使用AVSpeechUtterance实现iOS语音播报,选择语言为简体中文“zh-CN”,读取中文“袆”(hui 第一声)错误,读成了“祎”(yi 第一声),希望能优化。
Audio
RSS for tagDive into the technical aspects of audio on your device, including codecs, format support, and customization options.
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Hi!
I get personal recommendations MusicItemCollection using this code:
func getRecommendations() async throws -> MusicItemCollection<MusicPersonalRecommendation> {
let request = MusicPersonalRecommendationsRequest()
let response = try await request.response()
let recommendations = response.recommendations
return recommendations
}
However, all recommendations contain no more than 12 MusicItem's, while the Music.app application provides much more for some recommendations, for example, for the You recently listened recommendation, the Music.app application displays 40 items. Each recommendation has an items property that contains a collection of musical items MusicItemCollection<MusicPersonalRecommendation.Item>, the hasNextBatch property for these collections is always false. I expected that for some collections loading of new items would be available. Please tell me if I'm doing something wrong or is this a MusicKit bug?
Thank you!
I’ve been researching how to achieve a recording playback effect in iOS similar to the hands-free calling effect in the system’s phone app. How can this be implemented? I tried using the voice chat recording method, but found that the volume of the speaker output is too low. How should this issue be addressed? I couldn’t find a suitable API. Could you provide me with some documentation or sample code? Thank you.
Hello. I am attempting to display the music inside of my app in Now Playing. I've tried a few different methods and keep running into unknown issues. I'm new to Objective-C and Apple development so I'm at a loss of how to continue.
Currently, I have an external call to viewDidLoad upon initialization. Then, when I'm ready to play the music, I call playMusic. I have it hardcoded to play an mp3 called "1". I believe I have all the signing set up as the music plays after I exit the app. However, there is nothing in Now Playing. There are no errors or issues that I can see while the app is running. This is the only file I have in Xcode relating to this feature.
Please let me know where I'm going wrong or if there is another object I need to use!
#import <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <MediaPlayer/MediaPlayer.h>
#import <AVFoundation/AVFoundation.h>
@interface ViewController : UIViewController <AVAudioPlayerDelegate>
@property (nonatomic, strong) AVPlayer *player;
@property (nonatomic, strong) MPRemoteCommandCenter *commandCenter;
@property (nonatomic, strong) MPMusicPlayerController *controller;
@property (nonatomic, strong) MPNowPlayingSession *nowPlayingSession;
@end
@implementation ViewController
- (void)viewDidLoad {
[super viewDidLoad];
NSLog(@"viewDidLoad started.");
[self setupAudioSession];
[self initializePlayer];
[self createNowPlayingSession];
[self configureNowPlayingInfo];
NSLog(@"viewDidLoad completed.");
}
- (void)setupAudioSession {
AVAudioSession *audioSession = [AVAudioSession sharedInstance];
NSError *setCategoryError = nil;
if (![audioSession setCategory:AVAudioSessionCategoryPlayback error:&setCategoryError]) {
NSLog(@"Error setting category: %@", [setCategoryError localizedDescription]);
} else {
NSLog(@"Audio session category set.");
}
NSError *activationError = nil;
if (![audioSession setActive:YES error:&activationError]) {
NSLog(@"Error activating audio session: %@", [activationError localizedDescription]);
} else {
NSLog(@"Audio session activated.");
}
}
- (void)initializePlayer {
NSString *soundFilePath = [NSString stringWithFormat:@"%@/base/game/%@",[[NSBundle mainBundle] resourcePath], @"bgm/1.mp3"];
if (!soundFilePath) {
NSLog(@"Audio file not found.");
return;
}
NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath];
self.player = [AVPlayer playerWithURL:soundFileURL];
NSLog(@"Player initialized with URL: %@", soundFileURL);
}
- (void)createNowPlayingSession {
self.nowPlayingSession = [[MPNowPlayingSession alloc] initWithPlayers:@[self.player]];
NSLog(@"Now Playing Session created with players: %@", self.nowPlayingSession.players);
}
- (void)configureNowPlayingInfo {
MPNowPlayingInfoCenter *infoCenter = [MPNowPlayingInfoCenter defaultCenter];
CMTime duration = self.player.currentItem.duration;
Float64 durationSeconds = CMTimeGetSeconds(duration);
CMTime currentTime = self.player.currentTime;
Float64 currentTimeSeconds = CMTimeGetSeconds(currentTime);
NSDictionary *nowPlayingInfo = @{
MPMediaItemPropertyTitle: @"Example Title",
MPMediaItemPropertyArtist: @"Example Artist",
MPMediaItemPropertyPlaybackDuration: @(durationSeconds),
MPNowPlayingInfoPropertyElapsedPlaybackTime: @(currentTimeSeconds),
MPNowPlayingInfoPropertyPlaybackRate: @(self.player.rate)
};
infoCenter.nowPlayingInfo = nowPlayingInfo;
NSLog(@"Now Playing info configured: %@", nowPlayingInfo);
}
- (void)playMusic {
[self.player play];
[self createNowPlayingSession];
[self configureNowPlayingInfo];
}
- (void)pauseMusic {
[self.player pause];
[self configureNowPlayingInfo];
}
@end
iPadOS 18.3 beta 3 (22D5055b) fixed the issue for me and my 7th generation iPad.
I am trying to use AVAudioEngine for recording and playing for a voice chat kind of app, but when the speaker plays any audio while recording, the recording take the speaker audio as input. I want to filter that out. Are there any suggestions for the swift code
I have used AVQueuePlayer in my music app to play sequence of audios from a remote server, this how I have defined things my player in my ViewModel
Variables
private var cancellables = Set()
private let audioSession = AVAudioSession.sharedInstance()
private var avQueuePlayer: AVQueuePlayer?
@Published var playbackSpeed: Float = 1.0
before starting playback, I am making sure that audio session is set properly, the code snippet used for that is
do {
try audioSession.setCategory(.playback, mode: .default, options: [])
try audioSession.setActive(true, options: [])
} catch {
return
}
and this is the function I am using to update playback speed
func updatePlaybackSpeed(_ newSpeed: Float){
if newSpeed > 0.0, newSpeed <= 2.0{
playbackSpeed = newSpeed
avQueuePlayer?.rate = newSpeed
print("requested speed is (newSpeed) and actual speed is (String(describing: avQueuePlayer?.rate))")
}
}
sometimes whatever speed is set, player seems to play at the same speed as it was set,
e.g. Once I got "requested speed is 1.5 and actual speed is 1.5", and player also seemed to play at the speed of 1.5
but another time I got "requested speed is 2.0 and actual speed is 2.0", but player still seemed to play at the speed of 1.0
to observe changes in rate, I used this
**private func observeRateChanges() {
guard let avQueuePlayer = self.avQueuePlayer else { return }
NotificationCenter.default.publisher(for: AVQueuePlayer.rateDidChangeNotification, object: avQueuePlayer)
.compactMap { $0.userInfo?[AVPlayer.rateDidChangeReasonKey] as? AVPlayer.RateDidChangeReason }
.sink { reason in
switch reason {
case .appBackgrounded:
print("The app transitioned to the background.")
case .audioSessionInterrupted:
print("The system interrupts the app’s audio session.")
case .setRateCalled:
print("The app set the player’s rate.")
case .setRateFailed:
print("An attempt to change the player’s rate failed.")
default:
break
}
}
.store(in: &cancellables)
}**
when rate was set properly, I got this "The app set the player’s rate." from the above function, but when it wasn't, I got this "An attempt to change the player’s rate failed.,"
now I am not able to understand why rate is not being set, and if it gave "requested speed is 2.0 and actual speed is 2.0" from updatePlaybackSpeed function, why does the player seems to play with the speed of 1.0?
i have a CarPlay implementation eand I want to show previous/next track button on player UI
MPRemoteCommandCenter.shared().seekForwardCommand.isEnabled = false
MPRemoteCommandCenter.shared().seekBackwardCommand.isEnabled = false
MPRemoteCommandCenter.shared().previousTrackCommand.isEnabled = true
MPRemoteCommandCenter.shared().nextTrackCommand.isEnabled = true
It works correctly on CarPlay simulator , but on some car only SEEK button are shown .
