Convert usage of ARRAY_SIZE to arraysize.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 25 matching lines @@
 #include <algorithm>
 #include <cstdio>
 #include <string>
 #include <vector>
 
 #include "talk/media/base/audioframe.h"
 #include "talk/media/base/audiorenderer.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/streamparams.h"
 #include "talk/media/webrtc/webrtcvoe.h"
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ skipping 149 matching lines @@
 
 bool IsCodec(const AudioCodec& codec, const char* ref_name) {
   return (_stricmp(codec.name.c_str(), ref_name) == 0);
 }
 
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
   return (_stricmp(codec.plname, ref_name) == 0);
 }
 
 bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
-  for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
+  for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
     if (IsCodec(codec, kCodecPrefs[i].name) &&
         kCodecPrefs[i].clockrate == codec.plfreq) {
       return kCodecPrefs[i].is_multi_rate;
     }
   }
   return false;
 }
 
 bool FindCodec(const std::vector<AudioCodec>& codecs,
                       const AudioCodec& codec,
@@ skipping 220 matching lines @@
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
   for (int i = 0; i < ncodecs; ++i) {
     webrtc::CodecInst voe_codec;
     if (GetVoeCodec(i, &voe_codec)) {
       // Skip uncompressed formats.
       if (IsCodec(voe_codec, kL16CodecName)) {
         continue;
       }
 
       const CodecPref* pref = NULL;
-      for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
+      for (size_t j = 0; j < arraysize(kCodecPrefs); ++j) {
         if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
             kCodecPrefs[j].clockrate == voe_codec.plfreq &&
             kCodecPrefs[j].channels == voe_codec.channels) {
           pref = &kCodecPrefs[j];
           break;
         }
       }
 
       if (pref) {
         // Use the payload type that we've configured in our pref table;
         // use the offset in our pref table to determine the sort order.
         AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
                          voe_codec.rate, voe_codec.channels,
-                         ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
+                         (int)arraysize(kCodecPrefs) - (pref - kCodecPrefs));

[tommi (sloooow) - chröme, 2015/11/06 22:33:25]

use static_cast

[tfarina, 2015/11/10 23:02:45]

Done.

         LOG(LS_INFO) << ToString(codec);
         if (IsCodec(codec, kIsacCodecName)) {
           // Indicate auto-bitrate in signaling.
           codec.bitrate = 0;
         }
         if (IsCodec(codec, kOpusCodecName)) {
           // Only add fmtp parameters that differ from the spec.
           if (kPreferredMinPTime != kOpusDefaultMinPTime) {
             codec.params[kCodecParamMinPTime] =
                 rtc::ToString(kPreferredMinPTime);
@@ skipping 2406 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE