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Merge "res_pjsip_sdp_rtp.c: Fixup some whitespace."

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This commit is contained in:
Joshua Colp
2015-08-06 11:53:20 -05:00
committed by Gerrit Code Review
parent 3d4db97253 ba7dd38470
commit 7ebca6795b
1 changed files with 14 additions and 10 deletions
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24
res/res_pjsip_sdp_rtp.c
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@@ -237,13 +237,13 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
}
ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
if (strcmp(name,"telephone-event") == 0) {
tel_event++;
}
if (strcmp(name, "telephone-event") == 0) {
tel_event++;
}
ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]),
media, name, options, rtpmap->clock_rate);
ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL,
pj_strtoul(&stream->desc.fmt[i]), media, name, options, rtpmap->clock_rate);
/* Look for an optional associated fmtp attribute */
if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
continue;
@@ -270,8 +270,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
}
}
}
if ((tel_event==0) && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
if (!tel_event && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
/* Get the packetization, if it exists */
if ((attr = pjmedia_sdp_media_find_attr2(stream, "ptime", NULL))) {
@@ -329,7 +329,7 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
}
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
session_media->rtp);
ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
@@ -1138,8 +1138,12 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
/* Add non-codec formats */
if (media_type != AST_MEDIA_TYPE_VIDEO) {
for (index = 1LL; index <= AST_RTP_MAX; index <<= 1) {
if (!(noncodec & index) || (rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp),
0, NULL, index)) == -1) {
if (!(noncodec & index)) {
continue;
}
rtp_code = ast_rtp_codecs_payload_code(
ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index);
if (rtp_code == -1) {
continue;
}