Adds sanity check for sample rate 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 605 matching lines @@
          current_sample_rate,
          (unsigned long)current_frames_per_buffer);
 
   // Sample rate and buffer size are the same, no work to do.
   if (std::abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
       current_frames_per_buffer == session_frames_per_buffer) {
     RTCLog(@"Ignoring sample rate change since audio parameters are intact.");
     return;
   }
 
+  // Extra sanity check to ensure that the new sample rate is valid.
+  if (session_sample_rate <= 0.0) {
+    RTCLogError(@"Sample rate is invalid: %f", session_sample_rate);
+    return;
+  }
+
   // We need to adjust our format and buffer sizes.
   // The stream format is about to be changed and it requires that we first
   // stop and uninitialize the audio unit to deallocate its resources.
   RTCLog(@"Stopping and uninitializing audio unit to adjust buffers.");
   bool restart_audio_unit = false;
   if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
     audio_unit_->Stop();
     restart_audio_unit = true;
   }
   if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
@@ skipping 304 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