diff --git a/BUILD.gn b/BUILD.gn
index e7d4aa9c84..1fcc310669 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -434,10 +434,6 @@ rtc_source_set("webrtc_common") {
     "api/video:video_bitrate_allocation",
     "rtc_base:checks",
     "rtc_base:deprecation",
-
-    # TODO(nisse): Delete both of these together with STR_CASE_CMP and
-    # STR_NCASE_CMP.
-    "rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/strings",
   ]
 
diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
index a7d56d0c72..34ec2e4208 100644
--- a/api/audio_codecs/L16/BUILD.gn
+++ b/api/audio_codecs/L16/BUILD.gn
@@ -21,12 +21,12 @@ rtc_static_library("audio_encoder_L16") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -40,11 +40,11 @@ rtc_static_library("audio_decoder_L16") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_codecs/L16/audio_decoder_L16.cc b/api/audio_codecs/L16/audio_decoder_L16.cc
index a71e308700..be0c6b56b2 100644
--- a/api/audio_codecs/L16/audio_decoder_L16.cc
+++ b/api/audio_codecs/L16/audio_decoder_L16.cc
@@ -11,7 +11,7 @@
 #include "api/audio_codecs/L16/audio_decoder_L16.h"
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -23,7 +23,7 @@ absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
   Config config;
   config.sample_rate_hz = format.clockrate_hz;
   config.num_channels = rtc::checked_cast<int>(format.num_channels);
-  return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
+  return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
              ? absl::optional<Config>(config)
              : absl::nullopt;
 }
diff --git a/api/audio_codecs/L16/audio_encoder_L16.cc b/api/audio_codecs/L16/audio_encoder_L16.cc
index b516f62863..1bb552bc5c 100644
--- a/api/audio_codecs/L16/audio_encoder_L16.cc
+++ b/api/audio_codecs/L16/audio_encoder_L16.cc
@@ -11,7 +11,7 @@
 #include "api/audio_codecs/L16/audio_encoder_L16.h"
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -35,7 +35,7 @@ absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
       config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
     }
   }
-  return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
+  return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
              ? absl::optional<Config>(config)
              : absl::nullopt;
 }
diff --git a/api/audio_codecs/audio_format.cc b/api/audio_codecs/audio_format.cc
index bd2d3f2b5a..11788b91e8 100644
--- a/api/audio_codecs/audio_format.cc
+++ b/api/audio_codecs/audio_format.cc
@@ -10,7 +10,7 @@
 
 #include "api/audio_codecs/audio_format.h"
 
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 
 namespace webrtc {
 
@@ -32,7 +32,7 @@ SdpAudioFormat::SdpAudioFormat(absl::string_view name,
       parameters(param) {}
 
 bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
-  return STR_CASE_CMP(name.c_str(), o.name.c_str()) == 0 &&
+  return absl::EqualsIgnoreCase(name, o.name) &&
          clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
 }
 
@@ -41,7 +41,7 @@ SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
 SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
 
 bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
-  return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
+  return absl::EqualsIgnoreCase(a.name, b.name) &&
          a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
          a.parameters == b.parameters;
 }
diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
index ebd0b3299f..3b8f23c1bc 100644
--- a/api/audio_codecs/g711/BUILD.gn
+++ b/api/audio_codecs/g711/BUILD.gn
@@ -21,12 +21,12 @@ rtc_static_library("audio_encoder_g711") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -40,11 +40,11 @@ rtc_static_library("audio_decoder_g711") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_codecs/g711/audio_decoder_g711.cc b/api/audio_codecs/g711/audio_decoder_g711.cc
index cb16584cde..91599c416a 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.cc
+++ b/api/audio_codecs/g711/audio_decoder_g711.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
@@ -22,8 +22,8 @@ namespace webrtc {
 
 absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
     const SdpAudioFormat& format) {
-  const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
-  const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
   if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
       (is_pcmu || is_pcma)) {
     Config config;
diff --git a/api/audio_codecs/g711/audio_encoder_g711.cc b/api/audio_codecs/g711/audio_encoder_g711.cc
index 1d5e541276..0cc8dc4ddd 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.cc
+++ b/api/audio_codecs/g711/audio_encoder_g711.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
@@ -24,8 +24,8 @@ namespace webrtc {
 
 absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
     const SdpAudioFormat& format) {
-  const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
-  const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
   if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
       (is_pcmu || is_pcma)) {
     Config config;
diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
index 101c7a994d..e4321d2993 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@@ -29,12 +29,12 @@ rtc_static_library("audio_encoder_g722") {
   deps = [
     ":audio_encoder_g722_config",
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -48,11 +48,11 @@ rtc_static_library("audio_decoder_g722") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_codecs/g722/audio_decoder_g722.cc b/api/audio_codecs/g722/audio_decoder_g722.cc
index f1e2afbab5..2cc16c37d9 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.cc
+++ b/api/audio_codecs/g722/audio_decoder_g722.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
@@ -22,7 +22,7 @@ namespace webrtc {
 
 absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
     const SdpAudioFormat& format) {
-  return STR_CASE_CMP(format.name.c_str(), "G722") == 0 &&
+  return absl::EqualsIgnoreCase(format.name, "G722") &&
                  format.clockrate_hz == 8000 &&
                  (format.num_channels == 1 || format.num_channels == 2)
              ? absl::optional<Config>(
diff --git a/api/audio_codecs/g722/audio_encoder_g722.cc b/api/audio_codecs/g722/audio_encoder_g722.cc
index 0cf71639bf..6374ae802f 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
@@ -24,7 +24,7 @@ namespace webrtc {
 
 absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
     const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
+  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
       format.clockrate_hz != 8000) {
     return absl::nullopt;
   }
diff --git a/api/audio_codecs/ilbc/BUILD.gn b/api/audio_codecs/ilbc/BUILD.gn
index d4f504fd4f..d766fa4a97 100644
--- a/api/audio_codecs/ilbc/BUILD.gn
+++ b/api/audio_codecs/ilbc/BUILD.gn
@@ -29,11 +29,11 @@ rtc_static_library("audio_encoder_ilbc") {
   deps = [
     ":audio_encoder_ilbc_config",
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -47,10 +47,10 @@ rtc_static_library("audio_decoder_ilbc") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
index 1f4c475c40..4a00f8dbf0 100644
--- a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
@@ -14,14 +14,14 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 
 namespace webrtc {
 
 absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
     const SdpAudioFormat& format) {
-  return STR_CASE_CMP(format.name.c_str(), "ILBC") == 0 &&
+  return absl::EqualsIgnoreCase(format.name, "ILBC") &&
                  format.clockrate_hz == 8000 && format.num_channels == 1
              ? absl::optional<Config>(Config())
              : absl::nullopt;
diff --git a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
index efcef382ff..896ed238cc 100644
--- a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
@@ -40,7 +40,7 @@ int GetIlbcBitrate(int ptime) {
 
 absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
     const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 ||
+  if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
       format.clockrate_hz != 8000 || format.num_channels != 1) {
     return absl::nullopt;
   }
diff --git a/api/audio_codecs/isac/BUILD.gn b/api/audio_codecs/isac/BUILD.gn
index 9025360f2f..c7d6e43271 100644
--- a/api/audio_codecs/isac/BUILD.gn
+++ b/api/audio_codecs/isac/BUILD.gn
@@ -77,10 +77,10 @@ rtc_static_library("audio_encoder_isac_fix") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -94,10 +94,10 @@ rtc_static_library("audio_decoder_isac_fix") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -111,11 +111,11 @@ rtc_static_library("audio_encoder_isac_float") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -129,11 +129,11 @@ rtc_static_library("audio_decoder_isac_float") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_codecs/isac/audio_decoder_isac_fix.cc b/api/audio_codecs/isac/audio_decoder_isac_fix.cc
index 446640ff70..51ae572fa9 100644
--- a/api/audio_codecs/isac/audio_decoder_isac_fix.cc
+++ b/api/audio_codecs/isac/audio_decoder_isac_fix.cc
@@ -11,14 +11,14 @@
 #include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
 
 namespace webrtc {
 
 absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
     const SdpAudioFormat& format) {
-  return STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+  return absl::EqualsIgnoreCase(format.name, "ISAC") &&
                  format.clockrate_hz == 16000 && format.num_channels == 1
              ? absl::optional<Config>(Config())
              : absl::nullopt;
diff --git a/api/audio_codecs/isac/audio_decoder_isac_float.cc b/api/audio_codecs/isac/audio_decoder_isac_float.cc
index 1c1926f1a1..d9de3a08b3 100644
--- a/api/audio_codecs/isac/audio_decoder_isac_float.cc
+++ b/api/audio_codecs/isac/audio_decoder_isac_float.cc
@@ -11,14 +11,14 @@
 #include "api/audio_codecs/isac/audio_decoder_isac_float.h"
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
 
 namespace webrtc {
 
 absl::optional<AudioDecoderIsacFloat::Config>
 AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
       (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
       format.num_channels == 1) {
     Config config;
diff --git a/api/audio_codecs/isac/audio_encoder_isac_fix.cc b/api/audio_codecs/isac/audio_encoder_isac_fix.cc
index cd8975398b..a10d1ee61a 100644
--- a/api/audio_codecs/isac/audio_encoder_isac_fix.cc
+++ b/api/audio_codecs/isac/audio_encoder_isac_fix.cc
@@ -11,7 +11,7 @@
 #include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
 #include "rtc_base/string_to_number.h"
 
@@ -19,7 +19,7 @@ namespace webrtc {
 
 absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
     const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
       format.clockrate_hz == 16000 && format.num_channels == 1) {
     Config config;
     const auto ptime_iter = format.parameters.find("ptime");
diff --git a/api/audio_codecs/isac/audio_encoder_isac_float.cc b/api/audio_codecs/isac/audio_encoder_isac_float.cc
index 83d1faff2f..37982b1f15 100644
--- a/api/audio_codecs/isac/audio_encoder_isac_float.cc
+++ b/api/audio_codecs/isac/audio_encoder_isac_float.cc
@@ -11,7 +11,7 @@
 #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "rtc_base/string_to_number.h"
 
