Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.
This CL has been created with the following steps:
git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt
diff --new-line-format="" --unchanged-line-format="" \
/tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
/tmp/only_make_unique.txt /tmp/memory.txt | \
xargs grep -l "absl/memory" > /tmp/add-memory.txt
git grep -l "\babsl::make_unique\b" | \
xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"
git checkout PRESUBMIT.py abseil-in-webrtc.md
cat /tmp/add-memory.txt | \
xargs sed -i \
's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>
cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
xargs sed -i '/#include "absl\/memory\/memory.h"/d'
git ls-files | grep BUILD.gn | \
xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'
python tools_webrtc/gn_check_autofix.py \
-m tryserver.webrtc -b linux_rel
# Repead the gn_check_autofix step for other platforms
git ls-files | grep BUILD.gn | \
xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format
Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index cb57710..8c5fb00 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -103,7 +103,6 @@
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/memory",
]
}
@@ -764,7 +763,6 @@
"../../rtc_base:rtc_numerics",
"../../rtc_base:safe_minmax",
"../../system_wrappers:field_trial",
- "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -916,7 +914,6 @@
"../../rtc_base/system:file_wrapper",
"../../system_wrappers",
"../../system_wrappers:field_trial",
- "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -1024,7 +1021,6 @@
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -1098,7 +1094,6 @@
"../../test:rtp_test_utils",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/memory:memory",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -1181,7 +1176,6 @@
"../../rtc_base:rtc_base_approved",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [
@@ -1324,7 +1318,6 @@
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -1475,7 +1468,6 @@
"../../rtc_base:rtc_base_approved",
"../../test:audio_codec_mocks",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/memory",
]
}
@@ -1599,9 +1591,9 @@
testonly = true
deps = audio_coding_deps + [
+ "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
- "//third_party/abseil-cpp/absl/memory",
":audio_coding",
":audio_encoder_cng",
":neteq_input_audio_tools",
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 6b75c35..5b68085 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -17,7 +17,6 @@
#include <memory>
#include <vector>
-#include "absl/memory/memory.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
@@ -1663,7 +1662,7 @@
config.num_channels = 1;
config.payload_type = 0;
AudioEncoderPcmU encoder(config);
- auto mock_encoder = absl::make_unique<MockAudioEncoder>();
+ auto mock_encoder = std::make_unique<MockAudioEncoder>();
// Set expectations on the mock encoder and also delegate the calls to the
// real encoder.
EXPECT_CALL(*mock_encoder, SampleRateHz())
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index d6893cd..0a79484 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -14,9 +14,9 @@
#include <algorithm>
#include <cstdlib>
+#include <memory>
#include <utility>
-#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/rtc_event_log/rtc_event_log.h"
@@ -69,9 +69,9 @@
}
void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
- auto config_copy = absl::make_unique<AudioEncoderRuntimeConfig>(config);
- event_log_->Log(absl::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(config_copy)));
+ auto config_copy = std::make_unique<AudioEncoderRuntimeConfig>(config);
+ event_log_->Log(
+ std::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config_copy)));
last_logged_config_ = config;
}
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 9a29261..a4e0ffb 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -14,7 +14,6 @@
#include <memory>
#include <utility>
-#include "absl/memory/memory.h"
#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "rtc_base/checks.h"
@@ -317,7 +316,7 @@
std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
AudioEncoderCngConfig&& config) {
- return absl::make_unique<AudioEncoderCng>(std::move(config));
+ return std::make_unique<AudioEncoderCng>(std::move(config));
}
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
index 84a62a1..0614a0b 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -24,7 +24,6 @@
#include <string>
#include <vector>
-#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
#include "rtc_base/arraysize.h"
@@ -134,8 +133,8 @@
if (!config.IsOk()) {
return nullptr;
}
- return absl::make_unique<AudioEncoderMultiChannelOpusImpl>(config,
- payload_type);
+ return std::make_unique<AudioEncoderMultiChannelOpusImpl>(config,
+ payload_type);
}
AudioEncoderMultiChannelOpusImpl::AudioEncoderMultiChannelOpusImpl(
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index f901d3c..70081d7 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -12,10 +12,10 @@
#include <algorithm>
#include <iterator>
+#include <memory>
#include <string>
#include <utility>
-#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
@@ -244,15 +244,13 @@
if (sscanf(field_trial_string.c_str(), "Enabled-%d-%d-%f", &min_rate,
&max_rate, &slope) == 3 &&
IsValidPacketLossRate(min_rate) && IsValidPacketLossRate(max_rate)) {
- return absl::make_unique<
- AudioEncoderOpusImpl::NewPacketLossRateOptimizer>(
+ return std::make_unique<AudioEncoderOpusImpl::NewPacketLossRateOptimizer>(
ToFraction(min_rate), ToFraction(max_rate), slope);
}
RTC_LOG(LS_WARNING) << "Invalid parameters for "
<< kPacketLossOptimizationName
<< ", using default values.";
- return absl::make_unique<
- AudioEncoderOpusImpl::NewPacketLossRateOptimizer>();
+ return std::make_unique<AudioEncoderOpusImpl::NewPacketLossRateOptimizer>();
}
return nullptr;
}
@@ -300,7 +298,7 @@
const AudioEncoderOpusConfig& config,
int payload_type) {
RTC_DCHECK(config.IsOk());
- return absl::make_unique<AudioEncoderOpusImpl>(config, payload_type);
+ return std::make_unique<AudioEncoderOpusImpl>(config, payload_type);
}
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
@@ -417,7 +415,7 @@
return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
},
// We choose 5sec as initial time constant due to empirical data.
