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.
  */
 
 #include "webrtc/voice_engine/channel.h"
 
 #include <algorithm>
 #include <utility>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
@@ skipping 331 matching lines @@
                           const RTPFragmentationHeader* fragmentation) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
                " payloadSize=%" PRIuS ", fragmentation=0x%x)",
                frameType, payloadType, timeStamp, payloadSize, fragmentation);
 
   if (_includeAudioLevelIndication) {
     // Store current audio level in the RTP/RTCP module.
     // The level will be used in combination with voice-activity state
     // (frameType) to add an RTP header extension
-    _rtpRtcpModule->SetAudioLevel(rms_level_.RMS());
+    _rtpRtcpModule->SetAudioLevel(rms_level_.Average());
   }
 
   // Push data from ACM to RTP/RTCP-module to deliver audio frame for
   // packetization.
   // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
   if (!_rtpRtcpModule->SendOutgoingData(
           (FrameType&)frameType, payloadType, timeStamp,
           // Leaving the time when this frame was
           // received from the capture device as
           // undefined for voice for now.
@@ skipping 2379 matching lines @@
           _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);
+      rms_level_.AnalyzeMuted(length);
     } else {
-      rms_level_.Process(_audioFrame.data_, length);
+      rms_level_.Analyze(
+          rtc::ArrayView<const int16_t>(_audioFrame.data_, length));
     }
   }
   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