[svn-commits] file: branch group/media_formats r408269 - /team/group/media_formats/channels/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Feb 17 09:12:01 CST 2014
Author: file
Date: Mon Feb 17 09:11:56 2014
New Revision: 408269
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=408269
Log:
Move chan_motif over.
Since the RTP engine API isn't done yet this'll need to be tweaked again for packetization.
Modified:
team/group/media_formats/channels/chan_motif.c
Modified: team/group/media_formats/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/channels/chan_motif.c?view=diff&rev=408269&r1=408268&r2=408269
==============================================================================
--- team/group/media_formats/channels/chan_motif.c (original)
+++ team/group/media_formats/channels/chan_motif.c Mon Feb 17 09:11:56 2014
@@ -77,6 +77,7 @@
#include "asterisk/abstract_jb.h"
#include "asterisk/xmpp.h"
#include "asterisk/stasis_channels.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<configInfo name="chan_motif" language="en_US">
@@ -286,7 +287,6 @@
iksrule *rule; /*!< Active matching rule */
unsigned int maxicecandidates; /*!< Maximum number of ICE candidates we will offer */
unsigned int maxpayloads; /*!< Maximum number of payloads we will offer */
- struct ast_codec_pref prefs; /*!< Codec preferences */
struct ast_format_cap *cap; /*!< Formats to use */
ast_group_t callgroup; /*!< Call group */
ast_group_t pickupgroup; /*!< Pickup group */
@@ -309,7 +309,6 @@
char remote_original[XMPP_MAX_JIDLEN];/*!< Identifier of the original remote party (remote may have changed due to redirect) */
char remote[XMPP_MAX_JIDLEN]; /*!< Identifier of the remote party */
iksrule *rule; /*!< Session matching rule */
- struct ast_codec_pref prefs; /*!< Codec preferences */
struct ast_channel *owner; /*!< Master Channel */
struct ast_rtp_instance *rtp; /*!< RTP audio session */
struct ast_rtp_instance *vrtp; /*!< RTP video session */
@@ -454,8 +453,7 @@
ast_xmpp_client_unref(endpoint->connection);
}
- ast_format_cap_destroy(endpoint->cap);
-
+ ao2_cleanup(endpoint->cap);
ao2_ref(endpoint->state, -1);
ast_string_field_free_memory(endpoint);
@@ -519,7 +517,7 @@
ast_string_field_set(endpoint, name, cat);
- endpoint->cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
+ endpoint->cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
endpoint->transport = JINGLE_TRANSPORT_ICE_UDP;
return endpoint;
@@ -583,9 +581,9 @@
ast_rtp_instance_destroy(session->vrtp);
}
- ast_format_cap_destroy(session->cap);
- ast_format_cap_destroy(session->jointcap);
- ast_format_cap_destroy(session->peercap);
+ ao2_cleanup(session->cap);
+ ao2_cleanup(session->jointcap);
+ ao2_cleanup(session->peercap);
if (session->callid) {
ast_callid_unref(session->callid);
@@ -681,7 +679,7 @@
}
/* If there are no configured video codecs do not turn video support on, it just won't work */
- if (!ast_format_cap_has_type(session->cap, AST_FORMAT_TYPE_VIDEO)) {
+ if (!ast_format_cap_has_type(session->cap, AST_MEDIA_TYPE_VIDEO)) {
return;
}
@@ -695,7 +693,7 @@
ast_rtp_instance_set_channel_id(session->vrtp, ast_channel_uniqueid(session->owner));
ast_channel_set_fd(session->owner, 2, ast_rtp_instance_fd(session->vrtp, 0));
ast_channel_set_fd(session->owner, 3, ast_rtp_instance_fd(session->vrtp, 1));
- ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->vrtp), session->vrtp, &session->prefs);
+// ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->vrtp), session->vrtp, &session->prefs);
if (session->transport == JINGLE_TRANSPORT_GOOGLE_V2 && (ice = ast_rtp_instance_get_ice(session->vrtp))) {
ice->stop(session->vrtp);
@@ -741,15 +739,15 @@
session->connection = endpoint->connection;
session->transport = endpoint->transport;
- if (!(session->cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK)) ||
- !(session->jointcap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK)) ||
