Move early-return in TimeToSendPadding.

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 940 matching lines @@
   bool fec_enabled;
   uint8_t pt_red;
   uint8_t pt_fec;
   video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
+  if (bytes == 0)
+    return 0;
   {
     CriticalSectionScoped cs(send_critsect_.get());
     if (!sending_media_) return 0;
   }
-  if (bytes == 0)
-    return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes);
   if (bytes_sent < bytes)
     bytes_sent += TrySendPadData(bytes - bytes_sent);
   return bytes_sent;
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
 int32_t RTPSender::SendToNetwork(
     uint8_t *buffer, size_t payload_length, size_t rtp_header_length,
     int64_t capture_time_ms, StorageType storage,
@@ skipping 900 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc