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rtc::Event: Finalize migration to TimeDelta.

Browse source 
Bug: webrtc:14366
Change-Id: Icd8792a2f9efa5609dd13da2e175042fac101d36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272101
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37844}
This commit is contained in:
Markus Handell 2022-08-19 12:42:31 +00:00 committed by WebRTC LUCI CQ
parent 99d7d6b4f6
commit 0cd0dd3b07
7 changed files with 41 additions and 61 deletions
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28
modules/audio_device/android/audio_device_unittest.cc
View file

@ -60,7 +60,7 @@ namespace webrtc {
// an event indicating that the test was OK.
static const size_t kNumCallbacks = 10;
// Max amount of time we wait for an event to be set while counting callbacks.
static const int kTestTimeOutInMilliseconds = 10 * 1000;
static constexpr TimeDelta kTestTimeOut = TimeDelta::Seconds(10);
// Average number of audio callbacks per second assuming 10ms packet size.
static const size_t kNumCallbacksPerSecond = 100;
// Play out a test file during this time (unit is in seconds).
@ -69,7 +69,7 @@ static const size_t kBitsPerSample = 16;
static const size_t kBytesPerSample = kBitsPerSample / 8;
// Run the full-duplex test during this time (unit is in seconds).
// Note that first `kNumIgnoreFirstCallbacks` are ignored.
static const int kFullDuplexTimeInSec = 5;
static constexpr TimeDelta kFullDuplexTime = TimeDelta::Seconds(5);
// Wait for the callback sequence to stabilize by ignoring this amount of the
// initial callbacks (avoids initial FIFO access).
// Only used in the RunPlayoutAndRecordingInFullDuplex test.
@ -77,8 +77,8 @@ static const size_t kNumIgnoreFirstCallbacks = 50;
// Sets the number of impulses per second in the latency test.
static const int kImpulseFrequencyInHz = 1;
// Length of round-trip latency measurements. Number of transmitted impulses
// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
static const int kMeasureLatencyTimeInSec = 11;
// is kImpulseFrequencyInHz * kMeasureLatencyTime - 1.
static constexpr TimeDelta kMeasureLatencyTime = TimeDelta::Seconds(11);
// Utilized in round-trip latency measurements to avoid capturing noise samples.
static const int kImpulseThreshold = 1000;
static const char kTag[] = "[..........] ";
@ -877,7 +877,7 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopPlayout();
}
@ -896,7 +896,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartRecording();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopRecording();
}
@ -917,7 +917,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
StartRecording();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopRecording();
StopPlayout();
}
@ -937,7 +937,7 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
// SetMaxPlayoutVolume();
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopPlayout();
}
@ -967,13 +967,12 @@ TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
std::unique_ptr<FifoAudioStream> fifo_audio_stream(
new FifoAudioStream(playout_frames_per_10ms_buffer()));
mock.HandleCallbacks(&test_is_done_, fifo_audio_stream.get(),
kFullDuplexTimeInSec * kNumCallbacksPerSecond);
kFullDuplexTime.seconds() * kNumCallbacksPerSecond);
SetMaxPlayoutVolume();
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartRecording();
StartPlayout();
test_is_done_.Wait(
std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
test_is_done_.Wait(std::max(kTestTimeOut, kFullDuplexTime));
StopPlayout();
StopRecording();
@ -1000,20 +999,19 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
kMeasureLatencyTime.seconds() * kNumCallbacksPerSecond);
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
SetMaxPlayoutVolume();
DisableBuiltInAECIfAvailable();
StartRecording();
StartPlayout();
test_is_done_.Wait(
std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
test_is_done_.Wait(std::max(kTestTimeOut, kMeasureLatencyTime));
StopPlayout();
StopRecording();
// Verify that the correct number of transmitted impulses are detected.
EXPECT_EQ(latency_audio_stream->num_latency_values(),
static_cast<size_t>(
kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
kImpulseFrequencyInHz * kMeasureLatencyTime.seconds() - 1));
latency_audio_stream->PrintResults();
}

2
modules/audio_device/mac/audio_device_mac.cc
View file

@ -1329,7 +1329,7 @@ int32_t AudioDeviceMac::StopRecording() {
_recording = false;
_doStopRec = true; // Signal to io proc to stop audio device
mutex_.Unlock(); // Cannot be under lock, risk of deadlock
if (!_stopEventRec.Wait(2000)) {
if (!_stopEventRec.Wait(TimeDelta::Seconds(2))) {
MutexLock lockScoped(&mutex_);
RTC_LOG(LS_WARNING) << "Timed out stopping the capture IOProc."
"We may have failed to detect a device removal.";

