Replace CHECK(x && y) with two separate CHECK() calls

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 270 matching lines @@
   return 0;
 }
 
 // TODO(nisse): Delete this method, only used internally and by test code.
 void RTPSender::SetSendPayloadType(int8_t payload_type) {
   rtc::CritScope lock(&send_critsect_);
   payload_type_ = payload_type;
 }
 
 void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
-  // Sanity check.
-  RTC_DCHECK(max_packet_size >= 100 && max_packet_size <= IP_PACKET_SIZE)
-      << "Invalid max payload length: " << max_packet_size;
+  RTC_DCHECK_GE(max_packet_size, 100);
+  RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
   rtc::CritScope lock(&send_critsect_);
   max_packet_size_ = max_packet_size;
 }
 
 size_t RTPSender::MaxRtpPacketSize() const {
   return max_packet_size_;
 }
 
 void RTPSender::SetRtxStatus(int mode) {
   rtc::CritScope lock(&send_critsect_);
@@ skipping 992 matching lines @@
     rtc::CritScope lock(&send_critsect_);
     packet->SetTimestamp(last_rtp_timestamp_);
     packet->set_capture_time_ms(capture_time_ms_);
   }
   AssignSequenceNumber(packet.get());
   SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
                 RtpPacketSender::Priority::kLowPriority);
 }
 
 }  // namespace webrtc