RP2040 and SAMD51:
- Detect when DMA has finished, and stop DMA audio explicitly.
- Do not accidentally reuse `first_buffer` supplied by WaveFile or RawSample. Always realloc `first_buffer` and `second_buffer`
RP2040:
- When audio playing is stopped, write a final zero to the output register. This prevents residual PWM tones.
- Handle buffer size for 8-bit samples properly for 16-bit output.
- Fail on some edge cases (which may not be possible at the moment).
Instead of unrolling the code 16 times, unroll it 4 times and loop
over it 4 times. This gives the same 16 iterations, but at an expense
of less flash space.
* No weak link for modules. It only impacts _os and _time and is
already disabled for non-full builds.
* Turn off PA00 and PA01 because they are the crystal on the Metro
M0 Express.
* Change ejected default to false to move it to BSS. It is set on
USB connection anyway.
* Set sinc_filter to const. Doesn't help flash but keeps it out of
RAM.
This introduces the new macro SAM_D5X_E5X. This is mostly the same
as SAMD51 before, except in a few places where a special case for
SAME54 is required
Previously, we depended on allocated channels to always be
"dma_channel_enabled". However, (A) sometimes, many operations
would take place between find_free_audio_dma_channel and
audio_dma_enable_channel, and (B) some debugging I did led me to believe
that "dma_channel_enabled" would become false when the hardware ended
a scheduled DMA transaction, but while a CP object would still think it
owned the DMA channel.
((B) is not documented in the datasheet and I am not 100% convinced that
my debugging session was not simply missing where we were disabling the
channel, but in either case, it shows a need to directly track allocated
separately from enabled)
Therefore,
* Add audio_dma_{allocate,free}_channel.
* audio_dma_free_channel implies audio_dma_disable_channel
* track via a new array audio_dma_allocated[]
* clear all allocated flags on soft-reboot
* Convert find_free_audio_dma_channel to audio_dma_allocate_channel
* use audio_dma_allocated[] instead of dma_channel_enabled() to check
availability
* remove find_free_audio_dma_channel
* For each one, find a matching audio_dma_disable_channel to convert
to audio_dma_free_channel
Closes: #2058
When nrf pwm audio is introduced, it will be called `audiopwmio`. To
enable code sharing with the existing (dac-based) `audioio`, factor
the sample and mixer types to `audiocore`.
INCOMPATIBLE CHANGE: Now, `Mixer`, `RawSample` and `WaveFile` must
be imported from `audiocore`, not `audioio`.
This saves code space in builds which use link-time optimization.
The optimization drops the untranslated strings and replaces them
with a compressed_string_t struct. It can then be decompressed to
a c string.
Builds without LTO work as well but include both untranslated
strings and compressed strings.
This work could be expanded to include QSTRs and loaded strings if
a compress method is added to C. Its tracked in #531.
Its slimmed down by removing the qstr and bit packing TCC info.
The trinket m0 build actually grows by 20 bytes. The arduino zero
build shrinks by 188 bytes.
I2SOut.
The API is almost the same except the frequency attribute has been
renamed to sample_rate so that its less likely to be confused with
frequencies within the audio itself.
Fixes#263.
This evolves the API from 2.x (and breaks it). Playback devices are now
separate from the samples themselves. This allows for greater playback
flexibility. Two sample sources are audioio.RawSample and audioio.WaveFile.
They can both be mono or stereo. They can be output to audioio.AudioOut or
audiobusio.I2SOut.
Internally, the dma tracking has changed from a TC counting block transfers
to an interrupt generated by the block event sent to the EVSYS. This reduces
the overhead of each DMA transfer so multiple can occure without using up TCs.
Fixes#652. Fixes#522. Huge progress on #263
