Refactor RMSLevel and give it new functionality

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 2748 matching lines @@
     }
   }
 
   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);
+      rms_level_.Process(
+          rtc::ArrayView<const int16_t>(_audioFrame.data_, length));

[peah-webrtc, 2016/11/29 08:57:13]

You should probably add the include array_view.h. Now it is somehow implictly
included via another include as it is neither in channel.cc or channel.h.

[hlundin-webrtc, 2016/11/29 10:24:55]

Done.

     }
   }
   previous_frame_muted_ = is_muted;
 
   return 0;
 }
 
 uint32_t Channel::EncodeAndSend() {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::EncodeAndSend()");
@@ skipping 448 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