diff --git a/tools_webrtc/ubsan/blacklist.txt b/tools_webrtc/ubsan/blacklist.txt
index a65b160..3f4c80e 100644
--- a/tools_webrtc/ubsan/blacklist.txt
+++ b/tools_webrtc/ubsan/blacklist.txt
@@ -1,9 +1,9 @@
 #############################################################################
 # UBSan blacklist.
-# Please think twice before you add or remove these rules.
-
-# This is a stripped down copy of Chromium's blacklist.txt, to enable
-# adding WebRTC-specific blacklist entries.
+#
+# This is a WebRTC-specific replacement of Chromium's blacklist.txt.
+# Only exceptions for third party libraries go here. WebRTC's code should use
+# the RTC_NO_SANITIZE macro. Please think twice before adding new exceptions.
 
 #############################################################################
 # YASM does some funny things that UBsan doesn't like.
@@ -17,22 +17,3 @@
 #############################################################################
 # Ignore system libraries.
 src:*/usr/*
-
-#############################################################################
-# https://bugs.chromium.org/p/webrtc/issues/detail?id=5513
-fun:*FilterBanksTest*CalculateResidualEnergyTester*
-
-#############################################################################
-# https://bugs.chromium.org/p/webrtc/issues/detail?id=8200
-fun:*WebRtcAecm_ProcessBlock*
-
-#############################################################################
-# Ignore errors in common_audio.
-# https://bugs.chromium.org/p/webrtc/issues/detail?id=5486
-src:*/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
-src:*/webrtc/common_audio/signal_processing/resample_by_2_internal.c
-fun:*WebRtcSpl_AddSatW32*
-fun:*WebRtcSpl_SubSatW32*
-fun:*WebRtcSpl_DivW32HiLow*
-fun:*WebRtcSpl_LevinsonDurbin*
-fun:*GmmProbability*
diff --git a/webrtc/common_audio/signal_processing/division_operations.c b/webrtc/common_audio/signal_processing/division_operations.c
index eaa06a1..2d42052 100644
--- a/webrtc/common_audio/signal_processing/division_operations.c
+++ b/webrtc/common_audio/signal_processing/division_operations.c
@@ -22,6 +22,7 @@
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/rtc_base/sanitizer.h"
 
 uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den)
 {
@@ -97,7 +98,8 @@
     return div;
 }
 
-int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
+int32_t RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5486
+WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
 {
     int16_t approx, tmp_hi, tmp_low, num_hi, num_low;
     int32_t tmpW32;
@@ -110,6 +112,7 @@
     // tmpW32 = den * approx
 
     tmpW32 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
+    // UBSan: 2147483647 - -2 cannot be represented in type 'int'
 
     // Store tmpW32 in hi and low format
     tmp_hi = (int16_t)(tmpW32 >> 16);
diff --git a/webrtc/common_audio/signal_processing/levinson_durbin.c b/webrtc/common_audio/signal_processing/levinson_durbin.c
index c8e58ba..3e16b23 100644
--- a/webrtc/common_audio/signal_processing/levinson_durbin.c
+++ b/webrtc/common_audio/signal_processing/levinson_durbin.c
@@ -16,11 +16,13 @@
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/rtc_base/sanitizer.h"
 
 #define SPL_LEVINSON_MAXORDER 20
 
-int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
-                                 size_t order)
+int16_t RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5486
+WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
+                         size_t order)
 {
     size_t i, j;
     // Auto-correlation coefficients in high precision
@@ -44,6 +46,8 @@
     for (i = 0; i <= order; ++i)
     {
         temp1W32 = R[i] * (1 << norm);
+        // UBSan: 12 * 268435456 cannot be represented in type 'int'
+
         // Put R in hi and low format
         R_hi[i] = (int16_t)(temp1W32 >> 16);
         R_low[i] = (int16_t)((temp1W32 - ((int32_t)R_hi[i] * 65536)) >> 1);
diff --git a/webrtc/common_audio/signal_processing/resample_by_2_internal.c b/webrtc/common_audio/signal_processing/resample_by_2_internal.c
index 085069c..72bc0f9 100644
--- a/webrtc/common_audio/signal_processing/resample_by_2_internal.c
+++ b/webrtc/common_audio/signal_processing/resample_by_2_internal.c
@@ -15,6 +15,7 @@
  */
 
