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Ying Wang fb226af64d Remove some old logging in goog_cc for congestion window.
Bug: None
Change-Id: I05550e5099cd7b4bc9512d2ce4159222779c02a7
Reviewed-on: https://webrtc-review.googlesource.com/c/105326
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25118}
2018-10-11 16:08:57 +00:00
api Revert "Add ability to specify if rate controller of video encoder is trusted." 2018-10-11 15:37:40 +00:00
audio Makes AudioSendStream signal that it's part of allocation. 2018-10-10 09:47:46 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Delete post_encode_callback 2018-10-11 08:18:08 +00:00
common_audio Add support for many channels in push_resampler. 2018-10-02 13:11:51 +00:00
common_video Delete post_encode_callback 2018-10-11 08:18:08 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Delete unused includes of assert.h 2018-10-04 14:01:44 +00:00
infra Reland "Use new internal trybot in commit queue" 2018-09-25 11:02:03 +00:00
logging Use zero octets for rtp packet padding 2018-10-11 10:22:36 +00:00
media Revert "Add ability to specify if rate controller of video encoder is trusted." 2018-10-11 15:37:40 +00:00
modules Remove some old logging in goog_cc for congestion window. 2018-10-11 16:08:57 +00:00
p2p Revert "Use unique_ptr and ArrayView in SSLFingerprint" 2018-10-11 11:16:01 +00:00
pc Revert "Use unique_ptr and ArrayView in SSLFingerprint" 2018-10-11 11:16:01 +00:00
resources Drop legacy AEC metrics interface from ApmTest.Process 2018-09-14 08:16:43 +00:00
rtc_base Revert "Use unique_ptr and ArrayView in SSLFingerprint" 2018-10-11 11:16:01 +00:00
rtc_tools Print per-frame VMAF score instead of average. 2018-10-10 14:31:00 +00:00
sdk Add certificate gen/set functionality to bring Android closer to JS API 2018-10-10 13:37:47 +00:00
stats Move rtc_json code from API dir, enable unit test, unmark testonly 2018-10-02 15:21:26 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Remove noop system_wrappers_default build target. 2018-10-04 10:25:37 +00:00
test Adds flags indicating presence in allocation and feedback per packet. 2018-10-09 18:24:38 +00:00
tools_webrtc Remove ortc folder. 2018-10-10 09:06:57 +00:00
video Revert "Add ability to specify if rate controller of video encoder is trusted." 2018-10-11 15:37:40 +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 Reland "Compile frame analyzer for the host machine on perf tests." 2018-09-18 09:51:19 +00:00
.gn Re-enable gtest absl pretty printers. 2018-08-13 13:54:05 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Add a list of allowed and disallowed Abseil things 2018-08-27 08:37:20 +00:00
AUTHORS Generate module.modulemap file when building Mac Framework 2018-10-08 12:32:45 +00:00
BUILD.gn Add rtc_enable_symbol_export to incrementally create a WebRTC component. 2018-10-10 11:04:34 +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 Remove double declaration of 2 conversion functions. 2018-09-11 06:00:05 +00:00
DEPS Roll chromium_revision 2499289737..f34485ffde (598606:598711) 2018-10-11 10:33:01 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Add documentation about field_trial/metrics custom impl. 2018-09-18 11:27:59 +00:00
OWNERS Add a list of allowed and disallowed Abseil things 2018-08-27 08:37:20 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Allow the autoroller service account to commit 2018-10-04 09:25:03 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +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 a list of allowed and disallowed Abseil things 2018-08-27 08:37:20 +00:00
WATCHLISTS Adds srte to WATCHLISTS. 2018-10-11 11:52:28 +00:00
webrtc.gni Add rtc_enable_symbol_export to incrementally create a WebRTC component. 2018-10-10 11:04:34 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

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