Add a histogram metric tracking for how long audio RTP packets are sent

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index b9064431069213ee01d03cfb7c499d7fa4d70bc7..2216b622735f3547d59422e34777faeac2cfa4c7 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -233,7 +233,7 @@ struct ConfigHelper {
         .Times(1);
     EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
         .Times(1);
-    EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);

[ossu, 2017/07/14 11:13:15]

This is odd. Why would the test want to set the external transport to nullptr
(rather than DeRegistering the external transport)? Also, with a nullptr
external transport, I'd expect we wouldn't want to wrap it with a
TimedTransport.

[saza WebRTC, 2017/07/17 14:27:29]

Done.

+    EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(_)).Times(1);
     EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))