Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 16 insertions, 20 deletions

diff --git a/webrtc/modules/video_render/mac/video_render_nsopengl.mm b/webrtc/modules/video_render/mac/video_render_nsopengl.mm
index b5150eb668..b7683a96af 100644
--- a/webrtc/modules/video_render/mac/video_render_nsopengl.mm
+++ b/webrtc/modules/video_render/mac/video_render_nsopengl.mm
@@ -11,11 +11,11 @@
 #include "webrtc/engine_configurations.h"
 #if defined(COCOA_RENDERING)
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_render/mac/video_render_nsopengl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -378,8 +378,8 @@ _renderingIsPaused (FALSE),
 _windowRefSuperView(NULL),
 _windowRefSuperViewFrame(NSMakeRect(0,0,0,0))
 {
-    _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
-            this, "ScreenUpdateNSOpenGL");
+  _screenUpdateThread.reset(new rtc::PlatformThread(
+      ScreenUpdateThreadProc, this, "ScreenUpdateNSOpenGL"));
 }
 
 int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef)
@@ -427,15 +427,15 @@ int32_t VideoRenderNSOpenGL::StartRender()
         WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Restarting screenUpdateThread");
 
         // we already have the thread. Most likely StopRender() was called and they were paused
-        if(FALSE == _screenUpdateThread->Start() ||
-                FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
-        {
+        _screenUpdateThread->Start();
+        if (FALSE ==
+            _screenUpdateEvent->StartTimer(true, 1000 / MONITOR_FREQ)) {
             WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "Failed to restart screenUpdateThread or screenUpdateEvent");
             UnlockAGLCntx();
             return -1;
         }
 
-        _screenUpdateThread->SetPriority(kRealtimePriority);
+        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
         UnlockAGLCntx();
         return 0;
@@ -471,8 +471,8 @@ int32_t VideoRenderNSOpenGL::StopRender()
         return 0;
     }
 
-    if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
-    {
+    _screenUpdateThread->Stop();
+    if (FALSE == _screenUpdateEvent->StopTimer()) {
         _renderingIsPaused = FALSE;
 
         UnlockAGLCntx();
@@ -657,17 +657,15 @@ VideoRenderNSOpenGL::~VideoRenderNSOpenGL()
     }
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
         _screenUpdateEvent->StopTimer();
 
-        if (tmpPtr->Stop())
-        {
-            delete tmpPtr;
-        }
+        tmpPtr->Stop();
+        delete tmpPtr;
         delete _screenUpdateEvent;
         _screenUpdateEvent = NULL;
     }
@@ -716,7 +714,7 @@ int VideoRenderNSOpenGL::Init()
     }
 
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
@@ -864,17 +862,15 @@ int32_t VideoRenderNSOpenGL::GetChannelProperties(const uint16_t streamId,
 int VideoRenderNSOpenGL::StopThread()
 {
 
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
     WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
                  "%s Stopping thread ", __FUNCTION__, tmpPtr);
 
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
-        if (tmpPtr->Stop())
-        {
-            delete tmpPtr;
-        }
+        tmpPtr->Stop();
+        delete tmpPtr;
     }
 
     delete _screenUpdateEvent;