I have to suppose that it is that a problem on the car side , but I would ask about your opinion , maybe there is some pieces I'm missing
I'm trying to implement Ambisonic B-Format audio playback on Vision Pro with head tracking. So far audio plays, head tracking works, and the sound appears to be stereo. The problem is that it is not a proper binaural playback when compared to playing back the audiofile with a DAW. Has anyone successfully implemented B-Format playback on Vision Pro? Any suggestions on my current implementation:
func playAmbiAudioForum() async {
do {
try AVAudioSession.sharedInstance().setCategory(.playback)
try AVAudioSession.sharedInstance().setActive(true)
// AudioFile laoding/preperation
guard let testFileURL = Bundle.main.url(forResource: "audiofile", withExtension: "wav") else {
print("Test file not found")
return
}
let audioFile = try AVAudioFile(forReading: testFileURL)
let audioFileFormat = audioFile.fileFormat
// create AVAudioFormat with Ambisonics B Format
guard let layout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Ambisonic_B_Format) else {
print("layout failed")
return
}
let format = AVAudioFormat(
commonFormat: audioFile.processingFormat.commonFormat,
sampleRate: audioFile.fileFormat.sampleRate,
interleaved: false,
channelLayout: layout
)
// write audiofile to buffer
guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: UInt32(audioFile.length)) else {
print("buffer failed")
return
}
try audioFile.read(into: buffer)
playerNode.renderingAlgorithm = .HRTF
// connecting nodes
audioEngine.attach(playerNode)
audioEngine.connect(playerNode, to: audioEngine.outputNode, format: format)
audioEngine.prepare()
playerNode.scheduleBuffer(buffer, at: nil) {
print("File finished playing")
}
try audioEngine.start()
playerNode.play()
} catch {
print("Setup error:", error)
}
}
I’m facing a problem while trying to achieve spatial audio effects in my iOS 18 app. I have tried several approaches to get good 3D audio, but the effect never felt good enough or it didn’t work at all.
Also what mostly troubles me is I noticed that AirPods I have doesn’t recognize my app as one having spatial audio (in audio settings it shows "Spatial Audio Not Playing"). So i guess my app doesn't use spatial audio potential.
First approach uses AVAudioEnviromentNode with AVAudioEngine. Chaining position of player as well as changing listener’s doesn’t seem to change anything in how audio plays.
Here's simple how i initialize AVAudioEngine
import Foundation
import AVFoundation
class AudioManager: ObservableObject {
// important class variables
var audioEngine: AVAudioEngine!
var environmentNode: AVAudioEnvironmentNode!
var playerNode: AVAudioPlayerNode!
var audioFile: AVAudioFile?
...
//Sound set up
func setupAudio() {
do {
let session = AVAudioSession.sharedInstance()
try session.setCategory(.playback, mode: .default, options: [])
try session.setActive(true)
} catch {
print("Failed to configure AVAudioSession: \(error.localizedDescription)")
}
audioEngine = AVAudioEngine()
environmentNode = AVAudioEnvironmentNode()
playerNode = AVAudioPlayerNode()
audioEngine.attach(environmentNode)
audioEngine.attach(playerNode)
audioEngine.connect(playerNode, to: environmentNode, format: nil)
audioEngine.connect(environmentNode, to: audioEngine.mainMixerNode, format: nil)
environmentNode.listenerPosition = AVAudio3DPoint(x: 0, y: 0, z: 0)
environmentNode.listenerAngularOrientation = AVAudio3DAngularOrientation(yaw: 0, pitch: 0, roll: 0)
environmentNode.distanceAttenuationParameters.referenceDistance = 1.0 environmentNode.distanceAttenuationParameters.maximumDistance = 100.0
environmentNode.distanceAttenuationParameters.rolloffFactor = 2.0
// example.mp3 is mono sound
guard let audioURL = Bundle.main.url(forResource: "example", withExtension: "mp3") else {
print("Audio file not found")
return
}
do {
audioFile = try AVAudioFile(forReading: audioURL)
} catch {
print("Failed to load audio file: \(error)")
}
}
...
//Playing sound
func playSpatialAudio(pan: Float ) {
guard let audioFile = audioFile else { return }
// left side
playerNode.position = AVAudio3DPoint(x: pan, y: 0, z: 0)
playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil)
do {
try audioEngine.start()
playerNode.play()
} catch {
print("Failed to start audio engine: \(error)")
}
...
}
Second more complex approach using PHASE did better. I’ve made an exemplary app that allows players to move audio player in 3D space. I have added reverb, and sliders changing audio position up to 10 meters each direction from listener but audio seems to only really change left to right (x axis) - again I think it might be trouble with the app not being recognized as spatial.
//Crucial class Variables:
class PHASEAudioController: ObservableObject{
private var soundSourcePosition: simd_float4x4 = matrix_identity_float4x4
private var audioAsset: PHASESoundAsset!
private let phaseEngine: PHASEEngine
private let params = PHASEMixerParameters()
private var soundSource: PHASESource
private var phaseListener: PHASEListener!
private var soundEventAsset: PHASESoundEventNodeAsset?
// Initialization of PHASE
init{
do {
let session = AVAudioSession.sharedInstance()
try session.setCategory(.playback, mode: .default, options: [])
try session.setActive(true)
} catch {
print("Failed to configure AVAudioSession: \(error.localizedDescription)")
}
// Init PHASE Engine
phaseEngine = PHASEEngine(updateMode: .automatic)
phaseEngine.defaultReverbPreset = .mediumHall
phaseEngine.outputSpatializationMode = .automatic //nothing helps
// Set listener position to (0,0,0) in World space
let origin: simd_float4x4 = matrix_identity_float4x4
phaseListener = PHASEListener(engine: phaseEngine)
phaseListener.transform = origin
phaseListener.automaticHeadTrackingFlags = .orientation
try! self.phaseEngine.rootObject.addChild(self.phaseListener)
do{
try self.phaseEngine.start();
}
catch {
print("Could not start PHASE engine")
}
audioAsset = loadAudioAsset()
// Create sound Source
// Sphere
soundSourcePosition.translate(z:3.0)
let sphere = MDLMesh.newEllipsoid(withRadii: vector_float3(0.1,0.1,0.1), radialSegments: 14, verticalSegments: 14, geometryType: MDLGeometryType.triangles, inwardNormals: false, hemisphere: false, allocator: nil)
let shape = PHASEShape(engine: phaseEngine, mesh: sphere)
soundSource = PHASESource(engine: phaseEngine, shapes: [shape])
soundSource.transform = soundSourcePosition
print(soundSourcePosition)
do {
try phaseEngine.rootObject.addChild(soundSource)
}
catch {
print ("Failed to add a child object to the scene.")
}
let simpleModel = PHASEGeometricSpreadingDistanceModelParameters()
simpleModel.rolloffFactor = rolloffFactor
soundPipeline.distanceModelParameters = simpleModel
let samplerNode = PHASESamplerNodeDefinition(
soundAssetIdentifier: audioAsset.identifier,
mixerDefinition: soundPipeline,
identifier: audioAsset.identifier + "_SamplerNode")
samplerNode.playbackMode = .looping
do {soundEventAsset = try
phaseEngine.assetRegistry.registerSoundEventAsset(
rootNode: samplerNode,
identifier: audioAsset.identifier + "_SoundEventAsset")
} catch {
print("Failed to register a sound event asset.")
soundEventAsset = nil
}
}
//Playing sound
func playSound(){
// Fire new sound event with currently set properties
guard let soundEventAsset else { return }
params.addSpatialMixerParameters(
identifier: soundPipeline.identifier,
source: soundSource,
listener: phaseListener)
let soundEvent = try! PHASESoundEvent(engine: phaseEngine,
assetIdentifier: soundEventAsset.identifier,
mixerParameters: params)
soundEvent.start(completion: nil)
}
...
}
Also worth mentioning might be that I only own personal team account
I have spent a long time refactoring lots of older Swift code to compile without error in Swift 6.
The app is a v3 audio unit host and audio unit.
Having installed Sonoma and XCode 16 I compile the code using Swift 6 and it compiles and runs without any warnings or errors.
My host will load my AU no problem.
LOGIC PRO is still the ONLY audio unit host that will load native Mac V3 audio units and so I like to test my code using Logic.
In Sonoma with XCode 16...
My AU passes the most stringent AUVAL tests both in terminal and Logic pro.
If I compile the AU source in Swift 5 Logic will see the AU, load it and run it without problems.
But when I compile the AU in Swift 6 Logic sees the AU, will scan it and verify it passes the tests but will not load the AU. In XCode I see a log message that a "helper application failed to run" but the debugger never connects to the AU and I don't think Logic even gets as far as instantiating the AU.
So... what is causing this? I'm stumped..
Developing AUv3 is a brain-aching maze of undocumented hurdles and I'm hoping someone might have found a solution for this one. Meanwhile I guess my only option is to continue using the Swift 5 compiler.
(appending a little note just to mention that all the DSP code is written in C/C++, Swift is used mainly for the user interface and also does some offline thready work )
I recently got some plugins from Universal Audio, and have licensed them properly through both UA and iLok manager. Whenever I try to load up the plugins (specifically from UA) in GarageBand, it first says that
"NSCreateObjectFileImageFromMemory-p47UEwps” because the developper can not be verified.
