[asterisk-commits] mogorman: trunk r43243 - in /trunk: channels/
include/asterisk/ main/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Sep 18 16:32:57 MST 2006
Author: mogorman
Date: Mon Sep 18 18:32:57 2006
New Revision: 43243
URL: http://svn.digium.com/view/asterisk?rev=43243&view=rev
Log:
allow for packetization on rtp channel drivers, need to add
option for setting our own packetization as apposed to just doing
what is asked.
Modified:
trunk/channels/chan_sip.c
trunk/channels/chan_skinny.c
trunk/include/asterisk/frame.h
trunk/include/asterisk/rtp.h
trunk/main/frame.c
trunk/main/rtp.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=43243&r1=43242&r2=43243&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Sep 18 18:32:57 2006
@@ -542,6 +542,8 @@
static int regobjs = 0; /*!< Registry objects */
static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
+
+static int global_autoframing = 0;
/*! \brief Protect the SIP dialog list (of sip_pvt's) */
AST_MUTEX_DEFINE_STATIC(iflock);
@@ -966,6 +968,7 @@
struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
struct sip_pvt *next; /*!< Next dialog in chain */
struct sip_invite_param *options; /*!< Options for INVITE */
+ int autoframing;
} *iflist = NULL;
#define FLAG_RESPONSE (1 << 0)
@@ -1015,6 +1018,7 @@
struct ast_ha *ha; /*!< ACL setting */
struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
+ int autoframing;
};
/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
@@ -1077,6 +1081,7 @@
struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
struct sip_pvt *mwipvt; /*!< Subscription for MWI */
int lastmsg;
+ int autoframing;
};
@@ -2540,6 +2545,11 @@
if (option_debug)
ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", natflags ? "On" : "Off");
ast_udptl_setnat(dialog->udptl, natflags);
+ }
+ /* Set Frame packetization */
+ if (dialog->rtp) {
+ ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
+ dialog->autoframing = peer->autoframing;
}
ast_string_field_set(dialog, peername, peer->username);
ast_string_field_set(dialog, authname, peer->username);
@@ -4702,6 +4712,8 @@
*/
/* XXX This needs to be done per media stream, since it's media stream specific */
iterator = req->sdp_start;
+ int found_rtpmap_codecs[32];
+ int last_rtpmap_codec=0;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
if (option_debug > 1) {
@@ -4752,17 +4764,49 @@
} else if (!strcasecmp(a, "sendrecv")) {
sendonly = 0;
continue;
- } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
+ } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
+ char *tmp = strrchr(a, ':');
+ long int framing = 0;
+ if (tmp) {
+ tmp++;
+ framing = strtol(tmp, NULL, 10);
+ if (framing == LONG_MIN || framing == LONG_MAX) {
+ framing = 0;
+ ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
+ }
+ }
+ if (framing && last_rtpmap_codec) {
+ if (p->autoframing || global_autoframing) {
+ struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ int codec_n;
+ int format = 0;
+ for (codec_n = 0; codec_n < last_rtpmap_codec; codec_n++) {
+ format = ast_rtp_codec_getformat(found_rtpmap_codecs[codec_n]);
+ if (!format) /* non-codec or not found */
+ continue;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
+ ast_codec_pref_setsize(pref, format, framing);
+ }
+ ast_rtp_codec_setpref(p->rtp, pref);
+ }
+ }
+ memset(&found_rtpmap_codecs, 0, sizeof(found_rtpmap_codecs));
+ last_rtpmap_codec = 0;
continue;
- /* We have a rtpmap to handle */
- if (debug)
- ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
-
- /* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
+ } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
+ /* We have a rtpmap to handle */
+ if (debug)
+ ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
+ found_rtpmap_codecs[last_rtpmap_codec] = codec;
+ last_rtpmap_codec++;
+
+ /* Note: should really look at the 'freq' and '#chans' params too */
+ ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0);
- if (p->vrtp)
- ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
+ if (p->vrtp)
+ ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
+ }
}
if (udptlportno != -1) {
@@ -5593,12 +5637,19 @@
int debug)
