[asterisk-commits] oej: branch oej/sip-t140-red r116176 - in /team/oej/sip-t140-red: channels/ i...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed May 14 02:48:55 CDT 2008
Author: oej
Date: Wed May 14 02:48:54 2008
New Revision: 116176
URL: http://svn.digium.com/view/asterisk?view=rev&rev=116176
Log:
Update of patch
Modified:
team/oej/sip-t140-red/channels/chan_sip.c
team/oej/sip-t140-red/include/asterisk/frame.h
team/oej/sip-t140-red/include/asterisk/rtp.h
team/oej/sip-t140-red/main/frame.c
team/oej/sip-t140-red/main/rtp.c
Modified: team/oej/sip-t140-red/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/sip-t140-red/channels/chan_sip.c?view=diff&rev=116176&r1=116175&r2=116176
==============================================================================
--- team/oej/sip-t140-red/channels/chan_sip.c (original)
+++ team/oej/sip-t140-red/channels/chan_sip.c Wed May 14 02:48:54 2008
@@ -1298,6 +1298,9 @@
(A bit unsure of this, please correct if
you know more) */
struct sip_st_dlg *stimer; /*!< SIP Session-Timers */
+
+ char red_fmtp[100]; /*!< T.140 Redundancy response fmtp */
+ int red;
};
@@ -4929,16 +4932,20 @@
case AST_FRAME_TEXT:
if (p) {
sip_pvt_lock(p);
- if (p->trtp) {
- /* Activate text early media */
- if ((ast->_state != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ if (p->red) {
+ red_buffer_t140(p->trtp, frame);
+ } else {
+ if (p->trtp) {
+ /* Activate text early media */
+ if ((ast->_state != AST_STATE_UP) &&
+ !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
+ !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ }
+ p->lastrtptx = time(NULL);
+ res = ast_rtp_write(p->trtp, frame);
}
- p->lastrtptx = time(NULL);
- res = ast_rtp_write(p->trtp, frame);
}
sip_pvt_unlock(p);
}
@@ -6321,6 +6328,12 @@
char buf[SIPBUFSIZE];
int rua_version;
+
+ char *cp;
+ char red_fmtp[100] = "empty";
+ int red_data_pt[10];
+ int red_num_gen = 0;
+ int red_pt = 0;
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
@@ -6663,6 +6676,23 @@
memset(&found_rtpmap_codecs, 0, sizeof(found_rtpmap_codecs));
last_rtpmap_codec = 0;
continue;
+
+ } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
+ /* count numbers of generations in fmtp */
+ cp = &red_fmtp[strlen(red_fmtp)];
+ strncpy(red_fmtp, a, 100);
+
+ sscanf(cp, "%u", &red_data_pt[red_num_gen]);
+ cp = strtok(cp, "/");
+ while (cp && red_num_gen++ < RED_MAX_GENERATION) {
+ sscanf(cp, "%u", &red_data_pt[red_num_gen]);
+ cp = strtok(NULL, "/");
+ }
+ strncpy(p->red_fmtp, "a=", 2);
+ strncpy(&p->red_fmtp[2], a, 100);
+ cp = red_fmtp;
+ strncpy((cp = memchr(p->red_fmtp, '\0', 100)), "\r\n\0", 4);
+
} else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
/* We have a rtpmap to handle */
@@ -6679,10 +6709,18 @@
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
- } else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
+ } else if (!strncasecmp(mimeSubtype, "T140",4)) { /* Text */
+ ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
+ if (debug) {
+ ast_verbose("Adding t140 mimeSubtype to textrtp struct\n");
+ }
+ } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
if (p->trtp) {
- /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
+ red_pt = codec;
+ sprintf(red_fmtp, "fmtp:%d ", red_pt);
+ if (debug)
+ ast_verbose("Red submimetype has payload type: %d\n", red_pt);
}
} else { /* Must be audio?? */
if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
@@ -6856,6 +6894,13 @@
p->peercapability = newpeercapability; /* The other sides capability in latest offer */
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
+ if (p->jointcapability & AST_FORMAT_RED) {
+ p->red = 1;
+ rtp_red_init(p->trtp, 300, red_data_pt, 2);
+ } else {
+ p->red = 0;
+ }
+
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
ast_rtp_pt_copy(p->vrtp, newvideortp);
@@ -7768,6 +7813,22 @@
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
/* Add fmtp code here */
+
+ if (codec == AST_FORMAT_RED) {
+
+ ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
+ ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
+ ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
+ ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140));
+
+ /* if (p->red_fmtp[0] != '\0') {
+ ast_str_append(a_buf, 0, p->red_fmtp);
+ } else {
+ int c = ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140);
+ ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d\r\n", rtp_code, c,c);
+ }
+ */
+ }
}
Modified: team/oej/sip-t140-red/include/asterisk/frame.h
URL: http://svn.digium.com/view/asterisk/team/oej/sip-t140-red/include/asterisk/frame.h?view=diff&rev=116176&r1=116175&r2=116176
==============================================================================
--- team/oej/sip-t140-red/include/asterisk/frame.h (original)
+++ team/oej/sip-t140-red/include/asterisk/frame.h Wed May 14 02:48:54 2008
@@ -270,8 +270,12 @@
/*! MPEG4 Video */
#define AST_FORMAT_MP4_VIDEO (1 << 22)
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK))
-/*! T.140 Text format - ITU T.140, RFC 4351*/
-#define AST_FORMAT_T140 (1 << 25)
+/*! T.140 Text format - ITU T.140, RFC 4103 */
+#define AST_FORMAT_T140 (1 << 26)
+/*! T.140 RED Text format RFC 4103 */
+#define AST_FORMAT_RED (1 << 25)
+/*! Maximum text mask */
+#define AST_FORMAT_MAX_TEXT (1 << 27)
#define AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
enum ast_control_frame_type {
Modified: team/oej/sip-t140-red/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/team/oej/sip-t140-red/include/asterisk/rtp.h?view=diff&rev=116176&r1=116175&r2=116176
==============================================================================
--- team/oej/sip-t140-red/include/asterisk/rtp.h (original)
+++ team/oej/sip-t140-red/include/asterisk/rtp.h Wed May 14 02:48:54 2008
@@ -51,6 +51,9 @@
/*! Maxmum number of payload defintions for a RTP session */
#define MAX_RTP_PT 256
+/*! T.140 Redundancy Maxium number of generations */
+#define RED_MAX_GENERATION 5
+
#define FLAG_3389_WARNING (1 << 0)
enum ast_rtp_options {
@@ -67,6 +70,8 @@
};
struct ast_rtp;
+/*! T.140 Redundancy structure*/
+struct rtp_red;
/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem
*/
@@ -288,6 +293,22 @@
/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
+/*! \brief Initalize t.140 redudancy
+ * \param ti time between each t140red frame is sent
+ * \param red_pt payloadtype for RTP packet
+ * \param pt payloadtype numbers for each generation including primary data
+ * \param num_gen number of redundant generations, primary data excluded
+ */
+int rtp_red_init(struct ast_rtp *rtp, int ti, int *pt, int num_gen);
+
+void red_init(struct rtp_red *red, const struct ast_frame *f);
+
+
+/*! \brief Buffer t.140 data */
+void red_buffer_t140(struct ast_rtp *rtp, struct ast_frame *f);
+
+
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
Modified: team/oej/sip-t140-red/main/frame.c
URL: http://svn.digium.com/view/asterisk/team/oej/sip-t140-red/main/frame.c?view=diff&rev=116176&r1=116175&r2=116176
==============================================================================
--- team/oej/sip-t140-red/main/frame.c (original)
+++ team/oej/sip-t140-red/main/frame.c Wed May 14 02:48:54 2008
@@ -119,6 +119,7 @@
{ AST_FORMAT_H263_PLUS, "h263p", 0, "H.263+ Video" }, /*!< H.263plus passthrough support See format_h263.c */
{ AST_FORMAT_H264, "h264", 0, "H.264 Video" }, /*!< Passthrough support, see format_h263.c */
{ AST_FORMAT_MP4_VIDEO, "mpeg4", 0, "MPEG4 Video" }, /*!< Passthrough support for MPEG4 */
+ { AST_FORMAT_RED, "red", 1, "T.140 Realtime Text with redundancy"}, /*!< Redundant T.140 Realtime Text */
{ AST_FORMAT_T140, "t140", 0, "Passthrough T.140 Realtime Text" }, /*!< Passthrough support for T.140 Realtime Text */
};
@@ -585,12 +586,10 @@
all = strcasecmp(name, "all") ? 0 : 1;
for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
- if (all ||
- !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
