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Jonas Olsson d000b0a32e Move RTC_CHECK_OP error message construction out of header file. 
This simplifies the logic, prevents emitting code for every pair of
argument types to RTC_CHECK_OP and partially unblocks removing streams from
the check code altogether.

Bug: webrtc:8982
Change-Id: Ib6652ac9a342e4471c12574a79872833cc943407
Reviewed-on: https://webrtc-review.googlesource.com/86544
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23821}
2018-07-03 15:21:13 +00:00
api Removing deadbeef@ from OWNERS files. 2018-07-02 00:40:38 +00:00
audio Remove StreamStatistician::IsPacketInOrder 2018-06-28 08:44:40 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. 2018-06-27 10:33:40 +00:00
common_audio Extract fft4g into separate build target 2018-06-26 13:39:25 +00:00
common_video Add bit depth information to PlanarYuvBuffer 2018-06-26 20:23:37 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Remove usage of //build/config/clang:extra_warnings. 2018-07-03 12:35:49 +00:00
infra Flip luci.webrtc.try to production 2018-05-30 08:30:00 +00:00
logging Port RtcEventLog encoder unittests to the new parser API. 2018-06-28 15:31:23 +00:00
media Add sprang@ as owner for simulcast.cc/h 2018-07-03 15:00:33 +00:00
modules Discard frame self-dependency when parsing genric frame descriptor 2018-07-03 10:28:05 +00:00
ortc Removing deadbeef@ from OWNERS files. 2018-07-02 00:40:38 +00:00
p2p Fix a bug in TurnServer that causes flakiness in webrtc_perf_tests. 2018-07-02 18:51:23 +00:00
pc Add ADAPTER_TYPE_ANY in AdapterType. 2018-07-02 17:59:11 +00:00
resources AGC2 RNN VAD: Polishing. 2018-05-15 16:41:02 +00:00
rtc_base Move RTC_CHECK_OP error message construction out of header file. 2018-07-03 15:21:13 +00:00
rtc_tools Removes redundant delay based bwe. 2018-07-02 09:11:33 +00:00
sdk Remove usage of //build/config/clang:extra_warnings. 2018-07-03 12:35:49 +00:00
stats Adding "is_standardized" flag to RTCStatsMember. 2018-06-28 00:43:46 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Use the sparse histogram in RTC_HISTOGRAM_ENUMERATION_SPARSE. 2018-06-27 00:41:29 +00:00
test Discard frame self-dependency when parsing genric frame descriptor 2018-07-03 10:28:05 +00:00
tools_webrtc roll_deps: Accept any prefix (like 'git_revision:'), not only 'version:' for CIPD 2018-07-03 09:41:53 +00:00
video Remove StreamStatistician::IsPacketInOrder 2018-06-28 08:44:40 +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 format commit. 2018-06-20 09:26:44 +00:00
.gitignore Roll back checking in the third_party directory 2018-06-27 13:04:08 +00:00
.gn Set gtest_enable_absl_printers to true. 2018-06-29 09:36:17 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
AUTHORS Generalize SimulcastEncoderAdapter, use for H264 & VP8. 2018-06-21 15:57:43 +00:00
BUILD.gn A new PeerConnection level perf test. 2018-06-27 23:19:05 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
DEPS Roll chromium_revision f6935ecdd2..ce19c6d80b (572058:572160) 2018-07-03 10:09:25 +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
LICENSE_THIRD_PARTY Remove third party dependecies that are not more in the source code 2018-06-21 11:33:41 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add mbonadei@ to build configs OWNERS. 2018-06-20 12:39:11 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Roll back checking in the third_party directory 2018-06-27 13:04:08 +00:00
presubmit_test.py Roll chromium_revision 95336cb92b..191d55580e (557816:557824) 2018-05-11 11:17:05 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +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 style guidance about forward declarations. 2018-03-28 20:58:27 +00:00
typedefs.h Remove typedefs.h from webrtc/ root (part 1) 2018-05-23 12:07:10 +00:00
WATCHLISTS Adding mbonadei@ to build_files WATCHLIST. 2018-06-20 12:38:06 +00:00
webrtc.gni Don't call deprecated FFmpeg API. 2018-06-26 13:57:35 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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