Sam Zackrisson 7219d053d5 Split aec and aecm into separate build targets ... This clarifies dependencies and makes it easier to customize builds for different binaries. Also adds BUILD files in aec/ and aecm/. Moves unit tests to their own target, which subjects them to Chromium Clang style checks. The CL contains a fix for a thusly induced warning. Bug: webrtc:9488 Change-Id: I77b680b42a4dccc5f025005e0890f60b4eaf2961 Reviewed-on: https://webrtc-review.googlesource.com/87304 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23887}		2018-07-09 14:48:06 +00:00
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audio_coding	Making PacketDuration always consistent with Decode in FakeDecodeFromFile.	2018-07-06 13:30:47 +00:00
audio_device	Use absl::make_unique and absl::WrapUnique directly	2018-07-05 10:59:49 +00:00
audio_mixer	Calculate all audio samples in AudioMixerCalculateEnergy.	2018-06-29 14:47:13 +00:00
audio_processing	Split aec and aecm into separate build targets	2018-07-09 14:48:06 +00:00
bitrate_controller	Use absl::make_unique and absl::WrapUnique directly	2018-07-05 10:59:49 +00:00
congestion_controller	Field trial to initialize throughput estimate faster.	2018-07-05 12:14:04 +00:00
desktop_capture	Use absl::make_unique and absl::WrapUnique directly	2018-07-05 10:59:49 +00:00
include	Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.	2018-07-05 14:29:07 +00:00
pacing	Use absl::make_unique and absl::WrapUnique directly	2018-07-05 10:59:49 +00:00
remote_bitrate_estimator	Use absl::make_unique and absl::WrapUnique directly	2018-07-05 10:59:49 +00:00
rtp_rtcp	Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.	2018-07-05 14:29:07 +00:00
utility	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
video_capture	Delete unused file.	2018-06-28 12:53:17 +00:00
video_coding	Change test VideoProcessor kMaxBufferedInputFrames from 10 to 20.	2018-07-09 11:56:43 +00:00
video_processing	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
BUILD.gn	Replace rtc::Optional with absl::optional	2018-06-21 09:32:56 +00:00
module_common_types_unittest.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
OWNERS	Moving src/webrtc into src/.	2017-09-15 04:25:06 +00:00