I'm looking for the AVAudioEngine in Practice video from WWDC 2014 (session 502) but can't seem to find it anywhere. Does anyone have a link to this session video? I can only find the slides. Thanks.

I receive a buffer from[AVSpeechSynthesizer convertToBuffer:fromBuffer:] and want to schedule it on an AVPlayerNode. The player node's output format need to be something that the next node could handle and as far as I understand most nodes can handle a canonical format. The format provided by AVSpeechSynthesizer is not something thatAVAudioMixerNode supports. So the following:   AVAudioEngine *engine = [[AVAudioEngine alloc] init];   playerNode = [[AVAudioPlayerNode alloc] init];   AVAudioFormat *format = [[AVAudioFormat alloc] initWithSettings:utterance.voice.audioFileSettings];   [engine attachNode:self.playerNode];   [engine connect:self.playerNode to:engine.mainMixerNode format:format]; Throws an exception: Thread 1: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 \"(null)\"" I am looking for a way to obtain the canonical format for the platform so that I can use AVAudioConverter to convert the buffer. Since different platforms have different canonical formats, I imagine there should be some library way of doing this. Otherwise each developer will have to redefine it for each platform the code will run on (OSX, iOS etc) and keep it updated when it changes. I could not find any constant or function which can make such format, ASDB or settings. The smartest way I could think of, which does not work:   AudioStreamBasicDescription toDesc;   FillOutASBDForLPCM(toDesc, [AVAudioSession sharedInstance].sampleRate,                      2, 16, 16, kAudioFormatFlagIsFloat, kAudioFormatFlagsNativeEndian);   AVAudioFormat *toFormat = [[AVAudioFormat alloc] initWithStreamDescription:&toDesc]; Even the provided example for iPhone, in the documentation linked above, uses kAudioFormatFlagsAudioUnitCanonical and AudioUnitSampleType which are deprecated. So what is the correct way to do this?

I am currently working on planning a multi-component software system that consists of an Audio Server Plugin and an application for user interaction. I have very little experience with IPC/XPC and its performance implications, so I hope I can find a little guidance here. The Audio Server plugin publishes a number of multi-channel output devices on which it should perform computations and pass the result on to a different Core Audio device. My concerns here are: Can the plugin directly access other CoreAudio devices for audio output or is this prohibited by the sandboxing? If it cannot, would relaying the audio data via XPC be a good idea in terms of low latency stability? Can I use metal compute from within the Audio Server plugin? I have not found any information about metal related sandboxing entitlements. I am also concerned about performance implications as above. Regarding the user interface application, I would like to know: If a process that has not been started by launchd can communicate with the Audio Server plugin using XPC. If not, would a user agent instead of an app be a better choice? Or are there other communication channels that would work with sandboxing? Thank you very much! Andreas

At least under macOS Sonoma 14.2.1 kAudioFormatFlagIsBigEndian for 24bit audio doesn't seem to be supported by the CoreAudio engine when providing kAudioServerPlugInIOOperationWriteMix streaming buffers for our CoreAudio server plugin. Is that correct and to be expected? Or how should the AudioStreamBasicDescription be filled out on a kAudioStreamPropertyPhysicalFormat request to correctly announce 24bit big endian audio to CoreAudio? Thanks, hagen.

Does anyone have a working example on how to play OGG files with swift? I've been trying for over a year now. I was able to wrap the C Vorbis library in swift. I then used it to parse an OGG file successfully. Then I was required to use Obj-C\++ to fill the PCM because this method seems to only be available in C\++ and that part hangs my app for a good 40 seconds to several minutes depending on the audio file, it then plays for about 2 seconds and then crashes. I can't get the examples on the Vorbis site to work in objective-c and i tried every example on github I could find (most of which are for iOS - I want to play the files on mac) I also tried using Cricket Audio framework below. https://github.com/sjmerel/ck It has a swift example and it can play their proprietary soundbank format but it is also supposed to play OGG and it just doesn't do anything when trying to play OGG as you can see in the posted issue https://github.com/sjmerel/ck/issues/3 Right now I believe every player that can play OGGs on mac is written in Objective-C or C++. Anyway, any help/advice is appreciated. OGG format is very prevalent in the gaming community. I could use unity, which I believe plays oggs through the mono framework but I really really want to stay in swift.

I report here some messages from Apple Community for an untracked bug in macOS Sonoma (from 14.0 to 14.2 beta 4 at the time). https://discussions.apple.com/thread/255214328 Original message 1: I've finally noticed a pattern that occurs rather frequently on macOS Sonoma. I was blaming Bluetooth issues before, but it looks like it's more about audio in general. What happens is that at some point, all audio freezes. The hotkeys for the audio controls show "Stop" sign, like there are no audio outputs connected, the taskbar is completely unresponsive: Control Centre shows a spinning circle, the sidebar is not opening (Spotlight works, though). If you go to the System Settings, some menu items will be unresponsive: Sound doesn't open, Bluetooth does not open, Accessibility and Siri & Spotlights all do not open. Then, a new bug appeared that I've just started to notice recently. The screen is flashing like there's an Accessibility feature enabled that uses warning flash instead of sound. It appears just randomly, out of nowhere. Immediately after that, sound works just normally. When this is happening, and video/audio content in the browser and wherever does not work, Tidal shows many random errors, and Firefox just completely hangs when you try to play a video on YouTube. I've tried to stop coreaudiod and it did restart the daemon, but nothing else happened. The device is a very fresh M1 Max MacBook, and nothing like that was happening on Ventura. I've had audio cracks on another M1 Pro laptop, but this one didn't even have those. P.S. This is happened just when I was writing this post, and I've disabled Bluetooth just before. Now, the Bluetooth section in the Settings is opening, but others are still unresponsive. For reference - I have yabai and BetterSnapTool installed, which modify system behavior, but with system protection enabled. Siri is disabled. I've tried to stop a bunch of random processes when this happened, but none helped so far. This issue constantly haunts me since I've upgraded, and it's extremely annoying. Original message 2: Yes, I'm thinking it's a combination of Bluetooth and audio issues. I've got all apps that are trying to use audio crashing after I'm just connecting my Bluetooth earbuds. Now I see that the coreaudiod is just not running this time - I've tried to connect to a Slack Huddle, and it just hanged, sound is unresponsive again and the Settings app is not working as I mentioned before. I've checked the Activity Monitor and found that the process that works with audio on macOS (coreaudiod) is not running. I've attempted to launch coreaudiod with sudo launchctl load /system/library/launchdaemons/com.apple.audio.coreaudiod.plis, and got Load failed: 5: Input/output error as a response. After a while, when I disabled the earbuds it's started again on its own and coreaudiod is running, and the audio controls are working once more. Original message 3: Just accidentally looked at the Console App while looking for logs for other things, and found out that my codeaudiod crashing by cooldown every day 10 to 50 times with intervals from 1 second to a couple of hours, around 5 minutes on average. The crash is the following: Crashed Thread: 18 Dispatch queue: com.apple.audio.device.~:AMS2_StackedOutput:0.event Exception Type: EXC_BAD_ACCESS (SIGSEGV) Exception Codes: KERN_INVALID_ADDRESS at 0x0000000000000000 I also found that avconferenced fails too occasionally, though very rare - I believe that's the process that connects iPad as a second screen, and it _too fails with sigsegv on 0x0 - though not that it's some unique bug to attempt to read memory at 0, maybe just a coincidence. @Flomaster do you use Sidecar by chance? My message: I too have this problem on my MacBook Pro M2 Pro since upgrading to macOS Sonoma. It mainly occurs with AirPods Pro 2 but I have also had this happen to me using OnePlus Buds. The blockages are the same as you have experienced and, as I often work in video conferences, blocking MS Teams or Google Meet is really becoming a serious problem. Desperate, I tried installing macOS Sonoma 14.2 beta but none of the updates solved the problem. I even tried a full restore, re-importing the data from Time Machine but to no avail. Indeed, with Beta 4 the problem seems to have worsened because the AirPods now even struggle to connect.

I have a music player that is able to save and restore AU parameters using the kAudioUnitProperty_ClassInfo property. For non apple AUs, this works fine. But for any of the Apple units, the class info can be set only the first time after the audio graph is built. Subsequent sets of the property do not stick even though the OSStatus code is 0 upon return. Previously this had worked fine. But sometime, not sure when, the Apple provided AUs changed their behavior and is now causing me problems. Can anyone help shed light on this ? Thanks in advance for the help. Jeff Frey

I am trying to get the raw audio data from the system microphone using AudioToolbox and CoreFoundation frameworks. So far the writing packets to file logic works but when I try to capture the raw data into a file I am getting white noise. Callback function looks like this: static void MyAQInputCallback(void *inUserData, AudioQueueRef inQueue, AudioQueueBufferRef inBuffer, const AudioTimeStamp *inStartTime, UInt32 inNumPackets, const AudioStreamPacketDescription *inPacketDesc) { MyRecorder *recorder = (MyRecorder *)inUserData; if (inNumPackets > 0) { CheckError(AudioFileWritePackets(recorder->recordFile, FALSE, inBuffer->mAudioDataByteSize, inPacketDesc, recorder->recordPacket, &inNumPackets, inBuffer->mAudioData), "AudioFileWritePackets failed"); recorder->recordPacket += inNumPackets; int sampleCount = inBuffer->mAudioDataByteSize / sizeof(AUDIO_DATA_TYPE_FORMAT); AUDIO_DATA_TYPE_FORMAT* samples = (AUDIO_DATA_TYPE_FORMAT*)inBuffer->mAudioData; FILE *fp = fopen(filename, "a"); for (int i = 0; i < sampleCount; i++){ fprintf(fp, "%i;\n",samples[i]); } fclose(fp); } if (recorder->running) CheckError(AudioQueueEnqueueBuffer(inQueue, inBuffer, 0, NULL), "AudioQueueEnqueueBuffer failed"); } Some parameters: NumberRecordBuffers = 3 buffer duration = 0.1 format->mFramesPerPacket = 4096 samplerate = 44100 inNumPackets = 1 recordFormat.mFormatID = kAudioFormatAppleLossless; recordFormat.mChannelsPerFrame = 1; recordFormat.mBitsPerChannel = 16; Is this the correct way to do this? I could not find much information in the documentation. Any help is appreciated. Thank you in advance.

i have create one recording application, but user switch off or kill the application, so that time how to save ongoing record.

I've added a listener block for camera notifications. This works as expected: the listener block is invoked then the camera is activated/deactivated. However, when I call CMIOObjectRemovePropertyListenerBlock to remove the listener block, though the call succeeds, camera notifications are still delivered to the listener block. Since in the header file it states this function "Unregisters the given CMIOObjectPropertyListenerBlock from receiving notifications when the given properties change." I'd assume that once called, no more notifications would be delivered? Sample code: #import <Foundation/Foundation.h> #import <CoreMediaIO/CMIOHardware.h> #import <AVFoundation/AVCaptureDevice.h> int main(int argc, const char * argv[]) { AVCaptureDevice* camera = [AVCaptureDevice defaultDeviceWithMediaType:AVMediaTypeVideo]; OSStatus status = -1; CMIOObjectID deviceID = 0; CMIOObjectPropertyAddress propertyStruct = {0}; propertyStruct.mSelector = kAudioDevicePropertyDeviceIsRunningSomewhere; propertyStruct.mScope = kAudioObjectPropertyScopeGlobal; propertyStruct.mElement = kAudioObjectPropertyElementMain; deviceID = (UInt32)[camera performSelector:NSSelectorFromString(@"connectionID") withObject:nil]; CMIOObjectPropertyListenerBlock listenerBlock = ^(UInt32 inNumberAddresses, const CMIOObjectPropertyAddress addresses[]) { NSLog(@"Callback: CMIOObjectPropertyListenerBlock invoked"); }; status = CMIOObjectAddPropertyListenerBlock(deviceID, &propertyStruct, dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), listenerBlock); if(noErr != status) { NSLog(@"ERROR: CMIOObjectAddPropertyListenerBlock() failed with %d", status); return -1; } NSLog(@"Monitoring %@ (uuid: %@ / %x)", camera.localizedName, camera.uniqueID, deviceID); sleep(10); status = CMIOObjectRemovePropertyListenerBlock(deviceID, &propertyStruct, dispatch_get_main_queue(), listenerBlock); if(noErr != status) { NSLog(@"ERROR: 'AudioObjectRemovePropertyListenerBlock' failed with %d", status); return -1; } NSLog(@"Stopped monitoring %@ (uuid: %@ / %x)", camera.localizedName, camera.uniqueID, deviceID); sleep(10); return 0; } Compiling and running this code outputs: Monitoring FaceTime HD Camera (uuid: 3F45E80A-0176-46F7-B185-BB9E2C0E436A / 21) Callback: CMIOObjectPropertyListenerBlock invoked Callback: CMIOObjectPropertyListenerBlock invoked Stopped monitoring FaceTime HD Camera (uuid: 3F45E80A-0176-46F7-B185-BB9E2C0E436A / 21) Callback: CMIOObjectPropertyListenerBlock invoked Callback: CMIOObjectPropertyListenerBlock invoked Note the last two log messages showing that the CMIOObjectPropertyListenerBlock is still invoked ...even though CMIOObjectRemovePropertyListenerBlock has successfully been invoked. Am I just doing something wrong here? Or is the API broken?

