Version 0.3.0 from FTP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
parent
c78714436e
commit
e403f86b4b
23
frame.c
23
frame.c
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@ -11,6 +11,7 @@
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* the GNU General Public License
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*/
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#include <asterisk/lock.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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@ -34,23 +35,27 @@ struct ast_smoother {
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int size;
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int format;
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int readdata;
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float timeperbyte;
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float samplesperbyte;
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struct ast_frame f;
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char data[SMOOTHER_SIZE];
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char framedata[SMOOTHER_SIZE + AST_FRIENDLY_OFFSET];
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int len;
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};
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void ast_smoother_reset(struct ast_smoother *s, int size)
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{
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memset(s, 0, sizeof(struct ast_smoother));
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s->size = size;
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}
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struct ast_smoother *ast_smoother_new(int size)
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{
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struct ast_smoother *s;
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if (size < 1)
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return NULL;
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s = malloc(sizeof(struct ast_smoother));
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if (s) {
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memset(s, 0, sizeof(struct ast_smoother));
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s->size = size;
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}
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if (s)
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ast_smoother_reset(s, size);
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return s;
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}
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@ -62,7 +67,7 @@ int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f)
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}
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if (!s->format) {
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s->format = f->subclass;
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s->timeperbyte = (float)f->timelen / (float)f->datalen;
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s->samplesperbyte = (float)f->samples / (float)f->datalen;
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} else if (s->format != f->subclass) {
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ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
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return -1;
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@ -88,7 +93,7 @@ struct ast_frame *ast_smoother_read(struct ast_smoother *s)
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s->f.data = s->framedata + AST_FRIENDLY_OFFSET;
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s->f.offset = AST_FRIENDLY_OFFSET;
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s->f.datalen = s->size;
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s->f.timelen = s->size * s->timeperbyte;
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s->f.samples = s->size * s->samplesperbyte;
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/* Fill Data */
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memcpy(s->f.data, s->data, s->size);
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s->len -= s->size;
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@ -169,7 +174,7 @@ struct ast_frame *ast_frisolate(struct ast_frame *fr)
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out->frametype = fr->frametype;
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out->subclass = fr->subclass;
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out->datalen = 0;
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out->timelen = fr->timelen;
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out->samples = fr->samples;
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out->offset = 0;
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out->src = NULL;
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out->data = NULL;
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@ -299,6 +304,8 @@ int ast_getformatbyname(char *name)
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return AST_FORMAT_LPC10;
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else if (!strcasecmp(name, "adpcm"))
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return AST_FORMAT_ADPCM;
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else if (!strcasecmp(name, "g729"))
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return AST_FORMAT_G729A;
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else if (!strcasecmp(name, "speex"))
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return AST_FORMAT_SPEEX;
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else if (!strcasecmp(name, "all"))
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@ -33,8 +33,8 @@ struct ast_frame {
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int subclass;
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/*! Length of data */
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int datalen;
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/*! Amount of time associated with this frame */
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int timelen;
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/*! Number of 8khz samples in this frame */
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int samples;
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/*! Was the data malloc'd? i.e. should we free it when we discard the frame? */
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int mallocd;
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/*! How far into "data" the data really starts */
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@ -165,6 +165,10 @@ struct ast_frame_chain {
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#define AST_CONTROL_WINK 10
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/*! Set a low-level option */
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#define AST_CONTROL_OPTION 11
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/*! Key Radio */
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#define AST_CONTROL_RADIO_KEY 12
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/*! Un-Key Radio */
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#define AST_CONTROL_RADIO_UNKEY 13
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/* Option identifiers and flags */
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#define AST_OPTION_FLAG_REQUEST 0
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@ -181,6 +185,9 @@ struct ast_frame_chain {
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/* Put a compatible channel into TDD (TTY for the hearing-impared) mode */
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#define AST_OPTION_TDD 2
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/* Relax the parameters for DTMF reception (mainly for radio use) */
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#define AST_OPTION_RELAXDTMF 3
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struct ast_option_header {
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/* Always keep in network byte order */
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#if __BYTE_ORDER == __BIG_ENDIAN
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@ -280,6 +287,7 @@ struct ast_smoother;
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extern struct ast_smoother *ast_smoother_new(int bytes);
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extern void ast_smoother_free(struct ast_smoother *s);
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extern void ast_smoother_reset(struct ast_smoother *s, int bytes);
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extern int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f);
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extern struct ast_frame *ast_smoother_read(struct ast_smoother *s);
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170
rtp.c
170
rtp.c
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@ -31,6 +31,11 @@
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#include <asterisk/options.h>
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#include <asterisk/channel.h>
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#define TYPE_SILENCE 0x2
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#define TYPE_HIGH 0x0
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#define TYPE_LOW 0x1
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#define TYPE_MASK 0x3
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static int dtmftimeout = 300; /* 300 samples */
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struct ast_rtp {
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@ -41,6 +46,7 @@ struct ast_rtp {
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unsigned int ssrc;
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unsigned int lastts;
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unsigned int lastrxts;
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int lasttxformat;
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int dtmfcount;
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struct sockaddr_in us;
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struct sockaddr_in them;
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@ -56,6 +62,40 @@ struct ast_rtp {
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};
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static int g723_len(unsigned char buf)
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{
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switch(buf & TYPE_MASK) {
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case TYPE_MASK:
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case TYPE_SILENCE:
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return 4;
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break;
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case TYPE_HIGH:
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return 24;
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break;
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case TYPE_LOW:
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return 20;
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break;
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default:
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ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
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}
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return -1;
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}
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static int g723_samples(unsigned char *buf, int maxlen)
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{
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int pos = 0;
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int samples = 0;
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int res;
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while(pos < maxlen) {
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res = g723_len(buf[pos]);
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if (res < 0)
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break;
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samples += 240;
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pos += res;
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}
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return samples;
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}
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void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
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{
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rtp->data = data;
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@ -72,7 +112,7 @@ static void send_dtmf(struct ast_rtp *rtp)
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rtp->f.frametype = AST_FRAME_DTMF;
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rtp->f.subclass = rtp->resp;
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rtp->f.datalen = 0;
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rtp->f.timelen = 0;
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rtp->f.samples = 0;
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rtp->f.mallocd = 0;
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rtp->f.src = "RTP";
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rtp->resp = 0;
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@ -185,6 +225,9 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
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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 if (payloadtype == 100) {
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/* CISCO's notso proprietary DTMF bridge */
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process_rfc2833(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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@ -222,22 +265,25 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
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switch(rtp->f.subclass) {
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case AST_FORMAT_ULAW:
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case AST_FORMAT_ALAW:
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rtp->f.timelen = rtp->f.datalen / 8;
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rtp->f.samples = rtp->f.datalen;
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break;
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case AST_FORMAT_SLINEAR:
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rtp->f.timelen = rtp->f.datalen / 16;
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rtp->f.samples = rtp->f.datalen / 2;
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break;
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case AST_FORMAT_GSM:
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rtp->f.timelen = 20 * (rtp->f.datalen / 33);
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rtp->f.samples = 160 * (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 / 4;
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rtp->f.samples = rtp->f.datalen * 2;
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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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rtp->f.samples = rtp->f.datalen * 8;
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break;
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case AST_FORMAT_G723_1:
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rtp->f.samples = g723_samples(rtp->f.data, 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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ast_log(LOG_NOTICE, "Unable to calculate samples for format %d\n", rtp->f.subclass);
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break;
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}
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rtp->f.src = "RTP";
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@ -330,12 +376,27 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
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return rtp;
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}
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int ast_rtp_settos(struct ast_rtp *rtp, int tos)
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{
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int res;
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if ((res = setsockopt(rtp->s, SOL_IP, IP_TOS, &tos, sizeof(tos))))
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ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
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return res;
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}
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void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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{
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rtp->them.sin_port = them->sin_port;
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rtp->them.sin_addr = them->sin_addr;
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}
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void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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{
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them->sin_family = AF_INET;
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them->sin_port = rtp->them.sin_port;
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them->sin_addr = rtp->them.sin_addr;
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}
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void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
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{
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memcpy(us, &rtp->us, sizeof(rtp->us));
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@ -365,6 +426,67 @@ static unsigned int calc_txstamp(struct ast_rtp *rtp)
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return ms;
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}
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int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
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{
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unsigned int *rtpheader;
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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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int x;
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char data[256];
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if ((digit <= '9') && (digit >= '0'))
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digit -= '0';
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else if (digit == '*')
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digit = 10;
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else if (digit == '#')
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digit = 11;
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else if ((digit >= 'A') && (digit <= 'D'))
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digit = digit - 'A' + 12;
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else if ((digit >= 'a') && (digit <= 'd'))
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digit = digit - 'a' + 12;
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else {
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ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
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return -1;
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}
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/* If we have no peer, return immediately */
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if (!rtp->them.sin_addr.s_addr)
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return 0;
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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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/* Get a pointer to the header */
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rtpheader = (unsigned int *)data;
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rtpheader[0] = htonl((2 << 30) | (1 << 23) | (101 << 16) | (rtp->seqno++));
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rtpheader[1] = htonl(rtp->lastts);
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rtpheader[2] = htonl(rtp->ssrc);
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rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
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for (x=0;x<4;x++) {
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if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
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res = sendto(rtp->s, (void *)rtpheader, hdrlen + 4, 0, &rtp->them, sizeof(rtp->them));
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if (res <0)
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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));
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#if 0
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printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
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#endif
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}
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if (x ==0) {
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/* Clear marker bit and increment seqno */
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rtpheader[0] = htonl((2 << 30) | (101 << 16) | (rtp->seqno++));
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/* Make duration 240 */
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rtpheader[3] |= htonl((240));
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/* Set the End bit for the last 3 */
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rtpheader[3] |= htonl((1 << 23));
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}
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}
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return 0;
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}
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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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unsigned int *rtpheader;
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@ -387,6 +509,12 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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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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case AST_FORMAT_GSM:
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pred = rtp->lastts + f->datalen * 20 / 33;
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break;
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case AST_FORMAT_G723_1:
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pred = rtp->lastts + g723_samples(f->data, f->datalen);
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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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@ -423,8 +551,12 @@ int ast_rtp_write(struct ast_rtp *rtp, 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 we have no peer, return immediately */
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if (!rtp->them.sin_addr.s_addr)
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return 0;
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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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@ -436,6 +568,15 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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return -1;
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}
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if (rtp->lasttxformat != _f->subclass) {
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/* New format, reset the smoother */
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ast_log(LOG_DEBUG, "Ooh, format changed from %d to %d\n", rtp->lasttxformat, _f->subclass);
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rtp->lasttxformat = _f->subclass;
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if (rtp->smoother)
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ast_smoother_free(rtp->smoother);
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rtp->smoother = NULL;
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}
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switch(_f->subclass) {
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case AST_FORMAT_ULAW:
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@ -465,7 +606,18 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_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_GSM:
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if (!rtp->smoother) {
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rtp->smoother = ast_smoother_new(33);
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}
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if (!rtp->smoother) {
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ast_log(LOG_WARNING, "Unable to create GSM 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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