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authortilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-06 22:51:48 +0000
committertilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-06 22:51:48 +0000
commit4b2fc9d3e738dbb2e7dd297de2f35e571e350691 (patch)
treee611f1c508d4cbd9720858d06c92a4cfaa25dccb /channels/chan_phone.c
parentfed34a6362c74a4f65bae5fb89bcae3f58db5f27 (diff)
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_phone.c')
-rw-r--r--channels/chan_phone.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index 33f515d5e..17204fda5 100644
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -594,7 +594,7 @@ static struct ast_frame *phone_read(struct ast_channel *ast)
}
p->fr.samples = 240;
p->fr.datalen = res;
- p->fr.frametype = p->lastinput <= AST_FORMAT_MAX_AUDIO ?
+ p->fr.frametype = p->lastinput <= AST_FORMAT_AUDIO_MASK ?
AST_FRAME_VOICE :
p->lastinput <= AST_FORMAT_PNG ? AST_FRAME_IMAGE
: AST_FRAME_VIDEO;