4 commits

Author	SHA1	Message	Date
Ali Tofigh	714e3cbb48	Adopt absl::string_view in modules/audio_coding/ ... Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}	2022-07-20 13:34:23 +00:00
Jonas Olsson	a4d873786f	Format almost everything. ... This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}	2019-07-08 13:45:15 +00:00
Danil Chapovalov	b602123a5a	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}	2018-06-19 12:46:20 +00:00
Henrik Lundin	3ef3bfc2aa	Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent ... These two new histograms relate to the packet-loss concealment that happens when audio packets are lost or late for decoding, and the NetEq must resort to extrapolating audio from the previously decoded data. Bug: webrtc:9126 Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b Reviewed-on: https://webrtc-review.googlesource.com/67861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22812}	2018-04-10 21:32:55 +00:00