Removes flaky thread checker in AudioDeviceBuffer.

This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:

- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.

To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.

TBR=glaznev

Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}

modules/audio_device/android/aaudio_player.cc

@@ -219,7 +219,6 @@ void AAudioPlayer::HandleStreamDisconnected() {
   }
   // Perform a restart by first closing the disconnected stream and then start
   // a new stream; this time using the new (preferred) audio output device.
-  audio_device_buffer_->NativeAudioPlayoutInterrupted();
   StopPlayout();
   InitPlayout();
   StartPlayout();

modules/audio_device/android/aaudio_recorder.cc

@@ -212,7 +212,6 @@ void AAudioRecorder::HandleStreamDisconnected() {
   // TODO(henrika): resolve issue where a one restart attempt leads to a long
   // sequence of new calls to OnErrorCallback().
   // See b/73148976 for details.
-  audio_device_buffer_->NativeAudioRecordingInterrupted();
   StopRecording();
   InitRecording();
   StartRecording();

modules/audio_device/audio_device_buffer.cc

@@ -67,8 +67,6 @@ AudioDeviceBuffer::AudioDeviceBuffer()
   RTC_LOG(WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
 #endif
   WebRtcSpl_Init();
-  playout_thread_checker_.DetachFromThread();
-  recording_thread_checker_.DetachFromThread();
 }
 
 AudioDeviceBuffer::~AudioDeviceBuffer() {
@@ -99,7 +97,6 @@ void AudioDeviceBuffer::StartPlayout() {
     return;
   }
   RTC_LOG(INFO) << __FUNCTION__;
-  playout_thread_checker_.DetachFromThread();
   // Clear members tracking playout stats and do it on the task queue.
   task_queue_.PostTask([this] { ResetPlayStats(); });
   // Start a periodic timer based on task queue if not already done by the
@@ -119,7 +116,6 @@ void AudioDeviceBuffer::StartRecording() {
     return;
   }
   RTC_LOG(INFO) << __FUNCTION__;
-  recording_thread_checker_.DetachFromThread();
   // Clear members tracking recording stats and do it on the task queue.
   task_queue_.PostTask([this] { ResetRecStats(); });
   // Start a periodic timer based on task queue if not already done by the
@@ -222,30 +218,17 @@ size_t AudioDeviceBuffer::PlayoutChannels() const {
 }
 
 int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
-  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   typing_status_ = typing_status;
   return 0;
 }
 
-void AudioDeviceBuffer::NativeAudioPlayoutInterrupted() {
-  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
-  playout_thread_checker_.DetachFromThread();
-}
-
-void AudioDeviceBuffer::NativeAudioRecordingInterrupted() {
-  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
-  recording_thread_checker_.DetachFromThread();
-}
-
 void AudioDeviceBuffer::SetVQEData(int play_delay_ms, int rec_delay_ms) {
-  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   play_delay_ms_ = play_delay_ms;
   rec_delay_ms_ = rec_delay_ms;
 }
 
 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
                                              size_t samples_per_channel) {
-  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   // Copy the complete input buffer to the local buffer.
   const size_t old_size = rec_buffer_.size();
   rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
@@ -277,7 +260,6 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
 }
 
 int32_t AudioDeviceBuffer::DeliverRecordedData() {
-  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   if (!audio_transport_cb_) {
     RTC_LOG(LS_WARNING) << "Invalid audio transport";
     return 0;
@@ -297,7 +279,6 @@ int32_t AudioDeviceBuffer::DeliverRecordedData() {
 }
 
 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) {
-  RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   // The consumer can change the requested size on the fly and we therefore
   // resize the buffer accordingly. Also takes place at the first call to this
   // method.
@@ -342,7 +323,6 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) {
 }
 
 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
-  RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   RTC_DCHECK_GT(play_buffer_.size(), 0);
 #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
   const double phase_increment =
@@ -484,7 +464,6 @@ void AudioDeviceBuffer::ResetPlayStats() {
 
 void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs,
                                        size_t samples_per_channel) {
-  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   rtc::CritScope cs(&lock_);
   ++stats_.rec_callbacks;
   stats_.rec_samples += samples_per_channel;
@@ -495,7 +474,6 @@ void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs,
 
 void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs,
                                         size_t samples_per_channel) {
-  RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   rtc::CritScope cs(&lock_);
   ++stats_.play_callbacks;
   stats_.play_samples += samples_per_channel;

modules/audio_device/audio_device_buffer.h

@@ -104,13 +104,6 @@ class AudioDeviceBuffer {
 
   int32_t SetTypingStatus(bool typing_status);
 
-  // Called on iOS and Android where the native audio layer can be interrupted
-  // by other audio applications. These methods can then be used to reset
-  // internal states and detach thread checkers to allow for new audio sessions
-  // where native callbacks may come from a new set of I/O threads.
-  void NativeAudioPlayoutInterrupted();
-  void NativeAudioRecordingInterrupted();
-
  private:
   // Starts/stops periodic logging of audio stats.
   void StartPeriodicLogging();
@@ -145,12 +138,6 @@ class AudioDeviceBuffer {
   // Main thread on which this object is created.
   rtc::ThreadChecker main_thread_checker_;
 
-  // Native (platform specific) audio thread driving the playout side.
-  rtc::ThreadChecker playout_thread_checker_;
-
-  // Native (platform specific) audio thread driving the recording side.
-  rtc::ThreadChecker recording_thread_checker_;
-
   rtc::CriticalSection lock_;
 
   // Task queue used to invoke LogStats() periodically. Tasks are executed on a
@@ -183,18 +170,18 @@ class AudioDeviceBuffer {
   // Buffer used for audio samples to be played out. Size can be changed
   // dynamically. The 16-bit samples are interleaved, hence the size is
   // proportional to the number of channels.
-  rtc::BufferT<int16_t> play_buffer_ RTC_GUARDED_BY(playout_thread_checker_);
+  rtc::BufferT<int16_t> play_buffer_;
 
   // Byte buffer used for recorded audio samples. Size can be changed
   // dynamically.
-  rtc::BufferT<int16_t> rec_buffer_ RTC_GUARDED_BY(recording_thread_checker_);
+  rtc::BufferT<int16_t> rec_buffer_;
 
   // Contains true of a key-press has been detected.
-  bool typing_status_ RTC_GUARDED_BY(recording_thread_checker_);
+  bool typing_status_;
 
   // Delay values used by the AEC.
-  int play_delay_ms_ RTC_GUARDED_BY(recording_thread_checker_);
+  int play_delay_ms_;
-  int rec_delay_ms_ RTC_GUARDED_BY(recording_thread_checker_);
+  int rec_delay_ms_;
 
   // Counts number of times LogStats() has been called.
   size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
@@ -204,8 +191,8 @@ class AudioDeviceBuffer {
 
   // Counts number of audio callbacks modulo 50 to create a signal when
   // a new storage of audio stats shall be done.
-  int16_t rec_stat_count_ RTC_GUARDED_BY(recording_thread_checker_);
+  int16_t rec_stat_count_;
-  int16_t play_stat_count_ RTC_GUARDED_BY(playout_thread_checker_);
+  int16_t play_stat_count_;
 
   // Time stamps of when playout and recording starts.
   int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
@@ -231,7 +218,7 @@ class AudioDeviceBuffer {
 // Should *never* be defined in production builds. Only used for testing.
 // When defined, the output signal will be replaced by a sinus tone at 440Hz.
 #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
-  double phase_ RTC_GUARDED_BY(playout_thread_checker_);
+  double phase_;
 #endif
 };
 

modules/audio_device/ios/audio_device_ios.mm

@@ -903,12 +903,6 @@ void AudioDeviceIOS::PrepareForNewStart() {
   // which means that we must detach thread checkers here to be prepared for an
   // upcoming new audio stream.
   io_thread_checker_.DetachFromThread();
-  // The audio device buffer must also be informed about the interrupted
-  // state so it can detach its thread checkers as well.
-  if (audio_device_buffer_) {
-    audio_device_buffer_->NativeAudioPlayoutInterrupted();
-    audio_device_buffer_->NativeAudioRecordingInterrupted();
-  }
 }
 
 }  // namespace webrtc

modules/audio_device/win/core_audio_base_win.cc

@@ -582,8 +582,6 @@ bool CoreAudioBase::Stop() {
   // error callbacks.
   if (!IsRestarting()) {
     thread_checker_audio_.DetachFromThread();
-    IsOutput() ? audio_device_buffer_->NativeAudioPlayoutInterrupted()
-               : audio_device_buffer_->NativeAudioRecordingInterrupted();
   }
 
   // Release all allocated COM interfaces to allow for a restart without

sdk/android/src/jni/audio_device/aaudio_player.cc

@@ -235,7 +235,6 @@ void AAudioPlayer::HandleStreamDisconnected() {
   }
   // Perform a restart by first closing the disconnected stream and then start
   // a new stream; this time using the new (preferred) audio output device.
-  audio_device_buffer_->NativeAudioPlayoutInterrupted();
   StopPlayout();
   InitPlayout();
   StartPlayout();

sdk/android/src/jni/audio_device/aaudio_recorder.cc

@@ -224,7 +224,6 @@ void AAudioRecorder::HandleStreamDisconnected() {
   // TODO(henrika): resolve issue where a one restart attempt leads to a long
   // sequence of new calls to OnErrorCallback().
   // See b/73148976 for details.
-  audio_device_buffer_->NativeAudioRecordingInterrupted();
   StopRecording();
   InitRecording();
   StartRecording();

sdk/objc/native/src/audio/audio_device_ios.mm

@@ -915,12 +915,6 @@ void AudioDeviceIOS::PrepareForNewStart() {
   // which means that we must detach thread checkers here to be prepared for an
   // upcoming new audio stream.
   io_thread_checker_.DetachFromThread();
-  // The audio device buffer must also be informed about the interrupted
-  // state so it can detach its thread checkers as well.
-  if (audio_device_buffer_) {
-    audio_device_buffer_->NativeAudioPlayoutInterrupted();
-    audio_device_buffer_->NativeAudioRecordingInterrupted();
-  }
 }
 
 bool AudioDeviceIOS::IsInterrupted() {