Audit of kConstants missing the const qualifier

webrtc/media/engine/webrtcmediaengine.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 139 matching lines @@
     rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
   return CreateWebRtcMediaEngine(
       adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
       video_decoder_factory, audio_mixer, audio_processing);
 }
 
 namespace {
 // Remove mutually exclusive extensions with lower priority.
 void DiscardRedundantExtensions(
     std::vector<webrtc::RtpExtension>* extensions,
-    rtc::ArrayView<const char*> extensions_decreasing_prio) {
+    rtc::ArrayView<const char* const> extensions_decreasing_prio) {
   RTC_DCHECK(extensions);
   bool found = false;
   for (const char* uri : extensions_decreasing_prio) {
     auto it = std::find_if(
         extensions->begin(), extensions->end(),
         [uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; });
     if (it != extensions->end()) {
       if (found) {
         extensions->erase(it);
       }
@@ skipping 50 matching lines @@
   // Remove unnecessary extensions (used on send side).
   if (filter_redundant_extensions) {
     auto it = std::unique(
         result.begin(), result.end(),
         [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
           return rhs.uri == lhs.uri && rhs.encrypt == lhs.encrypt;
         });
     result.erase(it, result.end());
 
     // Keep just the highest priority extension of any in the following list.
-    static const char* kBweExtensionPriorities[] = {
+    static const char* const kBweExtensionPriorities[] = {
         webrtc::RtpExtension::kTransportSequenceNumberUri,
         webrtc::RtpExtension::kAbsSendTimeUri,
         webrtc::RtpExtension::kTimestampOffsetUri};
     DiscardRedundantExtensions(&result, kBweExtensionPriorities);
   }
 
   return result;
 }
 
 webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
@@ skipping 15 matching lines @@
   }
   if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
       bitrate_kbps > 0) {
     config.max_bitrate_bps = bitrate_kbps * 1000;
   } else {
     config.max_bitrate_bps = -1;
   }
   return config;
 }
 }  // namespace cricket