In PeerConnectionTestWrapper, put audio input on a separate thread.

This will prevent it from blocking network input when it falls behind,
which is happening when running with ThreadSanitizer.

BUG=webrtc:4663

Review URL: https://codereview.webrtc.org/1236023010

Cr-Commit-Position: refs/heads/master@{#9707}

talk/app/webrtc/peerconnection_unittest.cc

@@ -518,8 +518,7 @@ class PeerConnectionTestClientBase
     if (!allocator_factory_) {
       return false;
     }
-    fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
+    fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
-        rtc::Thread::Current());
 
     if (fake_audio_capture_module_ == NULL) {
       return false;

talk/app/webrtc/peerconnectionendtoend_unittest.cc

@@ -268,17 +268,7 @@ class PeerConnectionEndToEndTest
   DataChannelList callee_signaled_data_channels_;
 };
 
-// Disable for TSan v2, see
+TEST_F(PeerConnectionEndToEndTest, Call) {
-// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
-#if !defined(THREAD_SANITIZER)
-
-// Flaky on Windows. Disabled per issue 4464.
-#ifdef WEBRTC_WIN
-#define MAYBE_Call DISABLED_Call
-#else
-#define MAYBE_Call Call
-#endif
-TEST_F(PeerConnectionEndToEndTest, MAYBE_Call) {
   CreatePcs();
   GetAndAddUserMedia();
   Negotiate();
@@ -427,4 +417,3 @@ TEST_F(PeerConnectionEndToEndTest,
   EXPECT_EQ(1U, dc_1_observer->received_message_count());
   EXPECT_EQ(1U, dc_2_observer->received_message_count());
 }
-#endif // if !defined(THREAD_SANITIZER)

talk/app/webrtc/test/fakeaudiocapturemodule.cc

@@ -54,13 +54,11 @@ static const uint32_t kMaxVolume = 14392;
 enum {
   MSG_START_PROCESS,
   MSG_RUN_PROCESS,
-  MSG_STOP_PROCESS,
 };
 
-FakeAudioCaptureModule::FakeAudioCaptureModule(
+FakeAudioCaptureModule::FakeAudioCaptureModule()
-    rtc::Thread* process_thread)
     : last_process_time_ms_(0),
-      audio_callback_(NULL),
+      audio_callback_(nullptr),
       recording_(false),
       playing_(false),
       play_is_initialized_(false),
@@ -68,23 +66,20 @@ FakeAudioCaptureModule::FakeAudioCaptureModule(
       current_mic_level_(kMaxVolume),
       started_(false),
       next_frame_time_(0),
-      process_thread_(process_thread),
       frames_received_(0) {
 }
 
 FakeAudioCaptureModule::~FakeAudioCaptureModule() {
-  // Ensure that thread stops calling ProcessFrame().
+  if (process_thread_) {
-  process_thread_->Send(this, MSG_STOP_PROCESS);
+    process_thread_->Stop();
+  }
 }
 
-rtc::scoped_refptr<FakeAudioCaptureModule> FakeAudioCaptureModule::Create(
+rtc::scoped_refptr<FakeAudioCaptureModule> FakeAudioCaptureModule::Create() {
-    rtc::Thread* process_thread) {
-  if (process_thread == NULL) return NULL;
-
   rtc::scoped_refptr<FakeAudioCaptureModule> capture_module(
-      new rtc::RefCountedObject<FakeAudioCaptureModule>(process_thread));
+      new rtc::RefCountedObject<FakeAudioCaptureModule>());
   if (!capture_module->Initialize()) {
-    return NULL;
+    return nullptr;
   }
   return capture_module;
 }
@@ -601,9 +596,6 @@ void FakeAudioCaptureModule::OnMessage(rtc::Message* msg) {
     case MSG_RUN_PROCESS:
       ProcessFrameP();
       break;
-    case MSG_STOP_PROCESS:
-      StopProcessP();
-      break;
     default:
       // All existing messages should be caught. Getting here should never
       // happen.
@@ -650,14 +642,22 @@ bool FakeAudioCaptureModule::ShouldStartProcessing() {
 
 void FakeAudioCaptureModule::UpdateProcessing(bool start) {
   if (start) {
+    if (!process_thread_) {
+      process_thread_.reset(new rtc::Thread());
+      process_thread_->Start();
+    }
     process_thread_->Post(this, MSG_START_PROCESS);
   } else {
-    process_thread_->Send(this, MSG_STOP_PROCESS);
+    if (process_thread_) {
+      process_thread_->Stop();
+      process_thread_.reset(nullptr);
+    }
+    started_ = false;
   }
 }
 
 void FakeAudioCaptureModule::StartProcessP() {
-  ASSERT(rtc::Thread::Current() == process_thread_);
+  ASSERT(process_thread_->IsCurrent());
   if (started_) {
     // Already started.
     return;
@@ -666,27 +666,22 @@ void FakeAudioCaptureModule::StartProcessP() {
 }
 
 void FakeAudioCaptureModule::ProcessFrameP() {
-  ASSERT(rtc::Thread::Current() == process_thread_);
+  ASSERT(process_thread_->IsCurrent());
   if (!started_) {
     next_frame_time_ = rtc::Time();
     started_ = true;
   }
 
