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Merged
merged 27 commits into from
Nov 22, 2024
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61fc0e6
Update to m125. (#119)
cloudwebrtc Jun 12, 2024
776abe4
fix: duplicate simulcast entries
santhoshvai Nov 20, 2024
7d5245f
remove duplicate declaration
santhoshvai Nov 20, 2024
55bd50a
remove duplicate audioDeviceModule
santhoshvai Nov 20, 2024
912e3e4
Merge branch 'patch/m125' into patch/m125-livekit-merge
santhoshvai Nov 21, 2024
750735e
fix: removed livekit's external audio processor
santhoshvai Nov 21, 2024
14f453d
fix: add back simulcast factories
santhoshvai Nov 21, 2024
1bb29b2
Fix missing RTC_OBJC_TYPE macros
hiroshihorie Jun 14, 2024
d151996
Fix missing headers and Metal linking
hiroshihorie Jun 14, 2024
42dd9be
Fix Mac Catalyst `RTCCameraVideoCapturer` rotation (#126)
hiroshihorie Jun 20, 2024
72a67d8
Fix set frame transformer (#125)
hiroshihorie Jun 20, 2024
51729e8
Fix webrtc_voice_engine not notifying mute change (#128)
davidliu Jul 9, 2024
c6ed18c
android: Allow for skipping checking the audio playstate if needed (#…
davidliu Jul 9, 2024
1f65597
Allow to pass in capture session to RTCCameraVideoCapturer (#132)
hiroshihorie Jul 16, 2024
d0d7d32
Fix NetworkMonitor race condition when dispatching native observers (…
davidliu Jul 18, 2024
209fb1e
Support for Vision Pro (#131)
cloudwebrtc Aug 12, 2024
b966148
Multicam support (#137)
hiroshihorie Aug 20, 2024
a71a9b3
tvOS support (#139)
hiroshihorie Aug 22, 2024
863aa99
Add isDisposed to MediaStreamTrack (#140)
davidliu Sep 2, 2024
b89e03c
chore: handle invalid cipher from key size. (#142)
cloudwebrtc Sep 14, 2024
64a7c71
Allow software AEC for Simulator (#143)
hiroshihorie Sep 23, 2024
0019150
Fix AudioRenderer crash & expose AVAudioPCMBuffer (#144)
hiroshihorie Sep 23, 2024
6c402bc
fix: Fix bug for bypass voice processing. (#147)
cloudwebrtc Oct 4, 2024
e3ea019
chore: remove aes cbc for framecryptor. (#145)
cloudwebrtc Oct 7, 2024
ce58a11
Change audio renderer output format (#149)
hiroshihorie Oct 19, 2024
d48dd25
Fixed issue with missing network interfaces on iOS (#151)
hiroshihorie Oct 19, 2024
5121c69
Custom audio input for Android (#154)
davidliu Oct 31, 2024
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6 changes: 6 additions & 0 deletions .gitignore
Original file line number Diff line number Diff line change
Expand Up @@ -72,3 +72,9 @@
/xcodebuild
/.vscode
!webrtc/*
/tmp.patch
/out-release
/out-debug
/node_modules
/libwebrtc
/args.txt
1 change: 1 addition & 0 deletions AUTHORS
Original file line number Diff line number Diff line change
Expand Up @@ -33,6 +33,7 @@ Christophe Dumez <ch.dumez@samsung.com>
Chris Tserng <tserng@amazon.com>
Cody Barnes <conceptgenesis@gmail.com>
Colin Plumb
Corby Hoback <corby.hoback@gmail.com>
Cyril Lashkevich <notorca@gmail.com>
CZ Theng <cz.theng@gmail.com>
Danail Kirov <dkirovbroadsoft@gmail.com>
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2 changes: 1 addition & 1 deletion DEPS
Original file line number Diff line number Diff line change
Expand Up @@ -54,7 +54,7 @@ deps = {
'src/base':
'https://chromium.googlesource.com/chromium/src/base@738cf0c976fd3d07c5f1853f050594c5295300d8',
'src/build':
'https://chromium.googlesource.com/chromium/src/build@cab574b350bc82dc3e7a1f634fedeb3079bf9e9d',
'https://github.com/webrtc-sdk/build@6978bac6466311e4bee4c7a9fd395faa939e0fcd',
'src/buildtools':
'https://chromium.googlesource.com/chromium/src/buildtools@5eb927f0a922dfacf10cfa84ee76f39dcf2a7311',
# Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
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26 changes: 26 additions & 0 deletions NOTICE
Original file line number Diff line number Diff line change
@@ -0,0 +1,26 @@
###################################################################################

The following modifications follow Apache License 2.0 from shiguredo.

https://github.com/webrtc-sdk/webrtc/commit/dfec53e93a0a1cb93f444caf50f844ec0068c7b7
https://github.com/webrtc-sdk/webrtc/commit/403b4678543c5d4ac77bd1ea5753c02637b3bb89
https://github.com/webrtc-sdk/webrtc/commit/77d5d685a90fb4bded17835ae72ec6671b26d696

Apache License 2.0

Copyright 2019-2021, Wandbox LLC (Original Author)
Copyright 2019-2021, Shiguredo Inc.

Licensed under the Apache License, Version 2.0 (the "License");
you may not use this file except in compliance with the License.
You may obtain a copy of the License at

http://www.apache.org/licenses/LICENSE-2.0

Unless required by applicable law or agreed to in writing, software
distributed under the License is distributed on an "AS IS" BASIS,
WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
See the License for the specific language governing permissions and
limitations under the License.

#####################################################################################
1 change: 1 addition & 0 deletions api/BUILD.gn
Original file line number Diff line number Diff line change
Expand Up @@ -368,6 +368,7 @@ rtc_library("libjingle_peerconnection_api") {
"video:encoded_image",
"video:video_bitrate_allocator_factory",
"video:video_frame",
"video:yuv_helper",
"video:video_rtp_headers",
"video_codecs:video_codecs_api",

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18 changes: 18 additions & 0 deletions api/crypto/BUILD.gn
Original file line number Diff line number Diff line change
Expand Up @@ -16,6 +16,24 @@ group("crypto") {
]
}

rtc_library("frame_crypto_transformer") {
visibility = [ "*" ]
sources = [
"frame_crypto_transformer.cc",
"frame_crypto_transformer.h",
]

deps = [
"//api:frame_transformer_interface",
]

if (rtc_build_ssl) {
deps += [ "//third_party/boringssl" ]
} else {
configs += [ ":external_ssl_library" ]
}
}

rtc_library("options") {
visibility = [ "*" ]
sources = [
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