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move 'struct ast_rtp' back to rtp.c where it belongs

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-06-08 23:10:45 +00:00
parent 9cdd66dd01
commit 09778b268e
3 changed files with 90 additions and 78 deletions
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41
channels/chan_sip.c
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@@ -4112,7 +4112,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int sendonly = 0;
int numberofports;
struct ast_channel *bridgepeer = NULL;
struct ast_rtp newaudiortp, newvideortp; /* Buffers for codec handling */
struct ast_rtp *newaudiortp, *newvideortp; /* Buffers for codec handling */
int newjointcapability; /* Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
@@ -4125,10 +4125,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
memset(&newaudiortp, 0, sizeof(newaudiortp));
memset(&newvideortp, 0, sizeof(newvideortp));
ast_rtp_pt_default(&newaudiortp);
ast_rtp_pt_default(&newvideortp);
newaudiortp = alloca(ast_rtp_alloc_size());
memset(newaudiortp, 0, ast_rtp_alloc_size());
ast_rtp_pt_default(newaudiortp);
newvideortp = alloca(ast_rtp_alloc_size());
memset(newvideortp, 0, ast_rtp_alloc_size());
ast_rtp_pt_default(newvideortp);
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -4168,7 +4171,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
numberofmediastreams++;
if (p->vrtp)
ast_rtp_pt_clear(&newvideortp); /* Must be cleared in case no m=video line exists */
ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
numberofports = 1;
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
@@ -4176,7 +4179,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Found audio stream in this media definition */
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
ast_rtp_pt_clear(&newaudiortp);
ast_rtp_pt_clear(newaudiortp);
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -4184,7 +4187,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
if (debug)
ast_verbose("Found RTP audio format %d\n", codec);
ast_rtp_set_m_type(&newaudiortp, codec);
ast_rtp_set_m_type(newaudiortp, codec);
}
} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
@@ -4199,7 +4202,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
if (debug)
ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_set_m_type(&newvideortp, codec);
ast_rtp_set_m_type(newvideortp, codec);
}
} else
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
@@ -4305,14 +4308,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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);
ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype);
if (p->vrtp)
ast_rtp_set_rtpmap_type(&newvideortp, codec, "video", mimeSubtype);
ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype);
}
/* Now gather all of the codecs that we are asked for: */
ast_rtp_get_current_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
ast_rtp_get_current_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
newjointcapability = p->capability & (peercapability | vpeercapability);
newpeercapability = (peercapability | vpeercapability);
@@ -4346,15 +4349,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
p->peercapability = newpeercapability; /* The other sides capability in latest offer */
p->noncodeccapability = newnoncodeccapability; /* DTMF capabilities */
{
int i;
/* Copy payload types from source to destination */
for (i=0; i < MAX_RTP_PT; ++i) {
p->rtp->current_RTP_PT[i]= newaudiortp.current_RTP_PT[i];
if (p->vrtp)
p->vrtp->current_RTP_PT[i]= newvideortp.current_RTP_PT[i];
}
}
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
ast_rtp_pt_copy(p->vrtp, newvideortp);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);