Unify FrameType and VideoFrameType.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 17 matching lines @@
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 const size_t kMaxPaddingLength = 224;
 const int kSendSideDelayWindowMs = 1000;
 
 namespace {
 
 const size_t kRtpHeaderLength = 12;
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
-    case kFrameEmpty: return "empty";
+    case kEmptyFrame:
+      return "empty";
     case kAudioFrameSpeech: return "audio_speech";
     case kAudioFrameCN: return "audio_cn";
     case kVideoFrameKey: return "video_key";
     case kVideoFrameDelta: return "video_delta";
   }
   return "";
 }
 
 }  // namespace
 
@@ skipping 453 matching lines @@
   if (CheckPayloadType(payload_type, &video_type) != 0) {
     LOG(LS_ERROR) << "Don't send data with unknown payload type.";
     return -1;
   }
 
   int32_t ret_val;
   if (audio_configured_) {
     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 == kEmptyFrame);
 
     ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
                                 payload_data, payload_size, fragmentation);
   } else {
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
 
-    if (frame_type == kFrameEmpty)
+    if (frame_type == kEmptyFrame)
       return 0;
 
     ret_val =
         video_->SendVideo(video_type, frame_type, payload_type,
                           capture_timestamp, capture_time_ms, payload_data,
                           payload_size, fragmentation, rtp_hdr);
   }
 
   CriticalSectionScoped cs(statistics_crit_.get());
   // Note: This is currently only counting for video.
@@ skipping 1359 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc