mirror of
https://github.com/asterisk/asterisk.git
synced 2025-11-08 10:58:15 +00:00
- use '?' instead of if statements for assignment;
- fix indentation in a few places - use a variable to store the 'other' channel, thus removing the need for some duplicated code; - use '=' instead of memcpy to copy struct sockaddr_in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
53
rtp.c
53
rtp.c
@@ -783,15 +783,9 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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/* Get audio and video interface (if native bridge is possible) */
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destp = destpr->get_rtp_info(dest);
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if (destpr->get_vrtp_info)
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vdestp = destpr->get_vrtp_info(dest);
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else
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vdestp = NULL;
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vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
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srcp = srcpr->get_rtp_info(src);
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if (srcpr->get_vrtp_info)
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vsrcp = srcpr->get_vrtp_info(src);
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else
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vsrcp = NULL;
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vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
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/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
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if (!destp || !srcp) {
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@@ -832,13 +826,13 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
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int i;
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if (pt < 0 || pt > MAX_RTP_PT)
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return; /* bogus payload type */
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return; /* bogus payload type */
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for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
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if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
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strcasecmp(mimeType, mimeTypes[i].type) == 0) {
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rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
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return;
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return;
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}
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}
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}
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@@ -916,9 +910,8 @@ char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code)
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int i;
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for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
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if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat) {
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return mimeTypes[i].subtype;
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}
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if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat)
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return mimeTypes[i].subtype;
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}
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return "";
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}
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@@ -1119,7 +1112,7 @@ void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
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void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
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{
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memcpy(us, &rtp->us, sizeof(rtp->us));
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*us = rtp->us;
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}
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void ast_rtp_stop(struct ast_rtp *rtp)
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@@ -1563,7 +1556,7 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
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enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
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{
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struct ast_frame *f;
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struct ast_channel *who, *cs[3];
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struct ast_channel *who, *other, *cs[3];
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struct ast_rtp *p0, *p1; /* Audio RTP Channels */
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struct ast_rtp *vp0, *vp1; /* Video RTP channels */
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struct ast_rtp_protocol *pr0, *pr1;
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@@ -1615,15 +1608,9 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
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/* Get audio and video interface (if native bridge is possible) */
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p0 = pr0->get_rtp_info(c0);
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if (pr0->get_vrtp_info)
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vp0 = pr0->get_vrtp_info(c0);
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else
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vp0 = NULL;
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vp0 = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0) : NULL;
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p1 = pr1->get_rtp_info(c1);
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if (pr1->get_vrtp_info)
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vp1 = pr1->get_vrtp_info(c1);
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else
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vp1 = NULL;
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vp1 = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1) : NULL;
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/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
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if (!p0 || !p1) {
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@@ -1633,14 +1620,8 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
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return AST_BRIDGE_FAILED_NOWARN;
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}
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/* Get codecs from both sides */
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if (pr0->get_codec)
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codec0 = pr0->get_codec(c0);
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else
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codec0 = 0;
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if (pr1->get_codec)
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codec1 = pr1->get_codec(c1);
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else
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codec1 = 0;
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codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
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codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
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if (pr0->get_codec && pr1->get_codec) {
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/* Hey, we can't do reinvite if both parties speak different codecs */
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if (!(codec0 & codec1)) {
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@@ -1751,9 +1732,11 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
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continue;
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}
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f = ast_read(who);
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other = (who == c0) ? c1 : c0; /* the other channel */
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if (!f || ((f->frametype == AST_FRAME_DTMF) &&
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(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
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((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
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/* breaking out of the bridge. */
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*fo = f;
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*rc = who;
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if (option_debug)
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@@ -1770,7 +1753,7 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
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} else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
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if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
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(f->subclass == AST_CONTROL_VIDUPDATE)) {
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ast_indicate(who == c0 ? c1 : c0, f->subclass);
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ast_indicate(other, f->subclass);
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ast_frfree(f);
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} else {
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*fo = f;
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@@ -1783,11 +1766,7 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
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(f->frametype == AST_FRAME_VOICE) ||
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(f->frametype == AST_FRAME_VIDEO)) {
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/* Forward voice or DTMF frames if they happen upon us */
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if (who == c0) {
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ast_write(c1, f);
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} else if (who == c1) {
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ast_write(c0, f);
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}
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ast_write(other, f);
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}
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ast_frfree(f);
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}
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