Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.

Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 2ba551f..01a0e93 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -53,6 +53,7 @@
     "../../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 @@
   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 @@
            ":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_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_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 @@
       "../../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 311af2d..e109cc7 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 @@
   }
 
   // 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 @@
 
   // 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 @@
     // 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 2d7f296..c149ec1 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 @@
 // 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 @@
     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 f631746..93fa27c 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 @@
     }
     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 @@
     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 60afeb6..d675203 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 @@
   }
 
   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 f29e0f1..5094e21 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 @@
   }
 
   // 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 @@
 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 0a9ce11..b904eb8 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 @@
 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::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 @@
     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 e38aa33..bf99f8a 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 @@
 }  // 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 @@
 }
 
 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 c07abbe..f1aaf70 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 @@
 
 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 @@
 
 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 1fd8c03..bd5d719 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 @@
 }
 
 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 @@
 
 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::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 c6f3f8c..2408366 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 @@
 
   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 c56fed9..4866118 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -173,7 +173,7 @@
   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 fd69fd2..bf8e189 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 @@
   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 4f02eda..aa91ee0 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 @@
   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 a130ae3..c332fe0 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 @@
   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 83c25b5..53f8077 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 @@
       "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;
     }
   }