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/neteq/neteq_impl.cc

diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 688162b..40eae1b 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -422,14 +422,14 @@
   vad_->Disable();
 }
 
-rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
+absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
   rtc::CritScope lock(&crit_sect_);
   if (first_packet_ || last_mode_ == kModeRfc3389Cng ||
       last_mode_ == kModeCodecInternalCng) {
     // We don't have a valid RTP timestamp until we have decoded our first
     // RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
     // which is indicated by returning an empty value.
-    return rtc::nullopt;
+    return absl::nullopt;
   }
   return timestamp_scaler_->ToExternal(playout_timestamp_);
 }
@@ -439,12 +439,12 @@
   return last_output_sample_rate_hz_;
 }
 
-rtc::Optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
+absl::optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
   rtc::CritScope lock(&crit_sect_);
   const DecoderDatabase::DecoderInfo* di =
       decoder_database_->GetDecoderInfo(payload_type);
   if (!di) {
-    return rtc::nullopt;
+    return absl::nullopt;
   }
 
   // Create a CodecInst with some fields set. The remaining fields are zeroed,
@@ -460,13 +460,13 @@
   return ci;
 }
 
-rtc::Optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
+absl::optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
     int payload_type) const {
   rtc::CritScope lock(&crit_sect_);
   const DecoderDatabase::DecoderInfo* const di =
       decoder_database_->GetDecoderInfo(payload_type);
   if (!di) {
-    return rtc::nullopt;  // Payload type not registered.
+    return absl::nullopt;  // Payload type not registered.
   }
   return di->GetFormat();
 }
@@ -1957,7 +1957,7 @@
   // Packet extraction loop.
   do {
     timestamp_ = next_packet->timestamp;
-    rtc::Optional<Packet> packet = packet_buffer_->GetNextPacket();
+    absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
     // |next_packet| may be invalid after the |packet_buffer_| operation.
     next_packet = nullptr;
     if (!packet) {
@@ -2009,7 +2009,7 @@
     stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
 
     packet_list->push_back(std::move(*packet));  // Store packet in list.
-    packet = rtc::nullopt;  // Ensure it's never used after the move.
+    packet = absl::nullopt;  // Ensure it's never used after the move.
 
     // Check what packet is available next.
     next_packet = packet_buffer_->PeekNextPacket();