Fix a bug in computing audio delay on ios device. Converts seconds to

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 899 matching lines @@
     OSStatus result =
         AudioUnitGetProperty(_auVoiceProcessing, kAudioUnitProperty_Latency,
                              kAudioUnitScope_Global, 0, &f64, &size);
     if (0 != result) {
       LOG_F(LS_ERROR) << "AU latency error: " << result;
     }
     assert(f64 >= 0);
     totalDelaySeconds += f64;
 
     // To ms
-    _playoutDelay = static_cast<uint32_t>(totalDelaySeconds / 1000);
+    _playoutDelay = static_cast<uint32_t>(totalDelaySeconds * 1000);
 
     // Reset counter
     _playoutDelayMeasurementCounter = 0;
   }
 
   // todo: Add playout buffer?
 }
 
 void AudioDeviceIOS::UpdateRecordingDelay() {
   ++_recordingDelayMeasurementCounter;
@@ skipping 114 matching lines @@
     //       Device Buffer?
     timespec t;
     t.tv_sec = 0;
     t.tv_nsec = 5 * 1000 * 1000;
     nanosleep(&t, nullptr);
   }
   return true;
 }
 
 }  // namespace webrtc