commit 90ae408774c8d084dd5b52114fc3926bfb704b4e
parent 349a946d4d7b94dbf8027145aa4ed7540f0fff2e
Author: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Oct 2025 15:55:00 -0500
Bug 1993083 - Vendor libwebrtc from fa780492c9
Essentially a no-op since we're going to see this change
reverted when we vendor in a473b69c06.
Upstream commit: https://webrtc.googlesource.com/src/+/fa780492c902b0b0079386178b21b9c61221f25e
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}
Diffstat:
5 files changed, 294 insertions(+), 80 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 2025-10-08T20:53:46.152638+00:00.
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-08T20:54:50.810232+00:00.
# base of lastest vendoring
-adf991437c
+fa780492c9
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
@@ -0,0 +1 @@
+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
@@ -1,48 +1,259 @@
-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
+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)
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:
-> Add chrome-cherry-picker account to bot allowlist
+> Record audio timestamp across all ObjC ADMs.
+>
+> Added audio timestamp capture to ObjCAudioDeviceModule and extracted
+> nanoseconds from AudioTimeStamp construction into helper function.
>
-> 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.
+> The timestamp capture was initially implemented in the CL:
+> https://webrtc-review.googlesource.com/c/src/+/334720
>
-> Bug: chromium:414375466
-> Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
-> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400480
+> 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: Björn Terelius <terelius@webrtc.org>
-> Commit-Queue: Gennady Tsitovich <gtsitovich@google.com>
-> Cr-Commit-Position: refs/heads/main@{#45148}
+> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
+> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
+> Cr-Commit-Position: refs/heads/main@{#45000}
-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}
+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}
---
- PRESUBMIT.py | 2 ++
- 1 file changed, 2 insertions(+)
+ 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
-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/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/third_party/libwebrtc/moz-patch-stack/p0002.patch b/third_party/libwebrtc/moz-patch-stack/p0002.patch
@@ -1,46 +1,48 @@
-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
+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
-No behavior changes.
+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}
-(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}
+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}
---
- rtc_base/openssl_stream_adapter.cc | 10 ++++++----
- 1 file changed, 6 insertions(+), 4 deletions(-)
+ PRESUBMIT.py | 2 ++
+ 1 file changed, 2 insertions(+)
-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/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/p0003.patch
