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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 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 373 matching lines @@
               ? std::move(audio_network_adaptor_creator)
               : [this](const ProtoString& config_string,
                        RtcEventLog* event_log) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
                                                            event_log);
               }),
       bitrate_smoother_(bitrate_smoother
           ? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
               // We choose 5sec as initial time constant due to empirical data.
               new SmoothingFilterImpl(5000))) {
-  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
 
   // Sanity check of the redundant payload type field that we want to get rid
   // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
   RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
 
   RTC_CHECK(RecreateEncoderInstance(config));
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
     : AudioEncoderOpus(CreateConfig(codec_inst), codec_inst.pltype) {}
@@ skipping 361 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc