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Sebastian Jansson c3eb9fd49f Reland "Reland "Only include overhead if using send side bandwidth estimation.""
This is a reland of 086055d0fd

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
2020-01-29 18:42:34 +00:00
api Allow non-identical datagram transport parameters. 2020-01-29 18:14:24 +00:00
audio Reland "Reland "Only include overhead if using send side bandwidth estimation."" 2020-01-29 18:42:34 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Reland "Reland "Only include overhead if using send side bandwidth estimation."" 2020-01-29 18:42:34 +00:00
common_audio Reformat GN files. 2020-01-21 12:13:11 +00:00
common_video Reformat GN files. 2020-01-21 12:13:11 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs remove mention of prebuilt libraries from docs/ 2020-01-28 16:08:44 +00:00
examples Reformat GN files. 2020-01-21 12:13:11 +00:00
logging Reformat GN files. 2020-01-21 12:13:11 +00:00
media Adding absolute capture timestamp to AudioTrackSinkInterface. 2020-01-29 13:46:28 +00:00
modules Reland "Reland "Only include overhead if using send side bandwidth estimation."" 2020-01-29 18:42:34 +00:00
p2p Allow non-identical datagram transport parameters. 2020-01-29 18:14:24 +00:00
pc Allow non-identical datagram transport parameters. 2020-01-29 18:14:24 +00:00
resources Reformat GN files. 2020-01-21 12:13:11 +00:00
rtc_base Makes all units and operations constexpr 2020-01-29 10:57:54 +00:00
rtc_tools Reformat GN files. 2020-01-21 12:13:11 +00:00
sdk Remove iceRegatherIntervalRange 2020-01-28 19:16:18 +00:00
stats [Stats] Include fecPackets[Reeceived/Discarded] in Members() 2020-01-28 11:22:09 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reformat GN files. 2020-01-21 12:13:11 +00:00
test Pass and store PacketBuffer::Packet by unique_ptr 2020-01-29 11:48:55 +00:00
tools_webrtc Remove strip_absolute_paths_from_debug_symbols from mb. 2020-01-27 09:17:46 +00:00
video Remove task_queue dependency for QualityScaler 2020-01-29 12:14:10 +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 Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +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 Update Android camera switch API to allow specifying a name 2020-01-09 16:04:09 +00:00
BUILD.gn Fix video_replay to build and actually work 2020-01-22 13:16:28 +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 fa85f826d0..dd5a54c29b (736081:736224) 2020-01-29 04:51:45 +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 Add #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +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.py Extract an interface from the perf results logger. 2020-01-14 06:05:02 +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 Add saza to audio watchlists 2019-09-03 14:55:43 +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
webrtc.gni Reformat GN files. 2020-01-21 12:13:11 +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