@@ -19,7 +19,7 @@ namespace webrtc {
 
 absl::optional<AudioEncoderIsacFloat::Config>
 AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
       (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
       format.num_channels == 1) {
     Config config;
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index af3cd7fc44..5552c21fa5 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -46,6 +46,7 @@ rtc_source_set("audio_encoder_opus") {
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -59,11 +60,11 @@ rtc_static_library("audio_decoder_opus") {
   ]
   deps = [
     "..:audio_codecs_api",
-    "../../..:webrtc_common",
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_codecs/opus/audio_decoder_opus.cc b/api/audio_codecs/opus/audio_decoder_opus.cc
index 41397f04d2..2f1668b825 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.cc
+++ b/api/audio_codecs/opus/audio_decoder_opus.cc
@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 
 namespace webrtc {
@@ -35,7 +35,7 @@ absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
     }
     return 1;  // Default to mono.
   }();
-  if (STR_CASE_CMP(format.name.c_str(), "opus") == 0 &&
+  if (absl::EqualsIgnoreCase(format.name, "opus") &&
       format.clockrate_hz == 48000 && format.num_channels == 2 &&
       num_channels) {
     return Config{*num_channels};
diff --git a/common_types.h b/common_types.h
index 37e6e2522a..30a2b3a93e 100644
--- a/common_types.h
+++ b/common_types.h
@@ -15,7 +15,6 @@
 #include <string.h>
 #include <string>
 #include <vector>
-// TODO(nisse): Delete include together with STR_CASE_CMP
 #include "absl/strings/match.h"
 #include "api/array_view.h"
 // TODO(sprang): Remove this include when all usage includes it directly.
@@ -31,18 +30,6 @@
 
 #define RTP_PAYLOAD_NAME_SIZE 32u
 
-// Compares two strings without regard to case.
-// TODO(nisse): Delete, implementation using EqualsIgnoreCase is misleading
-// since this can't be used for sorting.
-#define STR_CASE_CMP(s1, s2) (absl::EqualsIgnoreCase(s1, s2) ? 0 : 1)
-
-#if defined(WEBRTC_WIN) || defined(WIN32)
-// Compares characters of two strings without regard to case.
-#define STR_NCASE_CMP(s1, s2, n) ::_strnicmp(s1, s2, n)
-#else
-#define STR_NCASE_CMP(s1, s2, n) ::strncasecmp(s1, s2, n)
-#endif
-
 namespace webrtc {
 
 enum FrameType {
@@ -210,7 +197,7 @@ struct CodecInst {
 
   bool operator==(const CodecInst& other) const {
     return pltype == other.pltype &&
-           (STR_CASE_CMP(plname, other.plname) == 0) &&
+           absl::EqualsIgnoreCase(plname, other.plname) &&
            plfreq == other.plfreq && pacsize == other.pacsize &&
            channels == other.channels && rate == other.rate;
   }
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index baad232a87..03cc543947 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -968,8 +968,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     const bool is_opus =
         config_.send_codec_spec &&
-        !STR_CASE_CMP(config_.send_codec_spec->format.name.c_str(),
-                      kOpusCodecName);
+        absl::EqualsIgnoreCase(config_.send_codec_spec->format.name,
+                               kOpusCodecName);
     if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
       config_.min_bitrate_bps = kOpusMinBitrateBps;
 
