Avoids crash at device switch on iOS

webrtc/modules/audio_device/audio_device_buffer.h

 /*
  *  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 55 matching lines @@
   // valid implementation. Investigate the possibility to either remove them
   // or add a proper implementation if needed.
   int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopInputFileRecording();
   int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopOutputFileRecording();
 
   int32_t SetTypingStatus(bool typing_status);
 
  private:
-  void AllocatePlayoutBufferIfNeeded();
-  void AllocateRecordingBufferIfNeeded();
+  // Playout and recording parameters can change on the fly. e.g. at device
+  // switch. These methods ensures that the callback methods always use the
+  // latest parameters.
+  void UpdatePlayoutParameters();
+  void UpdateRecordingParameters();
 
   // Posts the first delayed task in the task queue and starts the periodic
   // timer.
   void StartTimer();
 
   // Called periodically on the internal thread created by the TaskQueue.
   void LogStats();
 
   // Updates counters in each play/record callback but does it on the task
   // queue to ensure that they can be read by LogStats() without any locks since
@@ skipping 36 matching lines @@
   // Number of bytes per audio sample (2 or 4).
   size_t rec_bytes_per_sample_;
   size_t play_bytes_per_sample_;
 
   // Number of audio samples/bytes per 10ms.
   size_t rec_samples_per_10ms_;
   size_t rec_bytes_per_10ms_;
   size_t play_samples_per_10ms_;
   size_t play_bytes_per_10ms_;
 
-  // Buffer used for recorded audio samples. Size is given by
-  // |rec_bytes_per_10ms_| and the buffer is allocated in InitRecording() on the
-  // main/creating thread.
+  // Buffer used for recorded audio samples. Size is currently fixed
+  // but it should be changed to be dynamic and correspond to
+  // |play_bytes_per_10ms_|. TODO(henrika): avoid using fixed (max) size.
   std::unique_ptr<int8_t[]> rec_buffer_;
 
-  // Buffer used for audio samples to be played out. Size is given by
-  // |play_bytes_per_10ms_| and the buffer is allocated in InitPlayout() on the
-  // main/creating thread.
+  // Buffer used for audio samples to be played out. Size is currently fixed
+  // but it should be changed to be dynamic and correspond to
+  // |play_bytes_per_10ms_|. TODO(henrika): avoid using fixed (max) size.
   std::unique_ptr<int8_t[]> play_buffer_;
 
   // AGC parameters.
   uint32_t current_mic_level_;
   uint32_t new_mic_level_;
 
   // Contains true of a key-press has been detected.
   bool typing_status_;
 
   // Delay values used by the AEC.
@@ skipping 42 matching lines @@
   // milliseconds) between two successive playout callbacks, and the stored
   // value is the number of times a given time difference was found.
   // Writing to the array is done without a lock since it is only read once at
   // destruction when no audio is running.
   uint32_t playout_diff_times_[kMaxDeltaTimeInMs + 1] = {0};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_