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/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
index a7d56d0..34ec2e4 100644
--- a/api/audio_codecs/L16/BUILD.gn
+++ b/api/audio_codecs/L16/BUILD.gn
@@ -21,12 +21,12 @@
]
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 @@
]
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 a71e308..be0c6b5 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 @@
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 b516f62..1bb552b 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 @@
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 bd2d3f2..11788b9 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 @@
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=(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 ebd0b32..3b8f23c 100644
--- a/api/audio_codecs/g711/BUILD.gn
+++ b/api/audio_codecs/g711/BUILD.gn
@@ -21,12 +21,12 @@
]
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 @@
]
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 cb16584..91599c4 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 @@
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 1d5e541..0cc8dc4 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 @@
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 101c7a9..e4321d2 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@@ -29,12 +29,12 @@
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 @@
]
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 f1e2afb..2cc16c3 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 @@
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 0cf7163..6374ae8 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 @@
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 d4f504f..d766fa4 100644
--- a/api/audio_codecs/ilbc/BUILD.gn
+++ b/api/audio_codecs/ilbc/BUILD.gn
@@ -29,11 +29,11 @@
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 @@
]
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 1f4c475..4a00f8d 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 efcef38..896ed23 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 @@
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 9025360..c7d6e43 100644
--- a/api/audio_codecs/isac/BUILD.gn
+++ b/api/audio_codecs/isac/BUILD.gn
@@ -77,10 +77,10 @@
]
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 @@
]
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 @@
]
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 @@
]
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 446640f..51ae572 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 1c1926f..d9de3a0 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 cd89753..a10d1ee 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 @@
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 83d1faf..37982b1 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 @@
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 af3cd7f..5552c21 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -46,6 +46,7 @@
"../../../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 @@
]
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 41397f0..2f1668b 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 @@
}
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};