Adds thread safety annotations to the AudioDeviceBuffer class

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 20 matching lines @@
 static const size_t kTimerIntervalInMilliseconds =
     kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec;
 // Min time required to qualify an audio session as a "call". If playout or
 // recording has been active for less than this time we will not store any
 // logs or UMA stats but instead consider the call as too short.
 static const size_t kMinValidCallTimeTimeInSeconds = 10;
 static const size_t kMinValidCallTimeTimeInMilliseconds =
     kMinValidCallTimeTimeInSeconds * rtc::kNumMillisecsPerSec;
 
 AudioDeviceBuffer::AudioDeviceBuffer()
-    : audio_transport_cb_(nullptr),
-      task_queue_(kTimerQueueName),
-      playing_(false),
-      recording_(false),
+    : task_queue_(kTimerQueueName),
+      audio_transport_cb_(nullptr),
       rec_sample_rate_(0),
       play_sample_rate_(0),
       rec_channels_(0),
       play_channels_(0),
-      rec_bytes_per_sample_(0),
-      play_bytes_per_sample_(0),
+      playing_(false),
+      recording_(false),
       current_mic_level_(0),
       new_mic_level_(0),
       typing_status_(false),
       play_delay_ms_(0),
       rec_delay_ms_(0),
       clock_drift_(0),
       num_stat_reports_(0),
       rec_callbacks_(0),
       last_rec_callbacks_(0),
       play_callbacks_(0),
       last_play_callbacks_(0),
       rec_samples_(0),
       last_rec_samples_(0),
       play_samples_(0),
       last_play_samples_(0),
-      last_timer_task_time_(0),
       max_rec_level_(0),
       max_play_level_(0),
+      last_timer_task_time_(0),
       rec_stat_count_(0),
       play_stat_count_(0),
       play_start_time_(0),
       rec_start_time_(0),
       only_silence_recorded_(true) {
   LOG(INFO) << "AudioDeviceBuffer::ctor";
 }
 
 AudioDeviceBuffer::~AudioDeviceBuffer() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
   RTC_DCHECK(!playing_);
   RTC_DCHECK(!recording_);
   LOG(INFO) << "AudioDeviceBuffer::~dtor";
 }
 
 int32_t AudioDeviceBuffer::RegisterAudioCallback(
     AudioTransport* audio_callback) {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
   LOG(INFO) << __FUNCTION__;
-  rtc::CritScope lock(&lock_cb_);
+  if (playing_ || recording_) {
+    LOG(LS_ERROR) << "Failed to set audio transport since media was active";
+    return -1;
+  }
   audio_transport_cb_ = audio_callback;
   return 0;
 }
 
 void AudioDeviceBuffer::StartPlayout() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
   // TODO(henrika): allow for usage of DCHECK(!playing_) here instead. Today the
   // ADM allows calling Start(), Start() by ignoring the second call but it
   // makes more sense to only allow one call.
   if (playing_) {
     return;
   }
   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
   // recording side.
   if (!recording_) {
     StartPeriodicLogging();
   }
   const uint64_t now_time = rtc::TimeMillis();
   // Clear members that are only touched on the main (creating) thread.
   play_start_time_ = now_time;
-  last_playout_time_ = now_time;
   playing_ = true;
 }
 
 void AudioDeviceBuffer::StartRecording() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
   if (recording_) {
     return;
   }
   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
   // playout side.
   if (!playing_) {
     StartPeriodicLogging();
   }
   // Clear members that will be touched on the main (creating) thread.
   rec_start_time_ = rtc::TimeMillis();
   recording_ = true;
   // And finally a member which can be modified on the native audio thread.
   // It is safe to do so since we know by design that the owning ADM has not
   // yet started the native audio recording.
   only_silence_recorded_ = true;
 }
 
