Adding a some checks and switching out a few assert for RTC_[D]CHECK.

webrtc/voice_engine/channel.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 2975 matching lines @@
       _inputExternalMediaCallbackPtr->Process(
           _channelId, kRecordingPerChannel, (int16_t*)_audioFrame.data_,
           _audioFrame.samples_per_channel_, _audioFrame.sample_rate_hz_,
           isStereo);
     }
   }
 
   if (_includeAudioLevelIndication) {
     size_t length =
         _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
+    RTC_CHECK_LE(length, sizeof(_audioFrame.data_));
     if (is_muted && previous_frame_muted_) {
       rms_level_.ProcessMuted(length);
     } else {
       rms_level_.Process(_audioFrame.data_, length);
     }
   }
   previous_frame_muted_ = is_muted;
 
   return 0;
 }
@@ skipping 553 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc