Replace rtc::Optional with absl::optional in api
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
index 7026abb..52e1ee9 100644
--- a/api/audio_codecs/g711/BUILD.gn
+++ b/api/audio_codecs/g711/BUILD.gn
@@ -21,11 +21,11 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -38,9 +38,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
+ "//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 c715e80..e8afa60 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.cc
+++ b/api/audio_codecs/g711/audio_decoder_g711.cc
@@ -20,7 +20,7 @@
namespace webrtc {
-rtc::Optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
+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;
@@ -32,7 +32,7 @@
RTC_DCHECK(config.IsOk());
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -45,7 +45,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
switch (config.type) {
case Config::Type::kPcmU:
diff --git a/api/audio_codecs/g711/audio_decoder_g711.h b/api/audio_codecs/g711/audio_decoder_g711.h
index 5085283..8275a8c 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.h
+++ b/api/audio_codecs/g711/audio_decoder_g711.h
@@ -14,10 +14,10 @@
#include <memory>
#include <vector>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
-#include "api/optional.h"
namespace webrtc {
@@ -34,11 +34,11 @@
Type type;
int num_channels;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/g711/audio_encoder_g711.cc b/api/audio_codecs/g711/audio_encoder_g711.cc
index e5abc33..95595fa 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.cc
+++ b/api/audio_codecs/g711/audio_encoder_g711.cc
@@ -22,7 +22,7 @@
namespace webrtc {
-rtc::Optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
+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;
@@ -42,7 +42,7 @@
RTC_DCHECK(config.IsOk());
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -62,7 +62,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
const Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
switch (config.type) {
case Config::Type::kPcmU: {
diff --git a/api/audio_codecs/g711/audio_encoder_g711.h b/api/audio_codecs/g711/audio_encoder_g711.h
index 22a74b4..6b6eb5f 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.h
+++ b/api/audio_codecs/g711/audio_encoder_g711.h
@@ -14,10 +14,10 @@
#include <memory>
#include <vector>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
-#include "api/optional.h"
namespace webrtc {
@@ -36,14 +36,14 @@
int num_channels = 1;
int frame_size_ms = 20;
};
- static rtc::Optional<AudioEncoderG711::Config> SdpToConfig(
+ static absl::optional<AudioEncoderG711::Config> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc