The old formulation
* wouldn't work if there were ID3 tags at the end
* would choose whether to background-refill the inbuf
based on a check before skipping to the next sync word, which
could be incorrect.
I think it was aspect "B" that ended up triggering the erroneous EOF
problem fixed in the prior commit. This would depend on specific data
sizes and offsets occuring in the file such that a read would be
scheduled but then the buffer would be filled and left 100% full by
find_sync_word(). It's just lucky(?) that a particular person produced
such a file, and/or many files produced by Audacity have those
characteristics.
Some audio implementations, notably samd, really don't like it when
you return 0 samples of data. This was the case when reaching the
end of an MP3 file.
Now, we read forward in an MP3 file to the next sync word during
"get_buffer", so that we can accurately return GET_BUFFER_DONE when the
NEXT call WOULD HAVE resulted in 0 samples.
Tested with @gamblor21's "laugh.mp3" file on a Trellis M4 Express.
In testing, I saw that the decoded_samples value kept increasing when I
stopped and restarted playback, as I'd missed setting it back to zero
during the reset operation.
In my testing, there is no way to accurately know how far into a MP3 file
you're currently playing. You can use monotonic time, but that can have
drift versus the audio playback system, which may not be running at exactly
the expected sample rate.
To allow syncing animation with timestamps in a MP3 file, this presents a
new property, decoded_samples, that records the number of audio samples
sent out of the decoder. While this may not be a completely accurate time,
due to mixer delays, it's much better position that the monotonic clock
difference.
Implementation is keeping track of this value in the mp3file structure and
adding to it whenever data is sent out of the decoder. The property
implementation was a copy/paste from current properties in the audiomp3
files.
Before this, the mp3 file would be read into the in-memory buffer
only when new samples were actually needed. This meant that the time
to read mp3 content always counted against the ~22ms audio buffer length.
Now, when there's at least 1 full disk block of free space in the input
buffer, we can request that the buffer be filled _after_ returning from
audiomp3_mp3file_get_buffer and actually filling the DMA pointers. In
this way, the time taken for reading MP3 data from flash/SD is less
likely to cause an underrun of audio DMA.
The existing calls to fill the inbuf remain, but in most cases during
streaming these become no-ops because the buffer will be over half full.
After adding the ability to change files in an existing MP3File object,
it became apparent that at the beginning of a track some part of an
existing buffer was playing first.
I noticed that in get_buffer, the just-populated buffer wasn't being
returned, but the other one was. But still after fixing this, I heard
wrong audio at the beginning of a track, so I took the heavy duty approach
and zeroed the buffers out. That means there's a remaining bug to chase,
which is merely hidden by the memset()s.
This enables jeplayer to allocate just one MP3File at startup, rather
than have to make repeated large allocations while the application is
running.
The buffers have to be allocated their theoretical maximum, but that
doesn't matter much as all the real-life MP3 files I checked needed
that much allocation anyway.
Apparently sometimes, a proper "frame info" block is not found after
a "sync word". Keep looking for one as needed, instead of giving up
after one try.
This was one reason that the "bartlebeats" mp3s would not play.
There were several problems with the way this worked -- the read_count
approach was too complicated and I made a mistake "simplifying" it from
WaveFile. And when the right channel was returned, it was off by 1 byte,
making it into static.
Instead, directly track which is the "other" channel that has data
available, and by using the right data type make the "+ channel"
arithmetic give the right result.
This requires a double cast (int16_t*)(void*) due to an alignment warning;
the alignment is now ensured manually, but the compiler doesn't make the
necessary inference that the low address bit must be clear.
When a playing mp3 is deinitted, it's possible to reach get_buffer,
but all the internal pointers are NULL. This would lead to a hard fault.
Avoid it by returning GET_BUFFER_ERROR instead.
