Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.
BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 7e88666..b736a2f 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -225,6 +225,22 @@
Transport() {}
};
+struct RtcpStatistics {
+ public:
+ RtcpStatistics()
+ : fraction_lost(0),
+ cumulative_lost(0),
+ extended_max_sequence_number(0),
+ jitter(0),
+ max_jitter(0) {}
+
+ uint8_t fraction_lost;
+ uint32_t cumulative_lost;
+ uint32_t extended_max_sequence_number;
+ uint32_t jitter;
+ uint32_t max_jitter;
+};
+
// ==================================================================
// Voice specific types
// ==================================================================
