Hello. My team and I think we have an issue where our app is asked to gracefully shutdown with a following SIGTERM. As we’ve learned, this is normally not an issue. However, it seems to also be happening while our app (an audio streamer) is actively playing in the background. From our perspective, starting playback is indicating strong user intent. We understand that there can be extreme circumstances where the background audio needs to be killed, but should it be considered part of normal operation? We hope that’s not the case. All we see in the logs is the graceful shutdown request. We can say with high certainty that it’s happening though, as we know that playback is running within 0.5 seconds of the crash, without any other tracked user interaction. Can you verify if this is intended behavior, and if there’s something we can do about it from our end. From our logs it doesn’t look to be related to either memory usage within the app, or the system as a whole. Best, John

Hello! I've two mics connected to a USB-hub. The USB-hub is then connected to my iPad. Both mics are part of the audio session's list of available inputs. The problem is that regardless of which mic I select in my app (using setPreferredInput() on the audio session), the audio keeps coming from the mic that was last connected to the USB-hub. Anyone that knows if this is a limitation in iPadOS/iOS?

iOS 26.0 (23A5276f) – Bluetooth Call Audio Issue I’m experiencing a Bluetooth audio issue on iOS 26.0 (build 23A5276f). I cannot make or receive phone calls properly using Bluetooth devices — this affects both my car’s Bluetooth system and my AirPods Pro (2nd generation). Notably: Regular phone calls have no audio (either I can’t hear the other person, or they can’t hear me). WhatsApp and other VoIP apps work fine with the same Bluetooth devices. Media playback (music, video, etc.) works without issues over Bluetooth. It seems this bug is limited to the native Phone app or the system audio routing for regular cellular calls. Please advise if this is a known issue or if a fix is expected in upcoming beta releases.

Hey folks, I'm running into an odd issue suddenly with an app that had a working MusicKit integration before. I'm using ApplicationMusicPlayer to play Apple Music albums and songs. I'm testing on a physical device, signed in to Apple ID, and with a valid subscription. Apple Music via the first-party app works entirely fine on this device. Attempting to play back any content at all gives the log: <ICUserIdentityStoreACAccountBackend: 0x1070bf3e0> Failed to initialize primary apple account, error=Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store} [ICUserIdentityStore] - initializing account histories with activeAccountDSID = nil, activeLockerAccountDSID = nil, timestamp = 14605951908 [ICUserIdentityStore] Failed to fetch local store account with error: Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store}. The album artwork, track names, etc, all appear in the control center playback controls, but the music doesn't play. Trying to trigger playback with control center just results in it skipping to the next track, which doesn't play either. This exact code used to work. I have the MusicKit service selected in Apple Connect. Since this isn't entitlement-based, I'm not sure how else to check that I'm set up correctly. I've tried deleting/reinstalling the app, restarting the device, cleaning/rebuilding, and deleting DerivedData, to no avail. Any help? Running Xcode 16.4 (16F6), testing on iOS 18.5 (22F76)

I have a question regarding the behavior of AVAudioSession.sharedInstance().outputVolume. Observed behavior: When the app is in the foreground, I read audioSession.outputVolume (for example, 0.1). The app is then moved to the background. While the app is in the background, the user changes the system volume using the hardware buttons (for example, to 0.5). When the app returns to the foreground, audioSession.outputVolume still reports the previous value (0.1). From my testing, outputVolume only seems to update when the system volume is changed while the app is in the foreground. Volume changes made while the app is in the background are not reflected when the app returns to the foreground. Questions: According to Apple’s documentation for AVAudioSession.outputVolume: “The systemwide output volume set by the user.” https://developer.apple.com/documentation/avfaudio/avaudiosession/outputvolume However, based on our testing on iOS 18.6.2 and iOS 18.1, the observed behavior seems to differ from this description. Questions: The documentation states that outputVolume represents the system-wide volume set by the user. In our testing, the value does not reflect volume changes made while the app is in the background and only updates when the app is in the foreground.Is this the expected behavior of AVAudioSession.outputVolume? Is there any other recommended way in Swift to retrieve the current system volume that reflects user changes made both while the app is in the foreground and while it is in the background? Any clarification on the intended behavior or recommended handling would be greatly appreciated.

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!

Hi, our CourAudio server plugin utilizes the SystemConfiguration.framework to store and restore specific shared system wide settings. While our application can authenticate to utilize the SystemConfiguration.framework to gain write access to the shared configuration settings the CoreAudio server plugin obviously can't have any user interaction and therefor does not authenticate. Is it possible to authenticate the CoreAudio server plugin to gain write permissions? Are there any entitlements or other means that would allow this? Thanks!

