Formatting changes
Found a large amount of missing {} in the code before patching in another branch git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -6078,41 +6078,49 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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}
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}
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}
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if (!ast_strlen_zero(peer->fromuser))
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if (!ast_strlen_zero(peer->fromuser)) {
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ast_string_field_set(dialog, fromuser, peer->fromuser);
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if (!ast_strlen_zero(peer->language))
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}
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if (!ast_strlen_zero(peer->language)) {
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ast_string_field_set(dialog, language, peer->language);
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}
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/* Set timer T1 to RTT for this peer (if known by qualify=) */
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/* Minimum is settable or default to 100 ms */
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/* If there is a maxms and lastms from a qualify use that over a manual T1
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value. Otherwise, use the peer's T1 value. */
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if (peer->maxms && peer->lastms)
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if (peer->maxms && peer->lastms) {
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dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
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else
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} else {
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dialog->timer_t1 = peer->timer_t1;
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}
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/* Set timer B to control transaction timeouts, the peer setting is the default and overrides
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the known timer */
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if (peer->timer_b)
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if (peer->timer_b) {
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dialog->timer_b = peer->timer_b;
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else
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} else {
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dialog->timer_b = 64 * dialog->timer_t1;
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}
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if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
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(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
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(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
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dialog->noncodeccapability |= AST_RTP_DTMF;
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else
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} else {
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dialog->noncodeccapability &= ~AST_RTP_DTMF;
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}
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dialog->directmediaacl = ast_duplicate_acl_list(peer->directmediaacl);
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if (peer->call_limit)
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if (peer->call_limit) {
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ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
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if (!dialog->portinuri)
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}
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if (!dialog->portinuri) {
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dialog->portinuri = peer->portinuri;
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}
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dialog->chanvars = copy_vars(peer->chanvars);
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if (peer->fromdomainport)
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if (peer->fromdomainport) {
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dialog->fromdomainport = peer->fromdomainport;
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}
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return 0;
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}
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@ -2995,8 +2995,9 @@ static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, u
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f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
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f->samples = rtp->dtmfsamples * (rtp->lastrxformat.id ? (rtp_get_rate(&rtp->lastrxformat) / 1000) : 8);
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rtp->resp = 0;
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} else if (rtp->resp == resp)
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} else if (rtp->resp == resp) {
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rtp->dtmfsamples += 20 * (rtp->lastrxformat.id ? (rtp_get_rate(&rtp->lastrxformat) / 1000) : 8);
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}
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rtp->dtmf_timeout = 0;
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@ -3010,8 +3011,9 @@ static struct ast_frame *process_cn_rfc3389(struct ast_rtp_instance *instance, u
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/* Convert comfort noise into audio with various codecs. Unfortunately this doesn't
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totally help us out becuase we don't have an engine to keep it going and we are not
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guaranteed to have it every 20ms or anything */
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if (rtpdebug)
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if (rtpdebug) {
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ast_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", (int) rtp->lastrxformat.id, len);
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}
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if (ast_test_flag(rtp, FLAG_3389_WARNING)) {
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struct ast_sockaddr remote_address = { {0,} };
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@ -3024,8 +3026,9 @@ static struct ast_frame *process_cn_rfc3389(struct ast_rtp_instance *instance, u
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}
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/* Must have at least one byte */
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if (!len)
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if (!len) {
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return NULL;
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}
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if (len < 24) {
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rtp->f.data.ptr = rtp->rawdata + AST_FRIENDLY_OFFSET;
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rtp->f.datalen = len - 1;
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@ -3099,9 +3102,10 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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/* Send to whoever sent to us */
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if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
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ast_sockaddr_copy(&rtp->rtcp->them, &addr);
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if (rtpdebug)
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if (rtpdebug) {
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ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
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ast_sockaddr_stringify(&rtp->rtcp->them));
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}
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}
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}
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@ -3121,8 +3125,9 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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length &= 0xffff;
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if ((i + length) > packetwords) {
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if (rtpdebug)
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if (rtpdebug) {
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ast_debug(1, "RTCP Read too short\n");
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}
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return &ast_null_frame;
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}
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@ -3153,8 +3158,9 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
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}
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i += 5;
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if (rc < 1)
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if (rc < 1) {
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break;