{
int rtp_code;
+ struct ast_format_list fmt;
+
if (debug)
ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
return;
+ if (p->rtp) {
+ struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ fmt = ast_codec_pref_getsize(pref, codec);
+ } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
+ return;
ast_build_string(m_buf, m_size, " %d", rtp_code);
ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype(1, codec,
@@ -5608,9 +5659,12 @@
/* Indicate that we don't support VAD (G.729 annex B) */
ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
} else if (codec == AST_FORMAT_ILBC) {
- /* Add information about us using only 20 ms packetization */
- ast_build_string(a_buf, a_size, "a=fmtp:%d mode=20\r\n", rtp_code);
-
+ /* Add information about us using only 20/30 ms packetization */
+ ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
+ }
+
+ if (codec != AST_FORMAT_ILBC) {
+ ast_build_string(a_buf, a_size, "a=ptime:%d\r\n", fmt.cur_ms);
}
}
@@ -6084,6 +6138,11 @@
}
respprep(&resp, p, msg, req);
if (p->rtp) {
+ if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing from config on incoming call\n");
+ ast_rtp_codec_setpref(p->rtp, &p->prefs);
+ }
try_suggested_sip_codec(p);
add_sdp(&resp, p);
} else
@@ -7930,6 +7989,11 @@
ASTOBJ_UNREF(peer, sip_destroy_peer);
}
if (peer) {
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ }
if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)) {
ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
} else {
@@ -8663,6 +8727,11 @@
}
}
p->prefs = user->prefs;
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &p->prefs);
+ p->autoframing = user->autoframing;
+ }
/* replace callerid if rpid found, and not restricted */
if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
char *tmp;
@@ -8771,6 +8840,11 @@
peer = find_peer(NULL, &p->recv, 1);
if (peer) {
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ }
if (debug)
ast_verbose("Found peer '%s'\n", peer->name);
@@ -9528,6 +9602,7 @@
if (!codec)
break;
ast_cli(fd, "%s", ast_getformatname(codec));
+ ast_cli(fd, ":%d", pref->framing[x]);
if (x < 31 && ast_codec_pref_index(pref, x + 1))
ast_cli(fd, ",");
}
@@ -15167,6 +15242,8 @@
ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 0);
+ } else if (!strcasecmp(v->name, "autoframing")) {
+ user->autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "callingpres")) {
user->callingpres = ast_parse_caller_presentation(v->value);
if (user->callingpres == -1)
@@ -15446,6 +15523,8 @@
ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, 0);
+ } else if (!strcasecmp(v->name, "autoframing")) {
+ peer->autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "rtptimeout")) {
if ((sscanf(v->value, "%d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
@@ -15807,6 +15886,8 @@
ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 0);
+ } else if (!strcasecmp(v->name, "autoframing")) {
+ global_autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "allowexternaldomains")) {
allow_external_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "autodomain")) {
Modified: trunk/channels/chan_skinny.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_skinny.c?rev=43243&r1=43242&r2=43243&view=diff
==============================================================================
--- trunk/channels/chan_skinny.c (original)
+++ trunk/channels/chan_skinny.c Mon Sep 18 18:32:57 2006
@@ -81,7 +81,8 @@
static const char config[] = "skinny.conf";
/* Just about everybody seems to support ulaw, so make it a nice default */
-static int capability = AST_FORMAT_ULAW;
+static int default_capability = AST_FORMAT_ULAW;
+static struct ast_codec_pref default_prefs;
#define DEFAULT_SKINNY_PORT 2000
#define DEFAULT_SKINNY_BACKLOG 2
@@ -948,6 +949,7 @@
int hookstate;
int nat;
+ struct ast_codec_pref prefs;
struct skinny_subchannel *sub;
struct skinny_line *next;
struct skinny_device *parent;
@@ -980,11 +982,13 @@
int registered;
int lastlineinstance;
int lastcallreference;
+ int capability;
struct sockaddr_in addr;
struct in_addr ourip;
struct skinny_line *lines;
struct skinny_speeddial *speeddials;
struct skinny_addon *addons;
