commit 48a9e230e614972e4a73c49f911383474e50a368
parent 90ae408774c8d084dd5b52114fc3926bfb704b4e
Author: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Oct 2025 15:56:06 -0500
Bug 1993083 - Vendor libwebrtc from a473b69c06
We already cherry-picked this when we vendored fa780492c9.
Upstream commit: https://webrtc.googlesource.com/src/+/a473b69c069021883d485f68f8d46e8d6cc4e4b0
Revert "Record audio timestamp across all ObjC ADMs."
This reverts commit fa780492c902b0b0079386178b21b9c61221f25e.
Reason for revert: Breaks first party projects. Use #include instead of #import for "rtc_base/checks.h"
Bug: webrtc:13609
Original change's description:
> Record audio timestamp across all ObjC ADMs.
>
> Added audio timestamp capture to ObjCAudioDeviceModule and extracted
> nanoseconds from AudioTimeStamp construction into helper function.
>
> The timestamp capture was initially implemented in the CL:
> https://webrtc-review.googlesource.com/c/src/+/334720
>
> No-Try: True
> Bug: webrtc:13609
> Change-Id: I68d5c29ccc98cf817365988fa825875841e32ee7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/397160
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45000}
Bug: webrtc:13609
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ife0052ff113f70d67dc6290d63691842d977cce7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/397741
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45001}
Diffstat:
5 files changed, 80 insertions(+), 340 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-08T20:54:50.810232+00:00.
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-08T20:55:56.972570+00:00.
# base of lastest vendoring
-fa780492c9
+a473b69c06
diff --git a/third_party/libwebrtc/moz-patch-stack/a473b69c06.no-op-cherry-pick-msg b/third_party/libwebrtc/moz-patch-stack/a473b69c06.no-op-cherry-pick-msg
@@ -1 +0,0 @@
-We already cherry-picked this when we vendored fa780492c9.
diff --git a/third_party/libwebrtc/moz-patch-stack/p0001.patch b/third_party/libwebrtc/moz-patch-stack/p0001.patch
@@ -1,259 +1,48 @@
-From: Michael Froman <mjfroman@mac.com>
-Date: Wed, 8 Oct 2025 15:54:39 -0500
-Subject: (tmp-cherry-pick) Revert "Record audio timestamp across all ObjC
- ADMs." (a473b69c06)
+From: Gennady Tsitovich <gtsitovich@google.com>
+Date: Tue, 15 Jul 2025 08:24:50 +0000
+Subject: (cherry-pick-branch-heads/7258) [M139] Add chrome-cherry-picker
+ account to bot allowlist
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
-This reverts commit fa780492c902b0b0079386178b21b9c61221f25e.
-
-Reason for revert: Breaks first party projects. Use #include instead of #import for "rtc_base/checks.h"
-
-Bug: webrtc:13609
Original change's description:
-> Record audio timestamp across all ObjC ADMs.
->
-> Added audio timestamp capture to ObjCAudioDeviceModule and extracted
-> nanoseconds from AudioTimeStamp construction into helper function.
+> Add chrome-cherry-picker account to bot allowlist
>
-> The timestamp capture was initially implemented in the CL:
-> https://webrtc-review.googlesource.com/c/src/+/334720
+> chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com is
+> being by the Chrome Cherry Picker (go/chromecherrypicker) and needs to
+> be able to skip the author check for presubmits.
