4 Commits

Author	SHA1	Message	Date
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
Nico Weber	22f9925b3e	webrtc: Remove semicolons. ... Bug: chromium:926235 Change-Id: I66c10ab3df38adf87152d1f18cc8162afedca7e4 Reviewed-on: https://webrtc-review.googlesource.com/c/123560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26780}	2019-02-20 16:02:59 +00:00
Yves Gerey	988cc0870b	[Cleanup] Add missing #include. Remove useless ones. ... This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}	2018-10-23 11:32:56 +00:00
Alex Loiko	4bb1e4a1d5	Lower gain parameters for AGC2. ... The AdaptiveAgc often boosts the signal outside of Float S16 range. It is expected, which is why we have a limiter after it in the process chain. But it turns out that this happens regularly even for simple input examples. The output signal peaks can be as high as +4 dBFs for a single speaker example (which should be easy). It leads to excessive gain modulation by the limiter. This CL is a new tuning designed to produce a safer gain. After this, we shouldn't hit the saturation region of the limiter as often. But we will still maintain a high gain. We have a 'configurable kill-switch': the settings can be changed via field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin. Bug: webrtc:7494, chromium:892043 Change-Id: I5014377050c74c32ae8998282991141eae31cf58 Reviewed-on: https://webrtc-review.googlesource.com/c/102922 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25006}	2018-10-05 09:55:25 +00:00