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philipel e9a74c918b Public RtpVideoFrameAssembler 
This class takes RtpPacketReceived and assembles them into RtpFrameObjects.

Change-Id: Ia9785d069fecccc1d5b81efd257f33c8bd7a778b
Bug: webrtc:7408, webrtc:12579
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222580
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34364}
2021-06-24 15:20:42 +00:00
api Public RtpVideoFrameAssembler 2021-06-24 15:20:42 +00:00
audio ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
common_audio Avoid undefined behavior in a division operation. 2021-04-23 07:49:24 +00:00
common_video Update BitBuffer methods to style guide 2021-05-18 11:10:27 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Create a VideoFrameTrackingId RTP header extension. 2021-03-25 17:25:18 +00:00
examples Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. 2021-06-23 09:56:20 +00:00
g3doc Add RecursiveCriticalSection to the don't-use list of primitives 2021-06-22 09:40:47 +00:00
logging Reland "Deprecate microsecond timestamps in RTC event log." 2021-06-17 12:08:54 +00:00
media Minor code cleanup of WebRtcVideoReceiveStream. 2021-06-22 08:09:48 +00:00
modules AV1 OBU test helper. 2021-06-23 13:43:50 +00:00
net/dcsctp dcsctp: Add DcSctpSocketFactory 2021-06-18 09:59:40 +00:00
p2p Reland "Port: migrate to TaskQueue." 2021-06-21 22:21:04 +00:00
pc Revert "Fix echo return loss stats and add to RTCAudioSourceStats." 2021-06-22 08:10:50 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Update dependencies on deprecated target rtc_base:critical_section 2021-06-23 07:01:42 +00:00
rtc_tools Delete legacy RtpHeaderParser wrapper 2021-06-21 09:17:52 +00:00
sdk Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. 2021-06-21 22:47:44 +00:00
stats Revert "Fix echo return loss stats and add to RTCAudioSourceStats." 2021-06-22 08:10:50 +00:00
system_wrappers Make Clock::ConvertTimestampToNtpTime pure virtual 2021-05-21 09:55:14 +00:00
test ModuleRtpRtcpImpl2: update test code. 2021-06-21 23:36:49 +00:00
tools_webrtc Remove unnused build configs for M1 builder 2021-06-17 09:37:50 +00:00
video ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +00:00
.vpython Update six library version 2021-04-26 16:39:07 +00:00
AUTHORS Added PeerConnectionObserverJni::OnRemoveTrack() 2021-06-03 19:24:55 +00:00
BUILD.gn Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. 2021-06-23 09:56:20 +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 6ade74989a..6f7025c98c (893176:893293) 2021-06-17 05:00:35 +00:00
DIR_METADATA Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
g3doc.lua Improve webrtc documentation infra. Preview at: 2021-03-30 10:29:30 +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 the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +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 Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. 2021-06-23 09:56:20 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Triggering build after flaky builders (asan). 2021-06-22 17:20:54 +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 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