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chromium-webrtc-autoroll 447fc3f555 Roll chromium_revision f5f7594337..e7ad7ca1d5 (1146453:1146741) 
Change log: f5f7594337..e7ad7ca1d5
Full diff: f5f7594337..e7ad7ca1d5

Changed dependencies
* fuchsia_vesion: version:12.20230519.0.1..version:12.20230519.1.1
* src/base: 4033618985..6c9c867e38
* src/build: 211e7d5672..03c786d588
* src/buildtools/linux64: git_revision:6975103d9f59ed9f3cfbc6d2201298f2c2fde046..git_revision:e9e83d9095d3234adf68f3e2866f25daf766d5c7
* src/buildtools/mac: git_revision:6975103d9f59ed9f3cfbc6d2201298f2c2fde046..git_revision:e9e83d9095d3234adf68f3e2866f25daf766d5c7
* src/buildtools/third_party/libc++abi/trunk: a4c3a6dc75..982e218cb8
* src/buildtools/win: git_revision:6975103d9f59ed9f3cfbc6d2201298f2c2fde046..git_revision:e9e83d9095d3234adf68f3e2866f25daf766d5c7
* src/ios: 2c5ac40524..6ede0267ed
* src/testing: 4f25c142e6..668dd0d405
* src/third_party: 727639a237..32cd23a328
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ac9d6ede9a..c6222c27ba
* src/third_party/depot_tools: b0e8e7adbc..9c32c9240c
* src/third_party/freetype/src: 99dadd56a4..115e927540
* src/third_party/perfetto: a9c0e2b0a6..2a96a77e3d
* src/tools: c6326ae214..6276212737
DEPS diff: f5f7594337..e7ad7ca1d5/DEPS

No update to Clang.

BUG=None

Change-Id: I4f78a95fe0691f2fbf540366cd06ad24a538f692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305773
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#40093}
2023-05-19 22:55:57 +00:00
api
[Stats] Align RTCStatsMember<T> closer to absl::optional<T>.
2023-05-11 07:33:54 +00:00
audio
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
build_overrides
Define enable_safe_libcxx in build_overrides/build.gni.
2023-05-03 08:18:25 +00:00
call
Reduce precision of RTT in RtrpTransportControllerSend
2023-05-19 10:38:37 +00:00
common_audio
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
common_video
webrtc_libyuv: Add support for more video types for consistency
2023-04-24 19:06:25 +00:00
data
…
docs
fix some more minor typos
2023-05-11 12:26:25 +00:00
examples
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Add docs about adding a new test binary.
2023-03-07 11:12:33 +00:00
infra
Increase timeout to 3h for bots that compile Chromium.
2023-05-09 09:18:40 +00:00
logging
Use DD encoder/decoder in RTC event log encoder/parser.
2023-04-24 10:35:22 +00:00
media
Guard send_codec variable against receive channel access
2023-05-15 11:10:35 +00:00
modules
Change ReceivedFecPacket to have list of ssrcs, seq nums and masks.
2023-05-16 06:07:48 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
pc
Refactor HasDataChannels
2023-05-15 14:52:12 +00:00
resources
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +00:00
rtc_base
Remove WEBRTC_EXTERNAL_JSON.
2023-05-10 21:34:52 +00:00
rtc_tools
Format the rest
2023-05-03 12:56:39 +00:00
sdk
Reland "Avoid recreating VirtualDisplay on format changes."
2023-05-19 12:49:05 +00:00
stats
stats: remove media_type which was an alias for kind
2023-05-09 11:46:52 +00:00
system_wrappers
Format the rest
2023-05-03 12:56:39 +00:00
test
Format the rest
2023-05-03 12:56:39 +00:00
tools_webrtc
Rewrite 'generate_sslroots' w/o OpenSSL.
2023-05-10 12:57:37 +00:00
video
Delete unused member in VideoSendStream
2023-05-12 11:42:00 +00:00
.clang-format
…
.git-blame-ignore-revs
Add formatting CLs to .git-blame-ignore-revs
2023-05-07 09:27:47 +00:00
.gitignore
…
.gn
Set use_cxx to true.
2023-05-17 06:30:04 +00:00
.mailmap
…
.style.yapf
…
.vpython
…
.vpython3
Rewrite 'generate_sslroots' w/o OpenSSL.
2023-05-10 12:57:37 +00:00
AUTHORS
Expose setCodecPreferences/getCapabilities for android
2023-05-15 19:24:15 +00:00
BUILD.gn
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
codereview.settings
…
DEPS
Roll chromium_revision f5f7594337..e7ad7ca1d5 (1146453:1146741)
2023-05-19 22:55:57 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
…
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
Add infra owners file
2022-12-02 09:21:47 +00:00
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
…
PRESUBMIT.py
Update portaudio to the latest
2022-05-13 09:01:34 +00:00
pylintrc
…
README.chromium
…
README.md
…
WATCHLISTS
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
whitespace.txt
Trigger bots
2023-05-16 08:24:54 +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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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