Audit of kConstants missing the const qualifier

webrtc/config.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.
  */
 #include "webrtc/config.h"
@@ skipping 31 matching lines @@
   std::stringstream ss;
   ss << "{uri: " << uri;
   ss << ", id: " << id;
   if (encrypt) {
     ss << ", encrypt";
   }
   ss << '}';
   return ss.str();
 }
 
-const char* RtpExtension::kAudioLevelUri =
+const char RtpExtension::kAudioLevelUri[] =
     "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
 const int RtpExtension::kAudioLevelDefaultId = 1;
 
-const char* RtpExtension::kTimestampOffsetUri =
+const char RtpExtension::kTimestampOffsetUri[] =
     "urn:ietf:params:rtp-hdrext:toffset";
 const int RtpExtension::kTimestampOffsetDefaultId = 2;
 
-const char* RtpExtension::kAbsSendTimeUri =
+const char RtpExtension::kAbsSendTimeUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 const int RtpExtension::kAbsSendTimeDefaultId = 3;
 
-const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation";
+const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
 const int RtpExtension::kVideoRotationDefaultId = 4;
 
-const char* RtpExtension::kTransportSequenceNumberUri =
+const char RtpExtension::kTransportSequenceNumberUri[] =
     "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
 const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
 
 // This extension allows applications to adaptively limit the playout delay
 // on frames as per the current needs. For example, a gaming application
 // has very different needs on end-to-end delay compared to a video-conference
 // application.
-const char* RtpExtension::kPlayoutDelayUri =
+const char RtpExtension::kPlayoutDelayUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
 const int RtpExtension::kPlayoutDelayDefaultId = 6;
 
-const char* RtpExtension::kVideoContentTypeUri =
+const char RtpExtension::kVideoContentTypeUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
 const int RtpExtension::kVideoContentTypeDefaultId = 7;
 
-const char* RtpExtension::kVideoTimingUri =
+const char RtpExtension::kVideoTimingUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
 const int RtpExtension::kVideoTimingDefaultId = 8;
 
-const char* RtpExtension::kEncryptHeaderExtensionsUri =
+const char RtpExtension::kEncryptHeaderExtensionsUri[] =
     "urn:ietf:params:rtp-hdrext:encrypt";
 
 const int RtpExtension::kMinId = 1;
 const int RtpExtension::kMaxId = 14;
 
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }
 
@@ skipping 176 matching lines @@
 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
     const VideoCodecVP9& specifics)
     : specifics_(specifics) {}
 
 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
     VideoCodecVP9* vp9_settings) const {
   *vp9_settings = specifics_;
 }
 
 }  // namespace webrtc