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C++ 89.3%
Java 2.9%
C 2.8%
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Michael Morrison c2302e8e2e Fix compile error when rtc_enable_protobuf is false 
When configuring without protobuf this test fails to compile with the error:
perf_test_histogram_writer_no_protobuf.cc:20:1: error: non-void function does not return a value

Bug: None
Change-Id: I8e2676ee4b5284eac08e648fc43bdfc585fc5d64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182740
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32021}
2020-08-31 23:07:13 +00:00
api Move ABSL_MUST_USE_RESULT at the beginning of the method decl. 2020-08-31 08:19:29 +00:00
audio DTMF Event Sub-API on VoIP API 2020-08-20 17:10:02 +00:00
build_overrides set perfetto flag to default value of false 2020-07-22 10:14:53 +00:00
call Move FrameCounts and FrameCountObserver to common_video/frame_counts.h 2020-08-27 09:53:18 +00:00
common_audio Support AVX2/FMA intrinsics in audio FIR filter 2020-08-19 10:21:31 +00:00
common_video Move FrameCounts and FrameCountObserver to common_video/frame_counts.h 2020-08-27 09:53:18 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Updated IOS documentation with correct build_ios_libs.py path 2020-08-17 09:37:59 +00:00
examples Update gradle wrapper & gradle plugin. 2020-08-28 08:58:57 +00:00
logging Ensure RtcEventLogEncoderNewFormat::EncodeRemoteEstimate handles infite 2020-08-25 09:22:49 +00:00
media [adaptation] Expose target pixels and max framerate in VideoAdapter 2020-08-31 09:46:21 +00:00
modules Move ABSL_MUST_USE_RESULT at the beginning of the method decl. 2020-08-31 08:19:29 +00:00
p2p Add WebRTC-IceFieldTrial send_ping_on_selected_ice_controlling 2020-08-27 12:07:04 +00:00
pc Fix destruction order of PortAllocator and PacketSocketFactory. 2020-08-31 21:52:27 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Fix ABA problem when iterating epoll events. 2020-08-31 20:26:37 +00:00
rtc_tools Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools 2020-08-12 09:17:15 +00:00
sdk update JavaAudioDeviceModule.Builder.build() to return JavaAudioDeviceModule 2020-08-26 17:43:37 +00:00
stats Reland "Implement packets_(sent | received) for RTCTransportStats" 2020-07-10 11:50:59 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +00:00
test Fix compile error when rtc_enable_protobuf is false 2020-08-31 23:07:13 +00:00
tools_webrtc Fix missing isolated output directory. 2020-08-31 11:40:10 +00:00
video Reland "[Adaptation] Remove QualityScalerResource when disabled." 2020-08-26 06:33:43 +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 Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
AUTHORS Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder 2020-08-05 09:41:49 +00:00
BUILD.gn Delete unneeded dependencies on deprecated build target webrtc_common 2020-08-25 07:33:12 +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 FrameCounts and FrameCountObserver to common_video/frame_counts.h 2020-08-27 09:53:18 +00:00
DEPS Roll chromium_revision 4bf6d39883..a50ca66bb1 (802598:802712) 2020-08-28 18:42:54 +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
native-api.md Make transient suppression optionally excludable via defines 2020-04-02 11:44:07 +00:00
OWNERS Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +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_test_mocks.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +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
whitespace.txt Trigger CI bots. 2020-07-15 17:50:55 +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