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

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index c2b95420537f4763893d1d5cea4d188240e9d527..245398702e8b2fe9f5225c1333c2665aac2c1291 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2993,6 +2993,7 @@ uint32_t Channel::PrepareEncodeAndSend(int mixingFrequency) {
   if (_includeAudioLevelIndication) {
     size_t length =
         _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
+    RTC_CHECK(length <= sizeof(_audioFrame.data_));

[hlundin-webrtc, 2016/05/24 06:38:03]

RTC_CHECK_LE

[tommi, 2016/05/24 07:49:13]

Done.

     if (is_muted && previous_frame_muted_) {
       rms_level_.ProcessMuted(length);
     } else {