After clicking either 'show in finder' or 'okay', it opens the plugin in a form without its GUI and showing that it is not licensed (even though it is). It also displays error code 100001. I have tried only some basic stuff to troubleshoot like restarting the DAW/my computer and reinstalling/relicensing the softwares. I don't know if the macOS version has anything to do with it but for some reason I just can't get it to work.
Hi all,
I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in.
Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped.
Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played.
Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset?
I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity.
Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them
Thanks for any feedback!
Please Update Andorid MusicKit,the version 1.1.2 will complied fail。the error msg:•SDKUriHandlerActivity>. Apps targeting Android 12 and higher are required to specify an explicit value for android:exported when the corres
Please include the line below in follow-up emails for this request.
Case-ID: 11089799
When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16.
1.let utterance = AVSpeechUtterance(string: textView.text)
let voice = AVSpeechSynthesisVoice(language: "zh-CN")
utterance.voice = voice
2.In the phone settings, Siri is set to Cantonese
I am using an AVAudioPlayer to play a "tick" sound once per second in a SwiftUI app.
When running the app on an iPhone 16 (18.2.1) the tick sounds increase in volume after a few seconds. This does not happen in the simulator nor on an iPhone SE 2020 (18.1.1).
I’m currently developing an iOS metronome app using DispatchSourceTimer as the timer. The interval is set very small, around 50 milliseconds, and I’m using CFAbsoluteTimeGetCurrent to calculate the elapsed time to ensure the beat is played within a ±0.003-second margin.
The problem is that once the app goes to the background, the timing becomes unstable—it slows down noticeably, then recovers after 1–2 seconds.
When coming back to the foreground, it suddenly speeds up, and again, it takes 1–2 seconds to return to normal. It feels like the app is randomly “powering off” and then “overclocking.” It’s super frustrating.
I’ve noticed that some metronome apps in the App Store have similar issues, but there’s one called “Professional Metronome” that’s rock solid with no such problems. What kind of magic are they using? Any experts out there who can help? Thanks in advance!
P.S. I’ve already enabled background audio permissions.
The professional metronome that has no issues: https://link.zhihu.com/?target=https%3A//apps.apple.com/cn/app/pro-metronome-%25E4%25B8%2593%25E4%25B8%259A%25E8%258A%2582%25E6%258B%258D%25E5%2599%25A8/id477960671
I'm encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected.
Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers)
Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers)
When using paired input and output devices:
The setup works as expected.
Example: MacBook Pro microphone → MacBook Pro speakers.
When using mismatched devices:
AVAudioEngine setup fails during aggregate device construction.
Example: AirPods microphone → MacBook Pro speakers.
Error logs indicate a channel count mismatch.
Here are the partial logs. Due to the content limit, I cannot post the entire logs.
AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875)
AUVPAggregate.cpp:1036 err=-10875
AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875
AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts
AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U)
AggregateDevice.mm:182 error fetching default pair
AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts
AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U)
AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702]
AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID
AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702]
AggregateDevice.mm:182 error fetching default pair
AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts
AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U)
Is it possible to use voice processing with different input/output devices?
If yes, are there any specific configurations required to handle mismatched devices?
How can we resolve channel count mismatch errors during aggregate device construction?
Are there settings or API adjustments to enforce compatibility between input/output devices?
Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices?
For instance, can we force an intermediate channel configuration or downmix input/output formats?
I am having issues deploying my iOS app, that uses ShazamKit, to get working on a Mac with Apple silicon.
When uploading the archive to App Store Connect I do get
ITMS-90863: Macs with Apple silicon support issue - The app links with libraries that aren’t present in macOS:
/usr/lib/swift/libswiftShazamKit.dylib
Is ShazamKit not supported for iOS apps that can run on Macs with Apple silicon? Or is there something I should fix in my setup / deployment?
I recently installed a rear-view camera in my car, and ever since, I've been experiencing a frustrating issue with my CarPlay. After about 15 seconds of playing audio via Bluetooth, the sound stops coming out of the speakers, even though the song continues to run in the background.
For context, my stereo system is an aftermarket unit that I installed to enable CarPlay functionality. Everything worked perfectly before adding the rear-view camera. Unfortunately, my unit does not have a port for a wired connection, so I can't test the audio using a cable.
Has anyone experienced a similar issue? Could the camera installation be interfering with the Bluetooth or audio system somehow? Any advice or troubleshooting tips would be greatly appreciated!