iOS stability improvement for device switching

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  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 564 matching lines @@
           case AVAudioSessionRouteChangeReasonUnknown:
             LOG(LS_INFO) << " ReasonUnknown";
             break;
           case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
             LOG(LS_INFO) << " NewDeviceAvailable";
             break;
           case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
             LOG(LS_INFO) << " OldDeviceUnavailable";
             break;
           case AVAudioSessionRouteChangeReasonCategoryChange:
+            // It turns out that we see this notification (at least in iOS 9.2)
+            // when making a switch from a BT device to e.g. Speaker using the
+            // iOS Control Center and that we therefore must check if the sample
+            // rate has changed. And if so is the case, restart the audio unit.
             LOG(LS_INFO) << " CategoryChange";
             LOG(LS_INFO) << " New category: " << ios::GetAudioSessionCategory();
-            // Don't see this as route change since it can be triggered in
-            // combination with session interruptions as well.
-            valid_route_change = false;
             break;
           case AVAudioSessionRouteChangeReasonOverride:
             LOG(LS_INFO) << " Override";
             break;
           case AVAudioSessionRouteChangeReasonWakeFromSleep:
             LOG(LS_INFO) << " WakeFromSleep";
             break;
           case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
             LOG(LS_INFO) << " NoSuitableRouteForCategory";
             break;
           case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
-            // Ignore this type of route change since we are focusing
+            // The set of input and output ports has not changed, but their
+            // configuration has, e.g., a port’s selected data source has
+            // changed. Ignore this type of route change since we are focusing
             // on detecting headset changes.
-            LOG(LS_INFO) << " RouteConfigurationChange";
+            LOG(LS_INFO) << " RouteConfigurationChange (ignored)";
             valid_route_change = false;
             break;
         }
 
         if (valid_route_change) {
           // Log previous route configuration.
           AVAudioSessionRouteDescription* prev_route =
               notification.userInfo[AVAudioSessionRouteChangePreviousRouteKey];
           LOG(LS_INFO) << "Previous route:";
           LOG(LS_INFO) << ios::StdStringFromNSString(
@@ skipping 313 matching lines @@
   application_format_.mSampleRate = playout_parameters_.sample_rate();
   UInt32 size = sizeof(application_format_);
   AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
                        kAudioUnitScope_Output, 1, &application_format_, size);
   AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
                        kAudioUnitScope_Input, 0, &application_format_, size);
 
   // Prepare the audio unit to render audio again.
   LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(vpio_unit_),
                           "Failed to initialize the Voice-Processing I/O unit");
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now reinitialized";
 
   // Start rendering audio using the new format.
   LOG_AND_RETURN_IF_ERROR(AudioOutputUnitStart(vpio_unit_),
                           "Failed to start the Voice-Processing I/O unit");
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now restarted";
   return true;
 }
 
 bool AudioDeviceIOS::InitPlayOrRecord() {
   LOGI() << "InitPlayOrRecord";
   // Activate the audio session if not already activated.
   if (!ActivateAudioSession()) {
     return false;
   }
 
@@ skipping 75 matching lines @@
     const AudioTimeStamp* in_time_stamp,
     UInt32 in_bus_number,
     UInt32 in_number_frames) {
   OSStatus result = noErr;
   // Simply return if recording is not enabled.
   if (!rtc::AtomicOps::AcquireLoad(&recording_))
     return result;
   if (in_number_frames != record_parameters_.frames_per_buffer()) {
     // We have seen short bursts (1-2 frames) where |in_number_frames| changes.
     // Add a log to keep track of longer sequences if that should ever happen.
+    // Also return since calling AudioUnitRender in this state will only result
+    // in kAudio_ParamError (-50) anyhow.
     LOG(LS_WARNING) << "in_number_frames (" << in_number_frames
                     << ") != " << record_parameters_.frames_per_buffer();
+    return noErr;
   }
   // Obtain the recorded audio samples by initiating a rendering cycle.
   // Since it happens on the input bus, the |io_data| parameter is a reference
   // to the preallocated audio buffer list that the audio unit renders into.
   // TODO(henrika): should error handling be improved?
   AudioBufferList* io_data = &audio_record_buffer_list_;
   result = AudioUnitRender(vpio_unit_, io_action_flags, in_time_stamp,
                            in_bus_number, in_number_frames, io_data);
   if (result != noErr) {
-    LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+    LOG_F(LS_ERROR) << "AudioUnitRender failed: " << result;
     return result;
   }
   // Get a pointer to the recorded audio and send it to the WebRTC ADB.
   // Use the FineAudioBuffer instance to convert between native buffer size
   // and the 10ms buffer size used by WebRTC.
   const UInt32 data_size_in_bytes = io_data->mBuffers[0].mDataByteSize;
   RTC_CHECK_EQ(data_size_in_bytes / kBytesPerSample, in_number_frames);
   SInt8* data = static_cast<SInt8*>(io_data->mBuffers[0].mData);
   fine_audio_buffer_->DeliverRecordedData(data, data_size_in_bytes,
                                           kFixedPlayoutDelayEstimate,
@@ skipping 37 matching lines @@
   // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
   // the native I/O audio unit) to a preallocated intermediate buffer and
   // copy the result to the audio buffer in the |io_data| destination.
   SInt8* source = playout_audio_buffer_.get();
   fine_audio_buffer_->GetPlayoutData(source);
   memcpy(destination, source, dataSizeInBytes);
   return noErr;
 }
 
 }  // namespace webrtc