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

Among Japanese end users, audio issues during screen recording—primarily in game applications—have become a topic of discussion. We have confirmed that the trigger for this issue is highly likely to be related to changes to IOBufferDuration. When using setPreferredIOBufferDuration and the IOBufferDuration is set to a value smaller than the default, audio problems occur in the recorded screen capture video. Audio playback is performed using AudioUnit (RemoteIO). https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc This issue was not observed on iOS 18, and it appears to have started occurring after upgrading to iOS 26. We provide an audio middleware solution, and we had incorporated changes to IOBufferDuration into our product to achieve low-latency audio playback. As a result, developers using our product as well as their end users are being affected by this issue. We kindly request that this issue be investigated and addressed in a future update. “This document has been translated by AI. The original text is included below for reference.” 日本のエンドユーザー間で主にゲームアプリケーションにおける画面収録時の音声の問題が話題になっています。 こちらの症状のトリガーが、IOBufferDurationの変更によるものである可能性が高いことを確認しました。 setPreferredIOBufferDurationを使用し、IOBufferDurationがデフォルトより小さい状態の時、画面収録された動画の音声に問題が発生することをしています。 音声の再生にはAudioUnit(RemoteIO)を使用しています。 https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc iOS 18ではこのような問題は確認されておらず、iOS26になってから問題が発生しているようです。 私たちはオーディオミドルウェアを提供しており、低遅延の再生のためにIOBufferDurationの変更を製品に組み込んでいました。 そのため、弊社製品をご利用いただいている開発者およびエンドユーザーの皆様がこの不具合の影響を受けています。 こちらの不具合の調査及び修正対応を検討いただけますでしょうか。

I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!

Hello, I am building an iOS-only, commercial app that uses AVSpeechSynthesizer with system voices, strictly using the APIs provided by Apple. Before distributing the app, I want to ensure that my current implementation does not conflict with the iOS Software License Agreement (SLA) and is aligned with Apple’s intended usage. For a better playback experience (more accurate estimation of utterance duration and smoother skip forward/backward during playback), I currently synthesize speech using: AVSpeechSynthesizer.write(_:toBufferCallback:) Converting the received AVAudioPCMBuffer buffers into audio data Storing the audio inside the app sandbox Playing it back using AVAudioPlayer / AVAudioEngine The cached audio is: Generated fully on-device using system voices Stored only inside the app’s private container Used only for internal playback controls (timeline, seek, skip ±5 seconds) Never shared, exported, uploaded, or exposed outside the app The alternative approaches would be: Keeping the generated audio entirely in memory (RAM) for playback purposes, without writing it to the file system at any point Or using AVSpeechSynthesizer.speak(_:) and playing speech strictly in real time which has a poorer user experience compared to my approach I have reviewed the current iOS Software License Agreement: https://www.apple.com/legal/sla/docs/iOS18_iPadOS18.pdf In particular, section (f) mentions restrictions around System Characters, Live Captions, and Personal Voice, including the following excerpt: “…use … only for your personal, non-commercial use… No other creation or use of the System Characters, Live Captions, or Personal Voice is permitted by this License, including but not limited to the use, reproduction, display, performance, recording, publishing or redistribution in a … commercial context.” I do not see a specific reference in the SLA to system text-to-speech voices used via AVSpeechSynthesizer, and I want to be certain that temporarily caching synthesized speech for internal, non-exported playback is acceptable in a commercial app. My question is: Is caching AVSpeechSynthesizer system-voice output inside the app sandbox for internal playback acceptable, or is Apple’s recommended approach to rely only on real-time playback (speak(_:)) or strictly in-memory buffering without file storage? If this question falls outside DTS technical scope and is instead a policy or licensing matter, I would appreciate guidance on the authoritative Apple documentation or the correct Apple team/contact. Thank you.

Hello, Starting in iOS 17, our application started having some issue publishing to our video session. More specifically the video capture seems to be broken in some, but not all sessions. What's troubling is that we're seeing that it fails consistently every 4 sessions. It also fails silently, without reporting any problems to the app. We only notice that there are no frames being rendered or sent to the remote devices. Here's what shows-up in the console: <<<< FigCaptureSourceRemote >>>> Fig assert: "! storage->connectionDied" at bail (FigCaptureSourceRemote.m:235) - (err=0) <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:253) - (err=-16453) Anyone seeing this? Any idea what could be the cause? Our sessions work perfectly on iOS16 and below. Thanks

AVAudioFormat has no Swift concurrency annotations but the documentation states "Instances of this class are immutable." This made me always assume it was safe to pass AVAudioFormat instances around. Is this the case? If so can it be marked as Sendable? Am I missing something?

I noticed that while playing back the same tracks via MusicKit on different OSes I get different results regarding the audio files being streamed. Playing back a lossless file with 24Bit 48kHz and watching the Console for RemotePlayerService I get: on iPadOS: Lossless; groupID: audio-alac-stereo-48000-24; bitDepth: 24-bit; sampleRate: 48khz; codec: alac; channels: 2; layout: Stereo; on macOS: Creating AudioQueue with format:'paac', framesPerPacket:1024, sampleRate:44100 While the iPad looks perfect, the Mac does not. Is there a way to fix this issue on macOS. BTW: I switched the Audio-Midi Settings before, after and while the macOS App was lunched. I also switched to different output devices. I wasn't able to change the bad audio-output on the mac. I tested this under Sequoia 15.5 and Tahoe beta 1, Xcode 16.4 and 26 beta 1. The AudioVariants of the Album/Tracks are .dolbyAtmos, .lossless, .lossyStereo Apple Music displays Lossless 24 Bit/48 kHz ALAC when clicking on the playercontroll icon on macOS I hope there are only some missing or misconfigured properties to get macOS up to par. Thanks :-)

So experimenting with the new SpeechTranscriber, if I do: let transcriber = SpeechTranscriber( locale: locale, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange] ) only the final result has audio time ranges, not the volatile results. Is this a performance consideration? If there is no performance problem, it would be nice to have the option to also get speech time ranges for volatile responses. I'm not presenting the volatile text at all in the UI, I was just trying to keep statistics about the non-speech and the speech noise level, this way I can determine when the noise level falls under the noisefloor for a while. The goal here was to finalize the recording automatically, when the noise level indicate that the user has finished speaking.

Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël

I have a SwiftUI app - (https://youtu.be/VbAfUk_eYl0?si=JxUBh0Bpb-vc1E1U) - which I thought was almost ready for release - a manager for airdropped audio files from Logic Pro or other music creation applications. It uses AVAudioEngine and AVAudioPlayerNode to play audio, and the MediaPlayer API to integrate with car audio and similar, all of which works well. It does not currently have an explicit CarPlay integration (and I'm slightly horrified at the amount of work that is going to require). I had the good or bad luck of getting a loaner car with carplay while mine is being repaired yesterday, and lo and behold, when connected to the vehicle via CarPlay, there is no audio output in the vehicle at all. The now playing panel correctly shows the information my app provides about the currently playing song; the player node believes it is playing, the AVAudioSession is configured as it should be. But there is no sound. Obviously I cannot ship it in this state. I've tried fiddling with the parameters the AVAudioSession is configured with, in case there was some parameter that was preventing audio output, to no avail - currently: var options = AVAudioSession.CategoryOptions() options.insert(.allowAirPlay) options.insert(.allowBluetooth) options.insert(.allowBluetoothA2DP) try session.setCategory(.playback, mode: .default, options: options) try? session.setPreferredIOBufferDuration(0.002) // ~96 samples at 44.1kHz try? session.setPrefersNoInterruptionsFromSystemAlerts(true) try? session.setPrefersInterruptionOnRouteDisconnect(false) try session.setActive(true, options: [.notifyOthersOnDeactivation]) All diagnostics within the app show the player operating correctly - files are played and flushed; AVAudioPlayerNodeCompletionCallbacks are called when they should be. But the output is not audible in the vehicle. I would much prefer to ship this app without full-blown CarPlay integration, but with working audio when connected via CarPlay, and work on full CarPlay integration for the next release. Is there some secret handshake I am just missing to make this work?

AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

I'm trying to implement airplay into my app. I can successfully playback sound and trigger the airplay selector sheet. If the target device is a Bluetooth only device I can connect with no problem and stream the audio to the Bluetooth device, but if the audio device is a airplay specific device like a HomePod or an Apple TV when I select it, I get a spinning icon, indicating that it is trying to connect, and eventually it times out and stops without connecting. I don't believe it is an AirPlay audio issue because if I go to a different app, for example a podcast app and select my HomePods for output, and then switch back to my app. My audio will correctly stream to the HomePod. Not only that, I have it so that my icon will change color to indicate that it is connected via airplay and it is correctly indicating that it is connected via AirPlay. But I cannot then disconnect it using the Airplay selector. The issue appears to be in the AirPlay selection side, which I have spent several days attempting to troubleshoot mostly using ChatGPT to suggest code different than what I have to maybe work around the issue. Mostly it is focused on the audio player section, but it doesn't seem like that is really the route that is the problem.

Hello Apple Developer Community, I am seeking clarification on the intended display behavior of HLS audio tracks within the iOS 26 (or current beta) native player, specifically concerning the NAME and LANGUAGE attributes of the EXT-X-MEDIA tag. In our HLS manifests, we define alternative audio tracks using EXT-X-MEDIA tags, like so: #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="ja",NAME="AUDIO-1",DEFAULT=YES,AUTOSELECT=YES,URI="audio_ja.m3u8" #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="ja",NAME="AUDIO-2",URI="audio_en.m3u8" Our observation is that when an audio track is selected and its name is displayed in the native iOS media controls (e.g., Control Center or within a full-screen video player's UI), the value specified in the NAME attribute ("AUDIO-1", "AUDIO-2") does not seem to be used. Instead, the display appears to derive from the LANGUAGE attribute ("ja", "en"), often showing the system's localized string for that language (e.g., "Japanese", "English"). We would like to understand the official or intended behavior regarding this. Is it the expected behavior for the iOS native player to prioritize the LANGUAGE attribute (or its localized equivalent) over the NAME attribute for displaying the selected audio track's label? If this is the intended design, what is the recommended best practice for developers who wish to present a custom, human-readable name for audio tracks (beyond the standard language name) in the native iOS UI? Are there any specific AVPlayer properties or AVMediaSelectionOption considerations that would allow more granular control over this display, or is this entirely managed by the system based on the LANGUAGE attribute? Any insights or official guidance on this behavior in iOS 26 (and potentially previous versions) would be greatly appreciated. Thank you for your time and assistance.