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Merge topic 'misc_rtp_tweaks' into 13

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* changes:
  rtp_engine.c: Must protect mime_types_len with mime_types_lock.
  res_pjsip_sdp_rtp.c: Fixup some whitespace.
This commit is contained in:
Joshua Colp
2015-08-06 11:50:25 -05:00
committed by Gerrit Code Review
parent c07fa843ec 1519eb44a7
commit ba40b07ddc
2 changed files with 24 additions and 15 deletions
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15
main/rtp_engine.c
View File

@@ -603,6 +603,7 @@ void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
{
ast_rtp_codecs_payloads_destroy(codecs);
ast_rtp_codecs_payloads_initialize(codecs);
if (instance && instance->engine && instance->engine->payload_set) {
int i;
@@ -610,8 +611,6 @@ void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp
instance->engine->payload_set(instance, i, 0, NULL, 0);
}
}
ast_rtp_codecs_payloads_initialize(codecs);
}
void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
@@ -1716,12 +1715,16 @@ static void rtp_engine_mime_type_cleanup(int i)
static void set_next_mime_type(struct ast_format *format, int rtp_code, const char *type, const char *subtype, unsigned int sample_rate)
{
int x = mime_types_len;
if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
int x;
ast_rwlock_wrlock(&mime_types_lock);
x = mime_types_len;
if (ARRAY_LEN(ast_rtp_mime_types) <= x) {
ast_rwlock_unlock(&mime_types_lock);
return;
}
ast_rwlock_wrlock(&mime_types_lock);
/* Make sure any previous value in ast_rtp_mime_types is cleaned up */
memset(&ast_rtp_mime_types[x], 0, sizeof(struct ast_rtp_mime_type));
if (format) {
@@ -1734,6 +1737,7 @@ static void set_next_mime_type(struct ast_format *format, int rtp_code, const ch
ast_copy_string(ast_rtp_mime_types[x].subtype, subtype, sizeof(ast_rtp_mime_types[x].subtype));
ast_rtp_mime_types[x].sample_rate = sample_rate;
mime_types_len++;
ast_rwlock_unlock(&mime_types_lock);
}
@@ -2095,6 +2099,7 @@ static void rtp_engine_shutdown(void)
rtp_engine_mime_type_cleanup(x);
}
}
mime_types_len = 0;
ast_rwlock_unlock(&mime_types_lock);
}

24
res/res_pjsip_sdp_rtp.c
View File

@@ -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);
@@ -1137,8 +1137,12 @@ 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;
}