tor-browser

commit 4e7b688fbed26ad55ab72311434b38bd07f69f62
parent b50e673ee54c95379e0ccf3a5c42f2f473766141
Author: Dan Baker <dbaker@mozilla.com>
Date:   Mon,  1 Dec 2025 23:47:00 -0700

Bug 2000941 - Vendor libwebrtc from 70965bc749

Upstream commit: https://webrtc.googlesource.com/src/+/70965bc749dd4fce9a324eab438c6f637a24b7ea
    Make FakeAudioCaptureModule aware of the Init()/Terminate() methods.

    This is needed so that the Initialize() method can be checked from tests
    in between the Init() and Terminate() calls. Also checking for a call to
    Init() to have been done before creating the internal test thread and
    matching that by stopping processing and the thread from within
    Terminate().

    Bug: none
    Change-Id: Iaa25821a74afb6acba919df21c8750e8735f2623
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411620
    Reviewed-by: Per Åhgren <peah@webrtc.org>
    Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
    Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#45719}

Diffstat:
M
third_party/libwebrtc/README.mozilla.last-vendor
 | 
4
++--
M
third_party/libwebrtc/pc/test/fake_audio_capture_module.cc
 | 
22
+++++++++++++++-------
M
third_party/libwebrtc/pc/test/fake_audio_capture_module.h
 | 
3
+++
M
third_party/libwebrtc/pc/test/fake_audio_capture_module_unittest.cc
 | 
7
+++++++

4 files changed, 27 insertions(+), 9 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 2025-12-02T06:44:31.965986+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-02T06:46:46.911426+00:00.
 # base of lastest vendoring
-df16623a4b
+70965bc749
diff --git a/third_party/libwebrtc/pc/test/fake_audio_capture_module.cc b/third_party/libwebrtc/pc/test/fake_audio_capture_module.cc
@@ -52,9 +52,7 @@ FakeAudioCaptureModule::FakeAudioCaptureModule()
       frames_received_(0) {}
 
 FakeAudioCaptureModule::~FakeAudioCaptureModule() {
-  if (process_thread_) {
-    process_thread_->Stop();
-  }
+  RTC_DCHECK(!initialized_);
 }
 
 webrtc::scoped_refptr<FakeAudioCaptureModule> FakeAudioCaptureModule::Create() {
@@ -84,18 +82,23 @@ int32_t FakeAudioCaptureModule::RegisterAudioCallback(
 }
 
 int32_t FakeAudioCaptureModule::Init() {
-  // Initialize is called by the factory method. Safe to ignore this Init call.
+  // Initialize is called by the factory method.
+  initialized_ = true;
   return 0;
 }
 
 int32_t FakeAudioCaptureModule::Terminate() {
-  // Clean up in the destructor. No action here, just success.
+  StopPlayout();
+  StopRecording();
+  if (process_thread_) {
+    process_thread_->Stop();
+  }
+  initialized_ = false;
   return 0;
 }
 
 bool FakeAudioCaptureModule::Initialized() const {
-  RTC_DCHECK_NOTREACHED();
-  return 0;
+  return initialized_;
 }
 
 int16_t FakeAudioCaptureModule::PlayoutDevices() {
@@ -194,6 +197,8 @@ int32_t FakeAudioCaptureModule::StopPlayout() {
   bool start = false;
   {
     webrtc::MutexLock lock(&mutex_);
+    if (!playing_)
+      return 0;
     playing_ = false;
     start = ShouldStartProcessing();
   }
@@ -223,6 +228,8 @@ int32_t FakeAudioCaptureModule::StopRecording() {
   bool start = false;
   {
     webrtc::MutexLock lock(&mutex_);
+    if (!recording_)
+      return 0;
     recording_ = false;
     start = ShouldStartProcessing();
   }
@@ -424,6 +431,7 @@ bool FakeAudioCaptureModule::ShouldStartProcessing() {
 }
 
 void FakeAudioCaptureModule::UpdateProcessing(bool start) {
+  RTC_DCHECK(initialized_);
   if (start) {
     if (!process_thread_) {
       process_thread_ = webrtc::Thread::Create();
diff --git a/third_party/libwebrtc/pc/test/fake_audio_capture_module.h b/third_party/libwebrtc/pc/test/fake_audio_capture_module.h
@@ -224,6 +224,9 @@ class FakeAudioCaptureModule : public webrtc::AudioDeviceModule {
   // (e.g. by a jitter buffer).
   int frames_received_;
 
+  // Set to true when Init() is called.
+  bool initialized_ = false;
+
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   mutable webrtc::Mutex mutex_;
diff --git a/third_party/libwebrtc/pc/test/fake_audio_capture_module_unittest.cc b/third_party/libwebrtc/pc/test/fake_audio_capture_module_unittest.cc
@@ -35,7 +35,14 @@ class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport {
 
   void SetUp() override {
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
+    fake_audio_capture_module_->Init();
     EXPECT_TRUE(fake_audio_capture_module_.get() != nullptr);
+    EXPECT_TRUE(fake_audio_capture_module_->Initialized());
+  }
+
+  void TearDown() override {
+    fake_audio_capture_module_->Terminate();
+    EXPECT_FALSE(fake_audio_capture_module_->Initialized());
   }
 
   // Callbacks inherited from webrtc::AudioTransport.