I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns.
Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally.
var bufferedData = Data()
func parseBytes(data: Data) {
bufferedData.append(data)
// XXX: this buffering reduces glitching
// to rather infrequent. But why?
if bufferedData.count > 32768 {
bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in
guard let baseAddress = bytes.baseAddress else { return }
let result = AudioFileStreamParseBytes(audioStream!,
UInt32(bufferedData.count),
baseAddress,
[])
if result != noErr {
print("❌ error parsing stream: \(result)")
}
}
bufferedData = Data()
}
}
No errors are returned by AudioFileStream or AVAudioConverter.
func handlePackets(data: Data,
packetDescriptions: [AudioStreamPacketDescription]) {
guard let audioConverter else {
return
}
var maxPacketSize: UInt32 = 0
for packetDescription in packetDescriptions {
maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize)
if packetDescription.mDataByteSize == 0 {
print("EMPTY PACKET")
}
if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count {
print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)")
}
}
let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize))
bufferIn.byteLength = UInt32(data.count)
for i in 0 ..< Int(packetDescriptions.count) {
bufferIn.packetDescriptions![i] = packetDescriptions[i]
}
bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count)
_ = data.withUnsafeBytes { ptr in
memcpy(bufferIn.data, ptr.baseAddress, data.count)
}
if verbose {
print("handlePackets: \(data.count) bytes")
}
// Setup input provider closure
var inputProvided = false
let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in
if !inputProvided {
inputProvided = true
statusPtr.pointee = .haveData
return bufferIn
} else {
statusPtr.pointee = .noDataNow
return nil
}
}
// Loop until converter runs dry or is done
while true {
let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)!
bufferOut.frameLength = 0
var error: NSError?
let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock)
switch status {
case .haveData:
if verbose {
print("✅ convert returned haveData: \(bufferOut.frameLength) frames")
}
if bufferOut.frameLength > 0 {
if bufferOut.isSilent {
print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)")
}
outBuffers.append(bufferOut)
totalFrames += Int(bufferOut.frameLength)
}
case .inputRanDry:
if verbose {
print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames")
}
if bufferOut.frameLength > 0 {
if bufferOut.isSilent {
print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)")
}
outBuffers.append(bufferOut)
totalFrames += Int(bufferOut.frameLength)
}
return // wait for next handlePackets
case .endOfStream:
if verbose {
print("✅ convert returned endOfStream")
}
return
case .error:
if verbose {
print("❌ convert returned error")
}
if let error = error {
print("error converting: \(error.localizedDescription)")
}
return
@unknown default:
fatalError()
}
}
}
Explore the integration of media technologies within your app. Discuss working with audio, video, camera, and other media functionalities.
Selecting any option will automatically load the page
Post
Replies
Boosts
Views
Activity
Hello! I have been following the UsingAVFoundationToPlayAndPersistHTTPLiveStreams sample code in order to test persisting streams to disk. In addition to support for m3u8, I have noticed in testing that this also seems to work for MP3 Audio, simply by changing the plist entries to point to remote URLs with audio/mpeg content. Is this expected, or are there caveats that I should be aware of?
Thanks you!
I am getting high error rates from the Apple Music API. This has been happening for months now, and it is quite frustrating. It is a mix of 404, 504, and random 500 errors. I hit these endpoints all of the time, so it is not like I am hitting a resource that doesn't exist. Why is this happening? Is this a known issue that is getting worked on?
I use the Apple Music API to poll my listening history at regular intervals.
Every morning between 5:30AM and 7:30AM, I observe a strange pattern in the API responses. During this window, one or more of the regular polling intervals returns a response that differs significantly from the prior history response, even though I had no listening activity at that time.
I'm using this endpoint: https://api.music.apple.com/v1/me/recent/played/tracks?types=songs,library-songs&include[library-songs]=catalog&include[songs]=albums,artists
Here’s a concrete example from this morning:
Time: 5:45AM
Fetch 1 Tracks (subset):
1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ...
Time: 5:50AM
Fetch 2 Tracks (subset):
1799261990, 1739657416, 1786317143, 1623924746, 1635185172, 1574004238, 1198763630, 1621299055, ...