 #include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
+#include "webrtc/rtc_base/sanitizer.h"
 
 // allpass filter coefficients.
 static const int16_t kResampleAllpass[2][3] = {
@@ -28,8 +29,9 @@
 // output: int16_t (saturated) (of length len/2)
 // state:  filter state array; length = 8
 
-void WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
-                                 int32_t *state)
+void RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5486
+WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
+                            int32_t *state)
 {
     int32_t tmp0, tmp1, diff;
     int32_t i;
@@ -41,6 +43,8 @@
     {
         tmp0 = in[i << 1];
         diff = tmp0 - state[1];
+        // UBSan: -1771017321 - 999586185 cannot be represented in type 'int'
+
         // scale down and round
         diff = (diff + (1 << 13)) >> 14;
         tmp1 = state[0] + diff * kResampleAllpass[1][0];
@@ -121,10 +125,11 @@
 // output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len/2)
 // state:  filter state array; length = 8
 
-void WebRtcSpl_DownBy2ShortToInt(const int16_t *in,
-                                  int32_t len,
-                                  int32_t *out,
-                                  int32_t *state)
+void RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5486
+WebRtcSpl_DownBy2ShortToInt(const int16_t *in,
+                            int32_t len,
+                            int32_t *out,
+                            int32_t *state)
 {
     int32_t tmp0, tmp1, diff;
     int32_t i;
@@ -141,6 +146,8 @@
         tmp1 = state[0] + diff * kResampleAllpass[1][0];
         state[0] = tmp0;
         diff = tmp1 - state[2];
+        // UBSan: -1379909682 - 834099714 cannot be represented in type 'int'
+
         // scale down and truncate
         diff = diff >> 14;
         if (diff < 0)
@@ -549,8 +556,9 @@
 // input:  int32_t (shifted 15 positions to the left, + offset 16384)
 // output: int32_t (normalized, not saturated)
 // state:  filter state array; length = 8
-void WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
-                             int32_t* state)
+void RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5486
+WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
+                        int32_t* state)
 {
     int32_t tmp0, tmp1, diff;
     int32_t i;
@@ -594,6 +602,8 @@
     {
         tmp0 = in[i << 1];
         diff = tmp0 - state[5];
+        // UBSan: -794814117 - 1566149201 cannot be represented in type 'int'
+
         // scale down and round
         diff = (diff + (1 << 13)) >> 14;
         tmp1 = state[4] + diff * kResampleAllpass[0][0];
diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index da6197e..52c4390 100644
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -41,7 +41,6 @@
     EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
 
     EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
-    EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
     EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
     b = WEBRTC_SPL_WORD16_MAX >> 1;
     EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
index ce96e2c..c8e6ac8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
@@ -12,6 +12,7 @@
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "webrtc/rtc_base/sanitizer.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
@@ -19,8 +20,9 @@
 class FilterBanksTest : public testing::Test {
  protected:
   // Pass a function pointer to the Tester function.
-  void CalculateResidualEnergyTester(AllpassFilter2FixDec16
-                                     AllpassFilter2FixDec16Function) {
+  void RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5513
+  CalculateResidualEnergyTester(AllpassFilter2FixDec16
+                                AllpassFilter2FixDec16Function) {
     const int kSamples = QLOOKAHEAD;
     const int kState = 2;
     int16_t data_ch1[kSamples] = {0};
@@ -41,6 +43,7 @@
       sign *= -1;
       data_ch1[i] = sign * WEBRTC_SPL_WORD32_MAX / (i * i + 1);
       data_ch2[i] = sign * WEBRTC_SPL_WORD32_MIN / (i * i + 1);
+      // UBSan: -1 * -2147483648 cannot be represented in type 'int'
     };
 
     AllpassFilter2FixDec16Function(data_ch1,
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_c.cc b/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
index 32f2656..bf26277 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
@@ -24,6 +24,7 @@
 }
 
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/sanitizer.h"
 #include "webrtc/typedefs.h"
 
 // Square root of Hanning window in Q14.
@@ -276,11 +277,12 @@
   return time_signal_scaling;
 }
 
