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Marina Ciocea 3e9af7fe05 Insert audio frame transformer between depacketizer and decoder.
The frame transformer is passed from RTPReceiverInterface through the
library to be eventually set in ChannelReceive, where the frame
transformation will occur in the follow-up CL.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I5af06d1431047ef50d00e304cf95e92a832b4220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30956}
2020-04-01 08:15:53 +00:00
api Introduce TransformableFrameInterface. 2020-03-30 13:35:26 +00:00
audio Insert audio frame transformer between depacketizer and decoder. 2020-04-01 08:15:53 +00:00
build_overrides Purge phoglund from most OWNERS files. 2020-03-09 14:08:30 +00:00
call Insert audio frame transformer between depacketizer and decoder. 2020-04-01 08:15:53 +00:00
common_audio Changed fft4g to be built as C++ 2020-03-20 12:10:16 +00:00
common_video Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Rename index.md to README.md to make it automatically show up 2020-03-06 10:43:51 +00:00
examples Chromium refactor: Replace "resources_dirs" with "sources" 2020-03-25 19:32:15 +00:00
logging Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
media Insert audio frame transformer between depacketizer and decoder. 2020-04-01 08:15:53 +00:00
modules Make prioritized RTX padding default again 2020-04-01 07:20:36 +00:00
p2p Changing from hard to soft error when temporary DTLS buffer is full. 2020-04-01 07:50:17 +00:00
pc Insert audio frame transformer between depacketizer and decoder. 2020-04-01 08:15:53 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Add test for relay bandwidth capping. 2020-03-30 13:02:46 +00:00
rtc_tools Add packet rate plots to event_log_visualizer. 2020-03-30 14:46:41 +00:00
sdk Expose enableDscp in Obj-C API. 2020-03-31 19:58:15 +00:00
stats Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API 2020-03-11 12:08:32 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Add RTC_EXPORT to webrtc::Clock 2020-03-10 23:00:21 +00:00
test Remove the histogram flag and all Chart JSON code. 2020-03-28 13:44:43 +00:00
tools_webrtc Revert "Trigger CI bots." 2020-03-28 22:37:03 +00:00
video [InsertableStreams] Set video frame transformer if RTP stream already started. 2020-03-31 14:07:29 +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 Fix bad frees in error paths of WebRtcAecm_Create 2020-03-21 23:01:47 +00:00
BUILD.gn Audio egress implementation for initial voip api in api/voip. 2020-03-27 18:45:43 +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 Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery" 2020-02-07 08:23:58 +00:00
DEPS Roll chromium_revision a0d7df3386..4d555ede52 (754491:754603) 2020-03-30 21:15:24 +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 Delete unused I420 "codec" 2018-12-18 12:30:58 +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 Fix public_deps presubmit and gn format fighting each other. 2020-01-30 11:22:46 +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_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 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.gni Reformat GN files. 2020-01-21 12:13:11 +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 Whitespace change 2020-01-28 07:53:15 +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 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