Time: 5:55AM
Fetch 3 Tracks (subset):
1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ...
At 5:50, a materially different history is returned, then it returns back to the prior history at the next poll. I've listened to all of the tracks in each set, but the 5:50 history drops some tracks and returns some from further back in history.
I've connected other accounts and the behavior is consistent and repeatable every day across them. It appears the API is temporarily returning a different (possibly outdated or cached?) view of the user's history during that early morning window.
Has anyone seen this behavior before?
Is this a known issue with the Apple Music API or MusicKit backend? I'd love any insights into what might cause this, or recommendations on how to work around it.
Hi, I’ve been experiencing a strange issue with video playback on my iPhone. While watching videos, the image will suddenly shift it becomes more greyish, then sometimes briefly goes black, and then returns to normal bright quality. This can happen multiple times during a single video.
This is not limited to the Photos app. I’ve seen it happen:
In the Photos app when playing videos I recorded myself
In Snapchat when watching videos sent by others
Occasionally in other social media apps as well
Additional details:
HDR Video is turned off
Apple ProRes is turned off
Tried both 4K 60fps and 4K 30fps
Camera format set to “Most Compatible”
Low Power Mode is off
Issue happens whether the phone is cool or warm
Doesn’t seem related to the video file itself the same file exported to another device looks fine all the way through
These exact same videos play back completely normally on my iPad with no brightness or contrast shifts at all
I’m currently on the iOS 17 public beta, but this issue was happening before I installed the beta as well, so it’s not beta-specific
It almost feels like the display or system is switching between different brightness/contrast profiles mid playback, regardless of the app.
Has anyone else experienced this, and is there a way to disable this behavior so the brightness and color stay consistent during video playback?
Thanks in advance!
I am playing FairPlay + Multi-Key content (fMP4) in Safari browser.
I want to implement the implementation to distinguish between SD and HD video quality, and play it in HD if HDCP is supported, and in SD if HDCP is not supported.
I have already confirmed that HDCP support is the default, and that a black screen is output in non-HDCP environments.
What I want is to improve the user experience by appropriately switching to SD/HD depending on HDCP support when playing DRM content.
Question: Is there an API or function that can detect HDCP support in Safari through JavaScript or other methods? Or is there a way to indirectly guess it?
Topic:
Media Technologies
SubTopic:
Streaming
Tags:
FairPlay Streaming
WebKit
Safari
HTTP Live Streaming
Our Final Cut Pro workflow extension built with ProExtensionHost framework uses an advanced NSPasteboardItemDataProvider system with multi-version FCPXML support (1.9, 1.10, 1.13) and proper relative path
UIDs for Motion templates. We've implemented clip wrapper approach with placeholder assets and elements containing effects to enable direct timeline drag functionality. However, drag
and drop from our Final Cut Pro workflow extension directly to timeline is still not working despite proper element structure in our FCPXML. Our implementation creates valid clip elements with
effects applied, but Final Cut Pro timeline doesn't accept them during drag operations from our ProExtensionHost-based workflow extension.
Steps to Reproduce:
Create Final Cut Pro workflow extension using ProExtensionHost framework with NSPasteboardItemDataProvider implementation
Generate FCPXML with proper element structure:
Expected Result: Clip should be accepted by timeline and effect applied from workflow extension
Actual Result: Timeline rejects drag operation from ProExtensionHost-based workflow extension
Question: Are there additional requirements or ProExtensionHost API calls needed beyond standard NSPasteboardItemDataProvider for Final Cut Pro workflow extension timeline drag functionality?
I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other.
Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?
We have a React website build to scan qr codes. The website is properly working for Android devices but for Iphone we see a camera glitch causing delay in scan which is unexpected.
Website URL : https://react-qr-code-scanner-app.vercel.app/
Topic:
Media Technologies
SubTopic:
Photos & Camera
I'm using AVFoundation to make a multi-track editor app, which can insert multiple track and clip, including scale some clip to change the speed of the clip, (also I'm not sure whether AVFoundation the best choice for me) but after making the scale with scaleTimeRange API, there is some short noise sound in play back. Also, sometimes it's fine when play AVMutableCompostion using AVPlayer with AVPlayerItem, but after exporting with AVAssetReader, will catch some short noise sounds in result file.... Not sure why.
