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webrtc_m130/modules
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Alessio Bazzica 76714a6cc8 AGC2 minor code clean up 
Dead code removed plus const ref std::string to avoid copies.

Bug: webrtc:7494
Change-Id: Ic408a810ae310fea942f25fc697ab81017c8a739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201624
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32968}
2021-01-14 08:03:17 +00:00
..
async_audio_processing
Async audio processing API
2020-10-02 12:33:34 +00:00
audio_coding
Correction for the calculation of the abs max value
2021-01-12 16:28:00 +00:00
audio_device
Add support for toggling builtin voice processing on iOS
2020-12-02 16:07:01 +00:00
audio_mixer
Cleanup after sorting out dependencies of OutputRateCalculator
2020-10-20 13:01:29 +00:00
audio_processing
AGC2 minor code clean up
2021-01-14 08:03:17 +00:00
congestion_controller
Fix potential 32bit integer overflow on rtcp receiver report
2020-12-30 23:36:07 +00:00
desktop_capture
Refactor test window creation functions to be reusable.
2020-12-15 20:52:45 +00:00
include
Delete RTPFragmentationHeader as no longer used
2020-09-03 00:01:27 +00:00
pacing
Cleans up WebRTC-Pacer-SmallFirstProbePacket flag
2020-12-04 15:34:10 +00:00
remote_bitrate_estimator
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +00:00
rtp_rtcp
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +00:00
third_party
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
utility
Rename CriticalSection to RecursiveCriticalSection.
2020-07-17 09:19:50 +00:00
video_capture
Delete unneeded dependencies on the Module abstraction
2020-12-21 09:09:57 +00:00
video_coding
Fill fps allocation by LibaomAv1Encoder::GetEncoderInfo
2021-01-12 17:26:40 +00:00
video_processing
Delete unneeded dependencies on the Module abstraction
2020-12-21 09:09:57 +00:00
BUILD.gn
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
module_common_types_unittest.cc
…