I have an ios app about communicate audio and video. the Voice communication part use audio unit. there are two obj obj_audiounit ,one is use to record (voip),another is use to play audio (voip).and the audio unit obj of play,crash !!!.at the play part , i use a thread to start the audio unit . but it crash random.thread run{OSStatus status = AudioUnitInitialize(pThis->m_pAudioUnit); if (status != noErr) { NSLog(@"couldn't initialize AURemoteIO instance"); } status = AudioOutputUnitStart(pThis->m_pAudioUnit); if (status != noErr) { NSLog(@"error AudioOutputUnitStart"); }}libembeddsystemaus.dylib aumethodstart+160876 (void*)+64libembeddsystemaus.dylib aumethodstart+160856(void*)+44libsystem_pthread _pthread_startthread X crash with arm thread state....what's the reason?

Hardware Model: iPhone8,1Process: MobileTerminal [36987]Path: /private/var/containers/Bundle/Application/FDVersion: 20200323 (2.3.0)AppStoreTools: 11C29AppVariant: 1:iPhone8,1:12.2Beta: YESCode Type: ARM-64 (Native)Role: Non UIParent Process: launchd [1]Date/Time: 2020-03-31 16:01:00.2887 +0800Launch Time: 2020-03-31 14:48:59.3714 +0800OS Version: iPhone OS 12.4.1 (16G102)Baseband Version: 5.70.01Report Version: 104Exception Type: EXC_CRASH (SIGKILL)Exception Codes: 0x0000000000000000, 0x0000000000000000Exception Note: EXC_CORPSE_NOTIFYTermination Reason: Namespace SPRINGBOARD, Code 0x8badf00dTriggered by Thread: 0Thread 0 name: Dispatch queue: com.apple.main-threadThread 0 Crashed:0 libsystem_kernel.dylib 0x00000001a7d5ef2c __psynch_mutexwait + 81 libsystem_pthread.dylib 0x00000001a7ddaa84 _pthread_mutex_firstfit_lock_wait + 922 libsystem_pthread.dylib 0x00000001a7dda9f4 _pthread_mutex_firstfit_lock_slow$VARIANT$mp + 2723 AudioToolbox 0x00000001ac4da888 CAMutex::Lock+ 4950152 () + 724 AudioToolbox 0x00000001ac4f86bc AUMethodStop+ 5072572 (void*) + 445 MobileT 0x00000001032dd060 COpenAL::ReleaseSound() + 6033504 (7 MobileT 0x00000001032dd108 COpenAL::~COpenAL() + 6033672 (10 MobileT 0x00000001032f0ab4 ::~CAudioManager() + 611397211 MobileT 0x00000001032d708c DestroyHandle + 600897212 MobileT 0x00000001032f32c4 :Release() +15 MobileT 0x00000001032f3584 ::~CAudioPlay() +16 MobileT 0x000000010331c560 ::Close() +19 MobileT 0x0000000103349ec0 MClose(void*) +20 MobileT 0x000000010332046c PlayCloseStream21 MobileT 0x000000010332145c PlayOpenStream22 MobileT 0x0000000102daaa44 invocation function for block in PlayDataCallBack(int, unsigned int, unsigned char*, unsigned int, void*) + 584260 (VideoConferencePlayComponent.mm:88)23 libdispatch.dylib 0x00000001a7c017d4 _dispatch_client_callout + 1624 libdispatch.dylib 0x00000001a7bafcec _dispatch_async_and_wait_invoke + 9225 libdispatch.dylib 0x00000001a7c017d4 _dispatch_client_callout + 1626 libdispatch.dylib 0x00000001a7baf008 _dispatch_main_queue_callback_4CF$VARIANT$mp + 106827 CoreFoundation 0x00000001a815432c __CFRUNLOOP_IS_SERVICING_THE_MAIN_DISPATCH_QUEUE__ + 1228 CoreFoundation 0x00000001a814f264 __CFRunLoopRun + 192429 CoreFoundation 0x00000001a814e7c0 CFRunLoopRunSpecific + 43630 GraphicsServices 0x00000001aa34f79c GSEventRunModal + 10431 UIKitCore 0x00000001d4ad3c38 UIApplicationMain + 21232 MobileTerminal 0x0000000102d738ac main + 358572 (main.m:14)33 libdyld.dylib 0x00000001a7c128e0 start + 4....................................................................................................................Happen by chance,when i shart audiounit play ....please help thanks!

AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""

hii use audiounit vpio subtype to do the voip. audio unit + voiceprocessioingIO,when i use ipad mini 5 play sound ,the voice is big;but when i use the ipad pro whic have four speaker ,the sound voice is very small,and speak by bottom speaker .have any method to make the voice bigger ?use the ipad pro 2019when i use remoteio,the voice is bigger.but i need echocancellation@SoundOfMindipad :MTXP2CH/A

I use two audio unit obj (a,b)to do the intercom demo. and unitA to do the record .unitB to do the playback, use the AVAudioSession like this :  [sessionInstance setMode:AVAudioSessionModeDefault error:&error];     [sessionInstance setCategory:AVAudioSessionCategoryPlayAndRecord                      withOptions:AVAudioSessionCategoryOptionDefaultToSpeaker|AVAudioSessionCategoryOptionDuckOthers error:&error]; I use kAudioUnitSubTypeVoiceProcessingIO to do the echo cancellation. but When we talk in a quiet room ,the VPIO have echo remain ,at the end of every sentences. iphone1 talk with iphone2 ,and iphone1 can hear some voice of iphone1, It's not eliminated of iphone2 by the kAudioUnitSubTypeVoiceProcessingIO。 so have any function to change the delay-time when we use VoiceProcessingIO ?

I use audiounit （subtype:vpio),to play the audio . The data is sent to my playback program via netEQ. The audio sent over the network is relatively uniform, But the video is jittery. question: 1 .The audio unit starting time is not fixed，sometimes 250MS，sometimes 400MS (in different iPhone iPad ) 2 . the playback callback period not even sometime .such us 8K Sample_rate, Most of times are called (playback callback ) in more than 20 milliseconds.340~342BYTE. but occasionally, the callback time interval 130+ ms,And ioData->mBuffers[0].mDataByteSize= 340 byte alse. I didn't make the time-consuming operation in the render callback function . Just copy the data to iodata. and data is enough. PlayCallback(void *inRefCon,                                AudioUnitRenderActionFlags *ioActionFlags,                                const AudioTimeStamp *inTimeStamp,                                UInt32 inBusNumber,                                UInt32 inNumberFrames,                                AudioBufferList *ioData){copy data from my buffer } ------------------------------------------------------------I want to know what will affect your render callback cycle(period).

I use kNewTimePitchParampitch ; audiounit (in)---convert(pitch shift)---audiounit( out);I can get the result about pitch shift status =AudioUnitSetParameter(convert, kNewTimePitchParamPitch, kAudioUnitScopeGlobal, 0, pitchShift, 0); change param:AudioUnitSetParameter(conver, kNewTimePitchParamRate  , kAudioUnitScope_Global, 0, pitchShift, 0); the AudioUnitAddRenderNotify can not get the data which have been change speech. the data is 00. so how to use the rate change?

I use openal to record and use openal to play ; use QT demo to do the communication. When I put on the Bluetooth headset, as long as I turn on the audio record by BT, the audio playback will be blocked, unable to consume data. And the trumpet  is black-color at this time; But when the trumpet is gray, the record will not block the playback,i can play and record . It's a random event。 I wonder if there is a program that can detect or change this blocking?