commit b602123a5aae1a9f0b6b581e611acf44eb525c08
author Danil Chapovalov <danilchap@webrtc.org> Tue Jun 19 13:26:36 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Jun 19 12:46:20 2018 +0000
tree 30d0844e16256401ba98608dfa86f9770e37dc08
parent bbfcc703ad1b5ae141dcd9c6a951241f6fc03967

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

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}

modules/audio_coding/acm2/audio_coding_module.cc

diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 4545810..2d8827c 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -54,7 +54,7 @@
       rtc::FunctionView<void(const AudioEncoder*)> query) override;
 
   // Get current send codec.
-  rtc::Optional<CodecInst> SendCodec() const override;
+  absl::optional<CodecInst> SendCodec() const override;
 
   // Get current send frequency.
   int SendFrequency() const override;
@@ -142,7 +142,7 @@
   // Get current received codec.
   int ReceiveCodec(CodecInst* current_codec) const override;
 
-  rtc::Optional<SdpAudioFormat> ReceiveFormat() const override;
+  absl::optional<SdpAudioFormat> ReceiveFormat() const override;
 
   // Incoming packet from network parsed and ready for decode.
   int IncomingPacket(const uint8_t* incoming_payload,
@@ -160,7 +160,7 @@
 
   RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
 
-  rtc::Optional<uint32_t> PlayoutTimestamp() override;
+  absl::optional<uint32_t> PlayoutTimestamp() override;
 
   int FilteredCurrentDelayMs() const override;
 
@@ -603,7 +603,7 @@
 }
 
 // Get current send codec.
-rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
+absl::optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
   rtc::CritScope lock(&acm_crit_sect_);
   if (encoder_factory_) {
     auto* ci = encoder_factory_->codec_manager.GetCodecInst();
@@ -616,12 +616,12 @@
     if (enc) {
       return acm2::CodecManager::ForgeCodecInst(enc.get());
     }
-    return rtc::nullopt;
+    return absl::nullopt;
   } else {
     return encoder_stack_
-               ? rtc::Optional<CodecInst>(
+               ? absl::optional<CodecInst>(
                      acm2::CodecManager::ForgeCodecInst(encoder_stack_.get()))
-               : rtc::nullopt;
+               : absl::nullopt;
   }
 }
 
@@ -640,7 +640,7 @@
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
   rtc::CritScope lock(&acm_crit_sect_);
   if (encoder_stack_) {
-    encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, rtc::nullopt);
+    encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, absl::nullopt);
   }
 }
 
@@ -1062,7 +1062,7 @@
   return receiver_.LastAudioCodec(current_codec);
 }
 
-rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
+absl::optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
   rtc::CritScope lock(&acm_crit_sect_);
   return receiver_.LastAudioFormat();
 }
@@ -1180,14 +1180,14 @@
 }
 
 int32_t AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
-  rtc::Optional<uint32_t> ts = PlayoutTimestamp();
+  absl::optional<uint32_t> ts = PlayoutTimestamp();
   if (!ts)
     return -1;
   *timestamp = *ts;
   return 0;
 }
 
-rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
+absl::optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
   return receiver_.GetPlayoutTimestamp();
 }
 
@@ -1279,7 +1279,7 @@
                              CodecInst* codec,
                              int sampling_freq_hz,
                              size_t channels) {
-  rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
+  absl::optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
       payload_name, sampling_freq_hz, channels);
   if (ci) {
     *codec = *ci;
@@ -1299,12 +1299,12 @@
 int AudioCodingModule::Codec(const char* payload_name,
                              int sampling_freq_hz,
                              size_t channels) {
-  rtc::Optional<acm2::RentACodec::CodecId> ci =
+  absl::optional<acm2::RentACodec::CodecId> ci =
       acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
                                         channels);
   if (!ci)
     return -1;
-  rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
+  absl::optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
   return i ? *i : -1;
 }