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https://github.com/asterisk/asterisk.git
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Minor fixes to playback logic
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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18
file.c
18
file.c
@@ -601,12 +601,13 @@ int ast_closestream(struct ast_filestream *f)
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{
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/* Stop a running stream if there is one */
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if (f->owner) {
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f->owner->stream = NULL;
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if (f->fmt->format < AST_FORMAT_MAX_AUDIO) {
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f->owner->stream = NULL;
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if (f->owner->streamid > -1)
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ast_sched_del(f->owner->sched, f->owner->streamid);
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f->owner->streamid = -1;
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} else {
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f->owner->vstream = NULL;
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if (f->owner->vstreamid > -1)
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ast_sched_del(f->owner->sched, f->owner->vstreamid);
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f->owner->vstreamid = -1;
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@@ -766,7 +767,10 @@ char ast_waitstream(struct ast_channel *c, char *breakon)
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while(c->stream) {
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res = ast_sched_wait(c->sched);
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if (res < 0) {
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ast_closestream(c->stream);
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if (c->stream)
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ast_closestream(c->stream);
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if (c->vstream)
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ast_closestream(c->vstream);
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break;
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}
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/* Setup timeout if supported */
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@@ -822,7 +826,10 @@ char ast_waitstream_fr(struct ast_channel *c, char *breakon, char *forward, char
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while(c->stream) {
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res = ast_sched_wait(c->sched);
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if (res < 0) {
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ast_closestream(c->stream);
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if (c->stream)
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ast_closestream(c->stream);
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if (c->vstream)
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ast_closestream(c->vstream);
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break;
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}
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ast_settimeout(c, res);
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@@ -886,7 +893,10 @@ char ast_waitstream_full(struct ast_channel *c, char *breakon, int audiofd, int
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while(c->stream) {
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ms = ast_sched_wait(c->sched);
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if (ms < 0) {
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ast_closestream(c->stream);
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if (c->stream)
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ast_closestream(c->stream);
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if (c->vstream)
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ast_closestream(c->vstream);
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break;
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}
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ast_settimeout(c, ms);
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