From 0ab42bc3f6438db4194b3d77b66629413c7038da Mon Sep 17 00:00:00 2001
From: "andrew@webrtc.org" <andrew@webrtc.org>
Date: Wed, 17 Dec 2014 22:56:09 +0000
Subject: [PATCH] Make safe_conversions suitable for rtc_base_approved.

Since we want to use checked_cast in WavReader.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7937 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 webrtc/base/BUILD.gn            |  4 ++--
 webrtc/base/base.gyp            |  4 ++--
 webrtc/base/safe_conversions.h  | 17 ++++-------------
 webrtc/common_audio/wav_file.cc |  7 +++----
 4 files changed, 11 insertions(+), 21 deletions(-)

diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index acce9f0401..ef02817e55 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -113,6 +113,8 @@ static_library("rtc_base_approved") {
     "md5digest.h",
     "platform_file.cc",
     "platform_file.h",
+    "safe_conversions.h",
+    "safe_conversions_impl.h",
     "stringencode.cc",
     "stringencode.h",
     "stringutils.cc",
@@ -225,8 +227,6 @@ static_library("webrtc_base") {
     "ratelimiter.h",
     "ratetracker.cc",
     "ratetracker.h",
-    "safe_conversions.h",
-    "safe_conversions_impl.h",
     "scoped_autorelease_pool.h",
     "scoped_autorelease_pool.mm",
     "scoped_ptr.h",
diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp
index 380eee6ed5..645c1dcf4b 100644
--- a/webrtc/base/base.gyp
+++ b/webrtc/base/base.gyp
@@ -48,6 +48,8 @@
         'md5digest.h',
         'platform_file.cc',
         'platform_file.h',
+        'safe_conversions.h',
+        'safe_conversions_impl.h',
         'stringencode.cc',
         'stringencode.h',
         'stringutils.cc',
@@ -219,8 +221,6 @@
         'refcount.h',
         'referencecountedsingletonfactory.h',
         'rollingaccumulator.h',
-        'safe_conversions.h',
-        'safe_conversions_impl.h',
         'schanneladapter.cc',
         'schanneladapter.h',
         'scoped_autorelease_pool.h',
diff --git a/webrtc/base/safe_conversions.h b/webrtc/base/safe_conversions.h
index f6cb24e412..7fc67cb67a 100644
--- a/webrtc/base/safe_conversions.h
+++ b/webrtc/base/safe_conversions.h
@@ -15,20 +15,11 @@
 
 #include <limits>
 
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions_impl.h"
 
 namespace rtc {
 
-inline void Check(bool condition) {
-  if (!condition) {
-    LOG(LS_ERROR) << "CHECK failed.";
-    Break();
-    // The program should have crashed at this point.
-  }
-}
-
 // Convenience function that returns true if the supplied value is in range
 // for the destination type.
 template <typename Dst, typename Src>
@@ -41,7 +32,7 @@ inline bool IsValueInRangeForNumericType(Src value) {
 // overflow or underflow. NaN source will always trigger a CHECK.
 template <typename Dst, typename Src>
 inline Dst checked_cast(Src value) {
-  Check(IsValueInRangeForNumericType<Dst>(value));
+  CHECK(IsValueInRangeForNumericType<Dst>(value));
   return static_cast<Dst>(value);
 }
 
@@ -66,11 +57,11 @@ inline Dst saturated_cast(Src value) {
 
     // Should fail only on attempting to assign NaN to a saturated integer.
     case internal::TYPE_INVALID:
-      Check(false);
+      FATAL();
       return std::numeric_limits<Dst>::max();
   }
 
-  Check(false); // NOTREACHED();
+  FATAL();
   return static_cast<Dst>(value);
 }
 
diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
index cc96e41da3..de83b03380 100644
--- a/webrtc/common_audio/wav_file.cc
+++ b/webrtc/common_audio/wav_file.cc
@@ -15,6 +15,7 @@
 #include <limits>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/wav_header.h"
 
@@ -58,17 +59,15 @@ size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
 #error "Need to convert samples to big-endian when reading from WAV file"
 #endif
-  // TODO(ajm): Import Chromium's safe_conversions.h for this.
-  CHECK_LE(num_samples, std::numeric_limits<uint32_t>::max());
   // There could be metadata after the audio; ensure we don't read it.
-  num_samples = std::min(static_cast<uint32_t>(num_samples),
+  num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
                          num_samples_remaining_);
   const size_t read =
       fread(samples, sizeof(*samples), num_samples, file_handle_);
   // If we didn't read what was requested, ensure we've reached the EOF.
   CHECK(read == num_samples || feof(file_handle_));
   CHECK_LE(read, num_samples_remaining_);
-  num_samples_remaining_ -= static_cast<uint32_t>(read);
+  num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
   return read;
 }