 void AudioDeviceBuffer::StopPlayout() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
   if (!playing_) {
     return;
   }
   LOG(INFO) << __FUNCTION__;
   playing_ = false;
   // Stop periodic logging if no more media is active.
   if (!recording_) {
     StopPeriodicLogging();
   }
-  // Add diagnostic logging of delta times for playout callbacks. We are doing
-  // this wihout a lock since playout should be stopped by now and it a minor
-  // conflict during stop will not have a great impact on the total statistics.
-  const size_t time_since_start = rtc::TimeSince(play_start_time_);
-  if (time_since_start > kMinValidCallTimeTimeInMilliseconds) {
-    size_t total_diff_time = 0;
-    int num_measurements = 0;
-    LOG(INFO) << "[playout diff time => #measurements]";
-    for (size_t diff = 0; diff < arraysize(playout_diff_times_); ++diff) {
-      uint32_t num_elements = playout_diff_times_[diff];
-      if (num_elements > 0) {
-        total_diff_time += num_elements * diff;
-        num_measurements += num_elements;
-        LOG(INFO) << "[" << diff << " => " << num_elements << "]";
-      }
-    }
-    if (num_measurements > 0) {
-      LOG(INFO) << "total_diff_time: " << total_diff_time << ", "
-                << "num_measurements: " << num_measurements << ", "
-                << "average: "
-                << static_cast<float>(total_diff_time) / num_measurements;
-    }
-  }
-  LOG(INFO) << "total playout time: " << time_since_start;
+  LOG(INFO) << "total playout time: " << rtc::TimeSince(play_start_time_);
 }
 
 void AudioDeviceBuffer::StopRecording() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
   if (!recording_) {
     return;
   }
   LOG(INFO) << __FUNCTION__;
   recording_ = false;
   // Stop periodic logging if no more media is active.
   if (!playing_) {
     StopPeriodicLogging();
   }
   // Add UMA histogram to keep track of the case when only zeros have been
   // recorded. Measurements (max of absolute level) are taken twice per second,
   // which means that if e.g 10 seconds of audio has been recorded, a total of
   // 20 level estimates must all be identical to zero to trigger the histogram.
   // |only_silence_recorded_| can only be cleared on the native audio thread
   // that drives audio capture but we know by design that the audio has stopped
   // when this method is called, hence there should not be aby conflicts. Also,
   // the fact that |only_silence_recorded_| can be affected during the complete
   // call makes chances of conflicts with potentially one last callback very
   // small.
   const size_t time_since_start = rtc::TimeSince(rec_start_time_);
   if (time_since_start > kMinValidCallTimeTimeInMilliseconds) {
     const int only_zeros = static_cast<int>(only_silence_recorded_);
     RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros", only_zeros);
     LOG(INFO) << "HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): " << only_zeros;
   }
   LOG(INFO) << "total recording time: " << time_since_start;
 }
 
 int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")";
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   rec_sample_rate_ = fsHz;
   return 0;
 }
 
 int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   play_sample_rate_ = fsHz;
   return 0;
 }
 
 int32_t AudioDeviceBuffer::RecordingSampleRate() const {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   return rec_sample_rate_;
 }
 
 int32_t AudioDeviceBuffer::PlayoutSampleRate() const {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   return play_sample_rate_;
 }
 
 int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   LOG(INFO) << "SetRecordingChannels(" << channels << ")";
-  rtc::CritScope lock(&lock_);
   rec_channels_ = channels;
-  rec_bytes_per_sample_ = sizeof(int16_t) * channels;
   return 0;
 }
 
 int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
-  rtc::CritScope lock(&lock_);
   play_channels_ = channels;
-  play_bytes_per_sample_ = sizeof(int16_t) * channels;
   return 0;
 }
 
 int32_t AudioDeviceBuffer::SetRecordingChannel(
     const AudioDeviceModule::ChannelType channel) {
   LOG(INFO) << "SetRecordingChannel(" << channel << ")";
   LOG(LS_WARNING) << "Not implemented";
   // Add DCHECK to ensure that user does not try to use this API with a non-
   // default parameter.
   RTC_DCHECK_EQ(channel, AudioDeviceModule::kChannelBoth);
   return -1;
 }
 
 int32_t AudioDeviceBuffer::RecordingChannel(
     AudioDeviceModule::ChannelType& channel) const {
   LOG(LS_WARNING) << "Not implemented";
   return -1;
 }
 
 size_t AudioDeviceBuffer::RecordingChannels() const {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   return rec_channels_;
 }
 
 size_t AudioDeviceBuffer::PlayoutChannels() const {
+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
   return play_channels_;
 }
 
 int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level) {
+  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   current_mic_level_ = level;
   return 0;
 }
 
 int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
+  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   typing_status_ = typing_status;
   return 0;
 }
 
 uint32_t AudioDeviceBuffer::NewMicLevel() const {
+  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   return new_mic_level_;
 }
 