-  bool playing;
-  bool recording;
   {
     rtc::CritScope cs(&crit_);
-    playing = playing_;
-    recording = recording_;
-  }
-
     // Receive and send frames every kTimePerFrameMs.
-  if (playing) {
+    if (playing_) {
       ReceiveFrameP();
     }
-  if (recording) {
+    if (recording_) {
       SendFrameP();
     }
+  }
 
   next_frame_time_ += kTimePerFrameMs;
   const uint32 current_time = rtc::Time();
@@ -696,7 +691,7 @@ void FakeAudioCaptureModule::ProcessFrameP() {
 }
 
 void FakeAudioCaptureModule::ReceiveFrameP() {
-  ASSERT(rtc::Thread::Current() == process_thread_);
+  ASSERT(process_thread_->IsCurrent());
   {
     rtc::CritScope cs(&crit_callback_);
     if (!audio_callback_) {
@@ -727,7 +722,7 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
 }
 
 void FakeAudioCaptureModule::SendFrameP() {
-  ASSERT(rtc::Thread::Current() == process_thread_);
+  ASSERT(process_thread_->IsCurrent());
   rtc::CritScope cs(&crit_callback_);
   if (!audio_callback_) {
     return;
@@ -747,8 +742,3 @@ void FakeAudioCaptureModule::SendFrameP() {
   SetMicrophoneVolume(current_mic_level);
 }
 
-void FakeAudioCaptureModule::StopProcessP() {
-  ASSERT(rtc::Thread::Current() == process_thread_);
-  started_ = false;
-  process_thread_->Clear(this);
-}

talk/app/webrtc/test/fakeaudiocapturemodule.h

@@ -40,14 +40,13 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
 namespace rtc {
-
 class Thread;
-
 }  // namespace rtc
 
 class FakeAudioCaptureModule
@@ -62,10 +61,7 @@ class FakeAudioCaptureModule
   static const int kNumberBytesPerSample = sizeof(Sample);
 
   // Creates a FakeAudioCaptureModule or returns NULL on failure.
-  // |process_thread| is used to push and pull audio frames to and from the
+  static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
-  // returned instance. Note: ownership of |process_thread| is not handed over.
-  static rtc::scoped_refptr<FakeAudioCaptureModule> Create(
-      rtc::Thread* process_thread);
 
   // Returns the number of frames that have been successfully pulled by the
   // instance. Note that correctly detecting success can only be done if the
@@ -206,7 +202,7 @@ class FakeAudioCaptureModule
   // exposed in which case the burden of proper instantiation would be put on
   // the creator of a FakeAudioCaptureModule instance. To create an instance of
   // this class use the Create(..) API.
-  explicit FakeAudioCaptureModule(rtc::Thread* process_thread);
+  explicit FakeAudioCaptureModule();
   // The destructor is protected because it is reference counted and should not
   // be deleted directly.
   virtual ~FakeAudioCaptureModule();
@@ -239,8 +235,6 @@ class FakeAudioCaptureModule
   void ReceiveFrameP();
   // Pushes frames to the registered webrtc::AudioTransport.
   void SendFrameP();
-  // Stops the periodic calling of ProcessFrame() in a thread safe way.
-  void StopProcessP();
 
   // The time in milliseconds when Process() was last called or 0 if no call
   // has been made.
@@ -266,8 +260,7 @@ class FakeAudioCaptureModule
   bool started_;
   uint32 next_frame_time_;
 
-  // User provided thread context.
+  rtc::scoped_ptr<rtc::Thread> process_thread_;
-  rtc::Thread* process_thread_;
 
   // Buffer for storing samples received from the webrtc::AudioTransport.
   char rec_buffer_[kNumberSamples * kNumberBytesPerSample];

talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc

@@ -29,6 +29,7 @@
 
 #include <algorithm>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread.h"
@@ -48,8 +49,7 @@ class FakeAdmTest : public testing::Test,
   }
 
   virtual void SetUp() {
-    fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
+    fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
-        rtc::Thread::Current());
     EXPECT_TRUE(fake_audio_capture_module_.get() != NULL);
   }
 
@@ -65,6 +65,7 @@ class FakeAdmTest : public testing::Test,
                                   const uint32_t currentMicLevel,
                                   const bool keyPressed,
                                   uint32_t& newMicLevel) override {
+    rtc::CritScope cs(&crit_);
     rec_buffer_bytes_ = nSamples * nBytesPerSample;
     if ((rec_buffer_bytes_ == 0) ||
         (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
@@ -87,6 +88,7 @@ class FakeAdmTest : public testing::Test,
                            uint32_t& nSamplesOut,
                            int64_t* elapsed_time_ms,
                            int64_t* ntp_time_ms) override {
+    rtc::CritScope cs(&crit_);
     ++pull_iterations_;
     const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
     const uint32_t bytes_out = RecordedDataReceived() ?
@@ -98,8 +100,14 @@ class FakeAdmTest : public testing::Test,
     return 0;
   }
 
-  int push_iterations() const { return push_iterations_; }
+  int push_iterations() const {
-  int pull_iterations() const { return pull_iterations_; }
+    rtc::CritScope cs(&crit_);
+    return push_iterations_;
+  }
+  int pull_iterations() const {
+    rtc::CritScope cs(&crit_);
+    return pull_iterations_;
+  }
 
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
 
@@ -118,6 +126,8 @@ class FakeAdmTest : public testing::Test,
     return min_buffer_size;
   }
 
+  mutable rtc::CriticalSection crit_;
+
   int push_iterations_;
   int pull_iterations_;
 

talk/app/webrtc/test/peerconnectiontestwrapper.cc

@@ -75,8 +75,7 @@ bool PeerConnectionTestWrapper::CreatePc(
     return false;
   }
 
-  fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
+  fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
-      rtc::Thread::Current());
   if (fake_audio_capture_module_ == NULL) {
     return false;
   }

talk/app/webrtc/test/peerconnectiontestwrapper.h

@@ -33,7 +33,6 @@
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
 #include "webrtc/base/sigslot.h"
-#include "webrtc/base/thread.h"
 
 namespace webrtc {
 class DtlsIdentityStoreInterface;