Player sometime never leaves STKAudioPlayerStateBuffering state

[zsaraf]

I'm using this library to build a radio iOS app that has a directory of Icecast streams that the user can click on and listen to.

This library works very well 95% of the time, but 5% of the time, the player gets stuck in the STKAudioPlayerStateBuffering state and never changes, not reporting any errors at all. It seems like some kind of a race condition. Any idea why this could be happening?

- (BOOL)startAudioIO
{
    NSError *error = nil;
    
    AVAudioSessionCategory category = AVAudioSessionCategoryPlayAndRecord;
    AVAudioSessionCategoryOptions options = AVAudioSessionCategoryOptionAllowBluetooth | AVAudioSessionCategoryOptionDefaultToSpeaker;
    if ([AVAudioSession sharedInstance].category != category || [AVAudioSession sharedInstance].categoryOptions != options) {
        [[AVAudioSession sharedInstance] setCategory:category withOptions:options error:&error];
        if (error) {
            NSLog(@"STKAudioPlayer: Error setting category on audio session.");
            return NO;
        }
    }
    
    [[AVAudioSession sharedInstance] setActive:YES error:&error];
    if (error) {
        NSLog(@"STKAudioPlayer: Error starting audio session.");
        return NO;
    }
    
    return YES;
}

- (void)setupPlayer
{
    STKAudioPlayerOptions options;
    options.bufferSizeInSeconds = 2;

    _audioPlayer = [[STKAudioPlayer alloc] initWithOptions:options];
    _audioPlayer.delegate = self;
    
    [_audioPlayer appendFrameFilterWithName:@"FFT" block:^(UInt32 channelsPerFrame, UInt32 bytesPerFrame, UInt32 frameCount, void* frames)
     {
         uint16_t bytesPerSample = bytesPerFrame / channelsPerFrame;
         
         if (bytesPerSample == 2) {
             vDSP_vflt16((short *)frames, 1, self->_floatInputBuffer, 1, frameCount * channelsPerFrame);
         } else if (bytesPerSample == 4) {
             vDSP_vflt32((int *)frames, 1, self->_floatInputBuffer, 1, frameCount * channelsPerFrame);
         } else {
             NSLog(@"STKAudioPlayer: Other bytePerSample sizes are not supported. TODO(zsaraf): Test for this now!");
         }
         
         // 2^(n-1) because it is a signed number where n = 16 (short *)
         const float nBits = pow(2, (bytesPerSample * 8) - 1);
         vDSP_vsdiv(self->_floatInputBuffer, 1, &nBits, self->_floatNormalizedInputBuffer, 1, channelsPerFrame * frameCount);
         
         [[DFFTManager shared] processFFTWithInputBuffers:self->_fftArgBuffer inputChannels:channelsPerFrame numberOfSamples:frameCount interleaved:YES];
     }];
}

- (void)startStreamWithUrl:(NSString *)streamUrl
                streamName:(NSString *)streamName
               streamImage:(NSString *)streamImage
     streamCreatorUsername:(NSString *)streamCreatorUsername
{
    // Set values in case we get disconnected and need to reconnect.
    self->_streamUrl = streamUrl;
    self->_streamName = streamName;
    self->_streamImage = streamImage;
    self->_streamCreatorUsername = streamCreatorUsername;
    
    // Begin stream.
    dispatch_async(dispatch_get_main_queue(), ^{
        [self startAudioIO];
        
        if (self->_audioPlayer == nil) {
            [self setupPlayer];
        }
        
        [self->_audioPlayer play:streamUrl];
    });
}

- (void)stopStream
{
    dispatch_async(dispatch_get_main_queue(), ^{
        [self->_audioPlayer stop];
        
        [self cleanupStream];
   });
}

Any ideas at all on where to look?

Thanks Zach

[zsaraf]

I've found that this error comes up because of issues with starting the AUGraph. AUGraphStart gets called and returns a successful status of 0. BUT the OutputRenderCallback is not being called.

Any ideas why this could be?

[keba76]

Hi! Between finish and start stream put a mini delay for example in 0.1 - 0.2 seconds.

[zsaraf]

This hasn't helped at all unfortunately! The AUGraphStart call is not actually starting the graph/callback functions. Does anyone have any idea how to fix this?

[keba76]

In your method - (void)setupPlayer, try setting bufferSizeInSeconds to 10 seconds instead of 2 seconds.

[keba76]

As one of the options for the occurrence of such a problem, there may be initialization of AudioSession and Remote Control. These actions must be performed before the player is first initialized and preferably as early as possible.

[grzegorzkrukowski]

We are experience similar issues - it's for iOS 13 only and we are able to reproduce it when playing audios on 3G - buffering seems to be broken for both foreground and background playing. Sometimes StreamingKit goes into Buffering (reporting state change by delegate) and never recovers from that - it's also not reporting any errors for that case. Lot's of users reporting that problem lately - and we didn't change anything within our code around this - it has been just fine for a long long time.

[iDevelopper]

Is this issue relative to the #398 one?

[grzegorzkrukowski]

No - we are playing from valid URLa, but on crappy 3G connection buffering sometimes just does not resume playing and StreamingKit stucks after usually 2-3 minutes of playing - it goes to buffering and never recovers from there even by calling “play” again.

Sent from my iPhone

On 23 Nov 2019, at 08:06, Patrick [email protected] wrote:

 Is this issue relative to the #398 one?

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