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Jeremy Leconte 3d6e88e6ac Remove low_bandwidth_audio_test. 
Change-Id: Ide4d34e1dada9dc1448f89a79cc7b803ea4b5f46
Bug: b/284448060
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307160
Reviewed-by: Henrik Lundin <hlundin@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40191}
2023-06-01 07:20:38 +00:00
api Updating AsyncAudioProcessing API, part 1. 2023-05-31 14:40:35 +00:00
audio Remove low_bandwidth_audio_test. 2023-06-01 07:20:38 +00:00
build_overrides Define enable_safe_libcxx in build_overrides/build.gni. 2023-05-03 08:18:25 +00:00
call Update WebRTC code version (2023-06-01T04:12:34). 2023-06-01 05:57:57 +00:00
common_audio Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
common_video webrtc_libyuv: Add support for more video types for consistency 2023-04-24 19:06:25 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs docs: explain release note process 2023-05-24 14:09:54 +00:00
examples Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Add docs about adding a new test binary. 2023-03-07 11:12:33 +00:00
infra Remove low_bandwidth_audio_test. 2023-06-01 07:20:38 +00:00
logging Use DD encoder/decoder in RTC event log encoder/parser. 2023-04-24 10:35:22 +00:00
media Updating AsyncAudioProcessing API, part 1. 2023-05-31 14:40:35 +00:00
modules Update ReceiveStatistics to use Timestamp/TimeDelta to represent time 2023-05-31 16:07:30 +00:00
net/dcsctp Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
p2p Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
pc sdp: reject BUNDLE with RTP header extension id collisions 2023-05-30 10:58:27 +00:00
resources Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00
rtc_base Remove unused Win32Window class 2023-05-25 10:48:40 +00:00
rtc_tools Format the rest 2023-05-03 12:56:39 +00:00
sdk More systematic null checks before calling native methods 2023-05-30 09:06:21 +00:00
stats Delete RTC[NonStandard/Restricted]StatsMember. 2023-05-25 08:39:48 +00:00
system_wrappers Format the rest 2023-05-03 12:56:39 +00:00
test In RtcpReceiver remove redundand way to represent RTCP report blocks 2023-05-28 15:24:46 +00:00
tools_webrtc Remove low_bandwidth_audio_test. 2023-06-01 07:20:38 +00:00
video Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay 2023-05-26 13:34:09 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Set use_cxx to true. 2023-05-17 06:30:04 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Rewrite 'generate_sslroots' w/o OpenSSL. 2023-05-10 12:57:37 +00:00
AUTHORS Expose setCodecPreferences/getCapabilities for android 2023-05-15 19:24:15 +00:00
BUILD.gn Remove low_bandwidth_audio_test. 2023-06-01 07:20:38 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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
DEPS Add //third_party/cpu_features to DEPS 2023-05-31 08:41:26 +00:00
DIR_METADATA Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS Remove phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +00:00
LICENSE
license_template.txt
native-api.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add infra owners file 2022-12-02 09:21:47 +00:00
PATENTS
PRESUBMIT.py Remove low_bandwidth_audio_test. 2023-06-01 07:20:38 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
pylintrc tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c 2023-05-26 09:20:16 +00:00
WATCHLISTS Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
webrtc_lib_link_test.cc Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
whitespace.txt Trigger bots 2023-05-16 08:24:54 +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 here 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