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index 095d8b240b..eefa6f0822 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -11,6 +11,7 @@
 #include <memory>
 #include <utility>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/rtpparameters.h"
@@ -3467,7 +3468,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
   engine.Init();
   for (const cricket::AudioCodec& codec : engine.send_codecs()) {
     auto is_codec = [&codec](const char* name, int clockrate = 0) {
-      return STR_CASE_CMP(codec.name.c_str(), name) == 0 &&
+      return absl::EqualsIgnoreCase(codec.name, name) &&
              (clockrate == 0 || codec.clockrate == clockrate);
     };
     if (is_codec("CN", 16000)) {
@@ -3616,7 +3617,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
   auto find_codec = [&codecs](const webrtc::SdpAudioFormat& format) -> int {
     for (size_t i = 0; i != codecs.size(); ++i) {
       const cricket::AudioCodec& codec = codecs[i];
-      if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 &&
+      if (absl::EqualsIgnoreCase(codec.name, format.name) &&
           codec.clockrate == format.clockrate_hz &&
           codec.channels == format.num_channels) {
         return rtc::checked_cast<int>(i);
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 2ba551f511..01a0e939d5 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -53,6 +53,7 @@ rtc_static_library("audio_format_conversion") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:sanitizer",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -74,6 +75,7 @@ rtc_static_library("rent_a_codec") {
   deps = [
            "../../rtc_base:checks",
            "../../api:array_view",
+           "//third_party/abseil-cpp/absl/strings",
            "//third_party/abseil-cpp/absl/types:optional",
            "../../api/audio_codecs:audio_codecs_api",
            "../..:webrtc_common",
@@ -136,6 +138,7 @@ rtc_static_library("audio_coding") {
            ":rent_a_codec",
            "../../rtc_base:audio_format_to_string",
            "../../rtc_base:rtc_base_approved",
+           "//third_party/abseil-cpp/absl/strings",
            "//third_party/abseil-cpp/absl/types:optional",
            "../../logging:rtc_event_log_api",
          ]
@@ -829,6 +832,7 @@ rtc_static_library("webrtc_opus") {
     "../../rtc_base:safe_minmax",
     "../../system_wrappers:field_trial",
     "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
   public_deps = [
@@ -1068,6 +1072,7 @@ rtc_static_library("neteq") {
     "../../rtc_base/system:fallthrough",
     "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -1362,6 +1367,7 @@ if (rtc_include_tests) {
       "../../system_wrappers",
       "../../test:fileutils",
       "../../test:test_support",
+      "//third_party/abseil-cpp/absl/strings",
       "//third_party/abseil-cpp/absl/types:optional",
     ]
     defines = audio_coding_defines
diff --git a/modules/audio_coding/acm2/acm_codec_database.cc b/modules/audio_coding/acm2/acm_codec_database.cc
index 311af2df3f..e109cc7e02 100644
--- a/modules/audio_coding/acm2/acm_codec_database.cc
+++ b/modules/audio_coding/acm2/acm_codec_database.cc
@@ -17,6 +17,7 @@
 // references, where appropriate.
 #include "modules/audio_coding/acm2/acm_codec_database.h"
 
+#include "absl/strings/match.h"
 #include "rtc_base/checks.h"
 
 #if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
@@ -239,12 +240,12 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
   }
 
   // Comfort Noise is special case, packet-size & rate is not checked.
-  if (STR_CASE_CMP(database_[codec_id].plname, "CN") == 0) {
+  if (absl::EqualsIgnoreCase(database_[codec_id].plname, "CN")) {
     return codec_id;
   }
 
   // RED is special case, packet-size & rate is not checked.
-  if (STR_CASE_CMP(database_[codec_id].plname, "red") == 0) {
+  if (absl::EqualsIgnoreCase(database_[codec_id].plname, "red")) {
     return codec_id;
   }
 
@@ -272,12 +273,12 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
 
   // Check the validity of rate. Codecs with multiple rates have their own
   // function for this.
-  if (STR_CASE_CMP("isac", codec_inst.plname) == 0) {
+  if (absl::EqualsIgnoreCase("isac", codec_inst.plname)) {
     return IsISACRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
-  } else if (STR_CASE_CMP("ilbc", codec_inst.plname) == 0) {
+  } else if (absl::EqualsIgnoreCase("ilbc", codec_inst.plname)) {
     return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize) ? codec_id
                                                                 : kInvalidRate;
-  } else if (STR_CASE_CMP("opus", codec_inst.plname) == 0) {
+  } else if (absl::EqualsIgnoreCase("opus", codec_inst.plname)) {
     return IsOpusRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
   }
 
@@ -304,10 +305,10 @@ int ACMCodecDB::CodecId(const char* payload_name,
     // Payload name, sampling frequency and number of channels need to match.
     // NOTE! If |frequency| is -1, the frequency is not applicable, and is
     // always treated as true, like for RED.
-    name_match = (STR_CASE_CMP(ci.plname, payload_name) == 0);
+    name_match = absl::EqualsIgnoreCase(ci.plname, payload_name);
     frequency_match = (frequency == ci.plfreq) || (frequency == -1);
     // The number of channels must match for all codecs but Opus.
-    if (STR_CASE_CMP(payload_name, "opus") != 0) {
+    if (!absl::EqualsIgnoreCase(payload_name, "opus")) {
       channels_match = (channels == ci.channels);
     } else {
       // For opus we just check that number of channels is valid.
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 2d7f2960de..c149ec1b0b 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -14,6 +14,7 @@
 
 #include <memory>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
@@ -30,11 +31,11 @@ namespace {
 // Returns true if the codec should be registered, otherwise false. Changes
 // the number of channels for the Opus codec to always be 1.
 bool ModifyAndUseThisCodec(CodecInst* codec_param) {
-  if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
+  if (absl::EqualsIgnoreCase(codec_param->plname, "CN") &&
       codec_param->plfreq == 48000)
     return false;  // Skip 48 kHz comfort noise.
 
-  if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
+  if (absl::EqualsIgnoreCase(codec_param->plname, "telephone-event"))
     return false;  // Skip DTFM.
 
   return true;
@@ -65,39 +66,43 @@ bool RemapPltypeAndUseThisCodec(const char* plname,
     return false;  // Don't use non-mono codecs.
 
