Simplifies FineAudioBuffer by using rtc::Buffer

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 609 matching lines @@
   RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(),
                 record_parameters_.GetBytesPerBuffer());
 
   // Update the ADB parameters since the sample rate might have changed.
   UpdateAudioDeviceBuffer();
 
   // Create a modified audio buffer class which allows us to ask for,
   // 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
   // AudioBuffer structure, which holds a pointer to the actual data buffer
   // in |record_audio_buffer_|. Recorded audio will be rendered into this memory
   // at each input callback when calling AudioUnitRender().
   const int data_byte_size = record_parameters_.GetBytesPerBuffer();
   record_audio_buffer_.reset(new SInt8[data_byte_size]);
   memset(record_audio_buffer_.get(), 0, data_byte_size);
   audio_record_buffer_list_.mNumberBuffers = 1;
@@ skipping 179 matching lines @@
 
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   [session lockForConfiguration];
   UnconfigureAudioSession();
   [session endWebRTCSession:nil];
   [session unlockForConfiguration];
 }
 
 }  // namespace webrtc