This removes downscaling (halving-add) when multiple voices are
being mixed. To avoid clipping, and get similar behavior to before,
set the "level" of each voice to (1/voice_count).
Slow paths that were applicable to only M0 chips were removed.
As a side effect, the internal volume representation is now 0 ..
0x8000 (inclusive), which additionally makes a level of exactly 0.5
representable.
Testing performed, on PyGamer: For all 4 data cases, for stereo and
mono, for 1 and 2 voices, play pure sign waves represented as
RawSamples and view the result on a scope and through headphones.
Also, scope the amount of time spent in background tasks.
Code size: growth of +272 bytes
Performance (time in background task when mixing 2 stereo 16-bit voices):
76us per down from 135us (once per ~2.9ms long term average)
(Decrease from 4.7% to 2.4% of all CPU time)
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.
In conversion I missed these arguments were being passed, but noticed
it when an implausible value for audio_channel was sent to mp3's
reset_buffer method.
It's not clear whether there was any negative impact to this, but it
should be fixed!
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.
We weren't correctly collecting the start and stop sequences. As
a result, the GC would free the space and allocate other info
there.
Thanks to JacobT on Discord for the bug report!
