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Sam Zackrisson 6f38d25f11 Add Java support for AudioProcessing and PostProcessing injection 
This allows injection of a user-defined post processing module from
the Android layer.

Bug: webrtc:8163
Change-Id: If3a6b4726c34c5f82d186b8cf95373c283cbd3f6
Reviewed-on: https://webrtc-review.googlesource.com/7610
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20367}
2017-10-20 08:37:23 +00:00
api Revert "BWE allocation strategy" 2017-10-19 15:34:52 +00:00
audio Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Revert "BWE allocation strategy" 2017-10-19 15:34:52 +00:00
common_audio Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_video Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv" 2017-10-16 16:05:37 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Revert "BWE allocation strategy" 2017-10-19 15:34:52 +00:00
infra Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
logging Prevent unbounded memory consumption through RtcEventLogImpl::config_history_ 2017-10-13 10:47:26 +00:00
media Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
modules Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
ortc Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers. 2017-09-27 09:14:28 +00:00
p2p Rewrite WebRtcSession ICE tests as PeerConnection tests 2017-10-13 18:42:34 +00:00
pc Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
resources Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtc_base Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
rtc_tools header_usage.sh: Keep leading directory for grep. 2017-10-19 13:46:51 +00:00
sdk Add Java support for AudioProcessing and PostProcessing injection 2017-10-20 08:37:23 +00:00
stats Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
system_wrappers Delete static_instance.h and fix_interlocked_exchange_pointer_win.h 2017-10-05 13:19:21 +00:00
test Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
tools_webrtc Revert "Add suppression for issue 8405" 2017-10-19 18:08:27 +00:00
video Simplify setting/unsetting REMB in RtcpSender 2017-10-19 14:07:31 +00:00
voice_engine Revert "BWE allocation strategy" 2017-10-19 15:34:52 +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 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. 2017-10-02 16:57:09 +00:00
.gn Remove remaining mentions of gflags 2017-09-25 15:34:41 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) 2017-09-25 13:37:12 +00:00
BUILD.gn Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." 2017-09-29 13:48:29 +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 NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_types.h Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. 2017-10-06 13:41:14 +00:00
DEPS Roll chromium_revision 65e1b24a78..f4ecd4bed3 (508708:508787) 2017-10-16 11:00:27 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
OWNERS Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +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 Update README.md and codereview.settings for new source location 2017-09-13 19:54:59 +00:00
style-guide.md Style guide: Attempt to make the L2 and L3 headings more visually distinct 2017-09-09 03:52:23 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Add myself to the watchlist for webrtc/api/ and webrtc/base/ 2017-05-04 13:22:46 +00:00
webrtc.gni GN rtc_* templates: Forward global "visibility" 2017-10-16 11:04:07 +00:00
whitespace.txt Moving src/webrtc into src/. 2017-09-15 04:25:06 +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.

More info