Adds detection of audio glitches for playout on iOS.

Bug: b/38018041
Change-Id: If6b53d3909a52333543c8aade500fd4c26b47255
Reviewed-on: https://chromium-review.googlesource.com/522563
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18570}

webrtc/modules/audio_device/BUILD.gn

@@ -92,6 +92,9 @@ rtc_static_library("audio_device") {
   }
   defines = []
   cflags = []
+  if (rtc_audio_device_plays_sinus_tone) {
+    defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
+  }
   if (rtc_include_internal_audio_device) {
     sources += [
       "audio_device_data_observer.cc",

webrtc/modules/audio_device/audio_device_buffer.cc

@@ -9,6 +9,7 @@
  */
 
 #include <algorithm>
+#include <cmath>
 
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 
@@ -36,6 +37,9 @@ static const size_t kTimerIntervalInMilliseconds =
 static const size_t kMinValidCallTimeTimeInSeconds = 10;
 static const size_t kMinValidCallTimeTimeInMilliseconds =
     kMinValidCallTimeTimeInSeconds * rtc::kNumMillisecsPerSec;
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+static const double k2Pi = 6.28318530717959;
+#endif
 
 AudioDeviceBuffer::AudioDeviceBuffer()
     : task_queue_(kTimerQueueName),
@@ -60,6 +64,10 @@ AudioDeviceBuffer::AudioDeviceBuffer()
       only_silence_recorded_(true),
       log_stats_(false) {
   LOG(INFO) << "AudioDeviceBuffer::ctor";
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  phase_ = 0.0;
+  LOG(WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
+#endif
   playout_thread_checker_.DetachFromThread();
   recording_thread_checker_.DetachFromThread();
 }
@@ -391,9 +399,18 @@ 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);
-  const size_t bytes_per_sample = sizeof(int16_t);
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  const double phase_increment =
+      k2Pi * 440.0 / static_cast<double>(play_sample_rate_);
+  int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer);
+  for (size_t i = 0; i < play_buffer_.size(); ++i) {
+    destination_r[i] = static_cast<int16_t>((sin(phase_) * (1 << 14)));
+    phase_ += phase_increment;
+  }
+#else
   memcpy(audio_buffer, play_buffer_.data(),
-         play_buffer_.size() * bytes_per_sample);
+         play_buffer_.size() * sizeof(int16_t));
+#endif
   // Return samples per channel or number of frames.
   return static_cast<int32_t>(play_buffer_.size() / play_channels_);
 }

webrtc/modules/audio_device/audio_device_buffer.h

@@ -21,6 +21,7 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+
 // Delta times between two successive playout callbacks are limited to this
 // value before added to an internal array.
 const size_t kMaxDeltaTimeInMs = 500;
@@ -251,6 +252,12 @@ class AudioDeviceBuffer {
   // Setting this member to false prevents (possiby invalid) log messages from
   // being printed in the LogStats() task.
   bool log_stats_ ACCESS_ON(task_queue_);
+
+// 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_ ACCESS_ON(playout_thread_checker_);
+#endif
 };
 
 }  // namespace webrtc

webrtc/modules/audio_device/ios/audio_device_ios.h

@@ -189,6 +189,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   void HandleValidRouteChange();
   void HandleCanPlayOrRecordChange(bool can_play_or_record);
   void HandleSampleRateChange(float sample_rate);
+  void HandlePlayoutGlitchDetected();
 
   // Uses current |playout_parameters_| and |record_parameters_| to inform the
   // audio device buffer (ADB) about our internal audio parameters.
@@ -290,6 +291,13 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   // Set to true if we've activated the audio session.
   bool has_configured_session_;
 
+  // Counts number of detected audio glitches on the playout side.
+  int64_t num_detected_playout_glitches_;
+  int64_t last_playout_time_;
+
+  // Counts number of playout callbacks per call.
+  int64_t num_playout_callbacks_;
+
   // Exposes private members for testing purposes only.
   FRIEND_TEST_ALL_PREFIXES(AudioDeviceTest, testInterruptedAudioSession);
 };

webrtc/modules/audio_device/ios/audio_device_ios.mm

@@ -13,8 +13,6 @@
 
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
 
-#include <cmath>
-
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/bind.h"
@@ -23,8 +21,10 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 #import "WebRTC/RTCLogging.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h"
@@ -66,6 +66,7 @@ enum AudioDeviceMessageType : uint32_t {
   kMessageTypeInterruptionEnd,
   kMessageTypeValidRouteChange,
   kMessageTypeCanPlayOrRecordChange,
+  kMessageTypePlayoutGlitchDetected,
 };
 
 using ios::CheckAndLogError;
@@ -109,7 +110,10 @@ AudioDeviceIOS::AudioDeviceIOS()
       initialized_(false),
       audio_is_initialized_(false),
       is_interrupted_(false),
-      has_configured_session_(false) {
+      has_configured_session_(false),
+      num_detected_playout_glitches_(0),
+      last_playout_time_(0),
+      num_playout_callbacks_(0) {
   LOGI() << "ctor" << ios::GetCurrentThreadDescription();
   thread_ = rtc::Thread::Current();
   audio_session_observer_ =
@@ -220,6 +224,7 @@ int32_t AudioDeviceIOS::StartPlayout() {
     LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 1);
+  num_playout_callbacks_ = 0;
   return 0;
 }
 