8
modules/desktop_capture/win/wgc_capture_session.cc
View file

@ -42,8 +42,8 @@ constexpr auto kPixelFormat = ABI::Windows::Graphics::DirectX::
// The maximum time `GetFrame` will wait for a frame to arrive, if we don't have
// any in the pool.
constexpr int kMaxWaitForFrameMs = 50;
constexpr int kMaxWaitForFirstFrameMs = 500;
constexpr TimeDelta kMaxWaitForFrame = TimeDelta::Millis(50);
constexpr TimeDelta kMaxWaitForFirstFrame = TimeDelta::Millis(500);
// These values are persisted to logs. Entries should not be renumbered and
// numeric values should never be reused.
@ -213,8 +213,8 @@ HRESULT WgcCaptureSession::GetFrame(
RTC_DCHECK(is_capture_started_);
if (frames_in_pool_ < 1)
wait_for_frame_event_.Wait(first_frame_ ? kMaxWaitForFirstFrameMs
: kMaxWaitForFrameMs);
wait_for_frame_event_.Wait(first_frame_ ? kMaxWaitForFirstFrame
: kMaxWaitForFrame);
ComPtr<WGC::IDirect3D11CaptureFrame> capture_frame;
HRESULT hr = frame_pool_->TryGetNextFrame(&capture_frame);

20
rtc_base/event.h
View file

@ -50,29 +50,13 @@ class Event {
bool Wait(webrtc::TimeDelta give_up_after, webrtc::TimeDelta warn_after);
// Waits with the given timeout and a reasonable default warning timeout.
// TODO(bugs.webrtc.org/14366): De-template this after millisec-based Wait is
// removed.
template <class T>
bool Wait(T give_up_after) {
webrtc::TimeDelta duration = ToTimeDelta(give_up_after);
return Wait(duration, duration.IsPlusInfinity()
bool Wait(webrtc::TimeDelta give_up_after) {
return Wait(give_up_after, give_up_after.IsPlusInfinity()
? webrtc::TimeDelta::Seconds(3)
: kForever);
}
private:
// TODO(bugs.webrtc.org/14366): Remove after millisec-based Wait is removed.
static webrtc::TimeDelta ToTimeDelta(int duration) {
// SocketServer users can get here with SocketServer::kForever which is
// -1. Mirror the definition here to avoid dependence.
constexpr int kForeverMs = -1;
return duration == kForeverMs ? kForever
: webrtc::TimeDelta::Millis(duration);
}
static webrtc::TimeDelta ToTimeDelta(webrtc::TimeDelta duration) {
return duration;
}
#if defined(WEBRTC_WIN)
HANDLE event_handle_;
#elif defined(WEBRTC_POSIX)