Generate tones, etc. in code? Is there a way? Regards, Patrick

I am working on an app that uses Core Audio through JUCE library for audio. The problem I'm trying to solve is that when the app is using a full duplex audio interface such as one from Focusrite Scarlett series for output, the app shows a dialog requesting permission to use microphone. The root cause of the issue is that by default, Core Audio opens full duplex devices for both input and output. On previous macOS versions, I was able to work around the problem by disabling the input stream before starting the IOProc by setting AudioHardwareIOProcStreamUsage to all zero for input. On macOS Sonoma this disables input so that the microphone indicator is not shown, but the permission popup is still shown. What other reasons there are to show the popup? I have noticed that Chrome and Slack have the same problem that they show the microphone popup when trying to play sounds on the Focusrite, but for example Deezer manages without the popup.

I have a USB audio interface that is causing kernel traps and the audio output to "skip" or dropout every few seconds. This behavior occurs with a completely fresh install of Catalina as well as Big Sur with the stock Music app on a 2019 MacBook Pro 16 (full specs below). The Console logs show coreaudiod got an error from a kernel trap, a "USB Sound assertion" in AppleUSBAudio/AppleUSBAudio-401.4/KEXT/AppleUSBAudioDevice.cpp at line 6644, and the Music app "skipping cycle due to overload." I've added a short snippet from Console logs around the time of the audio skip/drop out. The more complete logs are at this gist: https://gist.github.com/djflux/08d9007e2146884e6df1741770de5105 I've also opened a Feedback Assistant ticket (FB9037528): https://feedbackassistant.apple.com/feedback/9037528 Does anyone know what could be causing this issue? Thanks for any help. Cheers, Flux aka Andy. Hardware Overview:  Model Name: MacBook Pro  Model Identifier: MacBookPro16,1  Processor Name: 8-Core Intel Core i9  Processor Speed: 2.4 GHz  Number of Processors: 1  Total Number of Cores: 8  L2 Cache (per Core): 256 KB  L3 Cache: 16 MB  Hyper-Threading Technology: Enabled  Memory: 64 GB  System Firmware Version: 1554.80.3.0.0 (iBridge: 18.16.14347.0.0,0) System Software Overview: System Version: macOS 11.2.3 (20D91)  Kernel Version: Darwin 20.3.0  Boot Volume: Macintosh HD  Boot Mode: Normal  Computer Name: mycomputername  User Name: myusername  Secure Virtual Memory: Enabled  System Integrity Protection: Enabled USB interface: Denon DJ DS1 Snippet of Console logs error 21:07:04.848721-0500 coreaudiod HALS_IOA1Engine::EndWriting: got an error from the kernel trap, Error: 0xE00002D7 default 21:07:04.848855-0500 Music HALC_ProxyIOContext::IOWorkLoop: skipping cycle due to overload default 21:07:04.857903-0500 kernel USB Sound assertion (Resetting engine due to error returned in Read Handler) in /AppleInternal/BuildRoot/Library/Caches/com.apple.xbs/Sources/AppleUSBAudio/AppleUSBAudio-401.4/KEXT/AppleUSBAudioDevice.cpp at line 6644 ... default 21:07:05.102746-0500 coreaudiod Audio IO Overload inputs: 'private' outputs: 'private' cause: 'Unknown' prewarming: no recovering: no default 21:07:05.102926-0500 coreaudiod   CAReportingClient.mm:508  message {   HostApplicationDisplayID = "com.apple.Music";   cause = Unknown;   deadline = 2615019;   "input_device_source_list" = Unknown;   "input_device_transport_list" = USB;   "input_device_uid_list" = "AppleUSBAudioEngine:Denon DJ:DS1:000:1,2";   "io_buffer_size" = 512;   "io_cycle" = 1;   "is_prewarming" = 0;   "is_recovering" = 0;   "issue_type" = overload;   lateness = "-535";   "output_device_source_list" = Unknown;   "output_device_transport_list" = USB;   "output_device_uid_list" = "AppleUSBAudioEngine:Denon DJ:DS1:000:1,2"; }: (null)

