Pull out the audio mixing code from desktop and add it to core.
This will allow other backends to use it (such as the web audio
backend) to get consistent audio across all platforms.
It was initialized on a separate thread since 6178dd9ef1,
as a workaround for Windows.
But now thanks to https://github.com/RustAudio/cpal/pull/597, this
is no longer needed, and cpal can be initialized on the same thread
as before.
* #[derive(Collect)] should be before #[collect]
* Replace redunant `&buf[..]` with `buf`
* Changes most cases of UPPERCase to UpperCase
* Allow upper_case_acronym on most SWF types, as they are from
SWF spec/more annoying to change.
The points in an SWF sound envelope are always in 44.1KHz. The
desktop audio backend would end up using this points in the sound's
sample rate, often causing the envelope to be delayed.
Now the envelope is applied after resampling, and the positions
are scaled based on the output sample rate.
This unifies the code path for event sounds and stream sounds.
Both `AudioBackend::start_stream` and `start_sound` return a
`SoundHandle`. `stop_stream` is removed (`stop_sound` can be
used for both cases).
Also removes references to `CharacterId` from the `AudioBackend`
(instead, an increasing ID is returned to identiy streams while
loading).
Add an AudioManager struct to handle this list of actively playing
sounds. This will maintain information for each sound instance,
such as the owning display object, AVM1 object, etc.
This will allow us to implement the awkward AVM1 Sound API in a
fairly backend-agnostic way.
sample renamed into dasp. Switch to this crate and also use the
latest git revision, which fixed some compiler errors on nightly.
Also remove unnecessary sample dependency in exporter Cargo.toml.
- removed default implementations for `play()` and `pause()` methods for AudioBackend trait
- Implemented `play()` and `pause()` methods for CPAL audio backend
- Implemented empty block for `play()` and `pause()` methods for NULL audio backend
When using ALSA, cpal was returning bogus sample rates from
`Device::supported_output_configs`, causing the audio stream to
fail to initialize.
Use `Device::default_output_config` instead.
The Windows build was erroring immediately when trying to
initialize cpal because of changes in winit 0.20:
https://github.com/RustAudio/cpal/pull/348
Moved the cpal initialization to a separate thread to work around
this issue.
When a stream sound uses ADPCM compression, the ADPCM header is
included in each SoundStreamBlock (as opposed to stream sounds
in the other formats). This header wasn't being parsed, resulting
in corrupted audio (see https://homestarrunner.com/main12.swf).
Renamed `AudioBackend::play_sound` to `start_sound`, and this
also takes a `SoundInfo` parameter with the event sound settings
from the SWF file.
Desktop now obeys the loop and start/end point settings. Envelopes
are still TODO.
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