>
-> No-Try: True
-> Bug: webrtc:13609
-> Change-Id: I68d5c29ccc98cf817365988fa825875841e32ee7
-> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/397160
+> Bug: chromium:414375466
+> Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
+> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400480
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
-> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
-> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
-> Cr-Commit-Position: refs/heads/main@{#45000}
+> Reviewed-by: Björn Terelius <terelius@webrtc.org>
+> Commit-Queue: Gennady Tsitovich <gtsitovich@google.com>
+> Cr-Commit-Position: refs/heads/main@{#45148}
-Bug: webrtc:13609
-No-Presubmit: true
-No-Tree-Checks: true
-No-Try: true
-Change-Id: Ife0052ff113f70d67dc6290d63691842d977cce7
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/397741
-Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
-Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
-Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
-Cr-Commit-Position: refs/heads/main@{#45001}
+Bug: chromium:431157710,chromium:414375466
+Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400700
+Commit-Queue: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
+Auto-Submit: Chrome Cherry Picker <chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com>
+Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
+Cr-Commit-Position: refs/branch-heads/7258@{#2}
+Cr-Branched-From: 74fa937f86ed8432c07676f7a1ce0e5e2812b3d5-refs/heads/main@{#44974}
---
- sdk/BUILD.gn | 23 ---------
- sdk/objc/helpers/AudioTimeStamp+Nanoseconds.h | 16 ------
- .../helpers/AudioTimeStamp+Nanoseconds.mm | 51 -------------------
- sdk/objc/native/src/audio/audio_device_ios.h | 3 ++
- sdk/objc/native/src/audio/audio_device_ios.mm | 8 +--
- sdk/objc/native/src/objc_audio_device.mm | 8 +--
- 6 files changed, 10 insertions(+), 99 deletions(-)
- delete mode 100644 sdk/objc/helpers/AudioTimeStamp+Nanoseconds.h
- delete mode 100644 sdk/objc/helpers/AudioTimeStamp+Nanoseconds.mm
+ PRESUBMIT.py | 2 ++
+ 1 file changed, 2 insertions(+)
-diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
-index 3db2086d9e..26fc06efa8 100644
---- a/sdk/BUILD.gn
-+++ b/sdk/BUILD.gn
-@@ -184,27 +184,6 @@ if (is_ios || is_mac) {
- }
- }
-
-- rtc_library("core_audio_helpers_objc") {
-- sources = [
-- "objc/helpers/AudioTimeStamp+Nanoseconds.h",
-- "objc/helpers/AudioTimeStamp+Nanoseconds.mm",
-- ]
--
-- deps = [
-- ":base_objc",
-- "../rtc_base:checks",
-- ]
--
-- frameworks = [ "CoreAudio.framework" ]
--
-- configs += [
-- "..:common_objc",
-- ":used_from_extension",
-- ]
--
-- public_configs = [ ":common_config_objc" ]
-- }
--
- if (!build_with_chromium) {
- rtc_library("callback_logger_objc") {
- sources = [
-@@ -331,7 +310,6 @@ if (is_ios || is_mac) {
- ":audio_objc",
- ":audio_session_observer",
- ":base_objc",
-- ":core_audio_helpers_objc",
- "../api:array_view",
- "../api:scoped_refptr",
- "../api:sequence_checker",
-@@ -523,7 +501,6 @@ if (is_ios || is_mac) {
-
- deps = [
- ":audio_device_api_objc",
-- ":core_audio_helpers_objc",
- "../api:array_view",
- "../api:make_ref_counted",
- "../api:refcountedbase",
-diff --git a/sdk/objc/helpers/AudioTimeStamp+Nanoseconds.h b/sdk/objc/helpers/AudioTimeStamp+Nanoseconds.h
-deleted file mode 100644
-index 46233d8223..0000000000
---- a/sdk/objc/helpers/AudioTimeStamp+Nanoseconds.h
-+++ /dev/null
-@@ -1,16 +0,0 @@
--/*
-- * Copyright 2025 The WebRTC project authors. All Rights Reserved.
-- *
-- * Use of this source code is governed by a BSD-style license
-- * that can be found in the LICENSE file in the root of the source
-- * tree. An additional intellectual property rights grant can be found
-- * in the file PATENTS. All contributing project authors may
-- * be found in the AUTHORS file in the root of the source tree.
-- */
--
--#import <CoreAudioTypes/CoreAudioTypes.h>
--
--#include <optional>
--
--std::optional<int64_t> AudioTimeStampGetNanoseconds(
-- const AudioTimeStamp* timeStamp);
-diff --git a/sdk/objc/helpers/AudioTimeStamp+Nanoseconds.mm b/sdk/objc/helpers/AudioTimeStamp+Nanoseconds.mm
-deleted file mode 100644
-index bb11dbdfaa..0000000000
---- a/sdk/objc/helpers/AudioTimeStamp+Nanoseconds.mm
-+++ /dev/null
-@@ -1,51 +0,0 @@
--/*
-- * Copyright 2025 The WebRTC project authors. All Rights Reserved.
-- *
-- * Use of this source code is governed by a BSD-style license
-- * that can be found in the LICENSE file in the root of the source
-- * tree. An additional intellectual property rights grant can be found
-- * in the file PATENTS. All contributing project authors may
-- * be found in the AUTHORS file in the root of the source tree.
-- */
--
--#import "AudioTimeStamp+Nanoseconds.h"
--
--#import <mach/mach_time.h>
--#import "rtc_base/checks.h"
--
--std::optional<int64_t> AudioTimeStampGetNanoseconds(
-- const AudioTimeStamp* timeStamp) {
-- if (!timeStamp || ((timeStamp->mFlags & kAudioTimeStampHostTimeValid) == 0) ||
-- timeStamp->mHostTime == 0) {
-- return std::nullopt;
-- }
--
-- static mach_timebase_info_data_t mach_timebase;
-- if (mach_timebase.denom == 0) {
-- if (mach_timebase_info(&mach_timebase) != KERN_SUCCESS) {
-- RTC_DCHECK_NOTREACHED() << "mach_timebase_info bad return code";
-- return std::nullopt;
-- }
-- }
--
-- if (mach_timebase.denom == 0 || mach_timebase.numer == 0) {
-- RTC_DCHECK_NOTREACHED() << "mach_timebase_info provided bad data";
-- return std::nullopt;
-- }
--
-- uint64_t time_scaled = 0;
-- if (__builtin_umulll_overflow(
-- timeStamp->mHostTime, mach_timebase.numer, &time_scaled)) {
-- RTC_DCHECK_NOTREACHED() << "numeric overflow computing scaled host time";
-- return std::nullopt;
-- }
--
-- uint64_t nanoseconds = time_scaled / mach_timebase.denom;
-- if (nanoseconds >
-- static_cast<uint64_t>(std::numeric_limits<int64_t>::max())) {
-- RTC_DCHECK_NOTREACHED() << "numeric overflow computing nanoseconds";
-- return std::nullopt;
-- }
--
-- return static_cast<int64_t>(nanoseconds);
--}
-diff --git a/sdk/objc/native/src/audio/audio_device_ios.h b/sdk/objc/native/src/audio/audio_device_ios.h
-index d1788a33d7..7dcffb07d5 100644
---- a/sdk/objc/native/src/audio/audio_device_ios.h
-+++ b/sdk/objc/native/src/audio/audio_device_ios.h
-@@ -333,6 +333,9 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
- std::atomic<uint64_t> total_playout_delay_ms_;
- std::atomic<double> hw_output_latency_;
- int last_hw_output_latency_update_sample_count_;
-+ // Ratio between mach tick units and nanosecond. Used to change mach tick
-+ // units to nanoseconds.
-+ double machTickUnitsToNanoseconds_;
- };
- } // namespace ios_adm
- } // namespace webrtc
-diff --git a/sdk/objc/native/src/audio/audio_device_ios.mm b/sdk/objc/native/src/audio/audio_device_ios.mm
-index d9f8ea62a4..53dec77c75 100644
---- a/sdk/objc/native/src/audio/audio_device_ios.mm
-+++ b/sdk/objc/native/src/audio/audio_device_ios.mm
-@@ -33,7 +33,6 @@
- #import "components/audio/RTCAudioSession.h"
- #import "components/audio/RTCAudioSessionConfiguration.h"
- #import "components/audio/RTCNativeAudioSessionDelegateAdapter.h"
--#import "helpers/AudioTimeStamp+Nanoseconds.h"
-
- namespace webrtc {
- namespace ios_adm {
-@@ -131,6 +130,9 @@ AudioDeviceIOS::AudioDeviceIOS(
-
- audio_session_observer_ =
- [[RTCNativeAudioSessionDelegateAdapter alloc] initWithObserver:this];
-+ mach_timebase_info_data_t tinfo;
-+ mach_timebase_info(&tinfo);
-+ machTickUnitsToNanoseconds_ = (double)tinfo.numer / tinfo.denom;
- }
-
- AudioDeviceIOS::~AudioDeviceIOS() {
-@@ -415,8 +417,8 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(
- // Get audio timestamp for the audio.
- // The timestamp will not have NTP time epoch, but that will be addressed by
- // the TimeStampAligner in AudioDeviceBuffer::SetRecordedBuffer().
-- std::optional<int64_t> capture_timestamp_ns =
-- AudioTimeStampGetNanoseconds(time_stamp);
-+ SInt64 capture_timestamp_ns =
-+ time_stamp->mHostTime * machTickUnitsToNanoseconds_;
-
- // Allocate AudioBuffers to be used as storage for the received audio.