- !(session->peercap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK)) ||
+ if (!(session->cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
+ !(session->jointcap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
+ !(session->peercap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!session->callid) {
ao2_ref(session, -1);
return NULL;
}
- ast_format_cap_copy(session->cap, endpoint->cap);
+ ast_format_cap_append_by_type(session->cap, endpoint->cap, AST_MEDIA_TYPE_UNKNOWN);
/* While we rely on res_xmpp for communication we still need a temporary ast_sockaddr to tell the RTP engine
* that we want IPv4 */
@@ -763,8 +761,6 @@
ast_rtp_instance_set_prop(session->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_instance_set_prop(session->rtp, AST_RTP_PROPERTY_DTMF, 1);
- memcpy(&session->prefs, &endpoint->prefs, sizeof(session->prefs));
-
session->maxicecandidates = endpoint->maxicecandidates;
session->maxpayloads = endpoint->maxpayloads;
@@ -776,13 +772,20 @@
{
struct ast_channel *chan;
const char *str = S_OR(title, session->remote);
- struct ast_format tmpfmt;
-
- if (ast_format_cap_is_empty(session->cap)) {
+ struct ast_format_cap *caps;
+ struct ast_format *tmpfmt;
+
+ if (!ast_format_cap_count(session->cap)) {
return NULL;
}
+ caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (!caps) {
+ return NULL;
+ }
+
if (!(chan = ast_channel_alloc(1, state, S_OR(title, ""), S_OR(cid_name, ""), "", "", "", linkedid, 0, "Motif/%s-%04lx", str, ast_random() & 0xffff))) {
+ ao2_ref(caps, -1);
return NULL;
}
@@ -794,15 +797,18 @@
ast_channel_callid_set(chan, session->callid);
- ast_format_cap_copy(ast_channel_nativeformats(chan), session->cap);
- ast_codec_choose(&session->prefs, session->cap, 1, &tmpfmt);
+ ast_format_cap_append_by_type(caps, session->cap, AST_MEDIA_TYPE_UNKNOWN);
+ ast_channel_nativeformats_set(chan, caps);
+ ao2_ref(caps, -1);
+
+ tmpfmt = ast_format_cap_get_format(session->cap, 0);
if (session->rtp) {
struct ast_rtp_engine_ice *ice;
ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(session->rtp, 0));
ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(session->rtp, 1));
- ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->rtp), session->rtp, &session->prefs);
+// ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->rtp), session->rtp, &session->prefs);
if (((session->transport == JINGLE_TRANSPORT_GOOGLE_V2) ||
(session->transport == JINGLE_TRANSPORT_GOOGLE_V1)) &&
@@ -818,11 +824,12 @@
ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
- ast_best_codec(ast_channel_nativeformats(chan), &tmpfmt);
- ast_format_copy(ast_channel_writeformat(chan), &tmpfmt);
- ast_format_copy(ast_channel_rawwriteformat(chan), &tmpfmt);
- ast_format_copy(ast_channel_readformat(chan), &tmpfmt);
- ast_format_copy(ast_channel_rawreadformat(chan), &tmpfmt);
+ ast_channel_set_writeformat(chan, tmpfmt);
+ ast_channel_set_rawwriteformat(chan, tmpfmt);
+ ast_channel_set_readformat(chan, tmpfmt);
+ ast_channel_set_rawreadformat(chan, tmpfmt);
+
+ ao2_ref(tmpfmt, -1);
ao2_lock(endpoint);
@@ -1300,30 +1307,24 @@
}
/*! \brief Internal helper function which adds payloads to a description */
-static int jingle_add_payloads_to_description(struct jingle_session *session, struct ast_rtp_instance *rtp, iks *description, iks **payloads, enum ast_format_type type)
-{
- struct ast_format format;
+static int jingle_add_payloads_to_description(struct jingle_session *session, struct ast_rtp_instance *rtp, iks *description, iks **payloads, enum ast_media_type type)
+{
int x = 0, i = 0, res = 0;
- for (x = 0; (x < AST_CODEC_PREF_SIZE) && (i < (session->maxpayloads - 2)); x++) {
+ for (x = 0; (x < ast_format_cap_count(session->jointcap)) && (i < (session->maxpayloads - 2)); x++) {
+ struct ast_format *format = ast_format_cap_get_format(session->jointcap, x);
int rtp_code;
iks *payload;
char tmp[32];
