Wire up statistics in video send stream of new video engine api

Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5559006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5344 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 17 insertions, 0 deletions

diff --git a/config.h b/config.h
index 3ff3bb86..f42895ae 100644
--- a/config.h
+++ b/config.h
@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -33,6 +34,22 @@ struct RtpStatistics {
   std::string c_name;
 };
 
+struct StreamStats {
+  StreamStats() : key_frames(0), delta_frames(0), bitrate_bps(0) {}
+  uint32_t key_frames;
+  uint32_t delta_frames;
+  int32_t bitrate_bps;
+  StreamDataCounters rtp_stats;
+  RtcpStatistics rtcp_stats;
+
+  bool operator==(const StreamStats& other) const {
+    return key_frames == other.key_frames &&
+           delta_frames == other.delta_frames &&
+           bitrate_bps == other.bitrate_bps && rtp_stats == other.rtp_stats &&
+           rtcp_stats == other.rtcp_stats;
+  }
+};
+
 // Settings for NACK, see RFC 4585 for details.
 struct NackConfig {
   NackConfig() : rtp_history_ms(0) {}