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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-03-06 00:16:30 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-03-06 00:16:30 +0000
commit1fb47926d2de3a02e5f452b8d7c9945ca2fa58b2 (patch)
tree2c2ea15fbaf06e62b0950f17e96140076eb7c75f /include
parentcb639936d1e86c30af8a0537928c74ee8c67d480 (diff)
Merged revisions 105933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines Merged revisions 105932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106310 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/frame.h11
1 files changed, 11 insertions, 0 deletions
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 012f7a952..3206f860f 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -611,6 +611,17 @@ int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
*/
int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2);
+/*!
+ * \brief Get the sample rate for a given format.
+ */
+static force_inline int ast_format_rate(int format)
+{
+ if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
+ return 16000;
+
+ return 8000;
+}
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif