commit 4ae0483bd8566d826d628330cd714dae2610170f
parent bb130f40398d7dad5709f8e7bdfa39b298e17e7f
Author: Dan Baker <dbaker@mozilla.com>
Date: Mon, 1 Dec 2025 22:40:10 -0700
Bug 2000941 - Vendor libwebrtc from e468766ed3
Upstream commit: https://webrtc.googlesource.com/src/+/e468766ed349244c237c612a8c6dca73012d0fb9
Allow constructing objects on one thread and use on another.
Allow AudioState and AudioDeviceBuffer to be constructed on one thread
but used and destroyed on another.
This is a prerequisite for allowing asynchronous termination of the
WebRtcVoiceEngine.
Bug: webrtc:42224720
Change-Id: I586f2ef84cd34137eee2d57b9d954f3ed45a8898
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/410862
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45697}
Diffstat:
9 files changed, 55 insertions(+), 15 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 /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-02T05:37:07.482993+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-02T05:39:57.348426+00:00.
# base of lastest vendoring
-c9b54ede45
+e468766ed3
diff --git a/third_party/libwebrtc/audio/BUILD.gn b/third_party/libwebrtc/audio/BUILD.gn
@@ -186,6 +186,7 @@ if (rtc_include_tests) {
"../api:mock_frame_encryptor",
"../api:mock_frame_transformer",
"../api:mock_transformable_audio_frame",
+ "../api:ref_count",
"../api:rtc_error_matchers",
"../api:rtp_headers",
"../api:rtp_parameters",
diff --git a/third_party/libwebrtc/audio/audio_state.h b/third_party/libwebrtc/audio/audio_state.h
@@ -21,6 +21,7 @@
#include "call/audio_state.h"
#include "rtc_base/checks.h"
#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/thread_annotations.h"
@@ -66,8 +67,10 @@ class AudioState : public webrtc::AudioState {
void UpdateAudioTransportWithSendingStreams();
void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_);
- SequenceChecker thread_checker_;
- SequenceChecker process_thread_checker_{SequenceChecker::kDetached};
+ RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_{
+ SequenceChecker::kDetached};
+ RTC_NO_UNIQUE_ADDRESS SequenceChecker process_thread_checker_{
+ SequenceChecker::kDetached};
const webrtc::AudioState::Config config_;
bool recording_enabled_ = true;
bool playout_enabled_ = true;
diff --git a/third_party/libwebrtc/audio/audio_state_unittest.cc b/third_party/libwebrtc/audio/audio_state_unittest.cc
@@ -26,6 +26,7 @@
#include "api/audio/audio_mixer.h"
#include "api/location.h"
#include "api/make_ref_counted.h"
+#include "api/ref_count.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
@@ -37,6 +38,7 @@
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -49,6 +51,7 @@ using ::testing::_;
using ::testing::InSequence;
using ::testing::Matcher;
using ::testing::NiceMock;
+using ::testing::NotNull;
using ::testing::StrictMock;
using ::testing::Values;
@@ -200,6 +203,20 @@ TEST_P(AudioStateTest, ConstructDestruct) {
make_ref_counted<internal::AudioState>(helper.config()));
}
+TEST_P(AudioStateTest, CreateUseDeleteOnDifferentThread) {
+ ConfigHelper helper(GetParam());
+ scoped_refptr<AudioState> audio_state = AudioState::Create(helper.config());
+ ASSERT_THAT(audio_state, NotNull());
+ TaskQueueForTest queue;
+ queue.SendTask([&] {
+ // Attach to the current TQ.
+ audio_state->SetStereoChannelSwapping(true);
+ // Ensure the object gets deleted on the current thread.
+ EXPECT_EQ(audio_state.release()->Release(),
+ RefCountReleaseStatus::kDroppedLastRef);
+ });
+}
+
TEST_P(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
ConfigHelper helper(GetParam());
diff --git a/third_party/libwebrtc/modules/audio_device/BUILD.gn b/third_party/libwebrtc/modules/audio_device/BUILD.gn
@@ -77,6 +77,7 @@ if (!build_with_mozilla) { # See Bug 1820869.
"../../rtc_base:timestamp_aligner",
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
+ "../../rtc_base/system:no_unique_address",
"../../system_wrappers",
"../../system_wrappers:metrics",
]
@@ -487,6 +488,7 @@ if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) {
"../../rtc_base:race_checker",
"../../rtc_base:rtc_event",
"../../rtc_base:safe_conversions",
+ "../../rtc_base:task_queue_for_test",
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h
@@ -23,6 +23,7 @@
#include "api/task_queue/task_queue_base.h"
#include "rtc_base/buffer.h"
#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/timestamp_aligner.h"
@@ -153,8 +154,9 @@ class AudioDeviceBuffer {
// edge cases and it is IMHO not worth the risk to use them in this class.