Here is the example project, which can build and run directly. https://github.com/luckysmg/daily_images/raw/refs/heads/main/TestDemo.zip
Since MacOS 26 Apple Music has inconsitent drops to the Quality of some Tracks indiscrimantly. I don't know if others Expereinced it. It doesn't happen on the Speakers or connected via Bluetooth, but the AUX I/O has it quite often. It is more noticable on Headphones with 48kHz and higher Frequency Bandwidth.
Here is the FB18062589
Please consider adding the ability to programatically download Premium and Enhanced voices. At the moment it is extremely inconvenient for our users, as they have to navigate to settings themselves to download voices. Our app relies heavily on SpeechSynthesis integration, and it would greatly benefit from this feature.
FB16307193
We encounter issue with avplayer in case of EXT-X-DISCONTINUITY misalignment between audio and video produced after insertion of gaps.
The initial objective is to introduce an EXT-X-DISCONTINUITY in audio playlist after some missing segments (EXT-X-GAP) which durations are aligned to video segments durations, to handle irregular audio durations.
Please find below an example of corresponding video and audio playlists:
video:
#EXTM3U
#EXT-X-VERSION:7
#EXT-X-MEDIA-SEQUENCE:872524632
#EXT-X-INDEPENDENT-SEGMENTS
#EXT-X-TARGETDURATION:2
#USP-X-TIMESTAMP-MAP:MPEGTS=7096045027,LOCAL=2025-05-09T12:38:32.369100Z
#EXT-X-MAP:URI="hls/StreamingBasic-video=979200.m4s"
#EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.369111Z
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524632.m4s
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524633.m4s
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524634.m4s
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524635.m4s
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524636.m4s
## Media sequence discontinuity
#EXT-X-GAP
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524637.m4s
## Media sequence discontinuity
#EXT-X-GAP
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524638.m4s
#EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.383111Z
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524639.m4s
#EXTINF:2.002, no desc
hls/StreamingBasic-video=979200-872524640.m4s
audio:
EXTM3U
#EXT-X-VERSION:7
#EXT-X-MEDIA-SEQUENCE:872524632
#EXT-X-INDEPENDENT-SEGMENTS
#EXT-X-TARGETDURATION:2
#USP-X-TIMESTAMP-MAP:MPEGTS=7096045867,LOCAL=2025-05-09T12:38:32.378400Z
#EXT-X-MAP:URI="hls/StreamingBasic-audio_99500_eng=98800.m4s"
#EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.378444Z
#EXTINF:2.0053, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524632.m4s
#EXTINF:2.0053, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524633.m4s
#EXTINF:2.0053, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524634.m4s
#EXTINF:1.984, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524635.m4s
#EXTINF:2.0053, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524636.m4s
## Media sequence discontinuity
#EXT-X-GAP
#EXTINF:2.002, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524637.m4s
## Media sequence discontinuity
#EXT-X-GAP
#EXTINF:2.002, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524638.m4s
#EXT-X-DISCONTINUITY
#EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.778444Z
#EXTINF:1.6213, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524639.m4s
#EXTINF:2.0053, no desc
hls/StreamingBasic-audio_99500_eng=98800-872524640.m4s
In this case playback is broken with avplayer.
Is it conformed to Http Live Streaming?
Is it an avplayer bug?
What are the guidelines to handle such gaps?
On Apple TV 4K 3rd generation, with tvOS 26 beta 2, when two HomePod 2 are paired to the device, music and movie sources with Dolby Atmos can only be listened to in stereo. dolby atmos not supported
Topic:
Media Technologies
SubTopic:
Audio
Hello everyone,
I’m new to Swift development and have been working on an audio module that plays a specific sound at regular intervals - similar to a workout timer that signals switching exercises every few minutes.
