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/audio_encoder_factory.h b/api/audio_codecs/audio_encoder_factory.h
index 7825953..fb4e23f 100644
--- a/api/audio_codecs/audio_encoder_factory.h
+++ b/api/audio_codecs/audio_encoder_factory.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"
#include "rtc_base/refcount.h"
namespace webrtc {
@@ -32,7 +32,7 @@
// Returns information about how this format would be encoded, provided it's
// supported. More format and format variations may be supported than those
// returned by GetSupportedEncoders().
- virtual rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
+ virtual absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) = 0;
// Creates an AudioEncoder for the specified format. The encoder will tags
@@ -50,7 +50,7 @@
virtual std::unique_ptr<AudioEncoder> MakeAudioEncoder(
int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) = 0;
+ absl::optional<AudioCodecPairId> codec_pair_id) = 0;
};
} // namespace webrtc