Merging upstream version 126.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

13 files changed, 80 insertions, 63 deletions

diff --git a/third_party/libwebrtc/audio/BUILD.gn b/third_party/libwebrtc/audio/BUILD.gn
index 09562b9131..7ece107407 100644
--- a/third_party/libwebrtc/audio/BUILD.gn
+++ b/third_party/libwebrtc/audio/BUILD.gn
@@ -97,7 +97,6 @@ rtc_library("audio") {
     "../rtc_base:refcount",
     "../rtc_base:rtc_event",
     "../rtc_base:rtc_numerics",
-    "../rtc_base:rtc_task_queue",
     "../rtc_base:safe_conversions",
     "../rtc_base:safe_minmax",
     "../rtc_base:stringutils",
@@ -175,7 +174,8 @@ if (rtc_include_tests) {
       "../api/audio_codecs/opus:audio_decoder_opus",
       "../api/audio_codecs/opus:audio_encoder_opus",
       "../api/crypto:frame_decryptor_interface",
-      "../api/rtc_event_log",
+      "../api/environment",
+      "../api/environment:environment_factory",
       "../api/task_queue:default_task_queue_factory",
       "../api/task_queue/test:mock_task_queue_base",
       "../api/units:time_delta",
@@ -223,7 +223,10 @@ if (rtc_include_tests) {
       "utility:utility_tests",
       "//testing/gtest",
     ]
-    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
   }
 
   rtc_library("channel_receive_unittest") {
@@ -247,6 +250,9 @@ if (rtc_include_tests) {
       "../test:test_support",
       "../test/time_controller",
     ]
-    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
   }
 }
diff --git a/third_party/libwebrtc/audio/audio_receive_stream.cc b/third_party/libwebrtc/audio/audio_receive_stream.cc
index c49b83f95f..32a8e1c172 100644
--- a/third_party/libwebrtc/audio/audio_receive_stream.cc
+++ b/third_party/libwebrtc/audio/audio_receive_stream.cc
@@ -186,6 +186,7 @@ void AudioReceiveStreamImpl::Start() {
   if (playing_) {
     return;
   }
+  RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Start: " << remote_ssrc();
   channel_receive_->StartPlayout();
   playing_ = true;
   audio_state()->AddReceivingStream(this);
@@ -196,6 +197,7 @@ void AudioReceiveStreamImpl::Stop() {
   if (!playing_) {
     return;
   }
+  RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Stop: " << remote_ssrc();
   channel_receive_->StopPlayout();
   playing_ = false;
   audio_state()->RemoveReceivingStream(this);
diff --git a/third_party/libwebrtc/audio/audio_send_stream.cc b/third_party/libwebrtc/audio/audio_send_stream.cc
index e7ebb2bf4e..8dc78b18fa 100644
--- a/third_party/libwebrtc/audio/audio_send_stream.cc
+++ b/third_party/libwebrtc/audio/audio_send_stream.cc
@@ -355,6 +355,7 @@ void AudioSendStream::Start() {
   if (sending_) {
     return;
   }
+  RTC_LOG(LS_INFO) << "AudioSendStream::Start: " << config_.rtp.ssrc;
   if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
       config_.max_bitrate_bps != -1 &&
       (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
@@ -376,7 +377,7 @@ void AudioSendStream::Stop() {
   if (!sending_) {
     return;
   }
-
+  RTC_LOG(LS_INFO) << "AudioSendStream::Stop: " << config_.rtp.ssrc;
   RemoveBitrateObserver();
   channel_send_->StopSend();
   sending_ = false;
diff --git a/third_party/libwebrtc/audio/audio_send_stream.h b/third_party/libwebrtc/audio/audio_send_stream.h
index 62ccd524cb..09fd712d40 100644
--- a/third_party/libwebrtc/audio/audio_send_stream.h
+++ b/third_party/libwebrtc/audio/audio_send_stream.h
@@ -28,7 +28,6 @@
 #include "rtc_base/experiments/struct_parameters_parser.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 
 namespace webrtc {
 class RtcEventLog;
diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h
index 37ff75c2e9..d572ffe8d1 100644
--- a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h
+++ b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h
@@ -16,8 +16,8 @@
 
 #include "api/frame_transformer_interface.h"
 #include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
 #include "rtc_base/system/no_unique_address.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/thread.h"
 
 namespace webrtc {
diff --git a/third_party/libwebrtc/audio/channel_send.cc b/third_party/libwebrtc/audio/channel_send.cc
index 3c59be52b4..ae264a4c77 100644
--- a/third_party/libwebrtc/audio/channel_send.cc
+++ b/third_party/libwebrtc/audio/channel_send.cc
@@ -39,7 +39,7 @@
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
+#include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/clock.h"
@@ -196,7 +196,7 @@ class ChannelSend : public ChannelSendInterface,
                        rtc::ArrayView<const uint8_t> payload,
                        int64_t absolute_capture_timestamp_ms,
                        rtc::ArrayView<const uint32_t> csrcs)
-      RTC_RUN_ON(encoder_queue_);
+      RTC_RUN_ON(encoder_queue_checker_);
 
   void OnReceivedRtt(int64_t rtt_ms);
 
@@ -207,7 +207,7 @@ class ChannelSend : public ChannelSendInterface,
   // specific threads we know about. The goal is to eventually split up
   // voe::Channel into parts with single-threaded semantics, and thereby reduce
   // the need for locks.
-  SequenceChecker worker_thread_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
   // Methods accessed from audio and video threads are checked for sequential-
   // only access. We don't necessarily own and control these threads, so thread
   // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
@@ -231,9 +231,9 @@ class ChannelSend : public ChannelSendInterface,
   absl::optional<int64_t> last_capture_timestamp_ms_
       RTC_GUARDED_BY(audio_thread_race_checker_);
 
-  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
+  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_checker_);
   bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
-  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_) = false;
+  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_checker_) = false;
 
   const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
 
@@ -242,7 +242,7 @@ class ChannelSend : public ChannelSendInterface,
   const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
   const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
-  SequenceChecker construction_thread_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker construction_thread_;
 
   std::atomic<bool> include_audio_level_indication_ = false;
   std::atomic<bool> encoder_queue_is_active_ = false;
@@ -250,7 +250,7 @@ class ChannelSend : public ChannelSendInterface,
 
   // E2EE Audio Frame Encryption
   rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
-      RTC_GUARDED_BY(encoder_queue_);
+      RTC_GUARDED_BY(encoder_queue_checker_);
   // E2EE Frame Encryption Options
   const webrtc::CryptoOptions crypto_options_;
 
@@ -258,15 +258,14 @@ class ChannelSend : public ChannelSendInterface,
   // receives callbacks with the transformed frames; delegates calls to
   // ChannelSend::SendRtpAudio to send the transformed audio.
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
-      frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_);
+      frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_checker_);
 
   mutable Mutex rtcp_counter_mutex_;
   RtcpPacketTypeCounter rtcp_packet_type_counter_
       RTC_GUARDED_BY(rtcp_counter_mutex_);
 
-  // Defined last to ensure that there are no running tasks when the other
-  // members are destroyed.
-  rtc::TaskQueue encoder_queue_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_;
 
   SdpAudioFormat encoder_format_;
 };
@@ -299,7 +298,7 @@ class RtpPacketSenderProxy : public RtpPacketSender {
   }
 
  private:
-  SequenceChecker thread_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
   Mutex mutex_;
   RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_);
 };
@@ -310,7 +309,7 @@ int32_t ChannelSend::SendData(AudioFrameType frameType,
                               const uint8_t* payloadData,
                               size_t payloadSize,
                               int64_t absolute_capture_timestamp_ms) {
-  RTC_DCHECK_RUN_ON(&encoder_queue_);
+  RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
   rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
   if (frame_transformer_delegate_) {
     // Asynchronously transform the payload before sending it. After the payload
@@ -438,6 +437,7 @@ ChannelSend::ChannelSend(
       encoder_queue_(task_queue_factory->CreateTaskQueue(
           "AudioEncoder",
           TaskQueueFactory::Priority::NORMAL)),
+      encoder_queue_checker_(encoder_queue_.get()),
       encoder_format_("x-unknown", 0, 0) {
   audio_coding_ = AudioCodingModule::Create();
 
@@ -490,6 +490,10 @@ ChannelSend::~ChannelSend() {
   StopSend();
   int error = audio_coding_->RegisterTransportCallback(NULL);
   RTC_DCHECK_EQ(0, error);
+
+  // Delete the encoder task queue first to ensure that there are no running
+  // tasks when the other members are destroyed.
+  encoder_queue_ = nullptr;
 }
 
 void ChannelSend::StartSend() {
@@ -519,8 +523,8 @@ void ChannelSend::StopSend() {
   // Wait until all pending encode tasks are executed and clear any remaining
   // buffers in the encoder.
   rtc::Event flush;
-  encoder_queue_.PostTask([this, &flush]() {
-    RTC_DCHECK_RUN_ON(&encoder_queue_);
+  encoder_queue_->PostTask([this, &flush]() {
+    RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
     CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); });
     flush.Set();
   });
@@ -794,9 +798,9 @@ void ChannelSend::ProcessAndEncodeAudio(
   // Profile time between when the audio frame is added to the task queue and
   // when the task is actually executed.
   audio_frame->UpdateProfileTimeStamp();
-  encoder_queue_.PostTask(
+  encoder_queue_->PostTask(
       [this, audio_frame = std::move(audio_frame)]() mutable {
-        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
         if (!encoder_queue_is_active_.load()) {
           return;
         }
@@ -858,8 +862,8 @@ int64_t ChannelSend::GetRTT() const {
 void ChannelSend::SetFrameEncryptor(
     rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  encoder_queue_.PostTask([this, frame_encryptor]() mutable {
-    RTC_DCHECK_RUN_ON(&encoder_queue_);
+  encoder_queue_->PostTask([this, frame_encryptor]() mutable {
+    RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
     frame_encryptor_ = std::move(frame_encryptor);
   });
 }
@@ -870,9 +874,9 @@ void ChannelSend::SetEncoderToPacketizerFrameTransformer(
   if (!frame_transformer)
     return;
 
-  encoder_queue_.PostTask(
+  encoder_queue_->PostTask(
       [this, frame_transformer = std::move(frame_transformer)]() mutable {
-        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
         InitFrameTransformerDelegate(std::move(frame_transformer));
       });
 }
@@ -885,7 +889,7 @@ void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
 
 void ChannelSend::InitFrameTransformerDelegate(
     rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
-  RTC_DCHECK_RUN_ON(&encoder_queue_);
+  RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
   RTC_DCHECK(frame_transformer);
   RTC_DCHECK(!frame_transformer_delegate_);
 
@@ -897,7 +901,7 @@ void ChannelSend::InitFrameTransformerDelegate(
              rtc::ArrayView<const uint8_t> payload,
              int64_t absolute_capture_timestamp_ms,
              rtc::ArrayView<const uint32_t> csrcs) {
-        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
         return SendRtpAudio(
             frameType, payloadType,
             rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload,
@@ -906,7 +910,7 @@ void ChannelSend::InitFrameTransformerDelegate(
   frame_transformer_delegate_ =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
           std::move(send_audio_callback), std::move(frame_transformer),
-          &encoder_queue_);
+          encoder_queue_.get());
   frame_transformer_delegate_->Init();
 }
 
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
index 2eea0d2387..ac32410aed 100644
--- a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
@@ -114,7 +114,7 @@ class TransformableOutgoingAudioFrame
 ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
     SendFrameCallback send_frame_callback,
     rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
-    rtc::TaskQueue* encoder_queue)
+    TaskQueueBase* encoder_queue)
     : send_frame_callback_(send_frame_callback),
       frame_transformer_(std::move(frame_transformer)),
       encoder_queue_(encoder_queue) {}
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h
index 97fc14f737..30e63ff98b 100644
--- a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h
@@ -16,10 +16,10 @@
 
 #include "api/frame_transformer_interface.h"
 #include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 
 namespace webrtc {
 
@@ -40,7 +40,7 @@ class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
   ChannelSendFrameTransformerDelegate(
       SendFrameCallback send_frame_callback,
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
-      rtc::TaskQueue* encoder_queue);
+      TaskQueueBase* encoder_queue);
 
   // Registers `this` as callback for `frame_transformer_`, to get the
   // transformed frames.
@@ -79,7 +79,7 @@ class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
   mutable Mutex send_lock_;
   SendFrameCallback send_frame_callback_ RTC_GUARDED_BY(send_lock_);
   rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
-  rtc::TaskQueue* encoder_queue_ RTC_GUARDED_BY(send_lock_);
+  TaskQueueBase* const encoder_queue_;
   bool short_circuit_ RTC_GUARDED_BY(send_lock_) = false;
 };
 
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc
index 4dcd15cd95..483a2cce78 100644
--- a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -78,7 +78,7 @@ std::unique_ptr<TransformableAudioFrameInterface> CreateFrame() {
   MockChannelSend mock_channel;
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
-          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+          mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
 
   std::unique_ptr<TransformableFrameInterface> frame;
   ON_CALL(*mock_frame_transformer, Transform)
@@ -131,7 +131,7 @@ TEST(ChannelSendFrameTransformerDelegateTest,
   MockChannelSend mock_channel;
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
-          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+          mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
   rtc::scoped_refptr<TransformedFrameCallback> callback;
   EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
       .WillOnce(SaveArg<0>(&callback));
@@ -160,7 +160,7 @@ TEST(ChannelSendFrameTransformerDelegateTest,
   MockChannelSend mock_channel;
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
-          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+          mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
   rtc::scoped_refptr<TransformedFrameCallback> callback;
   EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
       .WillOnce(SaveArg<0>(&callback));
@@ -192,7 +192,7 @@ TEST(ChannelSendFrameTransformerDelegateTest,
   MockChannelSend mock_channel;
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
-          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+          mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
 
   delegate->Reset();
   EXPECT_CALL(mock_channel, SendFrame).Times(0);
@@ -207,7 +207,7 @@ TEST(ChannelSendFrameTransformerDelegateTest, ShortCircuitingSkipsTransform) {
   MockChannelSend mock_channel;
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
-          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+          mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
 
   delegate->StartShortCircuiting();
 
diff --git a/third_party/libwebrtc/audio/channel_send_unittest.cc b/third_party/libwebrtc/audio/channel_send_unittest.cc
index 58d7c93c1e..c86dcefadc 100644
--- a/third_party/libwebrtc/audio/channel_send_unittest.cc
+++ b/third_party/libwebrtc/audio/channel_send_unittest.cc
@@ -14,7 +14,8 @@
 
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "api/scoped_refptr.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -53,18 +54,17 @@ class ChannelSendTest : public ::testing::Test {
  protected:
   ChannelSendTest()
       : time_controller_(Timestamp::Seconds(1)),
+        env_(CreateEnvironment(&field_trials_,
+                               time_controller_.GetClock(),
+                               time_controller_.CreateTaskQueueFactory())),
         transport_controller_(
-            time_controller_.GetClock(),
-            RtpTransportConfig{
-                .bitrate_config = GetBitrateConfig(),
-                .event_log = &event_log_,
-                .task_queue_factory = time_controller_.GetTaskQueueFactory(),
-                .trials = &field_trials_,
-            }) {
+            RtpTransportConfig{.env = env_,
+                               .bitrate_config = GetBitrateConfig()}) {
     channel_ = voe::CreateChannelSend(
         time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(),
-        &transport_, nullptr, &event_log_, nullptr, crypto_options_, false,
-        kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_, field_trials_);
+        &transport_, nullptr, &env_.event_log(), nullptr, crypto_options_,
+        false, kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_,
+        env_.field_trials());
     encoder_factory_ = CreateBuiltinAudioEncoderFactory();
     SdpAudioFormat opus = SdpAudioFormat("opus", kRtpRateHz, 2);
     std::unique_ptr<AudioEncoder> encoder =
@@ -94,7 +94,7 @@ class ChannelSendTest : public ::testing::Test {
 