 void AudioDeviceBuffer::SetVQEData(int play_delay_ms,
                                    int rec_delay_ms,
                                    int clock_drift) {
+  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   play_delay_ms_ = play_delay_ms;
   rec_delay_ms_ = rec_delay_ms;
   clock_drift_ = clock_drift;
 }
 
 int32_t AudioDeviceBuffer::StartInputFileRecording(
     const char fileName[kAdmMaxFileNameSize]) {
   LOG(LS_WARNING) << "Not implemented";
   return 0;
 }
 
 int32_t AudioDeviceBuffer::StopInputFileRecording() {
   LOG(LS_WARNING) << "Not implemented";
   return 0;
 }
 
 int32_t AudioDeviceBuffer::StartOutputFileRecording(
     const char fileName[kAdmMaxFileNameSize]) {
   LOG(LS_WARNING) << "Not implemented";
   return 0;
 }
 
 int32_t AudioDeviceBuffer::StopOutputFileRecording() {
   LOG(LS_WARNING) << "Not implemented";
   return 0;
 }
 
 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
                                              size_t num_samples) {
-  const size_t rec_channels = [&] {
-    rtc::CritScope lock(&lock_);
-    return rec_channels_;
-  }();
+  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   // Copy the complete input buffer to the local buffer.
-  const size_t size_in_bytes = num_samples * rec_channels * sizeof(int16_t);
+  const size_t size_in_bytes = num_samples * rec_channels_ * sizeof(int16_t);
   const size_t old_size = rec_buffer_.size();
   rec_buffer_.SetData(static_cast<const uint8_t*>(audio_buffer), size_in_bytes);
   // Keep track of the size of the recording buffer. Only updated when the
   // size changes, which is a rare event.
   if (old_size != rec_buffer_.size()) {
     LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size();
   }
   // Derive a new level value twice per second and check if it is non-zero.
   int16_t max_abs = 0;
   RTC_DCHECK_LT(rec_stat_count_, 50);
   if (++rec_stat_count_ >= 50) {
-    const size_t size = num_samples * rec_channels;
+    const size_t size = num_samples * rec_channels_;
     // Returns the largest absolute value in a signed 16-bit vector.
     max_abs = WebRtcSpl_MaxAbsValueW16(
         reinterpret_cast<const int16_t*>(rec_buffer_.data()), size);
     rec_stat_count_ = 0;
     // Set |only_silence_recorded_| to false as soon as at least one detection
     // of a non-zero audio packet is found. It can only be restored to true
     // again by restarting the call.
     if (max_abs > 0) {
       only_silence_recorded_ = false;
     }
   }
   // Update some stats but do it on the task queue to ensure that the members
   // are modified and read on the same thread. Note that |max_abs| will be
   // zero in most calls and then have no effect of the stats. It is only updated
   // approximately two times per second and can then change the stats.
   task_queue_.PostTask(
       [this, max_abs, num_samples] { UpdateRecStats(max_abs, num_samples); });
   return 0;
 }
 
 int32_t AudioDeviceBuffer::DeliverRecordedData() {
-  rtc::CritScope lock(&lock_cb_);
+  RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   if (!audio_transport_cb_) {
     LOG(LS_WARNING) << "Invalid audio transport";
     return 0;
   }
-  const size_t rec_bytes_per_sample = [&] {
-    rtc::CritScope lock(&lock_);
-    return rec_bytes_per_sample_;
-  }();
+  const size_t rec_bytes_per_sample = rec_channels_ * sizeof(int16_t);
   uint32_t new_mic_level(0);
   uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_;
   size_t num_samples = rec_buffer_.size() / rec_bytes_per_sample;
   int32_t res = audio_transport_cb_->RecordedDataIsAvailable(
-      rec_buffer_.data(), num_samples, rec_bytes_per_sample_, rec_channels_,
+      rec_buffer_.data(), num_samples, rec_bytes_per_sample, rec_channels_,
       rec_sample_rate_, total_delay_ms, clock_drift_, current_mic_level_,
       typing_status_, new_mic_level);
   if (res != -1) {
     new_mic_level_ = new_mic_level;
   } else {
     LOG(LS_ERROR) << "RecordedDataIsAvailable() failed";
   }
   return 0;
 }
 