   // Re-map pltypes to those used in the NetEq test files.
-  if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
+  if (absl::EqualsIgnoreCase(plname, "PCMU") && plfreq == 8000) {
     *pltype = 0;
-  } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
+  } else if (absl::EqualsIgnoreCase(plname, "PCMA") && plfreq == 8000) {
     *pltype = 8;
-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
+  } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 8000) {
     *pltype = 13;
-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
+  } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 16000) {
     *pltype = 98;
-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
+  } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 32000) {
     *pltype = 99;
-  } else if (STR_CASE_CMP(plname, "ILBC") == 0) {
+  } else if (absl::EqualsIgnoreCase(plname, "ILBC")) {
     *pltype = 102;
-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
+  } else if (absl::EqualsIgnoreCase(plname, "ISAC") && plfreq == 16000) {
     *pltype = 103;
-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
+  } else if (absl::EqualsIgnoreCase(plname, "ISAC") && plfreq == 32000) {
     *pltype = 104;
-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 8000) {
+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
+             plfreq == 8000) {
     *pltype = 106;
-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 16000) {
+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
+             plfreq == 16000) {
     *pltype = 114;
-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 32000) {
+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
+             plfreq == 32000) {
     *pltype = 115;
-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 48000) {
+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
+             plfreq == 48000) {
     *pltype = 116;
-  } else if (STR_CASE_CMP(plname, "red") == 0) {
+  } else if (absl::EqualsIgnoreCase(plname, "red")) {
     *pltype = 117;
-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
+  } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 8000) {
     *pltype = 93;
-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
+  } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 16000) {
     *pltype = 94;
-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
+  } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 32000) {
     *pltype = 95;
-  } else if (STR_CASE_CMP(plname, "G722") == 0) {
+  } else if (absl::EqualsIgnoreCase(plname, "G722")) {
     *pltype = 9;
   } else {
     // Don't use any other codecs.
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index f631746b88..93fa27cb8a 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -15,9 +15,9 @@
 #include <algorithm>  // sort
 #include <vector>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/acm2/acm_resampler.h"
 #include "modules/audio_coding/acm2/call_statistics.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
@@ -92,7 +92,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
     }
     receive_timestamp = NowInTimestamp(ci->plfreq);
 
-    if (STR_CASE_CMP(ci->plname, "cn") == 0) {
+    if (absl::EqualsIgnoreCase(ci->plname, "cn")) {
       if (last_audio_decoder_ && last_audio_decoder_->channels > 1) {
         // This is a CNG and the audio codec is not mono, so skip pushing in
         // packets into NetEq.
@@ -391,7 +391,7 @@ const absl::optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
     uint8_t first_payload_byte) const {
   const absl::optional<CodecInst> ci =
       neteq_->GetDecoder(rtp_header.payloadType);
-  if (ci && STR_CASE_CMP(ci->plname, "red") == 0) {
+  if (ci && absl::EqualsIgnoreCase(ci->plname, "red")) {
     // This is a RED packet. Get the payload of the audio codec.
     return neteq_->GetDecoder(first_payload_byte & 0x7f);
   } else {
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 60afeb69bd..d6752038da 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -12,6 +12,7 @@
 
 #include <algorithm>
 
+#include "absl/strings/match.h"
 #include "modules/audio_coding/acm2/acm_receiver.h"
 #include "modules/audio_coding/acm2/acm_resampler.h"
 #include "modules/audio_coding/acm2/codec_manager.h"
@@ -990,7 +991,7 @@ int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
   }
 
   AudioDecoder* isac_decoder = nullptr;
-  if (STR_CASE_CMP(codec.plname, "isac") == 0) {
+  if (absl::EqualsIgnoreCase(codec.plname, "isac")) {
     std::unique_ptr<AudioDecoder>& saved_isac_decoder =
         codec.plfreq == 16000 ? isac_decoder_16k_ : isac_decoder_32k_;
     if (!saved_isac_decoder) {
diff --git a/modules/audio_coding/acm2/codec_manager.cc b/modules/audio_coding/acm2/codec_manager.cc
index f29e0f1a1e..5094e21f07 100644
--- a/modules/audio_coding/acm2/codec_manager.cc
+++ b/modules/audio_coding/acm2/codec_manager.cc
@@ -10,6 +10,7 @@
 
 #include "modules/audio_coding/acm2/codec_manager.h"
 
+#include "absl/strings/match.h"
 #include "rtc_base/checks.h"
 //#include "rtc_base/format_macros.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
@@ -35,7 +36,7 @@ int IsValidSendCodec(const CodecInst& send_codec) {
   }
 
   // Telephone-event cannot be a send codec.
-  if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
+  if (absl::EqualsIgnoreCase(send_codec.plname, "telephone-event")) {
     RTC_LOG(LS_ERROR) << "telephone-event cannot be a send codec";
     return -1;
   }
@@ -53,7 +54,7 @@ int IsValidSendCodec(const CodecInst& send_codec) {
 bool IsOpus(const CodecInst& codec) {
   return
 #ifdef WEBRTC_CODEC_OPUS
-      !STR_CASE_CMP(codec.plname, "opus") ||
+      absl::EqualsIgnoreCase(codec.plname, "opus") ||
 #endif
       false;
 }
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
index 0a9ce117e2..b904eb8d7d 100644
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/modules/audio_coding/acm2/rent_a_codec.cc
@@ -13,10 +13,11 @@
 #include <memory>
 #include <utility>
 
