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Niels Möller 67309ef93c Add release callback and reference count to java EncodedImage class
Callback set by HardwareVideoEncoder, and wired to the codec's
releaseOutputBuffer. Intention is to move call of this method to the
destructor of a corresponding C++ class in a followup cl, and
eliminate an allocation and memcpy in the process.

Bug: webrtc:9378
Change-Id: I578480b63b68e6ac7a96cdde36379b3c50f05c3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142160
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29283}
2019-09-24 12:26:09 +00:00
api Delete AudioDecoder method IncomingPacket 2019-09-24 08:30:24 +00:00
audio Propagating TargetRate struct to BitrateAllocator. 2019-09-19 14:03:04 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Propagating TargetRate struct to BitrateAllocator. 2019-09-19 14:03:04 +00:00
common_audio Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
common_video Always pass arguments to INSTANTIATE_TEST_SUITE_P. 2019-09-24 08:56:24 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fixing some typos. 2019-09-10 10:03:50 +00:00
examples New build target api:media_interface 2019-09-19 09:32:27 +00:00
logging Adds remote estimates to rtc event log. 2019-09-19 09:22:37 +00:00
media Remove obsolete todo comment in simulcast.h 2019-09-20 14:17:20 +00:00
modules Always pass arguments to INSTANTIATE_TEST_SUITE_P. 2019-09-24 08:56:24 +00:00
p2p Delete the BasicPortAllocator constructor that enables gturn 2019-09-24 07:46:28 +00:00
pc Always pass arguments to INSTANTIATE_TEST_SUITE_P. 2019-09-24 08:56:24 +00:00
resources Use the AEC3 high-pass filter for the whole APM 2019-08-23 20:04:10 +00:00
rtc_base Handle macro _M_ARM64 for MSVC build 2019-09-24 08:34:04 +00:00
rtc_tools Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
sdk Add release callback and reference count to java EncodedImage class 2019-09-24 12:26:09 +00:00
stats Add qualityLimitationResolutionChanges stat 2019-09-09 15:22:57 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Move code related to VideoCodingModule to its own build target 2019-09-10 12:34:38 +00:00
test Delete AudioDecoder method IncomingPacket 2019-09-24 08:30:24 +00:00
tools_webrtc Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots. 2019-09-12 18:21:38 +00:00
video Always pass arguments to INSTANTIATE_TEST_SUITE_P. 2019-09-24 08:56:24 +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 Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. 2019-09-04 18:49:28 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +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 Update style guide for absl::make_unique. 2019-09-18 06:10:58 +00:00
AUTHORS Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5." 2019-09-06 05:36:23 +00:00
BUILD.gn Introduce api/crypto/BUILD.gn. 2019-09-13 17:21:47 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Roll chromium_revision 1d4ed9e21d..9f21b695c1 (699120:699240) 2019-09-24 08:35:09 +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 Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add juberti@ to webrtc root owners 2019-05-17 18:11:58 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py absl::make_unique presubmit check. 2019-09-17 17:47:31 +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 Update WebRTC's C++ style guide to reflect the switch to C++14. 2019-09-16 11:45:35 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +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