- absl::make_unique<SmoothingFilterImpl>(5000)) {}
+ std::make_unique<SmoothingFilterImpl>(5000)) {}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(
const AudioEncoderOpusConfig& config,
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 3870ecd..698b441 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -14,7 +14,6 @@
#include <memory>
#include <utility>
-#include "absl/memory/memory.h"
#include "common_audio/mocks/mock_smoothing_filter.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
@@ -55,7 +54,7 @@
std::unique_ptr<AudioEncoderOpusStates> CreateCodec(int sample_rate_hz,
size_t num_channels) {
std::unique_ptr<AudioEncoderOpusStates> states =
- absl::make_unique<AudioEncoderOpusStates>();
+ std::make_unique<AudioEncoderOpusStates>();
states->mock_audio_network_adaptor = nullptr;
states->fake_clock.reset(new rtc::ScopedFakeClock());
states->fake_clock->SetTime(Timestamp::us(kInitialTimeUs));
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index afa2a3f..3fda038 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -15,10 +15,10 @@
#include <stdlib.h>
#include <algorithm>
+#include <memory>
#include <numeric>
#include <string>
-#include "absl/memory/memory.h"
#include "modules/audio_coding/neteq/delay_peak_detector.h"
#include "modules/audio_coding/neteq/histogram.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
@@ -149,9 +149,9 @@
const HistogramMode mode = RELATIVE_ARRIVAL_DELAY;
DelayHistogramConfig config = GetDelayHistogramConfig();
const int quantile = config.quantile;
- std::unique_ptr<Histogram> histogram = absl::make_unique<Histogram>(
+ std::unique_ptr<Histogram> histogram = std::make_unique<Histogram>(
kDelayBuckets, config.forget_factor, config.start_forget_weight);
- return absl::make_unique<DelayManager>(
+ return std::make_unique<DelayManager>(
max_packets_in_buffer, base_minimum_delay_ms, quantile, mode,
enable_rtx_handling, peak_detector, tick_timer, statistics,
std::move(histogram));
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index 6979789..c691fd5 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -14,7 +14,8 @@
#include <math.h>
-#include "absl/memory/memory.h"
+#include <memory>
+
#include "modules/audio_coding/neteq/histogram.h"
#include "modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
#include "modules/audio_coding/neteq/mock/mock_histogram.h"
@@ -80,7 +81,7 @@
if (use_mock_histogram_) {
mock_histogram_ = new MockHistogram(kMaxIat, kForgetFactor);
std::unique_ptr<Histogram> histogram(mock_histogram_);
- dm_ = absl::make_unique<DelayManager>(
+ dm_ = std::make_unique<DelayManager>(
kMaxNumberOfPackets, kMinDelayMs, kDefaultHistogramQuantile,
histogram_mode_, enable_rtx_handling_, &detector_, &tick_timer_,
&stats_, std::move(histogram));
diff --git a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
index 21b15a9..49eb1cc 100644
--- a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
@@ -11,10 +11,10 @@
// Test to verify correct operation when using the decoder-internal PLC.
