Version 0.2.0 from FTP
authorMark Spencer <markster@digium.com>
Fri, 9 Aug 2002 17:17:54 +0000 (17:17 +0000)
committerMark Spencer <markster@digium.com>
Fri, 9 Aug 2002 17:17:54 +0000 (17:17 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

rtp.c

diff --git a/rtp.c b/rtp.c
index 3a87ad4..5162193 100755 (executable)
--- a/rtp.c
+++ b/rtp.c
@@ -26,6 +26,7 @@
 #include <fcntl.h>
 
 #include <asterisk/rtp.h>
+#include <asterisk/frame.h>
 #include <asterisk/logger.h>
 #include <asterisk/options.h>
 #include <asterisk/channel.h>
@@ -45,6 +46,7 @@ struct ast_rtp {
        struct sockaddr_in them;
        struct timeval rxcore;
        struct timeval txcore;
+       struct ast_smoother *smoother;
        int *ioid;
        unsigned short seqno;
        struct sched_context *sched;
@@ -104,6 +106,40 @@ static void process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len)
        rtp->dtmfcount = dtmftimeout;
 }
 
+static void process_type121(struct ast_rtp *rtp, unsigned char *data, int len)
+{
+       char resp = 0;
+       
+       unsigned char b0,b1,b2,b3,b4,b5,b6,b7;
+       
+       b0=*(data+0);b1=*(data+1);b2=*(data+2);b3=*(data+3);
+       b4=*(data+4);b5=*(data+5);b6=*(data+6);b7=*(data+7);
+//     printf("%u %u %u %u %u %u %u %u\n",b0,b1,b2,b3,b4,b5,b6,b7);
+       if (b2==32) {
+//             printf("Start %d\n",b3);
+               if (b4==0) {
+//                     printf("Detection point for DTMF %d\n",b3);
+                       if (b3<10) {
+                               resp='0'+b3;
+                       } else if (b3<11) {
+                               resp='*';
+                       } else if (b3<12) {
+                               resp='#';
+                       } else if (b3<16) {
+                               resp='A'+(b3-12);
+                       }
+                       rtp->resp=resp;
+                       send_dtmf(rtp);
+               }
+       }
+       if (b2==3) {
+//             printf("Stop(3) %d\n",b3);
+       }
+       if (b2==0) {
+//             printf("Stop(0) %d\n",b3);
+       }
+}
+
 static int rtpread(int *id, int fd, short events, void *cbdata)
 {
        struct ast_rtp *rtp = cbdata;
@@ -114,7 +150,6 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
        int payloadtype;
        int hdrlen = 12;
        unsigned int timestamp;
-       
        unsigned int *rtpheader;
        
        len = sizeof(sin);
@@ -147,8 +182,12 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
                if (payloadtype == 101) {
                        /* It's special -- rfc2833 process it */
                        process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
-               } else
+               } else if (payloadtype == 121) {
+                       /* CISCO proprietary DTMF bridge */
+                       process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
+               } else {
                        ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
+               }
                return 1;
        }
 
@@ -192,7 +231,10 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
                rtp->f.timelen = 20 * (rtp->f.datalen / 33);
                break;
        case AST_FORMAT_ADPCM:
-               rtp->f.timelen = rtp->f.datalen / 8;
+               rtp->f.timelen = rtp->f.datalen / 4;
+               break;
+       case AST_FORMAT_G729A:
+               rtp->f.timelen = rtp->f.datalen;
                break;
        default:
                ast_log(LOG_NOTICE, "Unable to calculate timelen for format %d\n", rtp->f.subclass);
@@ -207,13 +249,14 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
 static struct {
        int rtp;
        int ast;
+       char *label;
 } cmap[] = {
-       { 0, AST_FORMAT_ULAW },
-       { 3, AST_FORMAT_GSM },
-       { 4, AST_FORMAT_G723_1 },
-       { 5, AST_FORMAT_ADPCM },
-       { 8, AST_FORMAT_ALAW },
-       { 18, AST_FORMAT_G729A },
+       { 0, AST_FORMAT_ULAW, "PCMU" },
+       { 3, AST_FORMAT_GSM, "GSM" },
+       { 4, AST_FORMAT_G723_1, "G723" },
+       { 5, AST_FORMAT_ADPCM, "ADPCM" },
+       { 8, AST_FORMAT_ALAW, "PCMA" },
+       { 18, AST_FORMAT_G729A, "G729" },
 };
 
 int rtp2ast(int id)
@@ -236,6 +279,15 @@ int ast2rtp(int id)
        return -1;
 }
 
