Unify FrameType and VideoFrameType.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index ae292a0a8c81a2047e6831655889b73e018ee006..a77d31547e10d6a51ec818c3029c673ce71ec2a2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -35,7 +35,8 @@ const size_t kRtpHeaderLength = 12;
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
-    case kFrameEmpty: return "empty";
+    case kSkipFrame:
+      return "empty";

[stefan-webrtc, 2015/10/01 09:21:20]

skip?

[pbos-webrtc, 2015/10/06 15:42:45]

kEmptyFrame

     case kAudioFrameSpeech: return "audio_speech";
     case kAudioFrameCN: return "audio_cn";
     case kVideoFrameKey: return "video_key";
@@ -509,7 +510,7 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
-           frame_type == kFrameEmpty);
+           frame_type == kSkipFrame);
 
     ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
                                 payload_data, payload_size, fragmentation);
@@ -518,7 +519,7 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
 
-    if (frame_type == kFrameEmpty)
+    if (frame_type == kSkipFrame)
       return 0;
 
     ret_val =