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/modules/audio_coding/neteq4/interface/neteq.h b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
index ded87f5..6173930 100644
--- a/webrtc/modules/audio_coding/neteq4/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
@@ -15,6 +15,7 @@
#include <vector>
+#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
#include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "webrtc/typedefs.h"
@@ -24,14 +25,6 @@
// Forward declarations.
struct WebRtcRTPHeader;
-// RTCP statistics.
-struct RtcpStatistics {
- uint16_t fraction_lost;
- uint32_t cumulative_lost;
- uint32_t extended_max;
- uint32_t jitter;
-};
-
struct NetEqNetworkStatistics {
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t preferred_buffer_size_ms; // Target buffer size in ms.