[Asterisk-cvs] asterisk rtp.c,1.75,1.76
markster at lists.digium.com
markster at lists.digium.com
Tue Jun 29 16:25:06 CDT 2004
Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv19460
Modified Files:
rtp.c
Log Message:
Formatting fixes (bug #1951)
Index: rtp.c
===================================================================
RCS file: /usr/cvsroot/asterisk/rtp.c,v
retrieving revision 1.75
retrieving revision 1.76
diff -u -d -r1.75 -r1.76
--- rtp.c 29 Jun 2004 14:22:27 -0000 1.75
+++ rtp.c 29 Jun 2004 20:10:57 -0000 1.76
@@ -2,10 +2,12 @@
* Asterisk -- A telephony toolkit for Linux.
*
* Real-time Protocol Support
+ * Supports RTP and RTCP with Symmetric RTP support for NAT
+ * traversal
*
* Copyright (C) 1999, Mark Spencer
*
- * Mark Spencer <markster at linux-support.net>
+ * Mark Spencer <markster at digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
@@ -49,9 +51,9 @@
static int rtpstart = 0;
static int rtpend = 0;
-// The value of each payload format mapping:
+/* The value of each payload format mapping: */
struct rtpPayloadType {
- int isAstFormat; // whether the following code is an AST_FORMAT
+ int isAstFormat; /* whether the following code is an AST_FORMAT */
int code;
};
@@ -89,8 +91,7 @@
void *data;
ast_rtp_callback callback;
struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
- // a cache for the result of rtp_lookup_code():
- int rtp_lookup_code_cache_isAstFormat;
+ int rtp_lookup_code_cache_isAstFormat; /* a cache for the result of rtp_lookup_code(): */
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
struct ast_rtcp *rtcp;
@@ -457,7 +458,7 @@
#endif
rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
if (!rtpPT.isAstFormat) {
- // This is special in-band data that's not one of our codecs
+ /* This is special in-band data that's not one of our codecs */
if (rtpPT.code == AST_RTP_DTMF) {
/* It's special -- rfc2833 process it */
if (rtp->lasteventseqn <= seqno) {
@@ -541,7 +542,7 @@
break;
case AST_FORMAT_SPEEX:
rtp->f.samples = 160;
- // assumes that the RTP packet contained one Speex frame
+ /* assumes that the RTP packet contained one Speex frame */
break;
default:
ast_log(LOG_NOTICE, "Unable to calculate samples for format %s\n", ast_getformatname(rtp->f.subclass));
@@ -564,8 +565,8 @@
return &rtp->f;
}
-// The following array defines the MIME type (and subtype) for each
-// of our codecs, or RTP-specific data type.
+/* The following array defines the MIME Media type (and subtype) for each
+ of our codecs, or RTP-specific data type. */
static struct {
struct rtpPayloadType payloadType;
char* type;
@@ -596,18 +597,18 @@
table for transmission */
static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
[0] = {1, AST_FORMAT_ULAW},
- [2] = {1, AST_FORMAT_G726}, // Technically this is G.721, but if Cisco can do it, so can we...
+ [2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
[3] = {1, AST_FORMAT_GSM},
[4] = {1, AST_FORMAT_G723_1},
- [5] = {1, AST_FORMAT_ADPCM}, // 8 kHz
- [6] = {1, AST_FORMAT_ADPCM}, // 16 kHz
+ [5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
+ [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
[7] = {1, AST_FORMAT_LPC10},
[8] = {1, AST_FORMAT_ALAW},
- [10] = {1, AST_FORMAT_SLINEAR}, // 2 channels
- [11] = {1, AST_FORMAT_SLINEAR}, // 1 channel
+ [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
+ [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
[13] = {0, AST_RTP_CN},
- [16] = {1, AST_FORMAT_ADPCM}, // 11.025 kHz
- [17] = {1, AST_FORMAT_ADPCM}, // 22.050 kHz
+ [16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
+ [17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
[18] = {1, AST_FORMAT_G729A},
[26] = {1, AST_FORMAT_JPEG},
[31] = {1, AST_FORMAT_H261},
@@ -615,7 +616,7 @@
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},
- [121] = {0, AST_RTP_CISCO_DTMF}, // Must be type 121
+ [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
};
void ast_rtp_pt_clear(struct ast_rtp* rtp)
@@ -646,24 +647,24 @@
rtp->rtp_lookup_code_cache_result = 0;
}
-// Make a note of a RTP payload type that was seen in a SDP "m=" line.
-// By default, use the well-known value for this type (although it may
-// still be set to a different value by a subsequent "a=rtpmap:" line):
+/* Make a note of a RTP payload type that was seen in a SDP "m=" line. */
+/* By default, use the well-known value for this type (although it may */
+/* still be set to a different value by a subsequent "a=rtpmap:" line): */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
- if (pt < 0 || pt > MAX_RTP_PT) return; // bogus payload type
+ if (pt < 0 || pt > MAX_RTP_PT) return; /* bogus payload type */
if (static_RTP_PT[pt].code != 0) {
rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
}
}
-// Make a note of a RTP payload type (with MIME type) that was seen in
-// a SDP "a=rtpmap:" line.
+/* Make a note of a RTP payload type (with MIME type) that was seen in */
+/* a SDP "a=rtpmap:" line. */
void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
char* mimeType, char* mimeSubtype) {
int i;
- if (pt < 0 || pt > MAX_RTP_PT) return; // bogus payload type
+ if (pt < 0 || pt > MAX_RTP_PT) return; /* bogus payload type */
for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
@@ -674,8 +675,8 @@
}
}
-// Return the union of all of the codecs that were set by rtp_set...() calls
-// They're returned as two distinct sets: AST_FORMATs, and AST_RTPs
+/* Return the union of all of the codecs that were set by rtp_set...() calls */
+/* They're returned as two distinct sets: AST_FORMATs, and AST_RTPs */
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
int* astFormats, int* nonAstFormats) {
int pt;
@@ -693,9 +694,10 @@
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
{
struct rtpPayloadType result;
+
if (pt < 0 || pt > MAX_RTP_PT) {
result.isAstFormat = result.code = 0;
- return result; // bogus payload type
+ return result; /* bogus payload type */
}
/* Start with the negotiated codecs */
result = rtp->current_RTP_PT[pt];
@@ -705,14 +707,14 @@
return result;
}
+/* Looks up an RTP code out of our *static* outbound list */
int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code) {
int pt;
- /* Looks up an RTP code out of our *static* outbound list */
if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
code == rtp->rtp_lookup_code_cache_code) {
- // Use our cached mapping, to avoid the overhead of the loop below
+ /* Use our cached mapping, to avoid the overhead of the loop below */
return rtp->rtp_lookup_code_cache_result;
}
@@ -1046,7 +1048,7 @@
break;
case AST_FORMAT_SPEEX:
pred = rtp->lastts + 160;
- // assumes that the RTP packet contains one Speex frame
+ /* assumes that the RTP packet contains one Speex frame */
break;
default:
ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %s\n", ast_getformatname(f->subclass));
@@ -1207,12 +1209,12 @@
break;
default:
ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass));
- // fall through to...
+ /* fall through to... */
case AST_FORMAT_H261:
case AST_FORMAT_H263:
case AST_FORMAT_G723_1:
case AST_FORMAT_SPEEX:
- // Don't buffer outgoing frames; send them one-per-packet:
+ /* Don't buffer outgoing frames; send them one-per-packet: */
if (_f->offset < hdrlen) {
f = ast_frdup(_f);
} else {
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