-int WebRtcAecm_ProcessBlock(AecmCore* aecm,
-                            const int16_t* farend,
-                            const int16_t* nearendNoisy,
-                            const int16_t* nearendClean,
-                            int16_t* output) {
+int RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/8200
+WebRtcAecm_ProcessBlock(AecmCore* aecm,
+                        const int16_t* farend,
+                        const int16_t* nearendNoisy,
+                        const int16_t* nearendClean,
+                        int16_t* output) {
   int i;
 
   uint32_t xfaSum;
@@ -453,6 +455,7 @@
     // How much can we shift right to preserve resolution
     tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
     aecm->echoFilt[i] += (tmp32no1 * 50) >> 8;
+    // UBSan: 72293096 * 50 cannot be represented in type 'int'
 
     zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
     zeros16 = WebRtcSpl_NormW16(supGain) + 1;
diff --git a/webrtc/rtc_base/ipaddress_unittest.cc b/webrtc/rtc_base/ipaddress_unittest.cc
index 3389fef..61a067f 100644
--- a/webrtc/rtc_base/ipaddress_unittest.cc
+++ b/webrtc/rtc_base/ipaddress_unittest.cc
@@ -672,13 +672,7 @@
   EXPECT_EQ(addr, addr2);
 }
 
-// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5491
-#ifdef UNDEFINED_SANITIZER
-#define MAYBE_TestCountIPMaskBits DISABLED_TestCountIPMaskBits
-#else
-#define MAYBE_TestCountIPMaskBits TestCountIPMaskBits
-#endif
-TEST(IPAddressTest, MAYBE_TestCountIPMaskBits) {
+TEST(IPAddressTest, TestCountIPMaskBits) {
   IPAddress mask;
   // IPv4 on byte boundaries
   EXPECT_PRED2(CheckMaskCount, "255.255.255.255", 32);
diff --git a/webrtc/rtc_base/random.cc b/webrtc/rtc_base/random.cc
index c0cf7fe..f6bfc08 100644
--- a/webrtc/rtc_base/random.cc
+++ b/webrtc/rtc_base/random.cc
@@ -12,6 +12,7 @@
 #include <math.h>
 
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/safe_conversions.h"
 
 namespace webrtc {
 
@@ -40,11 +41,9 @@
 
 int32_t Random::Rand(int32_t low, int32_t high) {
   RTC_DCHECK(low <= high);
-  // We rely on subtraction (and addition) to be the same for signed and
-  // unsigned numbers in two-complement representation. Thus, although
-  // high - low might be negative as an int, it is the correct difference
-  // when interpreted as an unsigned.
-  return Rand(high - low) + low;
+  const int64_t low_i64{low};
+  return rtc::dchecked_cast<int32_t>(
+      Rand(rtc::dchecked_cast<uint32_t>(high - low_i64)) + low_i64);
 }
 
 template <>
diff --git a/webrtc/rtc_base/random_unittest.cc b/webrtc/rtc_base/random_unittest.cc
index 0af2733..12d17ad 100644
--- a/webrtc/rtc_base/random_unittest.cc
+++ b/webrtc/rtc_base/random_unittest.cc
@@ -199,13 +199,7 @@
   BucketTestUnsignedInterval(1000, 1000000, 0, 2147483999, 4, &prng);
 }
 
-// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5491
-#ifdef UNDEFINED_SANITIZER
-#define MAYBE_UniformSignedInterval DISABLED_UniformSignedInterval
-#else
-#define MAYBE_UniformSignedInterval UniformSignedInterval
-#endif
-TEST(RandomNumberGeneratorTest, MAYBE_UniformSignedInterval) {
+TEST(RandomNumberGeneratorTest, UniformSignedInterval) {
   Random prng(66260695729ull);
   BucketTestSignedInterval(2, 100000, 0, 1, 3, &prng);
   BucketTestSignedInterval(7, 100000, -2, 4, 3, &prng);
diff --git a/webrtc/rtc_base/sanitizer.h b/webrtc/rtc_base/sanitizer.h
index 49dc670..73099ac 100644
--- a/webrtc/rtc_base/sanitizer.h
+++ b/webrtc/rtc_base/sanitizer.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_RTC_BASE_SANITIZER_H_
 #define WEBRTC_RTC_BASE_SANITIZER_H_
 
+#include <stddef.h>  // for size_t
+
 #if defined(__has_feature)
 #if __has_feature(address_sanitizer)
 #define RTC_HAS_ASAN 1