Resolves issue with bad audio using BT headsets on iOS

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 413 matching lines @@
   RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample,
                num_frames);
   int8_t* destination = reinterpret_cast<int8_t*>(audio_buffer->mData);
   // Produce silence and give audio unit a hint about it if playout is not
   // activated.
   if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
     *flags |= kAudioUnitRenderAction_OutputIsSilence;
     memset(destination, 0, size_in_bytes);
     return noErr;
   }
+  // Produce silence and log a warning message for the case when Core Audio is
+  // asking for an invalid number of audio frames. I don't expect this to happen
+  // but it is done as a safety measure to avoid bad audio if such as case would
+  // ever be triggered e.g. in combination with BT devices.
+  const size_t frames_per_buffer = playout_parameters_.frames_per_buffer();
+  if (num_frames != frames_per_buffer) {
+    RTCLogWarning(@"Expected %u frames but got %u",
+                  static_cast<unsigned int>(frames_per_buffer),
+                  static_cast<unsigned int>(num_frames));
+    *flags |= kAudioUnitRenderAction_OutputIsSilence;
+    memset(destination, 0, size_in_bytes);
+    return noErr;
+  }
+
   // 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.
   int8_t* source = playout_audio_buffer_.get();
   fine_audio_buffer_->GetPlayoutData(source);
   memcpy(destination, source, size_in_bytes);
   return noErr;
 }
 
 void AudioDeviceIOS::OnMessage(rtc::Message *msg) {
@@ skipping 373 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