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authortilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b>2009-10-20 22:09:07 +0000
committertilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b>2009-10-20 22:09:07 +0000
commit38f43cba1be34203172d6f91be83968bb51c18fe (patch)
tree83526bb72e59a3a34f3de1aad8dbe0d8ced6dc73 /funcs
parentf7822860b4ee84f99ef058bf21caba0b2eb0a028 (diff)
Merged revisions 224855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224856 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'funcs')
-rw-r--r--funcs/func_speex.c21
1 files changed, 13 insertions, 8 deletions
diff --git a/funcs/func_speex.c b/funcs/func_speex.c
index d5246b0bd..699d4fd92 100644
--- a/funcs/func_speex.c
+++ b/funcs/func_speex.c
@@ -142,25 +142,24 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
struct ast_datastore *datastore = NULL;
struct speex_direction_info *sdi = NULL;
struct speex_info *si = NULL;
+ char source[80];
/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
- return 0;
+ return -1;
}
-
- ast_channel_lock(chan);
+
+ /* We are called with chan already locked */
if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
- ast_channel_unlock(chan);
- return 0;
+ return -1;
}
- ast_channel_unlock(chan);
si = datastore->data;
sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
if (!sdi) {
- return 0;
+ return -1;
}
if (sdi->samples != frame->samples) {
@@ -171,7 +170,7 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
return -1;
}
-
+
speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
if (sdi->agc) {
@@ -182,6 +181,12 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
}
speex_preprocess(sdi->state, frame->data.ptr, NULL);
+ snprintf(source, sizeof(source), "%s/speex", frame->src);
+ if (frame->mallocd & AST_MALLOCD_SRC) {
+ ast_free((char *) frame->src);
+ }
+ frame->src = ast_strdup(source);
+ frame->mallocd |= AST_MALLOCD_SRC;
return 0;
}