From d8bbc7858622b6d9c278469aab701ca0b609cddf Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Wed, 15 May 2024 05:35:49 +0200
Subject: Merging upstream version 126.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 third_party/libwebrtc/audio/voip/BUILD.gn        |  1 -
 third_party/libwebrtc/audio/voip/audio_egress.cc | 18 ++++++++++++------
 third_party/libwebrtc/audio/voip/audio_egress.h  | 10 +++++-----
 3 files changed, 17 insertions(+), 12 deletions(-)

(limited to 'third_party/libwebrtc/audio/voip')

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
-- 
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