How to remove static/skipping at the beginning of songs?

[cinderblockgames]

I know pops have been brought up with the advice to add silence, but I don't think this is the same thing. It's a harsh static: https://files.kga.earth/?file=kevin/Audio/VID_20251209_110257.mp4

Playing these same songs through a VS1053 does not have the same issue, so I'm hoping there's something I'm missing in the code to resolve it.

Code can be found here: https://github.com/cinderblockgames/uart-audio-module/tree/main/UART_Audio_Module
It's relatively simple and currently only supports MP3s because I haven't been able to get any other formats working yet. (If it ends up being MP3 only, that's fine.)

EDIT: Using a XIAO RP2040 + Adafruit Audio BFF. The VS1053 comparison setup is an Adafruit Feather RP2040 + Music Maker FeatherWing .

EDIT2: I did already increase the MP3_MAX_OUTPUT_SIZE and MP3_MAX_FRAME_SIZE per our earlier conversation, but the static happens on both 320k and 160k files, and increasing them even more doesn't make a difference.

EDIT3: Adding decoder.end(); in cleanUp and decoder.begin(); in startPlaying before copier.begin(decoder, audioFile); doesn't resolve it.

EDIT4: When it's skipping, I get this repeated on each stutter:

[I] AudioTypes.h : 126 - MP3DecoderHelix sample_rate: 24000 / channels: 1 / bits_per_sample: 16
[I] I2SStream.h : 96 - virtual void audio_tools::I2SStream::setAudioInfo(audio_tools::AudioInfo)
[I] AudioTypes.h : 126 - in: sample_rate: 24000 / channels: 1 / bits_per_sample: 16
[I] AudioTypes.h : 126 - out: sample_rate: 24000 / channels: 1 / bits_per_sample: 16
[I] AudioTypes.h : 126 - in: sample_rate: 24000 / channels: 1 / bits_per_sample: 16
[I] AudioTypes.h : 126 - out: sample_rate: 44100 / channels: 2 / bits_per_sample: 16
...
[I] AudioTypes.h : 126 - MP3DecoderHelix sample_rate: 16000 / channels: 2 / bits_per_sample: 16
[I] I2SStream.h : 96 - virtual void audio_tools::I2SStream::setAudioInfo(audio_tools::AudioInfo)
[I] AudioTypes.h : 126 - in: sample_rate: 16000 / channels: 2 / bits_per_sample: 16
[I] AudioTypes.h : 126 - out: sample_rate: 16000 / channels: 2 / bits_per_sample: 16
[I] AudioTypes.h : 126 - in: sample_rate: 16000 / channels: 2 / bits_per_sample: 16
[I] AudioTypes.h : 126 - out: sample_rate: 24000 / channels: 1 / bits_per_sample: 16

So it looks like it's taking a bit to settle into a working AudioInfo. Is there a way to force it so it doesn't bounce around? And would increasing the StreamCopy buffer size to larger than 1024 bytes help?

EDIT5: Tried #define USE_RESAMPLE_BUFFER false but that didn't prevent it.