#include <algorithm>
+#include <memory>
#include <utility>
#include <vector>
-#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
@@ -156,31 +156,31 @@
// The input is mostly useless. It sends zero-samples to a PCM16b encoder,
// but the actual encoded samples will never be used by the decoder in the
// test. See below about the decoder.
- auto generator = absl::make_unique<ZeroSampleGenerator>();
+ auto generator = std::make_unique<ZeroSampleGenerator>();
constexpr int kSampleRateHz = 32000;
constexpr int kPayloadType = 100;
AudioEncoderPcm16B::Config encoder_config;
encoder_config.sample_rate_hz = kSampleRateHz;
encoder_config.payload_type = kPayloadType;
- auto encoder = absl::make_unique<AudioEncoderPcm16B>(encoder_config);
+ auto encoder = std::make_unique<AudioEncoderPcm16B>(encoder_config);
constexpr int kRunTimeMs = 10000;
- auto input = absl::make_unique<EncodeNetEqInput>(
+ auto input = std::make_unique<EncodeNetEqInput>(
std::move(generator), std::move(encoder), kRunTimeMs);
// Wrap the input in a loss function.
auto lossy_input =
- absl::make_unique<LossyInput>(loss_cadence, std::move(input));
+ std::make_unique<LossyInput>(loss_cadence, std::move(input));
// Settinng up decoders.
NetEqTest::DecoderMap decoders;
// Using a fake decoder which simply reads the output audio from a file.
- auto input_file = absl::make_unique<InputAudioFile>(
+ auto input_file = std::make_unique<InputAudioFile>(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
AudioDecoderPlc dec(std::move(input_file), kSampleRateHz);
// Masquerading as a PCM16b decoder.
decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1));
// Output is simply a checksum calculator.
- auto output = absl::make_unique<AudioChecksumWithOutput>(checksum);
+ auto output = std::make_unique<AudioChecksumWithOutput>(checksum);
// No callback objects.
NetEqTest::Callbacks callbacks;
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index ded54bf..39c4e52 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -14,7 +14,6 @@
#include <utility>
#include <vector>
-#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/accelerate.h"
#include "modules/audio_coding/neteq/expand.h"
@@ -109,7 +108,7 @@
config_.max_packets_in_buffer, config_.min_delay_ms, 1020054733,
DelayManager::HistogramMode::INTER_ARRIVAL_TIME,
config_.enable_rtx_handling, delay_peak_detector_, tick_timer_,
- deps.stats.get(), absl::make_unique<Histogram>(50, 32745)));
+ deps.stats.get(), std::make_unique<Histogram>(50, 32745)));
mock_delay_manager_ = mock.get();
deps.delay_manager = std::move(mock);
}
@@ -1567,7 +1566,7 @@
new rtc::RefCountedObject<test::FunctionAudioDecoderFactory>(
[sampling_freq, speech_type]() {
std::unique_ptr<AudioDecoder> decoder =
- absl::make_unique<Decoder120ms>(sampling_freq, speech_type);
+ std::make_unique<Decoder120ms>(sampling_freq, speech_type);
RTC_CHECK_EQ(2, decoder->Channels());
return decoder;
});
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 688ce8d..0b638bf 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -12,7 +12,8 @@
#include "modules/audio_coding/neteq/packet_buffer.h"
-#include "absl/memory/memory.h"
+#include <memory>
+
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
@@ -704,7 +705,7 @@
Packet packet_1 = gen.NextPacket(kPayloadSizeBytes, nullptr);
std::unique_ptr<MockEncodedAudioFrame> mock_audio_frame =
- absl::make_unique<MockEncodedAudioFrame>();
+ std::make_unique<MockEncodedAudioFrame>();
EXPECT_CALL(*mock_audio_frame, Duration())
.WillRepeatedly(Return(kFrameSizeSamples));
Packet packet_2 =
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
index 8147142..3c3add4 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -22,7 +22,6 @@
#include <string>
#include <utility>
-#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
@@ -194,19 +193,19 @@
// If an output file is requested, open it.
std::unique_ptr<AudioSink> output;
if (!config.output_audio_filename.has_value()) {
- output = absl::make_unique<VoidAudioSink>();
+ output = std::make_unique<VoidAudioSink>();
std::cout << "No output audio file" << std::endl;
} else if (config.output_audio_filename->size() >= 4 &&
config.output_audio_filename->substr(
config.output_audio_filename->size() - 4) == ".wav") {
// Open a wav file with the known sample rate.
