Replace rtc::Optional with absl::optional in media, ortc, p2p
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'media ortc p2p':
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: I19167714af7cc1436d34cfcba6c8b3718d8e677b
Reviewed-on: https://webrtc-review.googlesource.com/83731
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23638}diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index a316c70..a9b586a 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -17,9 +17,9 @@
#include <utility>
#include <vector>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_options.h"
-#include "api/optional.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
@@ -62,7 +62,8 @@
const int kScreencastDefaultFps = 5;
template <class T>
-static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
+static std::string ToStringIfSet(const char* key,
+ const absl::optional<T>& val) {
std::string str;
if (val) {
str = key;
@@ -122,20 +123,20 @@
// Enable denoising? This flag comes from the getUserMedia
// constraint 'googNoiseReduction', and WebRtcVideoEngine passes it
// on to the codec options. Disabled by default.
- rtc::Optional<bool> video_noise_reduction;
+ absl::optional<bool> video_noise_reduction;
// Force screencast to use a minimum bitrate. This flag comes from
// the PeerConnection constraint 'googScreencastMinBitrate'. It is
// copied to the encoder config by WebRtcVideoChannel.
- rtc::Optional<int> screencast_min_bitrate_kbps;
+ absl::optional<int> screencast_min_bitrate_kbps;
// Set by screencast sources. Implies selection of encoding settings
// suitable for screencast. Most likely not the right way to do
// things, e.g., screencast of a text document and screencast of a
// youtube video have different needs.
- rtc::Optional<bool> is_screencast;
+ absl::optional<bool> is_screencast;
private:
template <typename T>
- static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
+ static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
if (o) {
*s = o;
}
@@ -328,7 +329,7 @@
float fraction_lost = 0.0f;
int64_t rtt_ms = 0;
std::string codec_name;
- rtc::Optional<int> codec_payload_type;
+ absl::optional<int> codec_payload_type;
std::vector<SsrcSenderInfo> local_stats;
std::vector<SsrcReceiverInfo> remote_stats;
};
@@ -374,7 +375,7 @@
int packets_lost = 0;
float fraction_lost = 0.0f;
std::string codec_name;
- rtc::Optional<int> codec_payload_type;
+ absl::optional<int> codec_payload_type;
std::vector<SsrcReceiverInfo> local_stats;
std::vector<SsrcSenderInfo> remote_stats;
};
@@ -468,7 +469,7 @@
int encode_usage_percent = 0;
uint32_t frames_encoded = 0;
bool has_entered_low_resolution = false;
- rtc::Optional<uint64_t> qp_sum;
+ absl::optional<uint64_t> qp_sum;
webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED;
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent
uint32_t huge_frames_sent = 0;
@@ -496,7 +497,7 @@
uint32_t frames_received = 0;
uint32_t frames_decoded = 0;
uint32_t frames_rendered = 0;
- rtc::Optional<uint64_t> qp_sum;
+ absl::optional<uint64_t> qp_sum;
int64_t interframe_delay_max_ms = -1;
webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED;
@@ -526,7 +527,7 @@
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
- rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info;
+ absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
};
struct DataSenderInfo : public MediaSenderInfo {