[Unit tests] Remove race condition and dangling pointer to mock.
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}
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@ -555,6 +555,13 @@ class MAYBE_AudioDeviceTest
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}
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}
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// This is needed by all tests using MockAudioTransport,
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// since there is no way to unregister it.
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// Without Terminate(), audio_device would still accesses
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// the destructed mock via "webrtc_audio_module_rec_thread".
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// An alternative would be for the mock to outlive audio_device.
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void PreTearDown() { EXPECT_EQ(0, audio_device_->Terminate()); }
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virtual ~MAYBE_AudioDeviceTest() {
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if (audio_device_) {
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EXPECT_EQ(0, audio_device_->Terminate());
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@ -937,6 +944,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithInternalRestart) {
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EXPECT_TRUE(audio_device()->Playing());
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// Stop playout and the audio thread after successful internal restart.
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StopPlayout();
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PreTearDown();
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}
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// Tests Start/Stop recording followed by a second session (emulates a restart
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@ -983,6 +991,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithInternalRestart) {
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EXPECT_TRUE(audio_device()->Recording());
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// Stop recording and the audio thread after successful internal restart.
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StopRecording();
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PreTearDown();
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}
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#endif // #ifdef WEBRTC_WIN
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@ -1016,6 +1025,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartRecordingVerifyCallbacks) {
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StartRecording();
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event()->Wait(kTestTimeOutInMilliseconds);
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StopRecording();
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PreTearDown();
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}
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// Start playout and recording (full-duplex audio) and verify that audio is
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@ -1035,6 +1045,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
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event()->Wait(kTestTimeOutInMilliseconds);
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StopRecording();
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StopPlayout();
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PreTearDown();
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}
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// Start playout and recording and store recorded data in an intermediate FIFO
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@ -1075,6 +1086,7 @@ TEST_P(MAYBE_AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
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// bots where relatively large average latencies can happen.
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EXPECT_LE(audio_stream.average_size(), 25u);
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PRINT("\n");
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PreTearDown();
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}
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// Runs audio in full duplex until user hits Enter. Intended as a manual test
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@ -1104,6 +1116,7 @@ TEST_P(MAYBE_AudioDeviceTest,
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} while (getchar() != '\n');
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StopRecording();
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StopPlayout();
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PreTearDown();
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}
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// Measures loopback latency and reports the min, max and average values for
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@ -1138,6 +1151,7 @@ TEST_P(MAYBE_AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
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kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 2));
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// Print out min, max and average delay values for debugging purposes.
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audio_stream.PrintResults();
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PreTearDown();
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}
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#ifdef WEBRTC_WIN
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