- if (!ast_codec_pref_index(&session->prefs, x, &format)) {
- break;
- }
-
- if (AST_FORMAT_GET_TYPE(format.id) != type) {
+ if (format->codec->type != type) {
+ ao2_ref(format, -1);
continue;
}
- if (!ast_format_cap_iscompatible(session->jointcap, &format)) {
- continue;
- }
-
- if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp), 1, &format, 0)) == -1) ||
+ if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp), 1, format, 0)) == -1) ||
(!(payload = iks_new("payload-type")))) {
+ ao2_ref(format, -1);
return -1;
}
@@ -1333,17 +1334,18 @@
snprintf(tmp, sizeof(tmp), "%d", rtp_code);
iks_insert_attrib(payload, "id", tmp);
- iks_insert_attrib(payload, "name", ast_rtp_lookup_mime_subtype2(1, &format, 0, 0));
+ iks_insert_attrib(payload, "name", ast_rtp_lookup_mime_subtype2(1, format, 0, 0));
iks_insert_attrib(payload, "channels", "1");
- if ((format.id == AST_FORMAT_G722) && ((session->transport == JINGLE_TRANSPORT_GOOGLE_V1) || (session->transport == JINGLE_TRANSPORT_GOOGLE_V2))) {
+ if ((ast_format_cmp(format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) &&
+ ((session->transport == JINGLE_TRANSPORT_GOOGLE_V1) || (session->transport == JINGLE_TRANSPORT_GOOGLE_V2))) {
iks_insert_attrib(payload, "clockrate", "16000");
} else {
- snprintf(tmp, sizeof(tmp), "%d", ast_rtp_lookup_sample_rate2(1, &format, 0));
+ snprintf(tmp, sizeof(tmp), "%d", ast_rtp_lookup_sample_rate2(1, format, 0));
iks_insert_attrib(payload, "clockrate", tmp);
}
- if ((type == AST_FORMAT_TYPE_VIDEO) && (session->transport == JINGLE_TRANSPORT_GOOGLE_V2)) {
+ if ((type == AST_MEDIA_TYPE_VIDEO) && (session->transport == JINGLE_TRANSPORT_GOOGLE_V2)) {
iks *parameter;
/* Google requires these parameters to be set, but alas we can not give accurate values so use some safe defaults */
@@ -1366,9 +1368,11 @@
iks_insert_node(description, payload);
payloads[i++] = payload;
+
+ ao2_ref(format, -1);
}
/* If this is for audio and there is room for RFC2833 add it in */
- if ((type == AST_FORMAT_TYPE_AUDIO) && (i < session->maxpayloads)) {
+ if ((type == AST_MEDIA_TYPE_AUDIO) && (i < session->maxpayloads)) {
iks *payload;
if ((payload = iks_new("payload-type"))) {
@@ -1390,7 +1394,7 @@
/*! \brief Helper function which adds content to a description */
static int jingle_add_content(struct jingle_session *session, iks *jingle, iks *content, iks *description, iks *transport,
- const char *name, enum ast_format_type type, struct ast_rtp_instance *rtp, iks **payloads)
+ const char *name, enum ast_media_type type, struct ast_rtp_instance *rtp, iks **payloads)
{
int res = 0;
@@ -1400,9 +1404,9 @@
iks_insert_node(jingle, content);
iks_insert_attrib(description, "xmlns", JINGLE_RTP_NS);
- if (type == AST_FORMAT_TYPE_AUDIO) {
+ if (type == AST_MEDIA_TYPE_AUDIO) {
iks_insert_attrib(description, "media", "audio");
- } else if (type == AST_FORMAT_TYPE_VIDEO) {
+ } else if (type == AST_MEDIA_TYPE_VIDEO) {
iks_insert_attrib(description, "media", "video");
} else {
return -1;
@@ -1469,7 +1473,7 @@
if (session->rtp && (audio = iks_new("content")) && (audio_description = iks_new("description")) &&
(audio_transport = iks_new("transport"))) {
res = jingle_add_content(session, jingle, audio, audio_description, audio_transport, session->audio_name,
- AST_FORMAT_TYPE_AUDIO, session->rtp, audio_payloads);
+ AST_MEDIA_TYPE_AUDIO, session->rtp, audio_payloads);
} else {
ast_log(LOG_ERROR, "Failed to allocate audio content stanzas for session '%s', hanging up\n", session->sid);
res = -1;
@@ -1479,7 +1483,7 @@
if ((video = iks_new("content")) && (video_description = iks_new("description")) &&
(video_transport = iks_new("transport"))) {
res = jingle_add_content(session, jingle, video, video_description, video_transport, session->video_name,
- AST_FORMAT_TYPE_VIDEO, session->vrtp, video_payloads);
+ AST_MEDIA_TYPE_VIDEO, session->vrtp, video_payloads);
} else {