Following AVFoundation documentation, I’m configuring my audio session like this:
let session = AVAudioSession.sharedInstance()
try session.setCategory(
.playback,
mode: .default,
options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers]
)
self.engine.attach(self.player)
self.engine.connect(self.player, to: self.engine.outputNode, format: self.audioFormat)
try? session.setActive(true)
When it’s time to play cues, I schedule playback on a DispatchQueue:
// scheduleAudio uses DispatchQueue
self.scheduleAudio(at: interval.start) {
do {
try audio.engine.start()
audio.node.play()
for sample in interval.samples {
audio.node.scheduleBuffer(sample.buffer, at: AVAudioTime(hostTime: sample.hostTime))
}
} catch {
print("Audio activation failed: \(error)")
}
}
This works perfectly in the foreground. But once the app goes into the background, the scheduled callback runs, yet the audio engine fails to start, resulting in an error with code 561015905.
Interestingly, if the app is already playing audio before going to the background, the scheduled sounds continue to play as expected.
I have added the required background audio mode to my Info plist file by including the key UIBackgroundModes with the value audio.
Is there anything else I should configure? What is the best practice to play periodic audio when the app runs in the background? How do apps like turn-by-turn navigation handle continuous audio playback in the background?
Any advice or pointers would be greatly appreciated!
Dear Apple Developer Community,
I'm encountering a critical issue with the MusicLibrary.shared.createPlaylist() method in MusicKit that's affecting our app's core functionality. Despite implementing all recommended authorization checks, the app consistently freezes for some users when this method is called.
What we've already verified before calling createPlaylist():
Network connectivity is properly checked and confirmed
Apple Music authorization is explicitly requested via MusicAuthorization.request()
User subscription status is verified using MusicSubscription.current.canPlayCatalogContent
Despite these precautions, many users report that their app completely freezes when attempting to create a playlist. This is particularly concerning as playlist creation is a core feature of our application.
User-reported workarounds (with mixed success):
Some users have resolved the issue by restarting their devices or reinstalling our app
Others report success after enabling "Sync Library" in Settings → Music Unfortunately, a significant number of users are still experiencing the issue even after trying both solutions above
We've reviewed the MusicKit documentation thoroughly and ensured our implementation follows all best practices. Our app correctly handles permissions and uses the async/await pattern as required by the API.
Is there a known issue with the createPlaylist() method that might cause it to block indefinitely? Are there additional authorization steps or settings we should be checking before calling this method? Could this be related to how MusicKit communicates with Apple Music servers?
Any insights from the developer community or official guidance would be greatly appreciated as this issue is severely impacting our user experience.
Thank you for your assistance
This is my native module code implementation
I'm getting base64 encoded string from server and passing this to my native module of pcm player to play audio
App.tsx
PcmPlayer.writeChunk(e.data);
PcmPlayer.swift
import AVFoundation
@objc(PcmPlayer)
class PcmPlayer: RCTEventEmitter {
private var engine: AVAudioEngine?
private var playerNode: AVAudioPlayerNode?
private var format: AVAudioFormat?
private var bufferQueue = [Data]()
private var isPlaying = false
private var hasEnded = false
private var scheduledBufferCount = 0
private let minBufferBytes = 50000
private let pcmQueue = DispatchQueue(label: "pcm.queue")
override init() {
super.init()
}
override func supportedEvents() -> [String]! {
return ["onStatus", "onMessage"]
}
@objc(initPlayer:channels:bitsPerSample:)
func initPlayer(_ sampleRate: NSNumber,
channels: NSNumber,
bitsPerSample: NSNumber) {
pcmQueue.async {
self.stopInternal()
let session = AVAudioSession.sharedInstance()
do {
try session.setCategory(.playback, mode: .default, options: [])
try session.setActive(true, options: .notifyOthersOnDeactivation)
try session.setMode(.default)
print("🔈 Audio session active. Output route:", session.currentRoute.outputs)
