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}
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@ -10,7 +10,8 @@
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include "absl/memory/memory.h"
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#include <memory>
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#include "absl/types/variant.h"
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#include "modules/rtp_rtcp/source/rtp_format_h264.h"
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#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
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@ -33,7 +34,7 @@ std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
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const RTPFragmentationHeader* fragmentation) {
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if (!type) {
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// Use raw packetizer.
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return absl::make_unique<RtpPacketizerGeneric>(payload, limits);
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return std::make_unique<RtpPacketizerGeneric>(payload, limits);
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}
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switch (*type) {
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@ -41,21 +42,21 @@ std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
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RTC_CHECK(fragmentation);
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const auto& h264 =
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absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header);
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return absl::make_unique<RtpPacketizerH264>(
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return std::make_unique<RtpPacketizerH264>(
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payload, limits, h264.packetization_mode, *fragmentation);
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}
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case kVideoCodecVP8: {
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const auto& vp8 =
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absl::get<RTPVideoHeaderVP8>(rtp_video_header.video_type_header);
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return absl::make_unique<RtpPacketizerVp8>(payload, limits, vp8);
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return std::make_unique<RtpPacketizerVp8>(payload, limits, vp8);
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}
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case kVideoCodecVP9: {
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const auto& vp9 =
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absl::get<RTPVideoHeaderVP9>(rtp_video_header.video_type_header);
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return absl::make_unique<RtpPacketizerVp9>(payload, limits, vp9);
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return std::make_unique<RtpPacketizerVp9>(payload, limits, vp9);
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}
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default: {
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return absl::make_unique<RtpPacketizerGeneric>(
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return std::make_unique<RtpPacketizerGeneric>(
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payload, limits, rtp_video_header, frame_type);
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}
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}
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