+#include "absl/strings/match.h"
 #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
-#include "rtc_base/logging.h"
 #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "rtc_base/logging.h"
 #ifdef WEBRTC_CODEC_ILBC
 #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"  // nogncheck
 #endif
@@ -109,7 +110,7 @@ absl::optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
 RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
     std::map<int, int>* pt_map,
     const CodecInst& codec_inst) {
-  if (STR_CASE_CMP(codec_inst.plname, "CN") != 0)
+  if (!absl::EqualsIgnoreCase(codec_inst.plname, "CN"))
     return RegistrationResult::kSkip;
   switch (codec_inst.plfreq) {
     case 8000:
@@ -126,7 +127,7 @@ RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
 RentACodec::RegistrationResult RentACodec::RegisterRedPayloadType(
     std::map<int, int>* pt_map,
     const CodecInst& codec_inst) {
-  if (STR_CASE_CMP(codec_inst.plname, "RED") != 0)
+  if (!absl::EqualsIgnoreCase(codec_inst.plname, "RED"))
     return RegistrationResult::kSkip;
   switch (codec_inst.plfreq) {
     case 8000:
@@ -145,30 +146,30 @@ std::unique_ptr<AudioEncoder> CreateEncoder(
     const CodecInst& speech_inst,
     const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
 #if defined(WEBRTC_CODEC_ISACFX)
-  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "isac"))
     return std::unique_ptr<AudioEncoder>(
         new AudioEncoderIsacFixImpl(speech_inst, bwinfo));
 #endif
 #if defined(WEBRTC_CODEC_ISAC)
-  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "isac"))
     return std::unique_ptr<AudioEncoder>(
         new AudioEncoderIsacFloatImpl(speech_inst, bwinfo));
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-  if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "opus"))
     return std::unique_ptr<AudioEncoder>(new AudioEncoderOpusImpl(speech_inst));
 #endif
-  if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "pcmu"))
     return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst));
-  if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "pcma"))
     return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst));
-  if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "l16"))
     return std::unique_ptr<AudioEncoder>(new AudioEncoderPcm16B(speech_inst));
 #ifdef WEBRTC_CODEC_ILBC
-  if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "ilbc"))
     return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbcImpl(speech_inst));
 #endif
-  if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
+  if (absl::EqualsIgnoreCase(speech_inst.plname, "g722"))
     return std::unique_ptr<AudioEncoder>(new AudioEncoderG722Impl(speech_inst));
   RTC_LOG_F(LS_ERROR) << "Could not create encoder of type "
                       << speech_inst.plname;
diff --git a/modules/audio_coding/codecs/audio_format_conversion.cc b/modules/audio_coding/codecs/audio_format_conversion.cc
index e38aa33335..bf99f8a823 100644
--- a/modules/audio_coding/codecs/audio_format_conversion.cc
+++ b/modules/audio_coding/codecs/audio_format_conversion.cc
@@ -12,6 +12,7 @@
 
 #include <string.h>
 
+#include "absl/strings/match.h"
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/checks.h"
@@ -41,11 +42,11 @@ CodecInst MakeCodecInst(int payload_type,
 }  // namespace
 
 SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
-  if (STR_CASE_CMP(ci.plname, "g722") == 0) {
+  if (absl::EqualsIgnoreCase(ci.plname, "g722")) {
     RTC_CHECK_EQ(16000, ci.plfreq);
     RTC_CHECK(ci.channels == 1 || ci.channels == 2);
     return {"g722", 8000, ci.channels};
-  } else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
+  } else if (absl::EqualsIgnoreCase(ci.plname, "opus")) {
     RTC_CHECK_EQ(48000, ci.plfreq);
     RTC_CHECK(ci.channels == 1 || ci.channels == 2);
     return ci.channels == 1
@@ -57,12 +58,12 @@ SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
 }
 
 CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) {
-  if (STR_CASE_CMP(audio_format.name.c_str(), "g722") == 0) {
+  if (absl::EqualsIgnoreCase(audio_format.name, "g722")) {
     RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
     RTC_CHECK(audio_format.num_channels == 1 || audio_format.num_channels == 2);
     return MakeCodecInst(payload_type, "g722", 16000,
                          audio_format.num_channels);
-  } else if (STR_CASE_CMP(audio_format.name.c_str(), "opus") == 0) {
+  } else if (absl::EqualsIgnoreCase(audio_format.name, "opus")) {
     RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
     RTC_CHECK_EQ(2, audio_format.num_channels);
     const int num_channels = [&] {
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index c07abbe03f..f1aaf70b38 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -15,7 +15,7 @@
 #include <utility>
 
