The EVSYS is used to generate an interrupt from the event. This
simplifies timing used in pulseio that conflicted with the
auto-reload countdown.
Fixes#3890
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).
With the previous change, stereo mp3 playback changed from needing
4 2304-byte allocations to needing 2 4604-byte allocations. This was
enough to cause MemoryErrors with regularity.
By using m_realloc() here, the existing memory region can be used.
m_realloc() also works on the first invocation, because m_realloc(NULL, sz)
just calls m_malloc of sz.
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
Some ports which actually don't have audioio or audiobusio were still
calling into audio_dma_background(). This wasn't an error until
the assignment to audio_dma_state in audio_dma_stop was added, though
it's not clear why.
audio_dma_stop can be reached twice in normal usage of AudioOut.
This may bear further investigation, but stop it here, by making the
function check for a previously freed channel number. This also prevents
the event channel from being disabled twice.
The first stop location is from audio_dma_get_playing, when the buffers
are exhausted; the second is from common_hal_audioio_audioout_stop when
checking the 'playing' flag.
As identified in #1908, when both AudioOut and PDMIn are used, hard
locks can occur. Because audio_dma_stop didn't clear audio_dma_state[],
a future call to audio_dma_load_next_block could occur using a DMA
object which belongs to PDMIn.
I believe that this Closes: #1908 though perhaps it is still not the full
story.
Testing performed: Loaded a sketch similar to the one on #1908 that
tends to reproduce the bug within ~30s. Ran for >300s without hard
lock. HOWEVER, while my cpx is no longer hard locking, it occasionally
(<1 / 200s) announces
Code done running. Waiting for reload.
(and does so), even though my main loop is surrounded by a 'while True:'
condition, so there are still gremlins nearby.
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.
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