 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t num_samples) {
-  // Measure time since last function call and update an array where the
-  // position/index corresponds to time differences (in milliseconds) between
-  // two successive playout callbacks, and the stored value is the number of
-  // times a given time difference was found.
-  int64_t now_time = rtc::TimeMillis();
-  size_t diff_time = rtc::TimeDiff(now_time, last_playout_time_);
-  // Truncate at 500ms to limit the size of the array.
-  diff_time = std::min(kMaxDeltaTimeInMs, diff_time);
-  last_playout_time_ = now_time;
-  playout_diff_times_[diff_time]++;
-
-  const size_t play_channels = [&] {
-    rtc::CritScope lock(&lock_);
-    return play_channels_;
-  }();
-
+  RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   // The consumer can change the request size on the fly and we therefore
   // resize the buffer accordingly. Also takes place at the first call to this
   // method.
-  const size_t play_bytes_per_sample = play_channels * sizeof(int16_t);
+  const size_t play_bytes_per_sample = play_channels_ * sizeof(int16_t);
   const size_t size_in_bytes = num_samples * play_bytes_per_sample;
   if (play_buffer_.size() != size_in_bytes) {
     play_buffer_.SetSize(size_in_bytes);
     LOG(LS_INFO) << "Size of playout buffer: " << play_buffer_.size();
   }
 
   size_t num_samples_out(0);
-  {
-    rtc::CritScope lock(&lock_cb_);
+  // It is currently supported to start playout without a valid audio
+  // transport object. Leads to warning and silence.
+  if (!audio_transport_cb_) {
+    LOG(LS_WARNING) << "Invalid audio transport";
+    return 0;
+  }
 
-    // It is currently supported to start playout without a valid audio
-    // transport object. Leads to warning and silence.
-    if (!audio_transport_cb_) {
-      LOG(LS_WARNING) << "Invalid audio transport";
-      return 0;
-    }
-
-    // Retrieve new 16-bit PCM audio data using the audio transport instance.
-    int64_t elapsed_time_ms = -1;
-    int64_t ntp_time_ms = -1;
-    uint32_t res = audio_transport_cb_->NeedMorePlayData(
-        num_samples, play_bytes_per_sample_, play_channels, play_sample_rate_,
-        play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms);
-    if (res != 0) {
-      LOG(LS_ERROR) << "NeedMorePlayData() failed";
-    }
+  // Retrieve new 16-bit PCM audio data using the audio transport instance.
+  int64_t elapsed_time_ms = -1;
+  int64_t ntp_time_ms = -1;
+  uint32_t res = audio_transport_cb_->NeedMorePlayData(
+      num_samples, play_bytes_per_sample, play_channels_, play_sample_rate_,
+      play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms);
+  if (res != 0) {
+    LOG(LS_ERROR) << "NeedMorePlayData() failed";
   }
 
   // Derive a new level value twice per second.
   int16_t max_abs = 0;
   RTC_DCHECK_LT(play_stat_count_, 50);
   if (++play_stat_count_ >= 50) {
-    const size_t size = num_samples * play_channels;
+    const size_t size = num_samples * play_channels_;
     // Returns the largest absolute value in a signed 16-bit vector.
     max_abs = WebRtcSpl_MaxAbsValueW16(
         reinterpret_cast<const int16_t*>(play_buffer_.data()), size);
     play_stat_count_ = 0;
   }
   // Update some stats but do it on the task queue to ensure that the members
   // are modified and read on the same thread. Note that |max_abs| will be
   // zero in most calls and then have no effect of the stats. It is only updated
   // approximately two times per second and can then change the stats.
   task_queue_.PostTask([this, max_abs, num_samples_out] {
     UpdatePlayStats(max_abs, num_samples_out);
   });
   return static_cast<int32_t>(num_samples_out);
 }
 
 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
+  RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   RTC_DCHECK_GT(play_buffer_.size(), 0u);
-  const size_t play_bytes_per_sample = [&] {
-    rtc::CritScope lock(&lock_);
-    return play_bytes_per_sample_;
-  }();
+  const size_t play_bytes_per_sample = play_channels_ * sizeof(int16_t);
   memcpy(audio_buffer, play_buffer_.data(), play_buffer_.size());
   return static_cast<int32_t>(play_buffer_.size() / play_bytes_per_sample);
 }
 
 void AudioDeviceBuffer::StartPeriodicLogging() {
   task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
                                  AudioDeviceBuffer::LOG_START));
 }
 
 void AudioDeviceBuffer::StopPeriodicLogging() {
   task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
                                  AudioDeviceBuffer::LOG_STOP));
 }
 
