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 fb2f65d339bf55bfaa8c9f5db2125d661dbb8bb8..96d95518f235877f6677731d1d71e351e9642411 100644 |
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm |
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm |
@@ -769,6 +769,7 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() { |
bool AudioDeviceIOS::CreateAudioUnit() { |
RTC_DCHECK(!audio_unit_); |
+ RTCLog(@"CreateAudioUnit"); |
audio_unit_.reset(new VoiceProcessingAudioUnit(this)); |
if (!audio_unit_->Init()) { |
@@ -927,6 +928,8 @@ bool AudioDeviceIOS::InitPlayOrRecord() { |
ConfigureAudioSession(); |
SetupAudioBuffersForActiveAudioSession(); |
audio_unit_->Initialize(playout_parameters_.sample_rate()); |
+ } else { |
+ RTCLogWarning(@"Not ready to start play or record yet."); |
} |
// Release the lock. |