| 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 3dcd48f2dcb97f3a5476344aae5959bb0cf29107..1d59aff46756ff51d86234b98c14383de1b098be 100644
|
| --- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| +++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| @@ -623,6 +623,12 @@ void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
|
| 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.
|
|
|