+ struct ast_codec_pref prefs;
struct ast_ha *ha;
struct skinnysession *session;
struct skinny_device *next;
@@ -1022,7 +1026,7 @@
static const struct ast_channel_tech skinny_tech = {
.type = "Skinny",
.description = tdesc,
- .capabilities = AST_FORMAT_ULAW,
+ .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = skinny_request,
.call = skinny_call,
@@ -1407,7 +1411,7 @@
req->data.openreceivechannel.conferenceId = htolel(0);
req->data.openreceivechannel.partyId = htolel(sub->callid);
- req->data.openreceivechannel.packets = htolel(20);
+ req->data.openreceivechannel.packets = htolel(l->prefs.framing[0]);
req->data.openreceivechannel.capability = htolel(convert_cap(l->capability));
req->data.openreceivechannel.echo = htolel(0);
req->data.openreceivechannel.bitrate = htolel(0);
@@ -1927,6 +1931,8 @@
} else {
ast_copy_string(d->name, cat, sizeof(d->name));
d->lastlineinstance = 1;
+ d->capability = default_capability;
+ d->prefs = default_prefs;
while(v) {
if (!strcasecmp(v->name, "host")) {
if (ast_get_ip(&d->addr, v->value)) {
@@ -1941,6 +1947,10 @@
d->ha = ast_append_ha(v->name, v->value, d->ha);
} else if (!strcasecmp(v->name, "context")) {
ast_copy_string(context, v->value, sizeof(context));
+ } else if (!strcasecmp(v->name, "allow")) {
+ ast_parse_allow_disallow(&d->prefs, &d->capability, v->value, 1);
+ } else if (!strcasecmp(v->name, "disallow")) {
+ ast_parse_allow_disallow(&d->prefs, &d->capability, v->value, 0);
} else if (!strcasecmp(v->name, "version")) {
ast_copy_string(d->version_id, v->value, sizeof(d->version_id));
} else if (!strcasecmp(v->name, "nat")) {
@@ -2040,7 +2050,8 @@
ast_verbose(VERBOSE_PREFIX_3 "Setting mailbox '%s' on %s@%s\n", mailbox, d->name, l->name);
}
l->msgstate = -1;
- l->capability = capability;
+ l->capability = d->capability;
+ l->prefs = d->prefs;
l->parent = d;
if (!strcasecmp(v->name, "trunk")) {
l->type = TYPE_TRUNK;
@@ -2167,6 +2178,10 @@
if (sub->vrtp) {
ast_rtp_setnat(sub->vrtp, l->nat);
}
+ /* Set Frame packetization */
+ if (sub->rtp)
+ ast_rtp_codec_setpref(sub->rtp, &l->prefs);
+
/* Create the RTP connection */
transmit_connect(d->session, sub);
ast_mutex_unlock(&sub->lock);
@@ -2680,7 +2695,7 @@
tmp->tech_pvt = sub;
tmp->nativeformats = l->capability;
if (!tmp->nativeformats)
- tmp->nativeformats = capability;
+ tmp->nativeformats = default_capability;
fmt = ast_best_codec(tmp->nativeformats);
if (skinnydebug)
ast_verbose("skinny_new: tmp->nativeformats=%d fmt=%d\n", tmp->nativeformats, fmt);
@@ -3566,7 +3581,7 @@
req->data.startmedia.passThruPartyId = htolel(sub->callid);
req->data.startmedia.remoteIp = htolel(d->ourip.s_addr);
req->data.startmedia.remotePort = htolel(ntohs(us.sin_port));
- req->data.startmedia.packetSize = htolel(20);
+ req->data.startmedia.packetSize = htolel(l->prefs.framing[0]);
req->data.startmedia.payloadType = htolel(convert_cap(l->capability));
req->data.startmedia.qualifier.precedence = htolel(127);
req->data.startmedia.qualifier.vad = htolel(0);
@@ -4277,7 +4292,7 @@
char *dest = data;
oldformat = format;
- format &= capability;
+ format &= default_capability;
if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", format);
return NULL;
@@ -4308,7 +4323,6 @@
int on = 1;
struct ast_config *cfg;
struct ast_variable *v;
- int format;
char *cat;
struct skinny_device *d;
int oldport = ntohs(bindaddr.sin_port);
@@ -4350,19 +4364,9 @@
} else if (!strcasecmp(v->name, "dateformat")) {
ast_copy_string(date_format, v->value, sizeof(date_format));
} else if (!strcasecmp(v->name, "allow")) {
- format = ast_getformatbyname(v->value);
- if (format < 1) {
- ast_log(LOG_WARNING, "Cannot allow unknown format '%s'\n", v->value);
- } else {
- capability |= format;
- }
+ ast_parse_allow_disallow(&default_prefs, &default_capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
- format = ast_getformatbyname(v->value);
- if (format < 1) {
- ast_log(LOG_WARNING, "Cannot disallow unknown format '%s'\n", v->value);
- } else {
- capability &= ~format;
- }
+ ast_parse_allow_disallow(&default_prefs, &default_capability, v->value, 0);
} else if (!strcasecmp(v->name, "bindport") || !strcasecmp(v->name, "port")) {
