Simplifies FineAudioBuffer by using rtc::Buffer

webrtc/modules/audio_device/ios/audio_device_ios.mm

Index: webrtc/modules/audio_device/ios/audio_device_ios.mm
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index bc7ceb84588d7b2c1ed6f48e4be65175af5d923f..5f9e0e75250752af86b92e4ef0d863185073f6ca 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -627,17 +627,11 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
   // or deliver, any number of samples (and not only multiple of 10ms) to match
   // the native audio unit buffer size.
   RTC_DCHECK(audio_device_buffer_);
+  const size_t buffer_size_in_bytes = playout_parameters_.GetBytesPerBuffer();
   fine_audio_buffer_.reset(new FineAudioBuffer(
-      audio_device_buffer_, playout_parameters_.GetBytesPerBuffer(),
+      audio_device_buffer_, buffer_size_in_bytes,
       playout_parameters_.sample_rate()));
-
-  // The extra/temporary playoutbuffer must be of this size to avoid
-  // unnecessary memcpy while caching data between successive callbacks.
-  const int required_playout_buffer_size =
-      fine_audio_buffer_->RequiredPlayoutBufferSizeBytes();
-  LOG(LS_INFO) << " required playout buffer size: "
-               << required_playout_buffer_size;
-  playout_audio_buffer_.reset(new SInt8[required_playout_buffer_size]);
+  playout_audio_buffer_.reset(new SInt8[buffer_size_in_bytes]);
 
   // Allocate AudioBuffers to be used as storage for the received audio.
   // The AudioBufferList structure works as a placeholder for the