Replace some usage of EventWrapper with rtc::Event.
Bug: webrtc:3380 Change-Id: Id33b19bf107273e6f838aa633784db73d02ae2c2 Reviewed-on: https://webrtc-review.googlesource.com/c/107888 Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25407}
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@ -35,6 +35,7 @@
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/event.h"
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#include "rtc_base/messagedigest.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/platform_thread.h"
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@ -42,7 +43,6 @@
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#include "rtc_base/system/arch.h"
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#include "rtc_base/thread_annotations.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/event_wrapper.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/gtest.h"
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#include "test/mock_audio_decoder.h"
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@ -481,7 +481,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
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send_thread_(CbSendThread, this, "send"),
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insert_packet_thread_(CbInsertPacketThread, this, "insert_packet"),
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pull_audio_thread_(CbPullAudioThread, this, "pull_audio"),
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test_complete_(EventWrapper::Create()),
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test_complete_(false, false),
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send_count_(0),
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insert_packet_count_(0),
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pull_audio_count_(0),
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@ -512,8 +512,8 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
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insert_packet_thread_.Stop();
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}
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EventTypeWrapper RunTest() {
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return test_complete_->Wait(10 * 60 * 1000); // 10 minutes' timeout.
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bool RunTest() {
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return test_complete_.Wait(10 * 60 * 1000); // 10 minutes' timeout.
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}
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virtual bool TestDone() {
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@ -538,12 +538,12 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
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SleepMs(1);
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if (HasFatalFailure()) {
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// End the test early if a fatal failure (ASSERT_*) has occurred.
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test_complete_->Set();
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test_complete_.Set();
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}
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++send_count_;
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InsertAudioAndVerifyEncoding();
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if (TestDone()) {
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test_complete_->Set();
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test_complete_.Set();
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}
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return true;
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}
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@ -592,7 +592,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
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rtc::PlatformThread send_thread_;
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rtc::PlatformThread insert_packet_thread_;
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rtc::PlatformThread pull_audio_thread_;
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const std::unique_ptr<EventWrapper> test_complete_;
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rtc::Event test_complete_;
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int send_count_;
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int insert_packet_count_;
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int pull_audio_count_ RTC_GUARDED_BY(crit_sect_);
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@ -607,7 +607,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
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#define MAYBE_DoTest DoTest
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#endif
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TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
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EXPECT_EQ(kEventSignaled, RunTest());
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EXPECT_TRUE(RunTest());
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}
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// This is a multi-threaded ACM test using iSAC. The test encodes audio
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@ -717,7 +717,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
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#endif
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#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
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TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
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EXPECT_EQ(kEventSignaled, RunTest());
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EXPECT_TRUE(RunTest());
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}
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#endif
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@ -734,7 +734,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
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codec_registration_thread_(CbCodecRegistrationThread,
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this,
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"codec_registration"),
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test_complete_(EventWrapper::Create()),
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test_complete_(false, false),
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codec_registered_(false),
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receive_packet_count_(0),
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next_insert_packet_time_ms_(0),
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@ -781,8 +781,8 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
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codec_registration_thread_.Stop();
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}
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EventTypeWrapper RunTest() {
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return test_complete_->Wait(10 * 60 * 1000); // 10 minutes' timeout.
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bool RunTest() {
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return test_complete_.Wait(10 * 60 * 1000); // 10 minutes' timeout.
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}
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static bool CbReceiveThread(void* context) {
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@ -845,7 +845,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
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SleepMs(1);
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if (HasFatalFailure()) {
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// End the test early if a fatal failure (ASSERT_*) has occurred.
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test_complete_->Set();
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test_complete_.Set();
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}
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rtc::CritScope lock(&crit_sect_);
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if (!codec_registered_ &&
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@ -856,14 +856,14 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
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codec_registered_ = true;
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}
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if (codec_registered_ && receive_packet_count_ > kNumPackets) {
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test_complete_->Set();
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test_complete_.Set();
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}
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return true;
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}
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rtc::PlatformThread receive_thread_;
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rtc::PlatformThread codec_registration_thread_;
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const std::unique_ptr<EventWrapper> test_complete_;
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rtc::Event test_complete_;
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rtc::CriticalSection crit_sect_;
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bool codec_registered_ RTC_GUARDED_BY(crit_sect_);
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int receive_packet_count_ RTC_GUARDED_BY(crit_sect_);
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@ -880,7 +880,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
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#endif
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#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
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TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
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EXPECT_EQ(kEventSignaled, RunTest());
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EXPECT_TRUE(RunTest());
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}
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#endif
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