if (sscanf(v->value, "%d", &ourport) == 1) {
bindaddr.sin_port = htons(ourport);
Modified: trunk/include/asterisk/frame.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/frame.h?rev=43243&r1=43242&r2=43243&view=diff
==============================================================================
--- trunk/include/asterisk/frame.h (original)
+++ trunk/include/asterisk/frame.h Mon Sep 18 18:32:57 2006
@@ -38,6 +38,7 @@
struct ast_codec_pref {
char order[32];
+ char framing[32];
};
/*! \page Def_Frame AST Multimedia and signalling frames
@@ -281,6 +282,7 @@
};
#define AST_SMOOTHER_FLAG_G729 (1 << 0)
+#define AST_SMOOTHER_FLAG_BE (1 << 1)
/* Option identifiers and flags */
#define AST_OPTION_FLAG_REQUEST 0
@@ -346,6 +348,23 @@
uint8_t data[0];
};
+
+/*! \brief Definition of supported media formats (codecs) */
+struct ast_format_list {
+ int visible; /*!< Can we see this entry */
+ int bits; /*!< bitmask value */
+ char *name; /*!< short name */
+ char *desc; /*!< Description */
+ int fr_len; /*!< Single frame length in bytes */
+ int min_ms; /*!< Min value */
+ int max_ms; /*!< Max value */
+ int inc_ms; /*!< Increment */
+ int def_ms; /*!< Default value */
+ unsigned int flags; /*!< Smoother flags */
+ int cur_ms; /*!< Current value */
+};
+
+
/*! \brief Requests a frame to be allocated
*
* \param source
@@ -436,6 +455,7 @@
struct ast_smoother *ast_smoother_new(int bytes);
void ast_smoother_set_flags(struct ast_smoother *smoother, int flags);
int ast_smoother_get_flags(struct ast_smoother *smoother);
+int ast_smoother_test_flag(struct ast_smoother *s, int flag);
void ast_smoother_free(struct ast_smoother *s);
void ast_smoother_reset(struct ast_smoother *s, int bytes);
int __ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f, int swap);
@@ -482,6 +502,14 @@
the format list is selected, otherwise 0 is returned. */
int ast_codec_choose(struct ast_codec_pref *pref, int formats, int find_best);
+/*! \brief Set packet size for codec
+*/
+int ast_codec_pref_setsize(struct ast_codec_pref *pref, int format, int framems);
+
+/*! \brief Get packet size for codec
+*/
+struct ast_format_list ast_codec_pref_getsize(struct ast_codec_pref *pref, int format);
+
/*! \brief Parse an "allow" or "deny" line in a channel or device configuration
and update the capabilities mask and pref if provided.
Video codecs are not added to codec preference lists, since we can not transcode
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?rev=43243&r1=43242&r2=43243&view=diff
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Mon Sep 18 18:32:57 2006
@@ -210,6 +210,12 @@
int ast_rtp_reload(void);
+int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
+
+struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
+
+int ast_rtp_codec_getformat(int pt);
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
Modified: trunk/main/frame.c
URL: http://svn.digium.com/view/asterisk/trunk/main/frame.c?rev=43243&r1=43242&r2=43243&view=diff
==============================================================================
--- trunk/main/frame.c (original)
+++ trunk/main/frame.c Mon Sep 18 18:32:57 2006
@@ -101,24 +101,19 @@
};
/*! \brief Definition of supported media formats (codecs) */
-static struct ast_format_list {
- int visible; /*!< Can we see this entry */
- int bits; /*!< bitmask value */
- char *name; /*!< short name */
- char *desc; /*!< Description */
-} AST_FORMAT_LIST[] = { /*!< Bit number: comment - Bit numbers are hard coded in show_codec() */
- { 1, AST_FORMAT_G723_1 , "g723" , "G.723.1"}, /*!< 1 */
- { 1, AST_FORMAT_GSM, "gsm" , "GSM"}, /*!< 2: codec_gsm.c */
- { 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law" }, /*!< 3: codec_ulaw.c */
- { 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law" }, /*!< 4: codec_alaw.c */
- { 1, AST_FORMAT_G726, "g726", "G.726 RFC3551" },/*!< 5: codec_g726.c */
- { 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM"}, /*!< 6: codec_adpcm.c */
- { 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM"}, /*!< 7 */
- { 1, AST_FORMAT_LPC10, "lpc10", "LPC10" }, /*!< 8: codec_lpc10.c */