} catch {
print("❌ Audio session setup failed:", error)
return
}
self.engine = AVAudioEngine()
self.playerNode = AVAudioPlayerNode()
guard let engine = self.engine, let playerNode = self.playerNode else {
print("❌ Engine or playerNode is nil")
return
}
engine.attach(playerNode)
self.format = AVAudioFormat(commonFormat: .pcmFormatFloat32,
sampleRate: sampleRate.doubleValue,
channels: AVAudioChannelCount(channels.uintValue),
interleaved: false)
guard let format = self.format else {
print("❌ Failed to create AVAudioFormat")
return
}
engine.connect(playerNode, to: engine.mainMixerNode, format: format)
do {
try engine.start()
playerNode.play()
engine.mainMixerNode.outputVolume = 1.0
print("✅ AVAudioEngine started with format:", format)
} catch {
print("❌ Engine start failed:", error)
}
self.hasEnded = false
}
}
@objc(writeChunk:)
func writeChunk(_ base64Pcm: String) {
pcmQueue.async {
guard base64Pcm.count >= 10 else {
print("⚠️ Skipping short base64 string")
return
}
var padded = base64Pcm
let mod4 = base64Pcm.count % 4
if mod4 > 0 {
padded += String(repeating: "=", count: 4 - mod4)
}
guard let data = Data(base64Encoded: padded, options: .ignoreUnknownCharacters) else {
print("❌ Failed to decode base64")
return
}
self.bufferQueue.append(data)
print("📥 Received PCM chunk (\(data.count) bytes)")
print("📥 writeChunk called. isPlaying=\(self.isPlaying), bufferQueue.count=\(self.bufferQueue.count)")
if !self.isPlaying {
self.isPlaying = true
self.waitForBufferAndStartPlayback()
} else if self.scheduledBufferCount == 0 {
self.isPlaying = true
self.waitForBufferAndStartPlayback()
}
}
}
private func waitForBufferAndStartPlayback() {
DispatchQueue.global().async {
while self.queueSize() < self.minBufferBytes && !self.hasEnded {
Thread.sleep(forTimeInterval: 0.01)
}
self.writeLoop()
}
}
private func writeLoop() {
DispatchQueue.global().async {
writeLoop: while self.isPlaying {
if self.bufferQueue.isEmpty {
for _ in 0..<100 {
Thread.sleep(forTimeInterval: 0.01)
if !self.bufferQueue.isEmpty { break }
}
if self.bufferQueue.isEmpty {
print("🔇 No more data to play after waiting")
self.isPlaying = false
break writeLoop
}
}
var data: Data?
self.pcmQueue.sync {
if !self.bufferQueue.isEmpty {
data = self.bufferQueue.removeFirst()
}
}
guard let chunk = data else {
print("⚠️ No data to process")
continue
}
if let buffer = self.pcmBufferFromData(chunk) {
self.scheduledBufferCount += 1
self.playerNode?.scheduleBuffer(buffer, completionHandler: {
self.pcmQueue.async {
self.scheduledBufferCount -= 1
if self.bufferQueue.isEmpty && self.scheduledBufferCount == 0 {
print("ℹ️ Playback idle - waiting for more data")
self.isPlaying = false
}
}
})
}
}
}
}
private func pcmBufferFromData(_ data: Data) -> AVAudioPCMBuffer? {
guard let format = self.format else { return nil }
let frameCount = UInt32(data.count / 2)
guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: frameCount) else {
print("❌ Failed to create AVAudioPCMBuffer")
return nil
}
buffer.frameLength = frameCount
guard let floatChannelData = buffer.floatChannelData?[0] else {
print("❌ floatChannelData is nil")
return nil
}
data.withUnsafeBytes { (rawBuffer: UnsafeRawBufferPointer) in
let int16Buffer = rawBuffer.bindMemory(to: Int16.self)
let count = min(int16Buffer.count, Int(frameCount))
for i in 0..<count {
floatChannelData[i] = Float32(int16Buffer[i]) / Float32(Int16.max)
}
}
return buffer
}
@objc(stopPlayer)
func stopPlayer() {
pcmQueue.async {
self.stopInternal()
}
}
private func stopInternal() {
print("🛑 stopInternal called")
self.playerNode?.stop()
self.engine?.stop()
self.engine?.reset()
self.playerNode = nil
self.engine = nil
self.format = nil
self.bufferQueue.removeAll()
self.isPlaying = false
self.hasEnded = true
self.scheduledBufferCount = 0
}
@objc(canWrite:rejecter:)
func canWrite(_ resolve: @escaping RCTPromiseResolveBlock,
rejecter reject: RCTPromiseRejectBlock) {
pcmQueue.async {
resolve(self.bufferQueue.count < 20)
}
}
@objc(flushPlayer:rejecter:)
func flushPlayer(_ resolve: @escaping RCTPromiseResolveBlock,
rejecter reject: RCTPromiseRejectBlock) {
pcmQueue.async {
self.bufferQueue.removeAll()
resolve(nil)
}
}
@objc
static override func requiresMainQueueSetup() -> Bool {
return false
}
private func queueSize() -> Int {
return pcmQueue.sync {
return self.bufferQueue.reduce(0) { $0 + $1.count }
}
}
}
I couldn't able to hear any audio via my real iOS device also it is working fine on emulator.
