Formatting changes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286271 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d877dea9bcabf7ff301e69f38ad29bb02bfc3cb6..115b72078201dc28d7c431f0feb7b10955986161 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6520,8 +6520,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
        ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
        ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
        ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
-       if (i->prefcodec)
+       if (i->prefcodec) {
                ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec));
+       }
 
        /* XXX Why are we choosing a codec from the native formats?? */
        fmt = ast_best_codec(tmp->nativeformats);
@@ -6546,10 +6547,11 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
                        needtext = i->jointcapability & AST_FORMAT_TEXT_MASK;   /* Inbound call */
        }
 
-       if (needvideo)
+       if (needvideo) {
                ast_debug(3, "This channel can handle video! HOLLYWOOD next!\n");
-       else
+       } else {
                ast_debug(3, "This channel will not be able to handle video.\n");
+       }
 
        enable_dsp_detect(i);
 
@@ -6573,13 +6575,16 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
                ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
                ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
        }
-       if (needtext && i->trtp)
+       if (needtext && i->trtp) {
                ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
-       if (i->udptl)
+       }
+       if (i->udptl) {
                ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
+       }
 
-       if (state == AST_STATE_RING)
+       if (state == AST_STATE_RING) {
                tmp->rings = 1;
+       }
        tmp->adsicpe = AST_ADSI_UNAVAILABLE;
 
        tmp->writeformat = fmt;
@@ -6596,14 +6601,18 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
        tmp->pickupgroup = i->pickupgroup;
        tmp->caller.id.name.presentation = i->callingpres;
        tmp->caller.id.number.presentation = i->callingpres;
-       if (!ast_strlen_zero(i->parkinglot))
+       if (!ast_strlen_zero(i->parkinglot)) {
                ast_string_field_set(tmp, parkinglot, i->parkinglot);
-       if (!ast_strlen_zero(i->accountcode))
+       }
+       if (!ast_strlen_zero(i->accountcode)) {
                ast_string_field_set(tmp, accountcode, i->accountcode);
-       if (i->amaflags)
+       }
+       if (i->amaflags) {
                tmp->amaflags = i->amaflags;
-       if (!ast_strlen_zero(i->language))
+       }
+       if (!ast_strlen_zero(i->language)) {
                ast_string_field_set(tmp, language, i->language);
+       }
        i->owner = tmp;
        ast_module_ref(ast_module_info->self);
        ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
@@ -6631,14 +6640,18 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
        }
 
        tmp->priority = 1;
-       if (!ast_strlen_zero(i->uri))
+       if (!ast_strlen_zero(i->uri)) {
                pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
-       if (!ast_strlen_zero(i->domain))
+       }
+       if (!ast_strlen_zero(i->domain)) {
                pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
-       if (!ast_strlen_zero(i->callid))
+       }
+       if (!ast_strlen_zero(i->callid)) {
                pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
-       if (i->rtp)
+       }
+       if (i->rtp) {
                ast_jb_configure(tmp, &global_jbconf);
+       }
 
        /* Set channel variables for this call from configuration */
        for (v = i->chanvars ; v ; v = v->next) {
@@ -6653,14 +6666,16 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
                tmp = NULL;
        }
 
-       if (i->do_history)
+       if (i->do_history) {
                append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
+       }
 
        /* Inform manager user about new channel and their SIP call ID */
-       if (sip_cfg.callevents)
+       if (sip_cfg.callevents) {
                manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
                        "Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
                        tmp->name, tmp->uniqueid, "SIP", i->callid, i->fullcontact);
+       }
 
        return tmp;
 }
@@ -6668,8 +6683,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 /*! \brief Reads one line of SIP message body */
 static char *get_body_by_line(const char *line, const char *name, int nameLen, char delimiter)
 {
-       if (!strncasecmp(line, name, nameLen) && line[nameLen] == delimiter)
+       if (!strncasecmp(line, name, nameLen) && line[nameLen] == delimiter) {
                return ast_skip_blanks(line + nameLen + 1);
+       }
 
        return "";
 }
@@ -6729,8 +6745,9 @@ static char *get_body(struct sip_request *req, char *name, char delimiter)
 
        for (x = 0; x < req->lines; x++) {
                r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[x]), name, len, delimiter);
-               if (r[0] != '\0')
+               if (r[0] != '\0') {
                        return r;
+               }
        }
 
        return "";
@@ -6794,8 +6811,9 @@ static const char *__get_header(const struct sip_request *req, const char *name,
                        const char *header = REQ_OFFSET_TO_STR(req, header[x]);
                        if (!strncasecmp(header, name, len)) {
                                const char *r = header + len;   /* skip name */
-                               if (sip_cfg.pedanticsipchecking)
+                               if (sip_cfg.pedanticsipchecking) {
                                        r = ast_skip_blanks(r);
+                               }
 
                                if (*r == ':') {
                                        *start = x+1;
@@ -6803,8 +6821,9 @@ static const char *__get_header(const struct sip_request *req, const char *name,
                                }
                        }
                }
-               if (pass == 0) /* Try aliases */
+               if (pass == 0) { /* Try aliases */
                        name = find_alias(name, NULL);
+               }
        }
 
        /* Don't return NULL, so get_header is always a valid pointer */
@@ -6849,11 +6868,13 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
                if (sipdebug_text) {
                        int i;
                        unsigned char* arr = f->data.ptr;
-                       for (i=0; i < f->datalen; i++)
+                       for (i=0; i < f->datalen; i++) {
                                ast_verbose("%c", (arr[i] > ' ' && arr[i] < '}') ? arr[i] : '.');
+                       }
                        ast_verbose(" -> ");
-                       for (i=0; i < f->datalen; i++)
+                       for (i=0; i < f->datalen; i++) {
                                ast_verbose("%02X ", arr[i]);
+                       }
                        ast_verbose("\n");
                }
                break;
@@ -6871,8 +6892,9 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
        }
 
        /* We already hold the channel lock */
-       if (!p->owner || (f && f->frametype != AST_FRAME_VOICE))
+       if (!p->owner || (f && f->frametype != AST_FRAME_VOICE)) {
                return f;
+       }
 
        if (f && f->subclass.codec != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
                if (!(f->subclass.codec & p->jointcapability)) {