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Replace rtc::Optional with absl::optional in modules/audio processing

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This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_processing'

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: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
This commit is contained in:
Danil Chapovalov
2018-06-19 10:50:11 +02:00
committed by Commit Bot
parent c66613daf7
commit db9f7ab9f9
65 changed files with 248 additions and 248 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
modules/audio_processing/include/audio_processing.h
View File

@ -22,9 +22,9 @@
#include <string.h>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
#include "api/optional.h"
#include "modules/audio_processing/beamformer/array_util.h"
#include "modules/audio_processing/include/audio_generator.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"

18
modules/audio_processing/include/audio_processing_statistics.h
View File

@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
#include "api/optional.h"
#include "absl/types/optional.h"
namespace webrtc {
// This version of the stats uses Optionals, it will replace the regular
@ -23,12 +23,12 @@ struct AudioProcessingStats {
// AEC Statistics.
// ERL = 10log_10(P_far / P_echo)
rtc::Optional<double> echo_return_loss;
absl::optional<double> echo_return_loss;
// ERLE = 10log_10(P_echo / P_out)
rtc::Optional<double> echo_return_loss_enhancement;
absl::optional<double> echo_return_loss_enhancement;
// Fraction of time that the AEC linear filter is divergent, in a 1-second
// non-overlapped aggregation window.
rtc::Optional<double> divergent_filter_fraction;
absl::optional<double> divergent_filter_fraction;
// The delay metrics consists of the delay median and standard deviation. It
// also consists of the fraction of delay estimates that can make the echo
@ -37,18 +37,18 @@ struct AudioProcessingStats {
// second. Note that if there are several clients pulling metrics from
// |GetStatistics()| during a session the first call from any of them will
// change to one second aggregation window for all.
rtc::Optional<int32_t> delay_median_ms;
rtc::Optional<int32_t> delay_standard_deviation_ms;
absl::optional<int32_t> delay_median_ms;
absl::optional<int32_t> delay_standard_deviation_ms;
// Residual echo detector likelihood.
rtc::Optional<double> residual_echo_likelihood;
absl::optional<double> residual_echo_likelihood;
// Maximum residual echo likelihood from the last time period.
rtc::Optional<double> residual_echo_likelihood_recent_max;
absl::optional<double> residual_echo_likelihood_recent_max;
// The instantaneous delay estimate produced in the AEC. The unit is in
// milliseconds and the value is the instantaneous value at the time of the
// call to |GetStatistics()|.
rtc::Optional<int32_t> delay_ms;
absl::optional<int32_t> delay_ms;
};
} // namespace webrtc