 #include "absl/memory/memory.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
@@ -316,7 +316,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
 
 absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
     const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
+  if (absl::EqualsIgnoreCase(format.name, GetPayloadName()) &&
       format.clockrate_hz == 48000 && format.num_channels == 2) {
     const size_t num_channels = GetChannelCount(format);
     const int bitrate =
@@ -348,7 +348,7 @@ AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
 
 absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
     const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
+  if (!absl::EqualsIgnoreCase(format.name, "opus") ||
       format.clockrate_hz != 48000 || format.num_channels != 2) {
     return absl::nullopt;
   }
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 1fd8c03ca2..bd5d7197a9 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -12,6 +12,7 @@
 
 #include <utility>  // pair
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -101,7 +102,7 @@ AudioDecoder* DecoderDatabase::DecoderInfo::GetDecoder() const {
 }
 
 bool DecoderDatabase::DecoderInfo::IsType(const char* name) const {
-  return STR_CASE_CMP(audio_format_.name.c_str(), name) == 0;
+  return absl::EqualsIgnoreCase(audio_format_.name, name);
 }
 
 bool DecoderDatabase::DecoderInfo::IsType(const std::string& name) const {
@@ -110,7 +111,7 @@ bool DecoderDatabase::DecoderInfo::IsType(const std::string& name) const {
 
 absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
 DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "CN") == 0) {
+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
     // CN has a 1:1 RTP clock rate to sample rate ratio.
     const int sample_rate_hz = format.clockrate_hz;
     RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
@@ -123,11 +124,11 @@ DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
 
 DecoderDatabase::DecoderInfo::Subtype
 DecoderDatabase::DecoderInfo::SubtypeFromFormat(const SdpAudioFormat& format) {
-  if (STR_CASE_CMP(format.name.c_str(), "CN") == 0) {
+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
     return Subtype::kComfortNoise;
-  } else if (STR_CASE_CMP(format.name.c_str(), "telephone-event") == 0) {
+  } else if (absl::EqualsIgnoreCase(format.name, "telephone-event")) {
     return Subtype::kDtmf;
-  } else if (STR_CASE_CMP(format.name.c_str(), "red") == 0) {
+  } else if (absl::EqualsIgnoreCase(format.name, "red")) {
     return Subtype::kRed;
   }
 
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index c6f3f8c9f2..240836688c 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -14,9 +14,9 @@
 #include <stdlib.h>
 #include <memory>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/test/utility.h"
@@ -248,11 +248,11 @@ void EncodeDecodeTest::Perform() {
 
   for (int n = 0; n < numCodecs; n++) {
     EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
-    if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
+    if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "telephone-event")) {
       numPars[n] = 0;
-    } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
+    } else if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "cn")) {
       numPars[n] = 0;
-    } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
+    } else if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "red")) {
       numPars[n] = 0;
     } else if (sendCodecTmp.channels == 2) {
       numPars[n] = 0;
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index c56fed95da..4866118748 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -173,7 +173,7 @@ void TestRedFec::RegisterSendCodec(
   auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
                                                     absl::nullopt);
   EXPECT_NE(encoder, nullptr);
-  if (STR_CASE_CMP(codec_format.name.c_str(), "opus") != 0) {
+  if (!absl::EqualsIgnoreCase(codec_format.name, "opus")) {
     if (vad_mode.has_value()) {
       AudioEncoderCng::Config config;
       config.speech_encoder = std::move(encoder);
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index fd69fd2626..bf8e1894d0 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -12,9 +12,9 @@
 
 #include <string>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "modules/audio_coding/test/utility.h"
@@ -626,14 +626,14 @@ void TestStereo::RegisterSendCodec(char side,
   ASSERT_TRUE(my_acm != NULL);
 
   auto encoder_factory = CreateBuiltinAudioEncoderFactory();
-  const int clockrate_hz = STR_CASE_CMP(codec_name, "g722") == 0
+  const int clockrate_hz = absl::EqualsIgnoreCase(codec_name, "g722")
                                ? sampling_freq_hz / 2
                                : sampling_freq_hz;
   const std::string ptime = rtc::ToString(rtc::CheckedDivExact(
       pack_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
   SdpAudioFormat::Parameters params = {{"ptime", ptime}};
   RTC_CHECK(channels == 1 || channels == 2);
-  if (STR_CASE_CMP(codec_name, "opus") == 0) {
+  if (absl::EqualsIgnoreCase(codec_name, "opus")) {
     if (channels == 2) {
       params["stereo"] = "1";
     }
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 4f02edaa83..aa91ee011a 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -12,6 +12,7 @@
 
