commit 31660fdfeacce1ebd27650e0fdf1121b72c8b6f2
author Danil Chapovalov <danilchap@webrtc.org> Fri Mar 22 12:59:48 2019 +0100
committer Commit Bot <commit-bot@chromium.org> Fri Mar 22 15:53:28 2019 +0000
tree 02200d211270d9ae912e4f64a7452cc925ec78bf
parent 0d617ccc1c0afb8f94135462ca7c9a512935e1f0

Avoid using global task queue factory in audio/ unittests

in particular replace rtc::TaskQueue with TaskQueueForTest class since
latter uses DefaultTaskQueueFactory() directly instead of through
GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27245}

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 1d946c5..0db569b 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -13,7 +13,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "api/task_queue/global_task_queue_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
 #include "api/test/mock_frame_encryptor.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
@@ -28,7 +28,7 @@
 #include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
 #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "rtc_base/task_queue.h"
+#include "rtc_base/task_queue_for_test.h"
 #include "system_wrappers/include/clock.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
@@ -128,10 +128,13 @@
 struct ConfigHelper {
   ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
       : clock_(1000000),
+        task_queue_factory_(CreateDefaultTaskQueueFactory()),
         stream_config_(/*send_transport=*/nullptr, /*media_transport=*/nullptr),
         audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
         bitrate_allocator_(&clock_, &limit_observer_),
-        worker_queue_("ConfigHelper_worker_queue"),
+        worker_queue_(task_queue_factory_->CreateTaskQueue(
+            "ConfigHelper_worker_queue",
+            TaskQueueFactory::Priority::NORMAL)),
         audio_encoder_(nullptr) {
     using testing::Invoke;
 
@@ -167,7 +170,7 @@
     return std::unique_ptr<internal::AudioSendStream>(
         new internal::AudioSendStream(
             Clock::GetRealTimeClock(), stream_config_, audio_state_,
-            &GlobalTaskQueueFactory(), &rtp_transport_, &bitrate_allocator_,
+            task_queue_factory_.get(), &rtp_transport_, &bitrate_allocator_,
             &event_log_, &rtcp_rtt_stats_, absl::nullopt,
             std::unique_ptr<voe::ChannelSendInterface>(channel_send_)));
   }
@@ -289,6 +292,7 @@
 
  private:
   SimulatedClock clock_;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockChannelSend>* channel_send_ = nullptr;
@@ -303,7 +307,7 @@
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
-  rtc::TaskQueue worker_queue_;
+  TaskQueueForTest worker_queue_;
   std::unique_ptr<AudioEncoder> audio_encoder_;
 };
 }  // namespace