// TODO(henrika): see if it is possible to refactor and annotate all members.
- // Main thread on which this object is created.
- SequenceChecker main_thread_checker_;
+ // Main thread for where this object is used.
+ RTC_NO_UNIQUE_ADDRESS SequenceChecker main_thread_checker_{
+ SequenceChecker::kDetached};
Mutex lock_;
diff --git a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl_test.cc b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl_test.cc
@@ -26,6 +26,7 @@
#include "modules/audio_device/include/test_audio_device.h"
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread_annotations.h"
#include "test/create_test_environment.h"
#include "test/gmock.h"
@@ -187,6 +188,20 @@ TEST(TestAudioDeviceTest, EnablingRecordingProducesAudio) {
ElementsAre(48000, 48000, 48000));
}
+// Construct an AudioDeviceBuffer on one thread, use it and delete it on a
+// different thread.
+TEST(TestAudioDeviceTest, AudioDeviceBufferOnDifferentThread) {
+ GlobalSimulatedTimeController time_controller(kStartTime);
+ const Environment env = CreateTestEnvironment({.time = &time_controller});
+ auto audio_buffer = std::make_unique<AudioDeviceBuffer>(env);
+ TaskQueueForTest queue;
+ queue.SendTask([&] {
+ TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording);
+ ASSERT_EQ(audio_buffer->RegisterAudioCallback(&audio_transport), 0);
+ audio_buffer = nullptr;
+ });
+}
+
TEST(TestAudioDeviceTest, RecordingIsAvailableWhenCapturerIsSet) {
GlobalSimulatedTimeController time_controller(kStartTime);
std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
diff --git a/third_party/libwebrtc/moz-patch-stack/s0027.patch b/third_party/libwebrtc/moz-patch-stack/s0027.patch
@@ -731,7 +731,7 @@ index 02c3ccde22..cf992635fe 100644
}
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
-index 8e8a4bff5f..e57e55882c 100644
+index c8b15e4040..7ace1135c8 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -30,6 +30,7 @@ rtc_source_set("audio_device_default") {
@@ -758,7 +758,7 @@ index 8e8a4bff5f..e57e55882c 100644
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
-@@ -78,6 +81,7 @@ rtc_library("audio_device_buffer") {
+@@ -79,6 +82,7 @@ rtc_library("audio_device_buffer") {
"../../system_wrappers:metrics",
]
}
@@ -766,7 +766,7 @@ index 8e8a4bff5f..e57e55882c 100644
rtc_library("audio_device_generic") {
sources = [
-@@ -256,6 +260,7 @@ if (!build_with_chromium) {
+@@ -257,6 +261,7 @@ if (!build_with_chromium) {
# Contains default implementations of webrtc::AudioDeviceModule for Windows,
# Linux, Mac, iOS and Android.
rtc_library("audio_device_impl") {
@@ -774,7 +774,7 @@ index 8e8a4bff5f..e57e55882c 100644
visibility = [ "*" ]
deps = [
":audio_device_buffer",
-@@ -302,9 +307,9 @@ rtc_library("audio_device_impl") {
+@@ -303,9 +308,9 @@ rtc_library("audio_device_impl") {
sources = [ "include/fake_audio_device.h" ]
if (build_with_mozilla) {
@@ -787,7 +787,7 @@ index 8e8a4bff5f..e57e55882c 100644
]
}
-@@ -407,6 +412,7 @@ rtc_library("audio_device_impl") {
+@@ -408,6 +413,7 @@ rtc_library("audio_device_impl") {
sources += [ "dummy/file_audio_device_factory.h" ]
}
}
@@ -795,7 +795,7 @@ index 8e8a4bff5f..e57e55882c 100644
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
-@@ -424,6 +430,7 @@ if (is_mac) {
+@@ -425,6 +431,7 @@ if (is_mac) {
}
}
@@ -803,7 +803,7 @@ index 8e8a4bff5f..e57e55882c 100644
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
-@@ -442,8 +449,10 @@ rtc_source_set("mock_audio_device") {
+@@ -443,8 +450,10 @@ rtc_source_set("mock_audio_device") {
"../../test:test_support",
]
}
diff --git a/third_party/libwebrtc/moz-patch-stack/s0102.patch b/third_party/libwebrtc/moz-patch-stack/s0102.patch
@@ -276,7 +276,7 @@ index c6040740ce..70ed476401 100644
rtc_library("scalability_mode") {
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
-index 3773ab8173..d2734974c9 100644
+index 3776dac967..0200d8c204 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -8,8 +8,8 @@
@@ -440,7 +440,7 @@ index 6896edbc34..ecbb7eae7e 100644
import("../webrtc.gni")
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
-index e57e55882c..80af18a5fa 100644
+index 7ace1135c8..b63880a296 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -9,8 +9,8 @@