- { 1, AST_FORMAT_G729A, "g729", "G.729A" }, /*!< 9: Binary commercial distribution */
- { 1, AST_FORMAT_SPEEX, "speex", "SpeeX" }, /*!< 10: codec_speex.c */
- { 1, AST_FORMAT_ILBC, "ilbc", "iLBC"}, /*!< 11: codec_ilbc.c */
- { 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2" }, /*!< 12: codec_g726.c */
+static struct ast_format_list AST_FORMAT_LIST[] = { /*!< Bit number: comment - Bit numbers are hard coded in show_codec() */
+ { 1, AST_FORMAT_G723_1 , "g723" , "G.723.1", 24, 30, 300, 30, 30 }, /*!< 1 */
+ { 1, AST_FORMAT_GSM, "gsm" , "GSM", 33, 20, 60, 20, 20 }, /*!< 2: codec_gsm.c */
+ { 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law", 80, 10, 30, 10, 20 }, /*!< 3: codec_ulaw.c */
+ { 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law", 80, 10, 30, 10, 20 }, /*!< 4: codec_alaw.c */
+ { 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 50, 10, 20 },/*!< 5: codec_g726.c */
+ { 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM", 40, 10, 30, 10, 20 }, /*!< 6: codec_adpcm.c */
+ { 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE }, /*!< 7 */
+ { 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 }, /*!< 8: codec_lpc10.c */
+ { 1, AST_FORMAT_G729A, "g729", "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 }, /*!< 9: Binary commercial distribution */
+ { 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 }, /*!< 10: codec_speex.c */
+ { 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 }, /*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
+ { 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 50, 10, 20 }, /*!< 12: codec_g726.c */
{ 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
{ 0, 0, "nothing", "undefined" },
@@ -162,6 +157,11 @@
void ast_smoother_set_flags(struct ast_smoother *s, int flags)
{
s->flags = flags;
+}
+
+int ast_smoother_test_flag(struct ast_smoother *s, int flag)
+{
+ return (s->flags & flag);
}
int __ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f, int swap)
@@ -1099,6 +1099,7 @@
struct ast_codec_pref oldorder;
int x, y = 0;
int slot;
+ int size;
if(!pref->order[0])
return;
@@ -1108,10 +1109,13 @@
for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
slot = oldorder.order[x];
+ size = oldorder.framing[x];
if(! slot)
break;
- if(AST_FORMAT_LIST[slot-1].bits != format)
- pref->order[y++] = slot;
+ if(AST_FORMAT_LIST[slot-1].bits != format) {
+ pref->order[y] = slot;
+ pref->framing[y++] = size;
+ }
}
}
@@ -1143,6 +1147,84 @@
}
+/*! \brief Set packet size for codec */
+int ast_codec_pref_setsize(struct ast_codec_pref *pref, int format, int framems)
+{
+ int x, index = -1;
+
+ for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+ if(AST_FORMAT_LIST[x].bits == format) {
+ index = x;
+ break;
+ }
+ }
+
+ if(index < 0)
+ return -1;
+
+ /* size validation */
+ if(!framems)
+ framems = AST_FORMAT_LIST[index].def_ms;
+
+ if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
+ framems -= framems % AST_FORMAT_LIST[index].inc_ms;
+
+ if(framems < AST_FORMAT_LIST[index].min_ms)
+ framems = AST_FORMAT_LIST[index].min_ms;
+
+ if(framems > AST_FORMAT_LIST[index].max_ms)
+ framems = AST_FORMAT_LIST[index].max_ms;
+
+
+ for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+ if(pref->order[x] == (index + 1)) {
+ pref->framing[x] = framems;
+ break;
+ }
+ }
+
+ return x;
+}
+
+/*! \brief Get packet size for codec */
+struct ast_format_list ast_codec_pref_getsize(struct ast_codec_pref *pref, int format)
+{
+ int x, index = -1, framems = 0;
+ struct ast_format_list fmt;
+
+ for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+ if(AST_FORMAT_LIST[x].bits == format) {
+ fmt = AST_FORMAT_LIST[x];
+ index = x;
+ break;
+ }
+ }
+
+ for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+ if(pref->order[x] == (index + 1)) {
+ framems = pref->framing[x];
+ break;
+ }
+ }
+
+ /* size validation */
+ if(!framems)
+ framems = AST_FORMAT_LIST[index].def_ms;
+
+ if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
+ framems -= framems % AST_FORMAT_LIST[index].inc_ms;
+
+ if(framems < AST_FORMAT_LIST[index].min_ms)
+ framems = AST_FORMAT_LIST[index].min_ms;
+
+ if(framems > AST_FORMAT_LIST[index].max_ms)