@@ -234,6 +239,19 @@ int32_t AudioDeviceIOS::StopPlayout() {
     audio_is_initialized_ = false;
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 0);
+
+  // Derive average number of calls to OnGetPlayoutData() between detected
+  // audio glitches and add the result to a histogram.
+  int average_number_of_playout_callbacks_between_glitches = 100000;
+  if (num_detected_playout_glitches_ > 0) {
+    average_number_of_playout_callbacks_between_glitches =
+        num_playout_callbacks_ / num_detected_playout_glitches_;
+  }
+  RTC_HISTOGRAM_COUNTS_100000(
+      "WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
+      average_number_of_playout_callbacks_between_glitches);
+  RTCLog(@"Average number of playout callbacks between glitches: %d",
+         average_number_of_playout_callbacks_between_glitches);
   return 0;
 }
 
@@ -419,6 +437,7 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
   RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
   AudioBuffer* audio_buffer = &io_data->mBuffers[0];
   RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
+
   // Get pointer to internal audio buffer to which new audio data shall be
   // written.
   const size_t size_in_bytes = audio_buffer->mDataByteSize;
@@ -433,10 +452,30 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
     return noErr;
   }
 
+  // Measure time since last call to OnGetPlayoutData() and see if it is larger
+  // than a well defined threshold. If so, we have a clear indication of a
+  // glitch in the output audio since the core audio layer will most likely run
+  // dry in this state.
+  ++num_playout_callbacks_;
+  const int64_t now_time = rtc::TimeMillis();
+  if (time_stamp->mSampleTime != num_frames) {
+    const int64_t delta_time = now_time - last_playout_time_;
+    const int glitch_threshold =
+        1.5 * playout_parameters_.GetBufferSizeInMilliseconds() - 1;
+    if (delta_time > glitch_threshold) {
+      RTCLogWarning(@"Playout audio glitch detected.\n"
+                     "  Time since last OnGetPlayoutData was %lld ms.",
+                    delta_time);
+      thread_->Post(RTC_FROM_HERE, this, kMessageTypePlayoutGlitchDetected);
+    }
+  }
+  last_playout_time_ = now_time;
+
   // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
   // the native I/O audio unit) and copy the result to the audio buffer in the
   // |io_data| destination.
-  fine_audio_buffer_->GetPlayoutData(rtc::ArrayView<int8_t>(destination, size_in_bytes));
+  fine_audio_buffer_->GetPlayoutData(
+      rtc::ArrayView<int8_t>(destination, size_in_bytes));
   return noErr;
 }
 
@@ -458,6 +497,9 @@ void AudioDeviceIOS::OnMessage(rtc::Message *msg) {
       delete data;
       break;
     }
+    case kMessageTypePlayoutGlitchDetected:
+      HandlePlayoutGlitchDetected();
+      break;
   }
 }
 
@@ -530,8 +572,10 @@ void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
           "  Session sample rate: %f frames_per_buffer: %lu\n"
           "  ADM sample rate: %f frames_per_buffer: %lu",
          sample_rate,
-         session_sample_rate, (unsigned long)session_frames_per_buffer,
-         current_sample_rate, (unsigned long)current_frames_per_buffer);;
+         session_sample_rate,
+         (unsigned long)session_frames_per_buffer,
+         current_sample_rate,
+         (unsigned long)current_frames_per_buffer);
 
   // Sample rate and buffer size are the same, no work to do.
   if (std::abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
@@ -572,6 +616,13 @@ void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
   RTCLog(@"Successfully handled sample rate change.");
 }
 
+void AudioDeviceIOS::HandlePlayoutGlitchDetected() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  num_detected_playout_glitches_++;
+  RTCLog(@"Number of detected playout glitches: %lld",
+         num_detected_playout_glitches_);
+}
+
 void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
   LOGI() << "UpdateAudioDevicebuffer";
   // AttachAudioBuffer() is called at construction by the main class but check

webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m

@@ -41,8 +41,8 @@ const double kRTCAudioSessionLowComplexitySampleRate = 16000.0;
 // ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
 // take care of any buffering required to convert between native buffers and
 // buffers used by WebRTC. It is beneficial for the performance if the native
-// size is as close to 10ms as possible since it results in "clean" callback
-// sequence without bursts of callbacks back to back.
+// size is as an even multiple of 10ms as possible since it results in "clean"
+// callback sequence without bursts of callbacks back to back.
 const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.01;
 
 // Use a larger buffer size on devices with only one core (e.g. iPhone 4).

webrtc/webrtc.gni

@@ -147,6 +147,12 @@ declare_args() {
   # use file-based audio playout and record.
   rtc_use_dummy_audio_file_devices = false
 
+  # When set to true, replace the audio output with a sinus tone at 440Hz.
+  # The ADM will ask for audio data from WebRTC but instead of reading real
+  # audio samples from NetEQ, a sinus tone will be generated and replace the
+  # real audio samples.
+  rtc_audio_device_plays_sinus_tone = false
+
   # When set to true, test targets will declare the files needed to run memcheck
   # as data dependencies. This is to enable memcheck execution on swarming bots.
   rtc_use_memcheck = false