- // The AudioBufferList structure works as a placeholder for the
-diff --git a/sdk/objc/native/src/objc_audio_device.mm b/sdk/objc/native/src/objc_audio_device.mm
-index 3044de9c9d..d17dd63811 100644
---- a/sdk/objc/native/src/objc_audio_device.mm
-+++ b/sdk/objc/native/src/objc_audio_device.mm
-@@ -13,7 +13,6 @@
- #include <memory>
-
- #import "components/audio/RTCAudioDevice.h"
--#import "helpers/AudioTimeStamp+Nanoseconds.h"
- #include "modules/audio_device/fine_audio_buffer.h"
- #include "objc_audio_device_delegate.h"
- #include "rtc_base/logging.h"
-@@ -450,8 +449,7 @@ OSStatus ObjCAudioDeviceModule::OnDeliverRecordedData(
- record_fine_audio_buffer_->DeliverRecordedData(
- webrtc::ArrayView<const int16_t>(
- static_cast<int16_t*>(audio_buffer->mData), num_frames),
-- cached_recording_delay_ms_.load(),
-- AudioTimeStampGetNanoseconds(time_stamp));
-+ cached_recording_delay_ms_.load());
- return noErr;
- }
- RTC_DCHECK(render_block != nullptr)
-@@ -494,9 +492,7 @@ OSStatus ObjCAudioDeviceModule::OnDeliverRecordedData(
- // Use the FineAudioBuffer instance to convert between native buffer size
- // and the 10ms buffer size used by WebRTC.
- record_fine_audio_buffer_->DeliverRecordedData(
-- record_audio_buffer_,
-- cached_recording_delay_ms_.load(),
-- AudioTimeStampGetNanoseconds(time_stamp));
-+ record_audio_buffer_, cached_recording_delay_ms_.load());
- return noErr;
- }
-
+diff --git a/PRESUBMIT.py b/PRESUBMIT.py
+index 96fa8abd9d..debc65fb24 100755
+--- a/PRESUBMIT.py
++++ b/PRESUBMIT.py
+@@ -991,6 +991,8 @@ def CommonChecks(input_api, output_api):
+ bot_allowlist=[
+ 'chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com',
+ 'webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com',
++ ('chrome-cherry-picker'
++ '@chops-service-accounts.iam.gserviceaccount.com'),
+ ]))
+ results.extend(
+ input_api.canned_checks.CheckChangeTodoHasOwner(
diff --git a/third_party/libwebrtc/moz-patch-stack/p0002.patch b/third_party/libwebrtc/moz-patch-stack/p0002.patch
@@ -1,48 +1,46 @@
-From: Gennady Tsitovich <gtsitovich@google.com>
-Date: Tue, 15 Jul 2025 08:24:50 +0000
-Subject: (cherry-pick-branch-heads/7258) [M139] Add chrome-cherry-picker
- account to bot allowlist
-MIME-Version: 1.0
-Content-Type: text/plain; charset=UTF-8
-Content-Transfer-Encoding: 8bit
+From: Guido Urdaneta <guidou@webrtc.org>
+Date: Thu, 24 Jul 2025 11:01:29 +0200
+Subject: (cherry-pick-branch-heads/7258) Use FieldTrialsView::IsEnabled for
+ DTLS 1.3
-Original change's description:
-> Add chrome-cherry-picker account to bot allowlist
->
-> chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com is
-> being by the Chrome Cherry Picker (go/chromecherrypicker) and needs to
-> be able to skip the author check for presubmits.
->
-> Bug: chromium:414375466
-> Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
-> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400480
-> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
-> Reviewed-by: Björn Terelius <terelius@webrtc.org>
-> Commit-Queue: Gennady Tsitovich <gtsitovich@google.com>
-> Cr-Commit-Position: refs/heads/main@{#45148}
+No behavior changes.