- output = absl::make_unique<OutputWavFile>(*config.output_audio_filename,
- *sample_rate_hz);
+ output = std::make_unique<OutputWavFile>(*config.output_audio_filename,
+ *sample_rate_hz);
std::cout << "Output WAV file: " << *config.output_audio_filename
<< std::endl;
} else {
// Open a pcm file.
- output = absl::make_unique<OutputAudioFile>(*config.output_audio_filename);
+ output = std::make_unique<OutputAudioFile>(*config.output_audio_filename);
std::cout << "Output PCM file: " << *config.output_audio_filename
<< std::endl;
}
@@ -254,9 +253,8 @@
std::unique_ptr<AudioDecoder> decoder =
decoder_factory->MakeAudioDecoder(format, codec_pair_id);
if (!decoder && format.name == "replacement") {
- decoder = absl::make_unique<FakeDecodeFromFile>(
- absl::make_unique<InputAudioFile>(
- config.replacement_audio_file),
+ decoder = std::make_unique<FakeDecodeFromFile>(
+ std::make_unique<InputAudioFile>(config.replacement_audio_file),
format.clockrate_hz, format.num_channels > 1);
}
return decoder;
@@ -274,11 +272,11 @@
// Create a text log file if needed.
std::unique_ptr<std::ofstream> text_log;
if (config.textlog_filename.has_value()) {
- text_log = absl::make_unique<std::ofstream>(*config.textlog_filename);
+ text_log = std::make_unique<std::ofstream>(*config.textlog_filename);
}
NetEqTest::Callbacks callbacks;
- stats_plotter_ = absl::make_unique<NetEqStatsPlotter>(
+ stats_plotter_ = std::make_unique<NetEqStatsPlotter>(
config.matlabplot, config.pythonplot, config.concealment_events,
config.plot_scripts_basename.value_or(""));
@@ -291,9 +289,9 @@
neteq_config.sample_rate_hz = *sample_rate_hz;
neteq_config.max_packets_in_buffer = config.max_nr_packets_in_buffer;
neteq_config.enable_fast_accelerate = config.enable_fast_accelerate;
- return absl::make_unique<NetEqTest>(neteq_config, decoder_factory, codecs,
- std::move(text_log), std::move(input),
- std::move(output), callbacks);
+ return std::make_unique<NetEqTest>(neteq_config, decoder_factory, codecs,
+ std::move(text_log), std::move(input),
+ std::move(output), callbacks);
}
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index f864aa1..30f4f95 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -14,10 +14,10 @@
#include <iostream>
#include <limits>
+#include <memory>
#include <set>
#include <utility>
-#include "absl/memory/memory.h"
#include "logging/rtc_event_log/rtc_event_processor.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "rtc_base/checks.h"
@@ -96,7 +96,7 @@
&packet_ssrcs](const webrtc::LoggedRtpPacketIncoming& incoming) {
if (!filter_.test(incoming.rtp.header.payloadType) &&
incoming.log_time_us() < first_log_end_time_us) {
- rtp_packets_.emplace_back(absl::make_unique<Packet>(
+ rtp_packets_.emplace_back(std::make_unique<Packet>(
incoming.rtp.header, incoming.rtp.total_length,
incoming.rtp.total_length - incoming.rtp.header_length,
static_cast<double>(incoming.log_time_ms())));
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.cc b/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 410af27..f578065 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -18,7 +18,6 @@
#include <memory>
-#include "absl/memory/memory.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "rtc_base/checks.h"
#include "test/rtp_file_reader.h"
@@ -66,7 +65,7 @@
std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
RtpUtility::RtpHeaderParser parser(packet_memory.get(), temp_packet.length);
- auto packet = absl::make_unique<Packet>(
+ auto packet = std::make_unique<Packet>(
packet_memory.release(), temp_packet.length,
temp_packet.original_length, temp_packet.time_ms, parser,
&rtp_header_extension_map_);
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 5155958..d2c8d8a 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -10,9 +10,9 @@
#include "modules/audio_coding/test/TestRedFec.h"
+#include <memory>
#include <utility>
-#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
@@ -190,7 +190,7 @@
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = std::move(encoder);
- encoder = absl::make_unique<AudioEncoderCopyRed>(std::move(config));
+ encoder = std::make_unique<AudioEncoderCopyRed>(std::move(config));
receive_codecs.emplace(
std::make_pair(red_payload_type,
SdpAudioFormat("red", codec_format.clockrate_hz, 1)));