Move ThreadWrapper to ProcessThread in base.

webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <string.h>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/md5digest.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
 
 using ::testing::AtLeast;
 using ::testing::Invoke;
 using ::testing::_;
 
 namespace webrtc {
 
 namespace {
@@ skipping 398 matching lines @@
 
 // A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz
 // codec, while the derive class AcmIsacMtTest is using iSAC.
 class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
  protected:
   static const int kNumPackets = 500;
   static const int kNumPullCalls = 500;
 
   AudioCodingModuleMtTestOldApi()
       : AudioCodingModuleTestOldApi(),
-        send_thread_(ThreadWrapper::CreateThread(CbSendThread, this, "send")),
-        insert_packet_thread_(ThreadWrapper::CreateThread(
-            CbInsertPacketThread, this, "insert_packet")),
-        pull_audio_thread_(ThreadWrapper::CreateThread(
-            CbPullAudioThread, this, "pull_audio")),
+        send_thread_(PlatformThread::CreateThread(CbSendThread, this, "send")),
+        insert_packet_thread_(PlatformThread::CreateThread(CbInsertPacketThread,
+                                                           this,
+                                                           "insert_packet")),
+        pull_audio_thread_(PlatformThread::CreateThread(CbPullAudioThread,
+                                                        this,
+                                                        "pull_audio")),
         test_complete_(EventWrapper::Create()),
         send_count_(0),
         insert_packet_count_(0),
         pull_audio_count_(0),
         crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
         next_insert_packet_time_ms_(0),
         fake_clock_(new SimulatedClock(0)) {
     clock_ = fake_clock_.get();
   }
 
@@ skipping 89 matching lines @@
         return true;
       }
       ++pull_audio_count_;
     }
     // Now we're not holding the crit sect when calling ACM.
     PullAudio();
     fake_clock_->AdvanceTimeMilliseconds(10);
     return true;
   }
 
-  rtc::scoped_ptr<ThreadWrapper> send_thread_;
-  rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
-  rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
+  rtc::scoped_ptr<PlatformThread> send_thread_;
+  rtc::scoped_ptr<PlatformThread> insert_packet_thread_;
+  rtc::scoped_ptr<PlatformThread> pull_audio_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
   int send_count_;
   int insert_packet_count_;
   int pull_audio_count_ GUARDED_BY(crit_sect_);
   const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
 
 TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
@@ skipping 107 matching lines @@
 class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
  protected:
   static const int kRegisterAfterNumPackets = 5;
   static const int kNumPackets = 10;
   static const int kPacketSizeMs = 30;
   static const int kPacketSizeSamples = kPacketSizeMs * 16;
 
   AcmReRegisterIsacMtTestOldApi()
       : AudioCodingModuleTestOldApi(),
         receive_thread_(
-            ThreadWrapper::CreateThread(CbReceiveThread, this, "receive")),
+            PlatformThread::CreateThread(CbReceiveThread, this, "receive")),
         codec_registration_thread_(
-            ThreadWrapper::CreateThread(CbCodecRegistrationThread,
-                                        this,
-                                        "codec_registration")),
+            PlatformThread::CreateThread(CbCodecRegistrationThread,
+                                         this,
+                                         "codec_registration")),
         test_complete_(EventWrapper::Create()),
         crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
         codec_registered_(false),
         receive_packet_count_(0),
         next_insert_packet_time_ms_(0),
         fake_clock_(new SimulatedClock(0)) {
     AudioEncoderIsac::Config config;
     config.payload_type = kPayloadType;
     isac_encoder_.reset(new AudioEncoderIsac(config));
     clock_ = fake_clock_.get();
@@ skipping 103 matching lines @@
       // Register the iSAC encoder.
       EXPECT_EQ(0, acm_->RegisterSendCodec(codec_));
       codec_registered_ = true;
     }
     if (codec_registered_ && receive_packet_count_ > kNumPackets) {
       test_complete_->Set();
     }
     return true;
   }
 
-  rtc::scoped_ptr<ThreadWrapper> receive_thread_;
-  rtc::scoped_ptr<ThreadWrapper> codec_registration_thread_;
+  rtc::scoped_ptr<PlatformThread> receive_thread_;
+  rtc::scoped_ptr<PlatformThread> codec_registration_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
   const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   bool codec_registered_ GUARDED_BY(crit_sect_);
   int receive_packet_count_ GUARDED_BY(crit_sect_);
   int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<AudioEncoderIsac> isac_encoder_;
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
   test::AudioLoop audio_loop_;
 };
 
@@ skipping 922 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc