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Sebastian Jansson 01cb965d34 Moved ostream operators for network units to test. 
Added ToString functions in a separate header and move the ostream
operators to a test only header.

Bug: webrtc:8982
Change-Id: If547173aa39bbae2244531e2d3091886f14eae31
Reviewed-on: https://webrtc-review.googlesource.com/65280
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22674}
2018-03-29 11:21:37 +00:00
api Further decrease the AEC3 look window in the nonlinear mode 2018-03-28 18:15:57 +00:00
audio Delete OnIncomingCSRCChanged and related code. 2018-03-27 13:18:35 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Add audio_ prefix to audio-related members of CallTest. 2018-03-28 13:49:46 +00:00
common_audio Move aligned memory utilities to rtc_base/memory/ 2018-03-22 14:13:24 +00:00
common_video Remove the public_deps to fileutils from test_support. 2018-03-16 09:06:27 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Android: Rename AudioDeviceModule to JavaAudioDeviceModule 2018-03-29 10:55:37 +00:00
infra Reland "Add linux_internal_compile_lite trybot to CQ" 2018-03-26 10:12:59 +00:00
logging Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods. 2018-03-20 18:06:18 +00:00
media Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" 2018-03-29 02:45:17 +00:00
modules Moved ostream operators for network units to test. 2018-03-29 11:21:37 +00:00
ortc Remove rtc::Optional::MoveValue 2018-03-28 11:58:06 +00:00
p2p Remove LOG_J and LOG_JV, tweak p2p logs. 2018-03-29 08:21:27 +00:00
pc Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" 2018-03-29 02:45:17 +00:00
resources Adding FourPeople_1280x720_30.yuv. 2018-02-12 15:55:00 +00:00
rtc_base Annotate rest of WebRTC with @Nullable. 2018-03-28 08:30:06 +00:00
rtc_tools Fix path to proto in py_event_log_analyzer/pb_parse.py 2018-03-19 07:42:35 +00:00
sdk Android: Rename AudioDeviceModule to JavaAudioDeviceModule 2018-03-29 10:55:37 +00:00
stats PeerConnectionInterface::GetStats() with selector argument added. 2018-03-26 12:08:20 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Lowercase Windows headers. 2018-03-26 11:38:10 +00:00
test Add audio_ prefix to audio-related members of CallTest. 2018-03-28 13:49:46 +00:00
tools_webrtc Add jsr305 as a dependency to AAR. 2018-03-29 09:56:07 +00:00
video Adding layering configurator and rate allocator for VP9 SVC. 2018-03-29 10:16:47 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
.gn Re-enabling 'gn check' on //examples/*. 2018-02-19 15:07:45 +00:00
.vpython vpython: Specify dependency on pywin32 2018-02-14 13:56:39 +00:00
AUTHORS Expose a link-local network interfaces enumeration option 2018-02-06 19:12:04 +00:00
BUILD.gn Removing -Wno-strict-overflow from main BUILD.gn. 2018-03-28 11:25:07 +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.cc Adding layering configurator and rate allocator for VP9 SVC. 2018-03-29 10:16:47 +00:00
common_types.h Adding layering configurator and rate allocator for VP9 SVC. 2018-03-29 10:16:47 +00:00
DEPS Roll chromium_revision b73c062f19..59284db4e1 (546590:546773) 2018-03-29 09:14:47 +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 custom MD5 / SHA-1 implementations. 2018-02-19 15:03:35 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add back OWNERS entris that went missing 2018-03-06 10:31:11 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Revert "Relaxing no-streams presubmit check (streams are allowed in tests)." 2018-03-19 10:32:02 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +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 Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix 2018-02-05 11:24:59 +00:00
WATCHLISTS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
webrtc.gni Annotate rest of WebRTC with @Nullable. 2018-03-28 08:30:06 +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