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 dde42129e5c9380475c6e820131ebf38e3b6741b..721ba77aaaafcf74236feeb84dc9bc2846341af9 100644 |
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm |
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm |
@@ -326,11 +326,13 @@ int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const { |
void AudioDeviceIOS::OnInterruptionBegin() { |
RTC_DCHECK(thread_); |
+ LOGI() << "OnInterruptionBegin"; |
thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionBegin); |
} |
void AudioDeviceIOS::OnInterruptionEnd() { |
RTC_DCHECK(thread_); |
+ LOGI() << "OnInterruptionEnd"; |
thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionEnd); |
} |
@@ -468,6 +470,7 @@ void AudioDeviceIOS::OnMessage(rtc::Message *msg) { |
void AudioDeviceIOS::HandleInterruptionBegin() { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+ RTCLog(@"Handling interruption begin."); |
if (audio_unit_ && |
audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) { |
RTCLog(@"Stopping the audio unit due to interruption begin."); |
@@ -481,8 +484,9 @@ void AudioDeviceIOS::HandleInterruptionBegin() { |
void AudioDeviceIOS::HandleInterruptionEnd() { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+ RTCLog(@"Interruption ended. IsInterrupted=%d. Updating audio unit state.", |
tkchin_webrtc
2017/03/29 20:02:51
nit: maybe for both these logs do
"Interruption ch
Chuck
2017/03/29 20:29:18
Done.
|
+ is_interrupted_); |
is_interrupted_ = false; |
- RTCLog(@"Interruption ended. Updating audio unit state."); |
UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord); |
} |