- !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name))) {
- format |= AST_FORMAT_LIST[x].bits;
- if (!all)
- break;
+ if (all || !strcasecmp(AST_FORMAT_LIST[x].name,name) || !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name))) {
+ format |= AST_FORMAT_LIST[x].bits;
+ if (!all)
+ break;
}
}
Modified: team/oej/sip-t140-red/main/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/sip-t140-red/main/rtp.c?view=diff&rev=116176&r1=116175&r2=116176
==============================================================================
--- team/oej/sip-t140-red/main/rtp.c (original)
+++ team/oej/sip-t140-red/main/rtp.c Wed May 14 02:48:54 2008
@@ -177,6 +177,25 @@
struct sockaddr_in strict_rtp_address; /*!< Remote address information for strict RTP purposes */
int set_marker_bit:1; /*!< Whether to set the marker bit or not */
+ struct rtp_red *red;
+};
+
+static struct ast_frame *red_t140_to_red(struct rtp_red *red);
+static int red_write(const void *data);
+
+struct rtp_red {
+ struct ast_frame t140; /*!< Primary data */
+ struct ast_frame t140red; /*!< Redundant t140*/
+ unsigned char pt[RED_MAX_GENERATION]; /*!< Payload types for redundancy data */
+ unsigned char ts[RED_MAX_GENERATION]; /*!< Time stamps */
+ unsigned char len[RED_MAX_GENERATION]; /*!< length of each generation */
+ int num_gen; /*!< Number of generations */
+ int schedid; /*!< Timer id */
+ int ti; /*!< How long to buffer data before send */
+ unsigned char t140red_data[64000];
+ unsigned char buf_data[64000]; /*!< buffered primary data */
+ int hdrlen;
+ long int prev_ts;
};
/* Forward declarations */
@@ -1392,6 +1411,7 @@
unsigned int *rtpheader;
struct rtpPayloadType rtpPT;
struct ast_rtp *bridged = NULL;
+ int prev_seqno;
/* If time is up, kill it */
if (rtp->sending_digit)
@@ -1541,6 +1561,8 @@
}
if ( (int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
rtp->cycles += RTP_SEQ_MOD;
+
+ prev_seqno = rtp->lastrxseqno;
rtp->lastrxseqno = seqno;
@@ -1604,6 +1626,61 @@
rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.seqno = seqno;
+
+ if (rtp->f.subclass == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
+ unsigned char *data = rtp->f.data;
+
+ memmove(rtp->f.data+3, rtp->f.data, rtp->f.datalen);
+ rtp->f.datalen +=3;
+ *data++ = 0xEF;
+ *data++ = 0xBF;
+ *data = 0xBD;
+ }
+
+ if (rtp->f.subclass == AST_FORMAT_RED) {
+ unsigned char *data = rtp->f.data;
+ unsigned char *header_end;
+ int num_generations;
+ int header_length;
+ int len;
+ int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
+ int x;
+
+ rtp->f.subclass = AST_FORMAT_T140;
+ header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
+ header_end++;
+
+ header_length = header_end - data;
+ num_generations = header_length / 4;
+ len = header_length;
+
+ if (!diff) {
+ for (x = 0; x < num_generations; x++)
+ len += data[x * 4 + 3];
+
+ if (!(rtp->f.datalen - len))
+ return &ast_null_frame;
+
+ rtp->f.data += len;
+ rtp->f.datalen -= len;
+ } else if (diff > num_generations && diff < 10) {
+ len -= 3;
+ rtp->f.data += len;
+ rtp->f.datalen -= len;
+
+ data = rtp->f.data;
+ *data++ = 0xEF;
+ *data++ = 0xBF;
+ *data = 0xBD;
+ } else {
+ for ( x = 0; x < num_generations - diff; x++)
+ len += data[x * 4 + 3];
+
+ rtp->f.data += len;
+ rtp->f.datalen -= len;
+ }
+ }
+
if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) {
rtp->f.samples = ast_codec_get_samples(&rtp->f);
if (rtp->f.subclass == AST_FORMAT_SLINEAR)
@@ -1674,6 +1751,7 @@
{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"},
{{1, AST_FORMAT_H264}, "video", "H264"},
{{1, AST_FORMAT_MP4_VIDEO}, "video", "MP4V-ES"},
+ {{1, AST_FORMAT_RED}, "text", "RED"},
{{1, AST_FORMAT_T140}, "text", "T140"},
};
@@ -1713,9 +1791,10 @@
[98] = {1, AST_FORMAT_H263_PLUS},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
- [102] = {1, AST_FORMAT_T140}, /* Real time text chat */
[103] = {1, AST_FORMAT_H263_PLUS},
[104] = {1, AST_FORMAT_MP4_VIDEO},
+ [105] = {1, AST_FORMAT_RED}, /* Real time text chat */
