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Gaurav Vaish b249d0a905 Allow AudioAttributes to be app/client configurable
WebRtcAudioTrack is hardcoded to configure AudioAttributes with
1. usage=USAGE_VOICE_COMMUNICATIOON
2. contentType=CONTENT_TYPE_SPEECH

This change allows AudioAttributes to be configured via the
 JavaAudioDeviceModule.

Bug: webrtc:12153
Change-Id: I67c7f6e572c5a9f3a8fde674b6600d2adaf17895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191941
Commit-Queue: Gaurav Vaish <gvaish@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32583}
2020-11-11 06:18:10 +00:00
api Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +00:00
audio Enable continuous audio polling from ADM after StopPlay in VoIP API 2020-11-04 19:18:03 +00:00
build_overrides set perfetto flag to default value of false 2020-07-22 10:14:53 +00:00
call Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +00:00
common_audio Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +00:00
common_video Reland "Add scaling interface to VideoFrameBuffer" 2020-10-09 08:30:50 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs doc: mention video_replay tool for reporting video bugs 2020-10-30 11:22:01 +00:00
examples Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
logging Fix incorrect ToUnsigned in RTC event log. 2020-11-02 16:50:36 +00:00
media Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +00:00
modules Revert "RNN VAD: pitch search optimizations (part 1)" 2020-11-10 20:31:28 +00:00
p2p Check for oversized TURN usernames 2020-11-02 13:46:16 +00:00
pc Eliminate sigslot from RtpTransmissionManager 2020-11-10 14:41:45 +00:00
resources Increased high frequency transparency 2020-11-02 10:42:10 +00:00
rtc_base CallbackList: Improve documentation 2020-11-10 19:08:45 +00:00
rtc_tools Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +00:00
sdk Allow AudioAttributes to be app/client configurable 2020-11-11 06:18:10 +00:00
stats Fix -Wrange-loop-analysis. 2020-11-05 15:55:33 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Delete use of RWLockWrapper from SimulatedClock 2020-11-04 08:01:08 +00:00
test Analyze quality of dropped frames in VideoProcessor. 2020-10-29 08:23:49 +00:00
tools_webrtc Add new try builder configs to mb_config.pyl. 2020-11-05 09:36:13 +00:00
video Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +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 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h 2020-09-07 08:37:14 +00:00
.vpython Add 'requests' to .vpython. 2020-09-09 14:36:03 +00:00
abseil-in-webrtc.md Polish the "Using Abseil in WebRTC" docs 2020-10-16 13:42:00 +00:00
AUTHORS Fix "control reaches end of non-void function" warnings 2020-10-27 10:22:23 +00:00
BUILD.gn Rename RoboCaller to CallbackList. 2020-10-23 15:14:22 +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
DEPS Roll chromium_revision cf72651802..56e883537d (825867:825974) 2020-11-10 22:53:38 +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 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.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: move bug reporting instructions to the repository 2020-10-21 14:47:49 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Reland "Remove placeholder Obj-C headers and use angle-bracketed headers." 2020-10-22 11:29:48 +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
whitespace.txt Trigger bots. 2020-10-22 19:13:15 +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