+ framems = AST_FORMAT_LIST[index].max_ms;
+
+ fmt.cur_ms = framems;
+
+ return fmt;
+}
+
/*! \brief Pick a codec */
int ast_codec_choose(struct ast_codec_pref *pref, int formats, int find_best)
{
@@ -1172,9 +1254,22 @@
char *parse;
char *this;
int format;
+ char *psize;
+ int framems;
parse = ast_strdupa(list);
while ((this = strsep(&parse, ","))) {
+ framems = 0;
+ if ((psize = strrchr(this, ':'))) {
+ *psize = '\0';
+ psize++;
+ if (option_debug)
+ ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize);
+ framems = strtol(psize,NULL,10);
+ if (framems == LONG_MIN || framems == LONG_MAX) {
+ framems = 0;
+ }
+ }
if (!(format = ast_getformatbyname(this))) {
ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
continue;
@@ -1192,8 +1287,10 @@
*/
if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
if (strcasecmp(this, "all")) {
- if (allowing)
+ if (allowing) {
ast_codec_pref_append(pref, format);
+ ast_codec_pref_setsize(pref, format, framems);
+ }
else
ast_codec_pref_remove(pref, format);
} else if (!allowing) {
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?rev=43243&r1=43242&r2=43243&view=diff
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Mon Sep 18 18:32:57 2006
@@ -160,6 +160,7 @@
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
struct ast_rtcp *rtcp;
+ struct ast_codec_pref pref;
struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
};
@@ -2480,6 +2481,35 @@
return 0;
}
+int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
+{
+ int x;
+ for (x = 0; x < 32; x++) { /* Ugly way */
+ rtp->pref.order[x] = prefs->order[x];
+ rtp->pref.framing[x] = prefs->framing[x];
+ }
+ if (rtp->smoother)
+ ast_smoother_free(rtp->smoother);
+ rtp->smoother = NULL;
+ return 0;
+}
+
+struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp)
+{
+ return &rtp->pref;
+}
+
+int ast_rtp_codec_getformat(int pt)
+{
+ if (pt < 0 || pt > MAX_RTP_PT)
+ return 0; /* bogus payload type */
+
+ if (static_RTP_PT[pt].isAstFormat)
+ return static_RTP_PT[pt].code;
+ else
+ return 0;
+}
+
int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
{
struct ast_frame *f;
@@ -2522,99 +2552,29 @@
rtp->smoother = NULL;
}
-
- switch(subclass) {
- case AST_FORMAT_SLINEAR:
- if (!rtp->smoother) {
- rtp->smoother = ast_smoother_new(320);
- }
- if (!rtp->smoother) {
- ast_log(LOG_WARNING, "Unable to create smoother :(\n");
- return -1;
- }
- ast_smoother_feed_be(rtp->smoother, _f);
-
+ if (!rtp->smoother) {
+ struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
+ if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
+ if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
+ ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+ return -1;
+ }
+ if (fmt.flags)
+ ast_smoother_set_flags(rtp->smoother, fmt.flags);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+ }
+ }
+ if (rtp->smoother) {
+ if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
+ ast_smoother_feed_be(rtp->smoother, _f);
+ } else {
+ ast_smoother_feed(rtp->smoother, _f);
+ }
+
while((f = ast_smoother_read(rtp->smoother)))
ast_rtp_raw_write(rtp, f, codec);
- break;
- 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_ADPCM:
- case AST_FORMAT_G726:
- case AST_FORMAT_G726_AAL2:
- if (!rtp->smoother) {
- rtp->smoother = ast_smoother_new(80);
- }
- 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_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729);
- }
- 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;
- 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;
- case AST_FORMAT_ILBC:
- if (!rtp->smoother) {
- rtp->smoother = ast_smoother_new(50);
- }
- if (!rtp->smoother) {
- ast_log(LOG_WARNING, "Unable to create ILBC 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 %s packets\n", ast_getformatname(subclass));
- /* fall through to... */
- case AST_FORMAT_H261:
- case AST_FORMAT_H263:
- case AST_FORMAT_H263_PLUS:
- case AST_FORMAT_H264:
- case AST_FORMAT_G723_1:
- case AST_FORMAT_LPC10:
- case AST_FORMAT_SPEEX:
+ } else {
/* Don't buffer outgoing frames; send them one-per-packet: */
if (_f->offset < hdrlen) {
f = ast_frdup(_f);
More information about the asterisk-commits
mailing list