zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

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
Download Patch File Download Diff File Expand all files Collapse all files

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();
}