I was watching a video when I noticed a periodical popping sound. Thought it was a problem with the video, but the issue persists on all other videos or sound tracks. Gets very annoying! Quite disappointed to find this flaw in a brand new Mac Book Pro.

Hello, I am an audio developer, currently using macOS version 14.1.1. I noticed that after disabling the microphone, the small yellow dot in the Control Center disappears immediately, but the one in the menu bar takes about 20 seconds to disappear. I tested the built-in Voice Memos app and found the same behavior. Our users may be concerned about their privacy being violated, even though the software is not using the microphone at that time. We believe this is a bug, and the microphone icon in the menu bar should disappear immediately after the microphone is no longer in use. Do you have plans to fix this issue in future versions? Additionally, is there any workaround for the current version? As a supplement, we are using CoreAudio API with AudioDeviceStart & AudioDeviceStop, not AudioUnit.

Hi, This topic is about Workgroups. I create child processes and I'd like to communicate a os_workgroup_t to my child process so they can join the work group as well. As far as I understand, the os_workgroup_t value is local to the process. I've found that one can use os_workgroup_copy_port() and os_workgroup_create_with_port(), but I'm not familiar at all with ports and I wonder what would be the minimal effort to achieve that. Thank you very much! Alex

After update XCode to 15, I encountered a crash with UnsafeMutableRawPointer. To recreate the problem I write this simple test code. class ViewController: UIViewController { override func viewDidLoad() { super.viewDidLoad() test() } private func test() { var abl = AudioBufferList() let capacity = 4096 let lp1 = UnsafeMutableAudioBufferListPointer(&amp;amp;abl) let outputBuffer1 = UnsafeMutablePointer&amp;lt;Int8&amp;gt;.allocate(capacity: capacity) let outputBuffer2 = UnsafeMutablePointer&amp;lt;Int8&amp;gt;.allocate(capacity: capacity) // It crashed here lp1[0].mData = UnsafeMutableRawPointer(outputBuffer1) lp1[0].mNumberChannels = 1 lp1[0].mDataByteSize = UInt32(capacity) lp1[1].mData = UnsafeMutableRawPointer(outputBuffer2) lp1[1].mNumberChannels = 1 lp1[1].mDataByteSize = UInt32(capacity) let lp2 = UnsafeMutableAudioBufferListPointer(&amp;amp;abl) let data = ( UnsafeMutablePointer&amp;lt;Int16&amp;gt;.allocate(capacity: 4096), packet: 1 ) lp2[0].mData = UnsafeMutableRawPointer(data.0) } } I checked the XCode 15 Release Notes and found out they did something with the pointer default initialization P1020R1 - Smart pointer creation with default initialization Is this causing the problem or I'm doing it wrong? Because it work perfectly fine with XCode 14.3.1 and below P/s: I can't provide the full crash logs cause it's company property but I can provide these: Exception Type: EXC_CRASH (SIGABRT) Exception Codes: 0x0000000000000000, 0x0000000000000000 Exception Note: EXC_CORPSE_NOTIFY Triggered by Thread: 5 Application Specific Information: stack buffer overflow Thread 5 name: Dispatch queue: com.apple.NSXPCConnection.user.endpoint Thread 5 Crashed: 0 libsystem_kernel.dylib 0x20419ab78 __pthread_kill + 8 1 libsystem_pthread.dylib 0x23de0c3bc pthread_kill + 268 2 libsystem_c.dylib 0x1d780c44c __abort + 128 3 libsystem_c.dylib 0x1d77f7868 __stack_chk_fail + 96 Clearly there are something wrong with the memory address after init UnsafeMutableRawPointer from the UnsafeMutablePointer&amp;lt;Int8&amp;gt;