ast_log(LOG_ERROR, "Failed to allocate video content stanzas for session '%s', hanging up\n", session->sid);
res = -1;
@@ -1668,17 +1672,24 @@
}
if (frame && frame->frametype == AST_FRAME_VOICE &&
- !ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format)) {
- if (!ast_format_cap_iscompatible(session->jointcap, &frame->subclass.format)) {
+ !ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format)) {
+ if (!ast_format_cap_iscompatible_format(session->jointcap, frame->subclass.format)) {
ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
- ast_getformatname(&frame->subclass.format), ast_channel_name(ast));
+ frame->subclass.format->codec->name, ast_channel_name(ast));
ast_frfree(frame);
frame = &ast_null_frame;
} else {
+ struct ast_format_cap *caps;
+
ast_debug(1, "Oooh, format changed to %s\n",
- ast_getformatname(&frame->subclass.format));
- ast_format_cap_remove_bytype(ast_channel_nativeformats(ast), AST_FORMAT_TYPE_AUDIO);
- ast_format_cap_add(ast_channel_nativeformats(ast), &frame->subclass.format);
+ frame->subclass.format->codec->name);
+
+ caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (caps) {
+ ast_format_cap_add(caps, frame->subclass.format, 0);
+ ast_channel_nativeformats_set(ast, caps);
+ ao2_ref(caps, -1);
+ }
ast_set_read_format(ast, ast_channel_readformat(ast));
ast_set_write_format(ast, ast_channel_writeformat(ast));
}
@@ -1696,13 +1707,13 @@
switch (frame->frametype) {
case AST_FRAME_VOICE:
- if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
+ if (!(ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format))) {
ast_log(LOG_WARNING,
"Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
- ast_getformatname(&frame->subclass.format),
+ frame->subclass.format->codec->name,
ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
- ast_getformatname(ast_channel_readformat(ast)),
- ast_getformatname(ast_channel_writeformat(ast)));
+ ast_channel_readformat(ast)->codec->name,
+ ast_channel_writeformat(ast)->codec->name);
return 0;
}
if (session && session->rtp) {
@@ -1845,7 +1856,7 @@
ast_setstate(ast, AST_STATE_RING);
/* Since we have no idea of the remote capabilities use ours for now */
- ast_format_cap_copy(session->jointcap, session->cap);
+ ast_format_cap_append_by_type(session->jointcap, session->cap, AST_MEDIA_TYPE_UNKNOWN);
/* We set up a hook so we can know when our session-initiate message was accepted or rejected */
session->rule = iks_filter_add_rule(session->connection->filter, jingle_outgoing_hook, session,
@@ -1908,7 +1919,7 @@
);
/* We require at a minimum one audio format to be requested */
- if (!ast_format_cap_has_type(cap, AST_FORMAT_TYPE_AUDIO)) {
+ if (!ast_format_cap_has_type(cap, AST_MEDIA_TYPE_AUDIO)) {
ast_log(LOG_ERROR, "Motif channel driver requires an audio format when dialing a destination\n");
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
return NULL;
@@ -2001,7 +2012,7 @@
}
/* If video was requested try to enable it on the session */
- if (ast_format_cap_has_type(cap, AST_FORMAT_TYPE_VIDEO)) {
+ if (ast_format_cap_has_type(cap, AST_MEDIA_TYPE_VIDEO)) {
jingle_enable_video(session);
}
@@ -2043,8 +2054,8 @@
ast_string_field_set(session, audio_name, name);
}
*rtp = session->rtp;
- ast_format_cap_remove_bytype(session->peercap, AST_FORMAT_TYPE_AUDIO);
- ast_format_cap_remove_bytype(session->jointcap, AST_FORMAT_TYPE_AUDIO);
+ ast_format_cap_remove_bytype(session->peercap, AST_MEDIA_TYPE_AUDIO);
+ ast_format_cap_remove_bytype(session->jointcap, AST_MEDIA_TYPE_AUDIO);
} else if (!strcasecmp(media, "video")) {
if (!ast_strlen_zero(name)) {
ast_string_field_set(session, video_name, name);
@@ -2060,8 +2071,8 @@
return -1;
}
- ast_format_cap_remove_bytype(session->peercap, AST_FORMAT_TYPE_VIDEO);
- ast_format_cap_remove_bytype(session->jointcap, AST_FORMAT_TYPE_VIDEO);
+ ast_format_cap_remove_bytype(session->peercap, AST_MEDIA_TYPE_VIDEO);
+ ast_format_cap_remove_bytype(session->jointcap, AST_MEDIA_TYPE_VIDEO);
