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317a1f09ed
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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|---|---|---|
| .. | ||
| audio_checksum.h | ||
| audio_loop.cc | ||
| audio_loop.h | ||
| audio_sink.cc | ||
| audio_sink.h | ||
| constant_pcm_packet_source.cc | ||
| constant_pcm_packet_source.h | ||
| DEPS | ||
| encode_neteq_input.cc | ||
| encode_neteq_input.h | ||
| fake_decode_from_file.cc | ||
| fake_decode_from_file.h | ||
| input_audio_file.cc | ||
| input_audio_file.h | ||
| input_audio_file_unittest.cc | ||
| neteq_delay_analyzer.cc | ||
| neteq_delay_analyzer.h | ||
| neteq_event_log_input.cc | ||
| neteq_event_log_input.h | ||
| neteq_input.cc | ||
| neteq_input.h | ||
| neteq_packet_source_input.cc | ||
| neteq_packet_source_input.h | ||
| neteq_performance_test.cc | ||
| neteq_performance_test.h | ||
| neteq_quality_test.cc | ||
| neteq_quality_test.h | ||
| neteq_replacement_input.cc | ||
| neteq_replacement_input.h | ||
| neteq_rtpplay.cc | ||
| neteq_rtpplay_test.sh | ||
| neteq_stats_getter.cc | ||
| neteq_stats_getter.h | ||
| neteq_stats_plotter.cc | ||
| neteq_stats_plotter.h | ||
| neteq_test.cc | ||
| neteq_test.h | ||
| neteq_test_factory.cc | ||
| neteq_test_factory.h | ||
| output_audio_file.h | ||
| output_wav_file.h | ||
| packet.cc | ||
| packet.h | ||
| packet_source.cc | ||
| packet_source.h | ||
| packet_unittest.cc | ||
| README.md | ||
| resample_input_audio_file.cc | ||
| resample_input_audio_file.h | ||
| rtc_event_log_source.cc | ||
| rtc_event_log_source.h | ||
| rtp_analyze.cc | ||
| rtp_encode.cc | ||
| rtp_file_source.cc | ||
| rtp_file_source.h | ||
| rtp_generator.cc | ||
| rtp_generator.h | ||
| rtp_jitter.cc | ||
| rtpcat.cc | ||
NetEQ RTP Play tool
Testing of the command line arguments
The command line tool neteq_rtpplay can be tested by running neteq_rtpplay_test.sh, which is not use on try bots, but it can be used before submitting any CLs that may break the behavior of the command line arguments of neteq_rtpplay.
Run neteq_rtpplay_test.sh as follows from the src/ folder:
src$ ./modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh \
out/Default/neteq_rtpplay \
resources/audio_coding/neteq_opus.rtp \
resources/short_mixed_mono_48.pcm
You can replace the RTP and PCM files with any other compatible files.
If you get an error using the files indicated above, try running gclient sync.
Requirements: awk and md5sum.