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Byoungchan Lee 10a7d23be5 Fix degradation_preference setting being ignored using RtpSender.SetParameters. 
RtpSenderBase::SetParametersInternal stores init_parameters_
if media_channel_ does not exist. When RtpSenderBase::SetSsrc is called,
init_parameters_ is used to set the initial encoding parameters and
degradation_preference. However, if no encoding parameter is specified,
degradation_preference will not be set.

This CL modifies the RtpSender so that degradation_preference is not
ignored even in this case.

Bug: webrtc:14279
Change-Id: I7e95ecdf5fcb19037e4f118981d1314d78ffca5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268960
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37574}
2022-07-20 13:48:27 +00:00
api Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
audio Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
common_audio Adopt absl::string_view in common_audio/ 2022-05-13 15:00:14 +00:00
common_video Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs [ios] Remove the support for bitcode 2022-07-04 09:01:52 +00:00
examples Delete ProcessThread and related Module interface 2022-07-04 10:20:35 +00:00
g3doc Clarify how to reference WebRTC bugs in TODOs 2022-07-01 08:03:34 +00:00
infra Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing." 2022-07-06 20:28:26 +00:00
logging Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
media Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
modules Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
net/dcsctp Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
p2p [TCPConnection] Check for valid port_ in OnClose and OnConnect. 2022-07-19 15:56:43 +00:00
pc Fix degradation_preference setting being ignored using RtpSender.SetParameters. 2022-07-20 13:48:27 +00:00
resources AEC3: Changing the default for the use_conservative_tail_frequency_response flag. 2021-12-21 17:35:26 +00:00
rtc_base Update TaskQueueWin implementation to absl::AnyInvocable 2022-07-20 12:49:44 +00:00
rtc_tools Add lower/upper link capacity to the outgoing bitrate graph. 2022-07-19 13:22:32 +00:00
sdk Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
stats Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
system_wrappers Delete rtc_base/atomic_ops.h 2022-06-28 08:32:13 +00:00
test Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
tools_webrtc Clobber win bots 2022-07-19 11:34:02 +00:00
video Cleanup configuration of max reordering threshold 2022-07-20 13:04:53 +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 Prevent jsoncpp from hiding deprecated declarations in WebRTC 2022-04-11 12:33:47 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Update protobuf-py2_py3 wheel. 2022-07-01 15:17:36 +00:00
AUTHORS Add missing header to fix build error when using linux system libraries 2022-07-19 12:25:42 +00:00
BUILD.gn SVC: Add end to end tests for VP8 and VP9 2022-06-22 11:07:01 +00:00
CODE_OF_CONDUCT.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +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 3a2eeb8205..bfeef78a75 (1026014:1026121) 2022-07-20 06:33:56 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +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 Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Fix add some eng prod owners to PRESUBMIT.py. 2022-03-18 13:19:07 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
pylintrc tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Update WATCHLISTS 2021-08-23 13:37:55 +00:00
webrtc.gni [Cast Convergence] Replace is_chromecast with new args 2022-06-16 00:50:08 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger CI bots 2021-12-16 17:45:31 +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