 #include <string>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/audio_decoder_factory_template.h"
 #include "api/audio_codecs/audio_encoder_factory_template.h"
 #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
@@ -81,7 +82,7 @@ bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
   auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
                                                     absl::nullopt);
   if (vad_mode.has_value() &&
-      STR_CASE_CMP(codec_format.name.c_str(), "opus") != 0) {
+      !absl::EqualsIgnoreCase(codec_format.name, "opus")) {
     AudioEncoderCng::Config config;
     config.speech_encoder = std::move(encoder);
     config.num_channels = 1;
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index a130ae3e66..c332fe08f1 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -23,6 +23,7 @@
 #include <time.h>
 #endif
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
@@ -92,12 +93,12 @@ void ISACTest::Setup() {
   for (codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs();
        codecCntr++) {
     EXPECT_EQ(0, AudioCodingModule::Codec(codecCntr, &codecParam));
-    if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
+    if (absl::EqualsIgnoreCase(codecParam.plname, "ISAC") &&
         codecParam.plfreq == 16000) {
       memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
       _idISAC16kHz = codecCntr;
     }
-    if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
+    if (absl::EqualsIgnoreCase(codecParam.plname, "ISAC") &&
         codecParam.plfreq == 32000) {
       memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
       _idISAC32kHz = codecCntr;
diff --git a/modules/audio_coding/test/utility.cc b/modules/audio_coding/test/utility.cc
index 83c25b537e..53f807790e 100644
--- a/modules/audio_coding/test/utility.cc
+++ b/modules/audio_coding/test/utility.cc
@@ -15,7 +15,7 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "common_types.h"  // NOLINT(build/include)
+#include "absl/strings/match.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "test/gtest.h"
 
@@ -268,7 +268,7 @@ bool FixedPayloadTypeCodec(const char* payloadName) {
       "G722", "QCELP", "CN",  "MPA",  "G728", "G729"};
 
   for (int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++) {
-    if (!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n])) {
+    if (absl::EqualsIgnoreCase(payloadName, fixPayloadTypeCodecs[n])) {
       return true;
     }
   }
diff --git a/pc/mediasession.cc b/pc/mediasession.cc
index d4a80c03a5..d84166278a 100644
--- a/pc/mediasession.cc
+++ b/pc/mediasession.cc
@@ -18,9 +18,9 @@
 #include <unordered_map>
 #include <utility>
 
+#include "absl/strings/match.h"
 #include "absl/types/optional.h"
 #include "api/cryptoparams.h"
-#include "common_types.h"  // NOLINT(build/include)
 #include "media/base/h264_profile_level_id.h"
 #include "media/base/mediaconstants.h"
 #include "p2p/base/p2pconstants.h"
@@ -653,7 +653,7 @@ static bool ContainsRtxCodec(const std::vector<C>& codecs) {
 
 template <class C>
 static bool IsRtxCodec(const C& codec) {
-  return STR_CASE_CMP(codec.name.c_str(), kRtxCodecName) == 0;
+  return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
 }
 
 template <class C>
@@ -668,7 +668,7 @@ static bool ContainsFlexfecCodec(const std::vector<C>& codecs) {
 
 template <class C>
 static bool IsFlexfecCodec(const C& codec) {
-  return STR_CASE_CMP(codec.name.c_str(), kFlexfecCodecName) == 0;
+  return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
 }
 
 // Create a media content to be offered for the given |sender_options|,
@@ -1041,9 +1041,8 @@ static void NegotiateRtpHeaderExtensions(
 static void StripCNCodecs(AudioCodecs* audio_codecs) {
   audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
                                      [](const AudioCodec& codec) {
-                                       return STR_CASE_CMP(
-                                                  codec.name.c_str(),
-                                                  kComfortNoiseCodecName) == 0;
+                                       return absl::EqualsIgnoreCase(
+                                           codec.name, kComfortNoiseCodecName);
                                      }),
                       audio_codecs->end());
 }
diff --git a/pc/peerconnectionendtoend_unittest.cc b/pc/peerconnectionendtoend_unittest.cc
index 3e0c2c39eb..52c6bf2a75 100644
--- a/pc/peerconnectionendtoend_unittest.cc
+++ b/pc/peerconnectionendtoend_unittest.cc
@@ -11,6 +11,7 @@
 #include <memory>
 
 #include "absl/memory/memory.h"
+#include "absl/strings/match.h"
 #include "api/audio_codecs/L16/audio_decoder_L16.h"
 #include "api/audio_codecs/L16/audio_encoder_L16.h"
 #include "api/audio_codecs/audio_codec_pair_id.h"
@@ -299,7 +300,7 @@ CreateForwardingMockDecoderFactory(
 struct AudioEncoderUnicornSparklesRainbow {
   using Config = webrtc::AudioEncoderL16::Config;
   static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
-    if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
+    if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
       const webrtc::SdpAudioFormat::Parameters expected_params = {
           {"num_horns", "1"}};
       EXPECT_EQ(expected_params, format.parameters);
@@ -336,7 +337,7 @@ struct AudioEncoderUnicornSparklesRainbow {
 struct AudioDecoderUnicornSparklesRainbow {
   using Config = webrtc::AudioDecoderL16::Config;
   static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
-    if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
+    if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
       const webrtc::SdpAudioFormat::Parameters expected_params = {
           {"num_horns", "1"}};
       EXPECT_EQ(expected_params, format.parameters);