[tsan] Guard audio_device_pulse_linux members from concurrent access.

This CL also fixes data races caused by tests themselves.

TBR= henrika@webrtc.org

Bug: webrtc:9751
Change-Id: Ie7c785b27142fd465f5b4dc9fb0628bd7274f1d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146600
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28829}

modules/audio_device/audio_device_unittest.cc

@@ -1010,6 +1010,7 @@ TEST_P(MAYBE_AudioDeviceTest, StartPlayoutVerifyCallbacks) {
   StartPlayout();
   event()->Wait(kTestTimeOutInMilliseconds);
   StopPlayout();
+  PreTearDown();
 }
 
 // Start recording and verify that the native audio layer starts providing real
@@ -1080,13 +1081,14 @@ TEST_P(MAYBE_AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
       std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec)));
   StopRecording();
   StopPlayout();
+  // Avoid concurrent access to audio_stream.
+  PreTearDown();
   // This thresholds is set rather high to accommodate differences in hardware
   // in several devices. The main idea is to capture cases where a very large
   // latency is built up. See http://bugs.webrtc.org/7744 for examples on
   // bots where relatively large average latencies can happen.
   EXPECT_LE(audio_stream.average_size(), 25u);
   PRINT("\n");
-  PreTearDown();
 }
 
 // Runs audio in full duplex until user hits Enter. Intended as a manual test
@@ -1145,13 +1147,14 @@ TEST_P(MAYBE_AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
       std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec)));
   StopRecording();
   StopPlayout();
+  // Avoid concurrent access to audio_stream.
+  PreTearDown();
   // Verify that a sufficient number of transmitted impulses are detected.
   EXPECT_GE(audio_stream.num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 2));
   // Print out min, max and average delay values for debugging purposes.
   audio_stream.PrintResults();
-  PreTearDown();
 }
 
 #ifdef WEBRTC_WIN

modules/audio_device/linux/audio_device_pulse_linux.cc

@@ -871,8 +871,11 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() {
   playSampleSpec.rate = sample_rate_hz_;
 
   // Create a new play stream
-  _playStream =
-      LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL);
+  {
+    rtc::CritScope lock(&_critSect);
+    _playStream =
+        LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL);
+  }
 
   if (!_playStream) {
     RTC_LOG(LS_ERROR) << "failed to create play stream, err="
@@ -941,8 +944,11 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() {
   LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this);
 
   // Mark playout side as initialized
-  _playIsInitialized = true;
-  _sndCardPlayDelay = 0;
+  {
+    rtc::CritScope lock(&_critSect);
+    _playIsInitialized = true;
+    _sndCardPlayDelay = 0;
+  }
 
   return 0;
 }

modules/audio_device/linux/audio_device_pulse_linux.h

@@ -298,7 +298,7 @@ class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
   bool update_speaker_volume_at_startup_;
   bool quit_ RTC_GUARDED_BY(&_critSect);
 
-  uint32_t _sndCardPlayDelay;
+  uint32_t _sndCardPlayDelay RTC_GUARDED_BY(&_critSect);
 
   int32_t _writeErrors;