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}
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/* Intentional fall through */
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case RTCP_PT_RR:
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/* Don't handle multiple reception reports (rc > 1) yet */
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@ -3182,13 +3188,16 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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if (comp - dlsr >= lsr) {
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rtp->rtcp->accumulated_transit += rttsec;
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if (rtp->rtcp->rtt_count == 0)
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if (rtp->rtcp->rtt_count == 0) {
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rtp->rtcp->minrtt = rttsec;
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}
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if (rtp->rtcp->maxrtt<rttsec)
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if (rtp->rtcp->maxrtt<rttsec) {
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rtp->rtcp->maxrtt = rttsec;
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if (rtp->rtcp->minrtt>rttsec)
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}
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if (rtp->rtcp->minrtt>rttsec) {
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rtp->rtcp->minrtt = rttsec;
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}
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normdevrtt_current = normdev_compute(rtp->rtcp->normdevrtt, rttsec, rtp->rtcp->rtt_count);
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@ -3210,14 +3219,17 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
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reported_jitter = (double) rtp->rtcp->reported_jitter;
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if (rtp->rtcp->reported_jitter_count == 0)
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if (rtp->rtcp->reported_jitter_count == 0) {
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rtp->rtcp->reported_minjitter = reported_jitter;
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}
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if (reported_jitter < rtp->rtcp->reported_minjitter)
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if (reported_jitter < rtp->rtcp->reported_minjitter) {
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rtp->rtcp->reported_minjitter = reported_jitter;
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}
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if (reported_jitter > rtp->rtcp->reported_maxjitter)
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if (reported_jitter > rtp->rtcp->reported_maxjitter) {
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rtp->rtcp->reported_maxjitter = reported_jitter;
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}
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reported_normdev_jitter_current = normdev_compute(rtp->rtcp->reported_normdev_jitter, reported_jitter, rtp->rtcp->reported_jitter_count);
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@ -3230,14 +3242,17 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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reported_lost = (double) rtp->rtcp->reported_lost;
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/* using same counter as for jitter */
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if (rtp->rtcp->reported_jitter_count == 0)
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if (rtp->rtcp->reported_jitter_count == 0) {
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rtp->rtcp->reported_minlost = reported_lost;
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}
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if (reported_lost < rtp->rtcp->reported_minlost)
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if (reported_lost < rtp->rtcp->reported_minlost) {
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rtp->rtcp->reported_minlost = reported_lost;
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}
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if (reported_lost > rtp->rtcp->reported_maxlost)
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if (reported_lost > rtp->rtcp->reported_maxlost) {
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rtp->rtcp->reported_maxlost = reported_lost;
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}
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reported_normdev_lost_current = normdev_compute(rtp->rtcp->reported_normdev_lost, reported_lost, rtp->rtcp->reported_jitter_count);
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rtp->rtcp->reported_stdev_lost = stddev_compute(rtp->rtcp->reported_stdev_lost, reported_lost, rtp->rtcp->reported_normdev_lost, reported_normdev_lost_current, rtp->rtcp->reported_jitter_count);
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@ -3254,8 +3269,9 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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ast_verbose(" Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
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ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
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ast_verbose(" DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
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if (rtt)
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if (rtt) {
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ast_verbose(" RTT: %lu(sec)\n", (unsigned long) rtt);
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}
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}
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if (rtt) {
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manager_event(EVENT_FLAG_REPORTING, "RTCPReceived", "From: %s\r\n"
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}
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break;
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case RTCP_PT_FUR:
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if (rtcp_debug_test_addr(&addr))
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if (rtcp_debug_test_addr(&addr)) {
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ast_verbose("Received an RTCP Fast Update Request\n");
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}
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rtp->f.frametype = AST_FRAME_CONTROL;
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rtp->f.subclass.integer = AST_CONTROL_VIDUPDATE;
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rtp->f.datalen = 0;
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f = &rtp->f;
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break;
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case RTCP_PT_SDES:
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if (rtcp_debug_test_addr(&addr))
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if (rtcp_debug_test_addr(&addr)) {
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ast_verbose("Received an SDES from %s\n",
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ast_sockaddr_stringify(&rtp->rtcp->them));
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}
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break;
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case RTCP_PT_BYE:
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if (rtcp_debug_test_addr(&addr))
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if (rtcp_debug_test_addr(&addr)) {
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ast_verbose("Received a BYE from %s\n",
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ast_sockaddr_stringify(&rtp->rtcp->them));
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}
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break;
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default:
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ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s\n",
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@ -3401,12 +3420,13 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int
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ast_sockaddr_stringify(&remote_address),
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strerror(errno));
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} else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
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if (option_debug || rtpdebug)
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if (option_debug || rtpdebug) {
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ast_log(LOG_WARNING,
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"RTP NAT: Can't write RTP to private "
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"address %s, waiting for other end to "
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"send audio...\n",
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ast_sockaddr_stringify(&remote_address));
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}
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ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
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}
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return 0;
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