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Replace rtc::Optional with absl::optional in modules/audio_coding

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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_coding'

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: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
This commit is contained in:
Danil Chapovalov
2018-06-19 13:26:36 +02:00
committed by Commit Bot
parent bbfcc703ad
commit b602123a5a
82 changed files with 452 additions and 459 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
modules/audio_coding/codecs/opus/audio_encoder_opus.h
View File

@ -16,10 +16,10 @@
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
#include "api/optional.h"
#include "common_audio/smoothing_filter.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
@ -40,13 +40,13 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
// defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
// Otherwise, returns the current complexity depending on whether the
// current bitrate is above or below complexity_threshold_bps.
static rtc::Optional<int> GetNewComplexity(
static absl::optional<int> GetNewComplexity(
const AudioEncoderOpusConfig& config);
// Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
// Returns empty if it is below, but bandwidth coincides with the desired one.
// Otherwise returns the desired bandwidth.
static rtc::Optional<int> GetNewBandwidth(
static absl::optional<int> GetNewBandwidth(
const AudioEncoderOpusConfig& config,
OpusEncInst* inst);
@ -69,7 +69,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
// Static interface for use by BuiltinAudioEncoderFactory.
static constexpr const char* GetPayloadName() { return "opus"; }
static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
static absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format);
int SampleRateHz() const override;
@ -98,7 +98,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> bwe_period_ms) override;
absl::optional<int64_t> bwe_period_ms) override;
void OnReceivedRtt(int rtt_ms) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
@ -125,7 +125,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
private:
class PacketLossFractionSmoother;
static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
const SdpAudioFormat& format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
@ -167,9 +167,9 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
rtc::Optional<size_t> overhead_bytes_per_packet_;
absl::optional<size_t> overhead_bytes_per_packet_;
const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
absl::optional<int64_t> bitrate_smoother_last_update_time_;
int consecutive_dtx_frames_;
friend struct AudioEncoderOpus;