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}diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index ce1e1f2..c0afbb1 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -16,9 +16,9 @@
#include <string>
#include <vector>
-#include "api/audio/audio_frame.h"
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
+#include "api/audio/audio_frame.h"
#include "common_audio/vad/include/webrtc_vad.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
@@ -159,7 +159,7 @@
// packet. If no packet of a registered non-CNG codec has been received, the
// return value is empty. Also, if the decoder was unregistered since the last
// packet was inserted, the return value is empty.
- rtc::Optional<int> last_packet_sample_rate_hz() const;
+ absl::optional<int> last_packet_sample_rate_hz() const;
// Returns last_output_sample_rate_hz from the NetEq instance.
int last_output_sample_rate_hz() const;
@@ -195,7 +195,7 @@
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
- rtc::Optional<uint32_t> GetPlayoutTimestamp();
+ absl::optional<uint32_t> GetPlayoutTimestamp();
// Returns the current total delay from NetEq (packet buffer and sync buffer)
// in ms, with smoothing applied to even out short-time fluctuations due to
@@ -215,7 +215,7 @@
//
int LastAudioCodec(CodecInst* codec) const;
- rtc::Optional<SdpAudioFormat> LastAudioFormat() const;
+ absl::optional<SdpAudioFormat> LastAudioFormat() const;
//
// Get a decoder given its registered payload-type.
@@ -273,22 +273,23 @@
int sample_rate_hz;
};
- const rtc::Optional<CodecInst> RtpHeaderToDecoder(const RTPHeader& rtp_header,
- uint8_t first_payload_byte)
- const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+ const absl::optional<CodecInst> RtpHeaderToDecoder(
+ const RTPHeader& rtp_header,
+ uint8_t first_payload_byte) const
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
rtc::CriticalSection crit_sect_;
- rtc::Optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
- rtc::Optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
const Clock* const clock_;
bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
- rtc::Optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
};
} // namespace acm2