigned int *)(f->data - hdrlen);
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rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++));
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@ -362,3 +416,65 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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}
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return 0;
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}
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int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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{
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struct ast_frame *f;
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int codec;
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int hdrlen = 12;
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/* Make sure we have enough space for RTP header */
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if (_f->frametype != AST_FRAME_VOICE) {
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ast_log(LOG_WARNING, "RTP can only send voice\n");
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return -1;
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}
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codec = ast2rtp(_f->subclass);
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if (codec < 0) {
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ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
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return -1;
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}
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switch(_f->subclass) {
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case AST_FORMAT_ULAW:
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case AST_FORMAT_ALAW:
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if (!rtp->smoother) {
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rtp->smoother = ast_smoother_new(160);
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}
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if (!rtp->smoother) {
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ast_log(LOG_WARNING, "Unable to create smoother :(\n");
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return -1;
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}
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ast_smoother_feed(rtp->smoother, _f);
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while((f = ast_smoother_read(rtp->smoother)))
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ast_rtp_raw_write(rtp, f, codec);
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break;
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case AST_FORMAT_G729A:
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if (!rtp->smoother) {
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rtp->smoother = ast_smoother_new(20);
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}
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if (!rtp->smoother) {
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ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n");
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return -1;
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}
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ast_smoother_feed(rtp->smoother, _f);
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while((f = ast_smoother_read(rtp->smoother)))
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ast_rtp_raw_write(rtp, f, codec);
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break;
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default:
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ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", _f->subclass);
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if (_f->offset < hdrlen) {
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f = ast_frdup(_f);
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} else {
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f = _f;
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}
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ast_rtp_raw_write(rtp, f, codec);
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}
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return 0;
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}
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