signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Per Åhgren 9136abb45a AEC3: Ensure that the data size in the reverb computer is not fixed 
This CL ensures that the no data vectors in the reverb computer code
are fixed. This allows arbitrary long filters to be used, and ensures
that a minimum required heap size is used.

Bug: webrtc:8671
Change-Id: I7085ed262a3f5965d796270434b6578f4030606e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162661
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30115}
2019-12-19 16:35:56 +00:00
api AEC3: Ensure that the data size in the reverb computer is not fixed 2019-12-19 16:35:56 +00:00
audio Minor fixes to ChannelSend. 2019-12-02 09:30:51 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Revert "Extracts ssrc based feedback tracking from feedback adapter." 2019-12-13 14:47:48 +00:00
common_audio Remove potential real-time reallocation in PushResampler 2019-12-11 13:16:37 +00:00
common_video Add ability to resize buffers pool in decoder and use it in IVF generator 2019-12-16 14:51:16 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Refresh some links in the docs folder. 2019-12-10 15:32:48 +00:00
examples Fix errorprone issues preventing Chromium Roll. 2019-11-27 12:52:48 +00:00
logging Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_. 2019-11-29 09:45:50 +00:00
media Parse max-fr and max-fs from SDP FMTP line 2019-12-19 15:00:41 +00:00
modules AEC3: Ensure that the data size in the reverb computer is not fixed 2019-12-19 16:35:56 +00:00
p2p Stop an SCTP connection when the DTLS transport closes. 2019-12-16 14:54:56 +00:00
pc Revert "Reland "Distinguish between send and receive video codecs"" 2019-12-16 15:28:41 +00:00
resources Make the high-pass filter operate in full-band 2019-12-18 16:01:24 +00:00
rtc_base Use a fake clock for rtc::Thread::PostDelayedTask test 2019-12-16 15:46:48 +00:00
rtc_tools Move audioproc_f to rtc_tools. 2019-12-10 12:08:56 +00:00
sdk Revert "Reland "Distinguish between send and receive video codecs"" 2019-12-16 15:28:41 +00:00
stats Add totalInterFrameDelay to RTCInboundRTPStreamStats 2019-11-25 10:50:37 +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 directive to make webrtc metrics optional. 2019-12-09 13:55:50 +00:00
test Don't allow creation of sockets for wild card IPs in emulated networks. 2019-12-17 10:16:59 +00:00
tools_webrtc [UBSan] Remove suppression for opus. 2019-12-10 08:59:30 +00:00
video Signal requested resolution alignment requirements from sinks to sources. 2019-12-19 10:39:04 +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 Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Fix typo in abseil-in-webrtc.md. 2019-12-18 14:27:34 +00:00
AUTHORS Revert "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""" 2019-12-09 11:26:20 +00:00
BUILD.gn Initial version of ResourceAdaptationProcessor and friends. 2019-12-10 15:31:43 +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 98124fd660..26cf7e7d6c (725465:725941) 2019-12-18 15:46:24 +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 #COMPONENT to WebRTC. 2019-10-08 12:20:39 +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 Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
PRESUBMIT.py Add possibility to skip check_includes presubmit check. 2019-10-22 19:35:31 +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 Add guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
webrtc.gni Do not disable metrics by default. 2019-12-11 08:08:58 +00:00
whitespace.txt Revert "Whitespace change" 2019-11-11 14:58:20 +00:00

README.md

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