hi, so i have a little bit of work left on the Asus Xonar family of audio devices. thanks to APPUL's samplepciaudiodriver code and their excellent documentation, Evegeny Gavrilov's kxAudio driver for MAC and Takashi Iwai's exceptional documentation of the ALSA API i have something that is ready for testing. the stats look good, but unfortunately i this is my second HDAV1.3 deluxe. the other one is also in the same room consuming all of my devices with powered audio outputs. no matter, i am in the process of acquiring another xonar sound card in this family. which brings me to my question: what is the benefit of getting an apple developer account for 99 dollars a year? will i be able to distribute a beta kext with my signature that will allow people to test the binary? i don't think others could run a self-signed kext built on one machine, on another, correct? so would a developer license allow others to test a binary built on my machine, assuming they're x86? my hope is that the developer program would allow me to test the binaries and solicit input from enthusiast mac pro owners WORLD WIDE. i them hope to create a new program that will give us the wealth mixers/controls this fantastic line is capable of providing.

Hello, I used kAudioDevicePropertyDeviceIsRunningSomewhere to check if an internal or external microphone is being used. My code works well for the internal microphone, and for microphones which are connected using a cable. External microphones which are connected using bluetooth are not reporting their status. The status is always requested successfully, but it is always reported as inactive. Main relevant parts in my code : static inline AudioObjectPropertyAddress makeGlobalPropertyAddress(AudioObjectPropertySelector selector) { AudioObjectPropertyAddress address = { selector, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster, }; return address; } static BOOL getBoolProperty(AudioDeviceID deviceID, AudioObjectPropertySelector selector) { AudioObjectPropertyAddress const address = makeGlobalPropertyAddress(selector); UInt32 prop; UInt32 propSize = sizeof(prop); OSStatus const status = AudioObjectGetPropertyData(deviceID, &address, 0, NULL, &propSize, &prop); if (status != noErr) { return 0; //this line never gets executed in my tests. The call above always succeeds, but it always gives back "false" status. } return static_cast<BOOL>(prop == 1); } ... __block BOOL microphoneActive = NO; iterateThroughAllInputDevices(^(AudioObjectID object, BOOL *stop) { if (getBoolProperty(object, kAudioDevicePropertyDeviceIsRunningSomewhere) != 0) { microphoneActive = YES; *stop = YES; } }); What could cause this and how could it be fixed? Thank you for your help in advance!

I am trying to migrate an Audio Unit host based on the AUv2 C API to the newer AUv3 API. While the migration itself was relatively straightforward (in terms of getting it to compile), the actual rendering fails at run-time with error -10876 aka. kAudioUnitErr_NoConnection. The app does not use AUGraph or AVAudioEngine, perhaps that is an issue? Since the AUv3 and the AUv2 API are bridged in both directions and the rendering works fine with the v2 API, I would expect there to be some way to make it work via the v3 API though. Perhaps someone has an idea why (or under which circumstances) the render block throws this error? For context, the app is Mixxx, an open-source DJing application, and here is the full diff between my AUv2 -> v3 migration: https://github.com/fwcd/mixxx/pull/5/files