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/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;