} else {
/* Unknown media type */
jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
@@ -2093,9 +2104,9 @@
}
ast_rtp_codecs_payload_formats(&codecs, session->peercap, &othercapability);
- ast_format_cap_joint_append(session->cap, session->peercap, session->jointcap);
-
- if (ast_format_cap_is_empty(session->jointcap)) {
+ ast_format_cap_get_compatible(session->cap, session->peercap, session->jointcap);
+
+ if (!ast_format_cap_count(session->jointcap)) {
/* We have no compatible codecs, so terminate the session appropriately */
jingle_queue_hangup_with_cause(session, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
ast_rtp_codecs_payloads_destroy(&codecs);
@@ -2355,12 +2366,20 @@
}
if ((chan = jingle_session_lock_full(session))) {
- struct ast_format fmt;
-
- ast_format_cap_copy(ast_channel_nativeformats(chan), session->jointcap);
- ast_codec_choose(&session->prefs, session->jointcap, 1, &fmt);
- ast_set_read_format(chan, &fmt);
- ast_set_write_format(chan, &fmt);
+ struct ast_format_cap *caps;
+ struct ast_format *fmt;
+
+ caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (caps) {
+ ast_format_cap_append_by_type(caps, session->jointcap, AST_MEDIA_TYPE_UNKNOWN);
+ ast_channel_nativeformats_set(chan, caps);
+ ao2_ref(caps, -1);
+ }
+
+ fmt = ast_format_cap_get_format(session->jointcap, 0);
+ ast_set_read_format(chan, fmt);
+ ast_set_write_format(chan, fmt);
+ ao2_ref(fmt, -1);
ast_channel_unlock(chan);
ast_channel_unref(chan);
@@ -2710,7 +2729,7 @@
*/
static int load_module(void)
{
- if (!(jingle_tech.capabilities = ast_format_cap_alloc(0))) {
+ if (!(jingle_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
return AST_MODULE_LOAD_DECLINE;
}
@@ -2726,8 +2745,8 @@
aco_option_register(&cfg_info, "musicclass", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, musicclass));
aco_option_register(&cfg_info, "parkinglot", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, parkinglot));
aco_option_register(&cfg_info, "accountcode", ACO_EXACT, endpoint_options, NULL, OPT_STRINGFIELD_T, 0, STRFLDSET(struct jingle_endpoint, accountcode));
- aco_option_register(&cfg_info, "allow", ACO_EXACT, endpoint_options, "ulaw,alaw", OPT_CODEC_T, 1, FLDSET(struct jingle_endpoint, prefs, cap));
- aco_option_register(&cfg_info, "disallow", ACO_EXACT, endpoint_options, "all", OPT_CODEC_T, 0, FLDSET(struct jingle_endpoint, prefs, cap));
+ aco_option_register(&cfg_info, "allow", ACO_EXACT, endpoint_options, "ulaw,alaw", OPT_CODEC_T, 1, FLDSET(struct jingle_endpoint, cap));
+ aco_option_register(&cfg_info, "disallow", ACO_EXACT, endpoint_options, "all", OPT_CODEC_T, 0, FLDSET(struct jingle_endpoint, cap));
aco_option_register_custom(&cfg_info, "connection", ACO_EXACT, endpoint_options, NULL, custom_connection_handler, 0);
aco_option_register_custom(&cfg_info, "transport", ACO_EXACT, endpoint_options, NULL, custom_transport_handler, 0);
aco_option_register(&cfg_info, "maxicecandidates", ACO_EXACT, endpoint_options, DEFAULT_MAX_ICE_CANDIDATES, OPT_UINT_T, PARSE_DEFAULT,
@@ -2735,7 +2754,7 @@
aco_option_register(&cfg_info, "maxpayloads", ACO_EXACT, endpoint_options, DEFAULT_MAX_PAYLOADS, OPT_UINT_T, PARSE_DEFAULT,
FLDSET(struct jingle_endpoint, maxpayloads), DEFAULT_MAX_PAYLOADS);
- ast_format_cap_add_all_by_type(jingle_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
+ ast_format_cap_add_all_by_type(jingle_tech.capabilities, AST_MEDIA_TYPE_AUDIO);
if (aco_process_config(&cfg_info, 0)) {
ast_log(LOG_ERROR, "Unable to read config file motif.conf. Module loaded but not running.\n");
@@ -2784,7 +2803,7 @@
static int unload_module(void)
{
ast_channel_unregister(&jingle_tech);
- ast_format_cap_destroy(jingle_tech.capabilities);
+ ao2_cleanup(jingle_tech.capabilities);
jingle_tech.capabilities = NULL;
ast_rtp_glue_unregister(&jingle_rtp_glue);
ast_sched_context_destroy(sched);
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