zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

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

Browse Source 
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}
This commit is contained in:
Niels Möller
2018-10-22 09:48:08 +02:00
committed by Commit Bot
parent 01cf44d397
commit 2edab4c026
42 changed files with 146 additions and 142 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
BUILD.gn
View File

@ -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",
]

4
api/audio_codecs/L16/BUILD.gn
View File

@ -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",
]
}

4
api/audio_codecs/L16/audio_decoder_L16.cc
View File

@ -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;
}

4
api/audio_codecs/L16/audio_encoder_L16.cc
View File

@ -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;
}

6
api/audio_codecs/audio_format.cc
View File

@ -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;
}

4
api/audio_codecs/g711/BUILD.gn
View File

@ -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",
]
}

6
api/audio_codecs/g711/audio_decoder_g711.cc
View File

@ -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;

6
api/audio_codecs/g711/audio_encoder_g711.cc
View File

@ -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;

4
api/audio_codecs/g722/BUILD.gn
View File

@ -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",
]
}

4
api/audio_codecs/g722/audio_decoder_g722.cc
View File

@ -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>(

4
api/audio_codecs/g722/audio_encoder_g722.cc
View File

@ -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;
}

4
api/audio_codecs/ilbc/BUILD.gn
View File

@ -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",
]
}

4
api/audio_codecs/ilbc/audio_decoder_ilbc.cc
View File

@ -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;

4
api/audio_codecs/ilbc/audio_encoder_ilbc.cc
View File

@ -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;
}

8
api/audio_codecs/isac/BUILD.gn
View File

@ -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",
]
}

4
api/audio_codecs/isac/audio_decoder_isac_fix.cc
View File

@ -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;

4
api/audio_codecs/isac/audio_decoder_isac_float.cc
View File

@ -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;

4
api/audio_codecs/isac/audio_encoder_isac_fix.cc
View File

@ -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");

4
api/audio_codecs/isac/audio_encoder_isac_float.cc
View File

@ -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;

3
api/audio_codecs/opus/BUILD.gn
View File

@ -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",
]
}

4
api/audio_codecs/opus/audio_decoder_opus.cc
View File

@ -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};

15
common_types.h
View File

@ -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;
}

2
media/engine/webrtcvoiceengine.cc
View File

@ -968,7 +968,7 @@ 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(),
absl::EqualsIgnoreCase(config_.send_codec_spec->format.name,
kOpusCodecName);
if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
config_.min_bitrate_bps = kOpusMinBitrateBps;

5
media/engine/webrtcvoiceengine_unittest.cc
View File

@ -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);

6
modules/audio_coding/BUILD.gn
View File

@ -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

15
modules/audio_coding/acm2/acm_codec_database.cc
View File

@ -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.

43
modules/audio_coding/acm2/acm_receive_test.cc
View File

@ -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.

6
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -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 {

3
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -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) {

5
modules/audio_coding/acm2/codec_manager.cc
View File

@ -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;
}

23
modules/audio_coding/acm2/rent_a_codec.cc
View File

@ -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;

9
modules/audio_coding/codecs/audio_format_conversion.cc
View File

@ -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 = [&] {

6
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -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;
}

11
modules/audio_coding/neteq/decoder_database.cc
View File

@ -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;
}

8
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -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;

2
modules/audio_coding/test/TestRedFec.cc
View File

@ -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);

6
modules/audio_coding/test/TestStereo.cc
View File

@ -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";
}

3
modules/audio_coding/test/TestVADDTX.cc
View File

@ -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;

5
modules/audio_coding/test/iSACTest.cc
View File

@ -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;

4
modules/audio_coding/test/utility.cc
View File

@ -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;
}
}

11
pc/mediasession.cc
View File

@ -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());
}

5
pc/peerconnectionendtoend_unittest.cc
View File

@ -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);