30
sdk/android/native_unittests/audio_device/audio_device_unittest.cc
View file

@ -57,7 +57,7 @@ namespace jni {
// an event indicating that the test was OK.
static const size_t kNumCallbacks = 10;
// Max amount of time we wait for an event to be set while counting callbacks.
static const int kTestTimeOutInMilliseconds = 10 * 1000;
static constexpr TimeDelta kTestTimeOut = TimeDelta::Seconds(10);
// Average number of audio callbacks per second assuming 10ms packet size.
static const size_t kNumCallbacksPerSecond = 100;
// Play out a test file during this time (unit is in seconds).
@ -66,7 +66,7 @@ static const size_t kBitsPerSample = 16;
static const size_t kBytesPerSample = kBitsPerSample / 8;
// Run the full-duplex test during this time (unit is in seconds).
// Note that first `kNumIgnoreFirstCallbacks` are ignored.
static const int kFullDuplexTimeInSec = 5;
static constexpr TimeDelta kFullDuplexTime = TimeDelta::Seconds(5);
// Wait for the callback sequence to stabilize by ignoring this amount of the
// initial callbacks (avoids initial FIFO access).
// Only used in the RunPlayoutAndRecordingInFullDuplex test.
@ -74,8 +74,8 @@ static const size_t kNumIgnoreFirstCallbacks = 50;
// Sets the number of impulses per second in the latency test.
static const int kImpulseFrequencyInHz = 1;
// Length of round-trip latency measurements. Number of transmitted impulses
// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
static const int kMeasureLatencyTimeInSec = 11;
// is kImpulseFrequencyInHz * kMeasureLatencyTime - 1.
static constexpr TimeDelta kMeasureLatencyTime = TimeDelta::Seconds(11);
// Utilized in round-trip latency measurements to avoid capturing noise samples.
static const int kImpulseThreshold = 1000;
static const char kTag[] = "[..........] ";
@ -880,7 +880,7 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopPlayout();
}
@ -897,7 +897,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartRecording();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopRecording();
}
@ -918,7 +918,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
StartRecording();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopRecording();
StopPlayout();
}
@ -938,7 +938,7 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
// SetMaxPlayoutVolume();
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartPlayout();
test_is_done_.Wait(kTestTimeOutInMilliseconds);
test_is_done_.Wait(kTestTimeOut);
StopPlayout();
}
@ -1059,7 +1059,7 @@ TEST_F(AudioDeviceTest, AudioParametersWithNonDefaultConstruction) {
// one packet on average. However, under more realistic conditions, the size
// of the FIFO will vary more due to an unbalance between the two sides.
// This test tries to verify that the device maintains a balanced callback-
// sequence by running in loopback for kFullDuplexTimeInSec seconds while
// sequence by running in loopback for kFullDuplexTime seconds while
// measuring the size (max and average) of the FIFO. The size of the FIFO is
// increased by the recording side and decreased by the playout side.
// TODO(henrika): tune the final test parameters after running tests on several
@ -1077,13 +1077,12 @@ TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
std::unique_ptr<FifoAudioStream> fifo_audio_stream(
new FifoAudioStream(playout_frames_per_10ms_buffer()));
mock.HandleCallbacks(&test_is_done_, fifo_audio_stream.get(),
kFullDuplexTimeInSec * kNumCallbacksPerSecond);
kFullDuplexTime.seconds() * kNumCallbacksPerSecond);
SetMaxPlayoutVolume();
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
StartRecording();
StartPlayout();
test_is_done_.Wait(
std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
test_is_done_.Wait(std::max(kTestTimeOut, kFullDuplexTime));
StopPlayout();
StopRecording();
@ -1110,20 +1109,19 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
kMeasureLatencyTime.seconds() * kNumCallbacksPerSecond);
EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
SetMaxPlayoutVolume();
DisableBuiltInAECIfAvailable();
StartRecording();
StartPlayout();
test_is_done_.Wait(
std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
test_is_done_.Wait(std::max(kTestTimeOut, kMeasureLatencyTime));
StopPlayout();
StopRecording();
// Verify that the correct number of transmitted impulses are detected.
EXPECT_EQ(latency_audio_stream->num_latency_values(),
static_cast<size_t>(
kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
kImpulseFrequencyInHz * kMeasureLatencyTime.seconds() - 1));
latency_audio_stream->PrintResults();
}

2
sdk/android/native_unittests/stacktrace/stacktrace_unittest.cc
View file

@ -263,7 +263,7 @@ TEST(Stacktrace, TestRtcEventDeadlockDetection) {
// The message should appear after 3 sec. We'll wait up to 10 sec in an
// attempt to not be flaky.
EXPECT_TRUE(sink.WhenFound().Wait(10000));
EXPECT_TRUE(sink.WhenFound().Wait(TimeDelta::Seconds(10)));
// Unblock the thread and shut it down.
ev.Set();

8
sdk/objc/unittests/RTCAudioSessionTest.mm
View file

@ -287,15 +287,15 @@ OCMLocation *OCMMakeLocation(id testCase, const char *fileCString, int line){
rtc::Event waitLock;
rtc::Event waitCleanup;
constexpr int timeoutMs = 5000;
thread->PostTask([audioSession, &waitLock, &waitCleanup] {
constexpr webrtc::TimeDelta timeout = webrtc::TimeDelta::Seconds(5);
thread->PostTask([audioSession, &waitLock, &waitCleanup, timeout] {
[audioSession lockForConfiguration];
waitLock.Set();
waitCleanup.Wait(timeoutMs);
waitCleanup.Wait(timeout);
[audioSession unlockForConfiguration];
});
waitLock.Wait(timeoutMs);
waitLock.Wait(timeout);
[audioSession setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:0 error:&error];
EXPECT_TRUE(error != nil);
EXPECT_EQ(error.domain, kRTCAudioSessionErrorDomain);