+ [106] = {1, AST_FORMAT_T140}, /* Real time text chat */
[110] = {1, AST_FORMAT_SPEEX},
[111] = {1, AST_FORMAT_G726},
[112] = {1, AST_FORMAT_G726_AAL2},
@@ -2380,6 +2459,11 @@
void ast_rtp_stop(struct ast_rtp *rtp)
{
AST_SCHED_DEL(rtp->sched, rtp->rtcp->schedid);
+ if (rtp->red) {
+ ast_sched_del(rtp->sched, rtp->red->schedid);
+ free(rtp->red);
+ rtp->red = NULL;
+ }
memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
@@ -3139,13 +3223,20 @@
return 0;
/* If there is no data length, return immediately */
- if (!_f->datalen)
+ if(!_f->datalen && !rtp->red)
return 0;
/* Make sure we have enough space for RTP header */
if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO) && (_f->frametype != AST_FRAME_TEXT)) {
ast_log(LOG_WARNING, "RTP can only send voice, video and text\n");
return -1;
+ }
+
+ if (rtp->red) {
+ /* return 0; */
+ /* no primary data or generations to send */
+ if ((_f = red_t140_to_red(rtp->red)) == NULL)
+ return 0;
}
/* The bottom bit of a video subclass contains the marker bit */
@@ -4265,3 +4356,112 @@
__ast_rtp_reload(0);
}
+/*! \brief Write t140 redundacy frame
+ * \param data primary data to be buffered
+ */
+static int red_write(const void *data)
+{
+ struct ast_rtp *rtp = (struct ast_rtp*) data;
+
+ ast_rtp_write(rtp, &rtp->red->t140);
+
+ return 1;
+}
+
+/*! \brief Construct a redundant frame
+ * \param red redundant data structure
+ */
+static struct ast_frame *red_t140_to_red(struct rtp_red *red) {
+ unsigned char *data = red->t140red.data;
+ int len = 0;
+ int i;
+
+ /* replace most aged generation */
+ if (red->len[0]) {
+ for (i = 1; i < red->num_gen+1; i++)
+ len += red->len[i];
+
+ memmove(&data[red->hdrlen], &data[red->hdrlen+red->len[0]], len);
+ }
+
+ /* Store length of each generation and primary data length*/
+ for (i = 0; i < red->num_gen; i++)
+ red->len[i] = red->len[i+1];
+ red->len[i] = red->t140.datalen;
+
+ /* write each generation length in red header */
+ len = red->hdrlen;
+ for (i = 0; i < red->num_gen; i++)
+ len += data[i*4+3] = red->len[i];
+
+ /* add primary data to buffer */
+ memcpy(&data[len], red->t140.data, red->t140.datalen);
+ red->t140red.datalen = len + red->t140.datalen;
+
+ /* no primary data and no generations to send */
+ if (len == red->hdrlen && !red->t140.datalen)
+ return NULL;
+
+ /* reset t.140 buffer */
+ red->t140.datalen = 0;
+
+ return &red->t140red;
+}
+
+/*! \brief Initialize t140 redundancy
+ * \param rtp
+ * \param ti buffer t140 for ti (msecs) before sending redundant frame
+ * \param red_data_pt Payloadtypes for primary- and generation-data
+ * \param num_gen numbers of generations (primary generation not encounted)
+ *
+*/
+int rtp_red_init(struct ast_rtp *rtp, int ti, int *red_data_pt, int num_gen)
+{
+ struct rtp_red *r;
+ int x;
+
+ if (!(r = ast_malloc(sizeof(struct rtp_red))))
+ return -1;
+
+ memset(r, 0, sizeof(struct rtp_red));
+ r->t140.frametype = AST_FRAME_TEXT;
+ r->t140.subclass = AST_FORMAT_RED;
+ r->t140.data = &r->buf_data;
+
+ r->t140.ts = 0;
+ r->t140red = r->t140;
+ r->t140red.data = &r->t140red_data;
+ r->t140red.datalen = 0;
+ r->ti = ti;
+ r->num_gen = num_gen;
+ r->hdrlen = num_gen * 4 + 1;
+ r->prev_ts = 0;
+
+ for (x = 0; x < num_gen; x++) {
+ r->pt[x] = red_data_pt[x];
+ r->pt[x] |= 1 << 7; /* mark redundant generations pt */
+ r->t140red_data[x*4] = r->pt[x];
+ }
+ r->t140red_data[x*4] = r->pt[x] = red_data_pt[x]; /* primary pt */
+ r->schedid = ast_sched_add(rtp->sched, ti, red_write, rtp);
+ rtp->red = r;
+
+ r->t140.datalen = 0;
+
+ return 0;
+}
+
+/*! \brief Buffer t140 from chan_sip
+ * \param rtp
+ * \param f frame
+ */
+void red_buffer_t140(struct ast_rtp *rtp, struct ast_frame *f)
+{
+ if( f->datalen > -1 ) {
+ struct rtp_red *red = rtp->red;
+ memcpy(&red->buf_data[red->t140.datalen], f->data, f->datalen);
+ red->t140.datalen += f->datalen;
+ red->t140.ts = f->ts;
+ }
+}
+
More information about the asterisk-commits
mailing list