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[Unit tests] Remove race condition and dangling pointer to mock.

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Lifetime issue: "webrtc_audio_module_rec_thread" was still accessing
                AudioTransport mock at and after its destruction.

Bug: webrtc:9751
Change-Id: I24308077cdeb77e570b8ec74098f1ae3397b7155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146217
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28635}
This commit is contained in:
Yves Gerey
2019-07-19 22:46:13 +02:00
committed by Commit Bot
parent e34d62caec
commit b93a2451e0
1 changed files with 14 additions and 0 deletions
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14
modules/audio_device/audio_device_unittest.cc
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@ -555,6 +555,13 @@ class MAYBE_AudioDeviceTest
} }
} }
// This is needed by all tests using MockAudioTransport,
// since there is no way to unregister it.
// Without Terminate(), audio_device would still accesses
// the destructed mock via "webrtc_audio_module_rec_thread".
// An alternative would be for the mock to outlive audio_device.
void PreTearDown() { EXPECT_EQ(0, audio_device_->Terminate()); }
virtual ~MAYBE_AudioDeviceTest() { virtual ~MAYBE_AudioDeviceTest() {
if (audio_device_) { if (audio_device_) {
EXPECT_EQ(0, audio_device_->Terminate()); EXPECT_EQ(0, audio_device_->Terminate());
@ -937,6 +944,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithInternalRestart) {
EXPECT_TRUE(audio_device()->Playing()); EXPECT_TRUE(audio_device()->Playing());
// Stop playout and the audio thread after successful internal restart. // Stop playout and the audio thread after successful internal restart.
StopPlayout(); StopPlayout();
PreTearDown();
} }
// Tests Start/Stop recording followed by a second session (emulates a restart // Tests Start/Stop recording followed by a second session (emulates a restart
@ -983,6 +991,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithInternalRestart) {
EXPECT_TRUE(audio_device()->Recording()); EXPECT_TRUE(audio_device()->Recording());
// Stop recording and the audio thread after successful internal restart. // Stop recording and the audio thread after successful internal restart.
StopRecording(); StopRecording();
PreTearDown();
} }
#endif // #ifdef WEBRTC_WIN #endif // #ifdef WEBRTC_WIN
@ -1016,6 +1025,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartRecordingVerifyCallbacks) {
StartRecording(); StartRecording();
event()->Wait(kTestTimeOutInMilliseconds); event()->Wait(kTestTimeOutInMilliseconds);
StopRecording(); StopRecording();
PreTearDown();
} }
// Start playout and recording (full-duplex audio) and verify that audio is // Start playout and recording (full-duplex audio) and verify that audio is
@ -1035,6 +1045,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
event()->Wait(kTestTimeOutInMilliseconds); event()->Wait(kTestTimeOutInMilliseconds);
StopRecording(); StopRecording();
StopPlayout(); StopPlayout();
PreTearDown();
} }
// Start playout and recording and store recorded data in an intermediate FIFO // Start playout and recording and store recorded data in an intermediate FIFO
@ -1075,6 +1086,7 @@ TEST_P(MAYBE_AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
// bots where relatively large average latencies can happen. // bots where relatively large average latencies can happen.
EXPECT_LE(audio_stream.average_size(), 25u); EXPECT_LE(audio_stream.average_size(), 25u);
PRINT("\n"); PRINT("\n");
PreTearDown();
} }
// Runs audio in full duplex until user hits Enter. Intended as a manual test // Runs audio in full duplex until user hits Enter. Intended as a manual test
@ -1104,6 +1116,7 @@ TEST_P(MAYBE_AudioDeviceTest,
} while (getchar() != '\n'); } while (getchar() != '\n');
StopRecording(); StopRecording();
StopPlayout(); StopPlayout();
PreTearDown();
} }
// Measures loopback latency and reports the min, max and average values for // Measures loopback latency and reports the min, max and average values for
@ -1138,6 +1151,7 @@ TEST_P(MAYBE_AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 2)); kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 2));
// Print out min, max and average delay values for debugging purposes. // Print out min, max and average delay values for debugging purposes.
audio_stream.PrintResults(); audio_stream.PrintResults();
PreTearDown();
} }
#ifdef WEBRTC_WIN #ifdef WEBRTC_WIN