Topic:
Media Technologies
SubTopic:
Streaming
Hello. My app uses AVAudioRecorder to generate recording files, which are consistently only 4kb in size. Most users generate audio files normally, with only a few users experiencing this phenomenon occasionally. After uninstalling and installing the app, it will work normally, but it will reappear after a period of time. I have compared that the problematic audio files generated each time are fixed and cannot be played. Added the audioRecorderDidFinishRecording proxy method, which shows that the recording was completed normally. The user also reported that the recording is normal, but there is a problem with the generated file. How should I handle this issue? Look forward to your reply.
- (void)startRecordWithOrderID:(NSString *)orderID {
AVAudioSession *audioSession = [AVAudioSession sharedInstance];
[audioSession setCategory:AVAudioSessionCategoryRecord error:nil];
[audioSession setActive:YES error:nil];
NSMutableDictionary *settings = [[NSMutableDictionary alloc] init];
[settings setObject:[NSNumber numberWithFloat: 8000.0] forKey:AVSampleRateKey];
[settings setObject:[NSNumber numberWithInt: kAudioFormatLinearPCM] forKey:AVFormatIDKey];
[settings setObject:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey];
[settings setObject:[NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey];
[settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey];
[settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey];
NSString *path = [WDUtility createDirInDocument:@"audios" withOrderID:orderID withPathExtension:@"wav"];
NSURL *tmpFile = [NSURL fileURLWithPath:path];
recorder = [[AVAudioRecorder alloc] initWithURL:tmpFile settings:settings error:nil];
[recorder setDelegate:self];
[recorder prepareToRecord];
[recorder record];
}
I'm creating an app that uses AVCaptureSession to pass camera input to AVCaptureMetadataOutput type set [metaout setMetadataObjectTypes:@[AVMetadataObjectTypeFace]] and scan Face.
After updating to OS 26 Beta2 and iOS 26 Beta2, an issue has occurred where the delegate method of AVCaptureMetadataOutputObjectsDelegate is not called on some devices. The following devices are experiencing this issue.
iPad (9th Gen)
iPad air (4th Gen)
iPhone 15
This issue has not occur on any other devices I have.
I tried running the AVFoundation sample code on the Apple Developer site on the above device. The same problem still occurs. https://developer.apple.com/documentation/avfoundation/capture_setup/avcambarcode_detecting_barcodes_and_faces
Are any additional settings required after OS 26 beta and iOS 26 beta? Or is there some problem on the OS side?
Hello, I'm trying to write a shortcut using Toolbox Pro that gets triggered by an accessibility trigger and then favorites the currently playing song. It's working pretty well, but I noticed that for some artists, especially asian ones, it simply doesn't work. While debugging, I noticed that the tool uses the same song ID, artist ID, everything as it should to search for the song and favorite it. However, I noticed that Apple Music treats artists with romanized names as two separate artists!
https://music.apple.com/br/artist/王菲/41760704
https://music.apple.com/br/artist/faye-wong/41760704?l=en-GB
You can see that the ID is the same (41760704). It seems that, when I search for the artist, the first artist (王菲) returns, so that when I open URLs on the web for the artist I can see a star next to the song name, meaning that it got a like. However, the romanized artist (faye-wong) doesn't have a like on the same song.
This is very weird, right?