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Per Åhgren d4e6904d40 AEC3: Reducing the complexity and heap usage of the adaptive filter
This CL reduces the complexity and heap usage of the adaptive filter
in AEC3 by avoiding to compute these for the shadow
filter. In particular it
-Moves to compute the ERL, frequency response and impulse response
 on an on-demand basis.
-Stores the ERL, frequency response and impulse response outside
 of the adaptive filter.

All the changes have been tested for bitexactness on a sizeable
amount of recordings.

Bug: webrtc:10913
Change-Id: If83c236a6e3f2e489be129b9ebf6143a72f521d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151138
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29081}
2019-09-05 14:30:49 +00:00
api Delete deprecated rtc_event_log_factory_interface.h 2019-09-05 08:57:36 +00:00
audio Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
common_audio Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +00:00
common_video Add new FrameRateEstimator utility class for more precis FPS estimation. 2019-08-14 12:15:06 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Make the RtpHeaderParserImpl available to tests and tools only. 2019-08-29 15:56:40 +00:00
logging Delete deprecated rtc_event_log_factory_interface.h 2019-09-05 08:57:36 +00:00
media Buffers non atomic message send with usrsctp lib. 2019-09-03 16:30:21 +00:00
modules AEC3: Reducing the complexity and heap usage of the adaptive filter 2019-09-05 14:30:49 +00:00
p2p Use the sanitized pair when surfacing the candidate pair change event. 2019-09-03 17:17:49 +00:00
pc Delete old version of PeerConnection::SetConfiguration 2019-09-04 08:23:18 +00:00
resources Use the AEC3 high-pass filter for the whole APM 2019-08-23 20:04:10 +00:00
rtc_base Add support for stable bitrate target in SvcRateAllocator 2019-09-04 14:22:43 +00:00
rtc_tools Revert "Always create output_dir in setup_apprtc.py." 2019-09-04 18:19:12 +00:00
sdk Add support of AudioRecord.Builder in the ADM for Android 2019-09-05 07:59:30 +00:00
stats Add support for RTCTransportStats.selectedCandidatePairChanges 2019-08-28 13:22:08 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Add helper functions to convert between integer milliseconds and fixed-point seconds. 2019-08-16 14:49:46 +00:00
test Revert "Make relative arrival delay mode default in NetEq delay manager." 2019-09-05 11:59:53 +00:00
tools_webrtc Add license for android_ndk 2019-08-28 14:57:28 +00:00
video Add support for stable bitrate target in SvcRateAllocator 2019-09-04 14:22:43 +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 new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. 2019-09-04 18:49:28 +00:00
.gn Remove last mention of ortc from the codebase. 2019-05-25 07:28:05 +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 Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
AUTHORS Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly 2019-06-28 19:12:14 +00:00
BUILD.gn Make the RtpHeaderParserImpl available to tests and tools only. 2019-08-29 15:56:40 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Roll chromium_revision e96090c328..7e5c36432b (693514:693630) 2019-09-05 11:08:49 +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 Add juberti@ to webrtc root owners 2019-05-17 18:11:58 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
PRESUBMIT.py Disable RunPythonTests on rtc_tools. 2019-08-06 12:48:33 +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 Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +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