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Per K ce2b49552e Set webrtc::PacketOptions.packet_id from
RtpPacketToSend::transport_sequence_number

packed_id is set to be 64 bit to align with rtc::PacketOptions.
packet_id is only set to RtpPacketToSend::transport_sequence_number if
TransportSequenceNumber header extension is not used in order to not
change current behaviour.

Bug: webrtc:15368
Change-Id: Ia532714226421422bdb292f8dd34b175560e9dc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41950}
2024-03-22 11:56:57 +00:00
api Set webrtc::PacketOptions.packet_id from 2024-03-22 11:56:57 +00:00
audio Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
build_overrides Roll chromium_revision 2024-03-15 07:58:00 +00:00
call Update WebRTC code version (2024-03-22T04:11:01). 2024-03-22 05:14:52 +00:00
common_audio Provide test output path with OutputPathWithRandomDirectory 1/n 2024-02-15 07:35:00 +00:00
common_video Deprecate VideoFrame::timestamp() and set_timestamp 2024-03-13 11:08:37 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs add xctest to gn args for ios sim 2024-03-05 08:37:00 +00:00
examples Remove remaining .cc files from rtc_media_base 2024-03-12 14:09:38 +00:00
experiments Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
g3doc Update TODO example in the style guide. 2023-11-21 23:11:09 +00:00
infra Re-enable msan bots 2024-03-15 10:47:02 +00:00
logging Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
media Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
modules Set webrtc::PacketOptions.packet_id from 2024-03-22 11:56:57 +00:00
net/dcsctp Reapply "dcsctp: Add per-stream-limit, refactor limits." 2024-03-15 13:27:37 +00:00
p2p Fix ubsan warning in ParseError testcase 2024-03-21 15:58:22 +00:00
pc pc: Remove additional buffering in SctpDataChannel 2024-03-22 09:25:11 +00:00
resources Ignore .binarypb files. 2023-10-30 14:56:36 +00:00
rtc_base Use propagated field trials for WebRTC-NormalizeSimulcastResolution experiment 2024-03-20 16:58:59 +00:00
rtc_tools Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
sdk Fix UIDeviceOrientation enums. 2024-03-21 11:07:28 +00:00
stats [Stats] Migrate from the RTCStatsMember type alias to absl::optional. 2024-01-25 21:56:08 +00:00
system_wrappers Use //third_party/cpu_features directly 2023-06-02 07:17:36 +00:00
test Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
tools_webrtc Add directory for ChromiumOS specific tools 2024-02-26 03:46:59 +00:00
video DoesUtilizeUlpfecForVp9WithNackEnabled is flaky. 2024-03-22 09:30:10 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Disable checks for googletest for WebRTC 2024-02-27 12:05:12 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Configure YAPF to follow PEP-8 altogether 2023-09-22 10:32:11 +00:00
.vpython3 Update to vpython 3.11 and remove .vpython (v2.x) 2024-01-25 11:12:20 +00:00
AUTHORS Expose setCodecPreferences/getCapabilities for iOS 2024-01-23 13:54:26 +00:00
BUILD.gn Add environment_construction poison 2023-11-27 11:44:50 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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 cbc33d5567..566136c383 (1274686:1274799) 2024-03-19 10:56:42 +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
LICENSE
license_template.txt
native-api.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
PATENTS
PRESUBMIT.py Implement Newline Check in the Presubmit 2024-01-23 07:50:56 +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 Configure Pylint to follow PEP-8 2023-09-25 15:56:09 +00:00
pylintrc_old_style Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
README.chromium [ssci] Added Shipped field to READMEs 2023-07-12 07:31:06 +00:00
README.md doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c 2023-05-26 09:20:16 +00:00
WATCHLISTS Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Add environment_construction poison 2023-11-27 11:44:50 +00:00
webrtc_lib_link_test.cc Deprecate RtcEventLogFactory constructor taking unused parameter 2023-12-07 21:46:56 +00:00
whitespace.txt Revert "Test new tree." 2024-01-31 08:50:04 +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 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