-Bug: chromium:431157710,chromium:414375466
-Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400700
-Commit-Queue: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
-Auto-Submit: Chrome Cherry Picker <chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com>
-Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
-Cr-Commit-Position: refs/branch-heads/7258@{#2}
+(cherry picked from commit 5ff715d5666106e01d27205c1775d1e2d07ea254)
+
+Bug: webrtc:383141571, chromium:433885045, chromium:434133034
+Change-Id: Ice5f3e5cbd245ddea407248a6f29c61c646e6a72
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/401740
+Reviewed-by: Harald Alvestrand <hta@webrtc.org>
+Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
+Cr-Original-Commit-Position: refs/heads/main@{#45206}
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/402200
+Cr-Commit-Position: refs/branch-heads/7258@{#3}
Cr-Branched-From: 74fa937f86ed8432c07676f7a1ce0e5e2812b3d5-refs/heads/main@{#44974}
---
- PRESUBMIT.py | 2 ++
- 1 file changed, 2 insertions(+)
+ rtc_base/openssl_stream_adapter.cc | 10 ++++++----
+ 1 file changed, 6 insertions(+), 4 deletions(-)
-diff --git a/PRESUBMIT.py b/PRESUBMIT.py
-index 96fa8abd9d..debc65fb24 100755
---- a/PRESUBMIT.py
-+++ b/PRESUBMIT.py
-@@ -991,6 +991,8 @@ def CommonChecks(input_api, output_api):
- bot_allowlist=[
- 'chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com',
- 'webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com',
-+ ('chrome-cherry-picker'
-+ '@chops-service-accounts.iam.gserviceaccount.com'),
- ]))
- results.extend(
- input_api.canned_checks.CheckChangeTodoHasOwner(
+diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
+index 7d7466b1cc..604a9465c7 100644
+--- a/rtc_base/openssl_stream_adapter.cc
++++ b/rtc_base/openssl_stream_adapter.cc
+@@ -144,13 +144,15 @@ int GetForceDtls13(const FieldTrialsView* field_trials) {
+ return kForceDtls13Off;
+ }
+ #ifdef DTLS1_3_VERSION
+- auto mode = field_trials->Lookup("WebRTC-ForceDtls13");
+- RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13: " << mode;
+- if (mode == "Enabled") {
++ if (field_trials->IsEnabled("WebRTC-ForceDtls13")) {
++ RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13 Enabled";
+ return kForceDtls13Enabled;
+- } else if (mode == "Only") {
++ }
++ if (field_trials->Lookup("WebRTC-ForceDtls13") == "Only") {
++ RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13 Only";
+ return kForceDtls13Only;
+ }
++ RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13 Disabled";
+ #endif
+ return kForceDtls13Off;
+ }
diff --git a/third_party/libwebrtc/moz-patch-stack/p0003.patch b/third_party/libwebrtc/moz-patch-stack/p0003.patch
@@ -1,46 +0,0 @@
-From: Guido Urdaneta <guidou@webrtc.org>
-Date: Thu, 24 Jul 2025 11:01:29 +0200
-Subject: (cherry-pick-branch-heads/7258) Use FieldTrialsView::IsEnabled for
- DTLS 1.3
-
-No behavior changes.
-
-(cherry picked from commit 5ff715d5666106e01d27205c1775d1e2d07ea254)
-
-Bug: webrtc:383141571, chromium:433885045, chromium:434133034
-Change-Id: Ice5f3e5cbd245ddea407248a6f29c61c646e6a72
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/401740
-Reviewed-by: Harald Alvestrand <hta@webrtc.org>
-Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
-Cr-Original-Commit-Position: refs/heads/main@{#45206}
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/402200
-Cr-Commit-Position: refs/branch-heads/7258@{#3}
-Cr-Branched-From: 74fa937f86ed8432c07676f7a1ce0e5e2812b3d5-refs/heads/main@{#44974}
----
- rtc_base/openssl_stream_adapter.cc | 10 ++++++----
- 1 file changed, 6 insertions(+), 4 deletions(-)
-
-diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
-index 7d7466b1cc..604a9465c7 100644
---- a/rtc_base/openssl_stream_adapter.cc
-+++ b/rtc_base/openssl_stream_adapter.cc
-@@ -144,13 +144,15 @@ int GetForceDtls13(const FieldTrialsView* field_trials) {
- return kForceDtls13Off;
- }
- #ifdef DTLS1_3_VERSION
-- auto mode = field_trials->Lookup("WebRTC-ForceDtls13");
-- RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13: " << mode;
-- if (mode == "Enabled") {
-+ if (field_trials->IsEnabled("WebRTC-ForceDtls13")) {
-+ RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13 Enabled";
- return kForceDtls13Enabled;
-- } else if (mode == "Only") {
-+ }
-+ if (field_trials->Lookup("WebRTC-ForceDtls13") == "Only") {
-+ RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13 Only";
- return kForceDtls13Only;
- }
-+ RTC_LOG(LS_WARNING) << "WebRTC-ForceDtls13 Disabled";
- #endif
- return kForceDtls13Off;
- }
