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Erik Språng cf6544aef7 Avoids unnecessary calls to audio encoder.
As of this CL:
https://webrtc-review.googlesource.com/c/src/+/173704
...we now call AudioEncoder::OnReceivedOverhead() too often, since we
don't check if overhead has actually changed.
This CL rectifies that.

Bug: webrtc:10809
Change-Id: I0b86e0296a7860dde3e62e817f9b941fa82afe4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175009
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31279}
2020-05-15 15:26:46 +00:00
api Reland "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase." 2020-05-15 10:20:03 +00:00
audio Avoids unnecessary calls to audio encoder. 2020-05-15 15:26:46 +00:00
build_overrides Purge phoglund from most OWNERS files. 2020-03-09 14:08:30 +00:00
call In call/ replace mock macros with unified MOCK_METHOD macro 2020-05-15 13:36:00 +00:00
common_audio Merge OOURA BUILD.gn files. 2020-04-21 09:06:37 +00:00
common_video In GenericFrameInfo do not set frame_id and frame_diff 2020-05-15 09:43:34 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs docs: Improve Android testing docs 2020-05-14 08:05:37 +00:00
examples Wrap WebRTC OBJC API types with RTC_OBJC_TYPE. 2020-05-04 15:01:26 +00:00
logging Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
media FakeNetworkInterface: remove lock recursions. 2020-05-15 09:49:17 +00:00
modules Introduce layering controller interface for av1 encoder 2020-05-15 15:25:42 +00:00
p2p Clear address:port in icecandidateerror for tcp servers with private IP 2020-05-15 11:30:20 +00:00
pc Clear address:port in icecandidateerror for tcp servers with private IP 2020-05-15 11:30:20 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Reland "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase." 2020-05-15 10:20:03 +00:00
rtc_tools Exclude frame_analyzer_host build on iOS. 2020-04-20 21:53:54 +00:00
sdk objc: Export RTCStatistics and RTCStatisticsReport 2020-05-14 13:26:01 +00:00
stats Reland "Improve outbound-rtp statistics for simulcast" 2020-05-05 20:22:19 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Do not trigger -Wimplicit-int-float-conversion 2020-04-09 11:43:14 +00:00
test Re-enable absl FailureSignalHandler in tests. 2020-05-15 13:54:16 +00:00
tools_webrtc Fix the name of the file generated by generate_sslroots.py. 2020-04-25 13:37:30 +00:00
video In GenericFrameInfo do not set frame_id and frame_diff 2020-05-15 09:43:34 +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 Fix typo in abseil-in-webrtc.md. 2019-12-18 14:27:34 +00:00
AUTHORS docs: Improve Android testing docs 2020-05-14 08:05:37 +00:00
BUILD.gn VoIP API implementation on top of AudioIngress/Egress 2020-05-05 19:55:29 +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 Replaces OverheadObserver with simple getter. 2020-05-07 17:33:45 +00:00
DEPS Roll chromium_revision 9330db6b76..8ffd72401d (768590:768712) 2020-05-14 11:12:54 +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_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +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.py Fix public_deps presubmit and gn format fighting each other. 2020-01-30 11:22:46 +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 Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md Add guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +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
webrtc.gni Android: Remove min_sdk_version in GN (reland) 2020-05-13 15:46:45 +00:00
whitespace.txt Trigger bots again. 2020-05-11 05:38:59 +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 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