 void AudioDeviceBuffer::LogStats(LogState state) {
-  RTC_DCHECK(task_queue_.IsCurrent());
+  RTC_DCHECK_RUN_ON(&task_queue_);
   int64_t now_time = rtc::TimeMillis();
   if (state == AudioDeviceBuffer::LOG_START) {
     // Reset counters at start. We will not add any logging in this state but
     // the timer will started by posting a new (delayed) task.
     num_stat_reports_ = 0;
     last_timer_task_time_ = now_time;
   } else if (state == AudioDeviceBuffer::LOG_STOP) {
     // Stop logging and posting new tasks.
     return;
   } else if (state == AudioDeviceBuffer::LOG_ACTIVE) {
     // Default state. Just keep on logging.
   }
 
   int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
   int64_t time_since_last = rtc::TimeDiff(now_time, last_timer_task_time_);
   last_timer_task_time_ = now_time;
 
+  // TODO(henrika): FIX

[kwiberg-webrtc, 2016/11/03 13:15:17]

A more descriptive comment before you commit this, please. :-)

[henrika_webrtc, 2016/11/03 14:36:32]

Sorry. Done.

+
+  const size_t rec_sample_rate = 48000;
+  const size_t play_sample_rate = 48000;
+
   // Log the latest statistics but skip the first round just after state was
   // set to LOG_START. Hence, first printed log will be after ~10 seconds.
   if (++num_stat_reports_ > 1 && time_since_last > 0) {
     uint32_t diff_samples = rec_samples_ - last_rec_samples_;
     float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
     LOG(INFO) << "[REC : " << time_since_last << "msec, "
-              << rec_sample_rate_ / 1000
+              << rec_sample_rate / 1000
               << "kHz] callbacks: " << rec_callbacks_ - last_rec_callbacks_
               << ", "
               << "samples: " << diff_samples << ", "
               << "rate: " << static_cast<int>(rate + 0.5) << ", "
               << "level: " << max_rec_level_;
 
     diff_samples = play_samples_ - last_play_samples_;
     rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
     LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
-              << play_sample_rate_ / 1000
+              << play_sample_rate / 1000
               << "kHz] callbacks: " << play_callbacks_ - last_play_callbacks_
               << ", "
               << "samples: " << diff_samples << ", "
               << "rate: " << static_cast<int>(rate + 0.5) << ", "
               << "level: " << max_play_level_;
   }
 
   last_rec_callbacks_ = rec_callbacks_;
   last_play_callbacks_ = play_callbacks_;
   last_rec_samples_ = rec_samples_;
   last_play_samples_ = play_samples_;
   max_rec_level_ = 0;
   max_play_level_ = 0;
 
   int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
   RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
 
   // Keep posting new (delayed) tasks until state is changed to kLogStop.
   task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
                                         AudioDeviceBuffer::LOG_ACTIVE),
                               time_to_wait_ms);
 }
 
 void AudioDeviceBuffer::ResetRecStats() {
-  RTC_DCHECK(task_queue_.IsCurrent());
+  RTC_DCHECK_RUN_ON(&task_queue_);
   rec_callbacks_ = 0;
   last_rec_callbacks_ = 0;
   rec_samples_ = 0;
   last_rec_samples_ = 0;
   max_rec_level_ = 0;
 }
 
 void AudioDeviceBuffer::ResetPlayStats() {
-  RTC_DCHECK(task_queue_.IsCurrent());
+  RTC_DCHECK_RUN_ON(&task_queue_);
   play_callbacks_ = 0;
   last_play_callbacks_ = 0;
   play_samples_ = 0;
   last_play_samples_ = 0;
   max_play_level_ = 0;
 }
 
 void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs, size_t num_samples) {
-  RTC_DCHECK(task_queue_.IsCurrent());
+  RTC_DCHECK_RUN_ON(&task_queue_);
   ++rec_callbacks_;
   rec_samples_ += num_samples;
   if (max_abs > max_rec_level_) {
     max_rec_level_ = max_abs;
   }
 }
 
 void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs, size_t num_samples) {
-  RTC_DCHECK(task_queue_.IsCurrent());
+  RTC_DCHECK_RUN_ON(&task_queue_);
   ++play_callbacks_;
   play_samples_ += num_samples;
   if (max_abs > max_play_level_) {
     max_play_level_ = max_abs;
   }
 }
 
 }  // namespace webrtc