   GlobalSimulatedTimeController time_controller_;
   webrtc::test::ScopedKeyValueConfig field_trials_;
-  RtcEventLogNull event_log_;
+  Environment env_;
   NiceMock<MockTransport> transport_;
   CryptoOptions crypto_options_;
   RtpTransportControllerSend transport_controller_;
diff --git a/third_party/libwebrtc/audio/voip/BUILD.gn b/third_party/libwebrtc/audio/voip/BUILD.gn
index e807e2276b..75f20a6ed2 100644
--- a/third_party/libwebrtc/audio/voip/BUILD.gn
+++ b/third_party/libwebrtc/audio/voip/BUILD.gn
@@ -94,7 +94,6 @@ rtc_library("audio_egress") {
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:logging",
-    "../../rtc_base:rtc_task_queue",
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
     "../../rtc_base/system:no_unique_address",
diff --git a/third_party/libwebrtc/audio/voip/audio_egress.cc b/third_party/libwebrtc/audio/voip/audio_egress.cc
index 95a1a3351e..09396cd28d 100644
--- a/third_party/libwebrtc/audio/voip/audio_egress.cc
+++ b/third_party/libwebrtc/audio/voip/audio_egress.cc
@@ -13,6 +13,7 @@
 #include <utility>
 #include <vector>
 
+#include "api/sequence_checker.h"
 #include "rtc_base/logging.h"
 
 namespace webrtc {
@@ -25,12 +26,17 @@ AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp,
       audio_coding_(AudioCodingModule::Create()),
       encoder_queue_(task_queue_factory->CreateTaskQueue(
           "AudioEncoder",
-          TaskQueueFactory::Priority::NORMAL)) {
+          TaskQueueFactory::Priority::NORMAL)),
+      encoder_queue_checker_(encoder_queue_.get()) {
   audio_coding_->RegisterTransportCallback(this);
 }
 
 AudioEgress::~AudioEgress() {
   audio_coding_->RegisterTransportCallback(nullptr);
+
+  // Delete first to ensure that there are no running tasks when the other
+  // members are destroyed.
+  encoder_queue_ = nullptr;
 }
 
 bool AudioEgress::IsSending() const {
@@ -73,9 +79,9 @@ void AudioEgress::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
   RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
   RTC_DCHECK_LE(audio_frame->num_channels_, 8);
 
-  encoder_queue_.PostTask(
+  encoder_queue_->PostTask(
       [this, audio_frame = std::move(audio_frame)]() mutable {
-        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
         if (!rtp_rtcp_->SendingMedia()) {
           return;
         }
@@ -112,7 +118,7 @@ int32_t AudioEgress::SendData(AudioFrameType frame_type,
                               uint32_t timestamp,
                               const uint8_t* payload_data,
                               size_t payload_size) {
-  RTC_DCHECK_RUN_ON(&encoder_queue_);
+  RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
 
   rtc::ArrayView<const uint8_t> payload(payload_data, payload_size);
 
@@ -175,8 +181,8 @@ bool AudioEgress::SendTelephoneEvent(int dtmf_event, int duration_ms) {
 }
 
 void AudioEgress::SetMute(bool mute) {
-  encoder_queue_.PostTask([this, mute] {
-    RTC_DCHECK_RUN_ON(&encoder_queue_);
+  encoder_queue_->PostTask([this, mute] {
+    RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
     encoder_context_.mute_ = mute;
   });
 }
diff --git a/third_party/libwebrtc/audio/voip/audio_egress.h b/third_party/libwebrtc/audio/voip/audio_egress.h
index 989e5bda59..6d1489db34 100644
--- a/third_party/libwebrtc/audio/voip/audio_egress.h
+++ b/third_party/libwebrtc/audio/voip/audio_egress.h
@@ -16,6 +16,7 @@
 
 #include "api/audio_codecs/audio_format.h"
 #include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "audio/audio_level.h"
 #include "audio/utility/audio_frame_operations.h"
@@ -25,7 +26,7 @@
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
 #include "modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
+#include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/time_utils.h"
 
 namespace webrtc {
@@ -146,11 +147,10 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback {
     bool previously_muted_ = false;
   };
 
-  EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_);
+  EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_checker_);
 
-  // Defined last to ensure that there are no running tasks when the other
-  // members are destroyed.
-  rtc::TaskQueue encoder_queue_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_;
 };
 
 }  // namespace webrtc