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Jonas Oreland 70a59b6900 Flip default value of AndroidNetworkMonitor field trials 
getAllNetworksFromCache => true (stop using old Android API)
requestVPN => true (is default in old api)

They have been enabled using field trial
for more than a year.

Bug: webrtc:13741
Change-Id: I288c4067193e95251f79d51e935dce555f6eb198
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361581
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42945}
2024-09-04 11:20:34 +00:00
api Delete deprecated AudioDecoderFactory::MakeAudioDecoder 2024-09-04 07:17:59 +00:00
audio Apply include-cleaner to call/ 2024-09-03 07:51:03 +00:00
build_overrides build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
call Add FrameInstrumentationData to RTPVideoHeader and CodecSpecificInfo 2024-09-04 07:21:02 +00:00
common_audio Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
common_video Rename is_key_frame to communicate_upper_bits in FrameInstrumentation*Data 2024-09-03 10:42:18 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fix formatting for corruption detection header explainer. 2024-08-27 15:18:27 +00:00
examples Adding ChannelStatistics Logs 2024-09-02 20:50:58 +00:00
experiments Add WebRTC-MixedCodecSimulcast field trial 2024-09-04 08:45:44 +00:00
g3doc Update docs about supported platforms and compilers. 2024-09-04 10:30:16 +00:00
infra Revert "Enable 'iwyu_verifier' bot." 2024-09-02 10:14:35 +00:00
logging Fix lint issues in logging/ 2024-09-04 07:58:47 +00:00
media Add WebRTC-MixedCodecSimulcast field trial 2024-09-04 08:45:44 +00:00
modules Add FrameInstrumentationData to RTPVideoHeader and CodecSpecificInfo 2024-09-04 07:21:02 +00:00
net/dcsctp Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
p2p Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
pc Add WebRTC-MixedCodecSimulcast field trial 2024-09-04 08:45:44 +00:00
resources Delete unused YUV files 2024-07-11 20:26:16 +00:00
rtc_base rewrite SSLInfoCallback logging 2024-09-02 17:17:20 +00:00
rtc_tools Delete deprecated AudioDecoderFactory::MakeAudioDecoder 2024-09-04 07:17:59 +00:00
sdk Flip default value of AndroidNetworkMonitor field trials 2024-09-04 11:20:34 +00:00
stats Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
system_wrappers Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
test Delete deprecated AudioDecoderFactory::MakeAudioDecoder 2024-09-04 07:17:59 +00:00
tools_webrtc Fix gtest/gmock includes in apply-include-cleaner script. 2024-09-03 10:47:16 +00:00
video Add fieldtrials WebRTC-QCM-Static-{AV1, VP8, VP9} 2024-09-03 11:27:39 +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 Roll chromium_revision ba1ae79f58..6f9b3224db (1319128:1338914) 2024-08-08 09:20:02 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Configure YAPF to follow PEP-8 altogether 2023-09-22 10:32:11 +00:00
.vpython3 Update to vpython 3.11 and remove .vpython (v2.x) 2024-01-25 11:12:20 +00:00
AUTHORS Adding ChannelStatistics Logs 2024-09-02 20:50:58 +00:00
BUILD.gn build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +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 Roll chromium_revision 01d6daf051..66e155a442 (1350197:1350363) 2024-09-03 20:49:21 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Update Abseil instructions for absl::optional 2024-09-02 14:32:11 +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 Configure Pylint to follow PEP-8 2023-09-25 15:56:09 +00:00
pylintrc_old_style Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
README.chromium Add Security Critical field to README.chromium. 2024-08-27 07:38:26 +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 AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
webrtc_lib_link_test.cc Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
whitespace.txt Test CQ 2024-05-27 12:46:57 +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