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Gustaf Ullberg 2bab5ad3b1 AEC3: Avoid using filter output in suppression gain computation in non-linear mode
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.

This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.

Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
2019-04-15 16:08:41 +00:00
api Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
audio Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
common_audio WebRtcSpl AffineTransform: make input const 2019-04-15 10:27:55 +00:00
common_video Add explicit stride options to I420BufferPool. 2019-04-10 17:53:19 +00:00
crypto Adding new top-level directory crypto/ 2019-03-08 00:35:05 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Remove loopback video quality analysis test. 2019-04-12 13:03:22 +00:00
logging Add RtcEventLogFactory factory with explicit TaskQueueFactory 2019-04-11 16:05:09 +00:00
media Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
modules AEC3: Avoid using filter output in suppression gain computation in non-linear mode 2019-04-15 16:08:41 +00:00
p2p Make ExtraICEPing send slightly fewer extras 2019-04-12 18:54:53 +00:00
pc Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
resources AEC3: Avoid using filter output in suppression gain computation in non-linear mode 2019-04-15 16:08:41 +00:00
rtc_base Allow encoder target bitrate to reach media rate if there is headroom. 2019-04-15 15:11:39 +00:00
rtc_tools Add flag to enable shared x-axis for local event log visualization. 2019-04-12 09:12:46 +00:00
sdk Revert "Make negotiationneeded processing in PeerConnection spec compliant." 2019-04-12 16:14:07 +00:00
stats Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Delete unneeded direct includes of common_types.h 2019-04-01 07:18:13 +00:00
test Adds more stats to CallStatsCollector. 2019-04-15 14:47:56 +00:00
tools_webrtc Remove loopback video quality test from configs 2019-04-12 12:32:42 +00:00
video Allow encoder target bitrate to reach media rate if there is headroom. 2019-04-15 15:11:39 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore format commit. 2018-06-20 09:26:44 +00:00
.gitignore Add Visual Studio Code project folder to gitignore file. 2019-01-21 18:42:33 +00:00
.gn Bump iOS min supported version to 10.0 2019-03-07 13:08:17 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Ban absl::StrSplit and absl::StrJoin 2019-02-26 00:45:11 +00:00
AUTHORS Import proto_library.gni when rtc_enable_protobuf is true 2019-02-27 09:56:42 +00:00
BUILD.gn Replacing SequencedTaskChecker with SequenceChecker. 2019-04-09 12:28:04 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Move enum VideoType out of common_types.h 2019-04-08 09:47:54 +00:00
DEPS Roll chromium_revision 1a9381db11..8d55ca9363 (650638:650742) 2019-04-15 11:36:15 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Clean up root OWNERS. 2018-11-09 14:23:59 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Add a presubmit check for absl/memory/memory.h inclusion for WrapUnique 2019-02-28 14:12:48 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Modify pc/ WATCHLISTS definition 2019-01-31 22:09:16 +00:00
webrtc.gni Testing no /DUNICODE assumptions with Win more configs bots. 2019-03-28 08:46:37 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info