+char *ast2rtpn(int id)
+{
+       int x;
+       for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
+               if (cmap[x].ast == id)
+                       return cmap[x].label;
+       }
+       return "";
+}
 struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
 {
        struct ast_rtp *rtp;
@@ -291,6 +343,8 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
 
 void ast_rtp_destroy(struct ast_rtp *rtp)
 {
+       if (rtp->smoother)
+               ast_smoother_free(rtp->smoother);
        if (rtp->ioid)
                ast_io_remove(rtp->io, rtp->ioid);
        if (rtp->s > -1)
@@ -307,46 +361,46 @@ static unsigned int calc_txstamp(struct ast_rtp *rtp)
        }
        gettimeofday(&now, NULL);
        ms = (now.tv_sec - rtp->txcore.tv_sec) * 1000;
-       ms += (now.tv_usec - rtp->txcore.tv_usec);
+       ms += (now.tv_usec - rtp->txcore.tv_usec) / 1000;
        return ms;
 }
 
-int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
+static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
 {
+       unsigned int *rtpheader;
        int hdrlen = 12;
-       struct ast_frame *f;
-       int codec;
        int res;
-       unsigned int ms;
-       unsigned int *rtpheader;
-       
-       /* 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;
-       }
-
-       codec = ast2rtp(_f->subclass);
-       if (codec < 0) {
-               ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
-               return -1;
-       }
+       int ms;
+       int pred;
 
-       if (_f->offset < hdrlen) {
-               f = ast_frdup(_f);
-       } else
-               f = _f;
-               
-       ms = calc_txstamp(rtp) * 8;
+       ms = calc_txstamp(rtp);
+       /* Default prediction */
+       pred = ms * 8;
+       
        switch(f->subclass) {
        case AST_FORMAT_ULAW:
        case AST_FORMAT_ALAW:
+               /* If we're within +/- 20ms from when where we
+                  predict we should be, use that */
+               pred = rtp->lastts + f->datalen;
+               break;
+       case AST_FORMAT_G729A:
+               pred = rtp->lastts + f->datalen * 8;
                break;
        default:
                ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %d\n", f->subclass);
        }
-       rtp->lastts += f->datalen;
+
+       /* Re-calculate last TS */
+       rtp->lastts = ms * 8;
+       
+       /* If it's close to ou prediction, go for it */
+       if (abs(rtp->lastts - pred) < 640)
+               rtp->lastts = pred;
+#if 0
+       else
+               printf("Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
+#endif                 
        /* Get a pointer to the header */
        rtpheader = (unsigned int *)(f->data - hdrlen);
        rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++));
@@ -362,3 +416,65 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
        }
        return 0;
 }
+
+int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
+{
+       struct ast_frame *f;
+       int codec;
+       int hdrlen = 12;
+       
+       /* 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;
+       }
+
+       codec = ast2rtp(_f->subclass);
+       if (codec < 0) {
+               ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
+               return -1;
+       }
+
+
+       switch(_f->subclass) {
+       case AST_FORMAT_ULAW:
+       case AST_FORMAT_ALAW:
+               if (!rtp->smoother) {
+                       rtp->smoother = ast_smoother_new(160);
+               }
+               if (!rtp->smoother) {
+                       ast_log(LOG_WARNING, "Unable to create 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;
+       case AST_FORMAT_G729A:
+               if (!rtp->smoother) {
+                       rtp->smoother = ast_smoother_new(20);
+               }
+               if (!rtp->smoother) {
+                       ast_log(LOG_WARNING, "Unable to create g729 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) {
+                       f = ast_frdup(_f);
+               } else {
+                       f = _f;
+               }
+               ast_rtp_raw_write(rtp, f, codec);
+       }
+               
+       return 0;
+}