I've been wanting to do this for a long time. It's time we start being
consistent about how this stuff works.
The new convention is:
- "LibFoo" is a userspace library that provides the "Foo" namespace.
That's it :^) This was pretty tedious to convert and I didn't even
start on LibGUI yet. But it's coming up next.
As suggested by Joshua, this commit adds the 2-clause BSD license as a
comment block to the top of every source file.
For the first pass, I've just added myself for simplicity. I encourage
everyone to add themselves as copyright holders of any file they've
added or modified in some significant way. If I've added myself in
error somewhere, feel free to replace it with the appropriate copyright
holder instead.
Going forward, all new source files should include a license header.
Use a pthread_cond_t to have the ASMixer thread wait until a client has
connected and added a buffer queue to the "pending mixing" vector.
This solves the long-standing issue of the system "idling" at ~8% CPU.
This patch adds muting to ASMixer, which works by substituting what we
would normally send to the sound card with zero-filled memory instead.
We do it this way to ensure that the queued sample buffers keep getting
played (silently.)
This is obviously not the perfect way of doing this, and in the future
we should improve on this, and also find a way to utilize any hardware
mixing functions in the sound card.
Give the mixer a main volume value (percent) that we scale all the
outgoing samples by (before clipping.)
Also add a simple "avol" program for querying and setting the volume:
- "avol" prints the current volume.
- "avol 200" sets the main mix volume to 200%
Each client connection now sets up an ASBufferQueue, which is basically a
queue of ABuffers. This allows us to immediately start streaming the next
pending buffer whenever our current buffer runs out of samples.
This makes the majority of the skippiness go away for me. :^)
Also get rid of the old PlayBuffer API, since we don't need it anymore.
This allows us to carry the same buffer all the way from the WAV loader
to the AudioServer mixer.
This alleviates some of the stutter, but there's still a noticeable
skip when switching buffers. We're gonna need to do better. :^)
I had to solve a bunch of things simultaneously to make this work.
Refactor AWavLoader to be a streaming loader rather than a one-shot one.
The constructor parses the header, and if everything looks good, you can
repeatedly ask the AWavLoader for sample buffers until it runs out.
Also send a message from AudioServer when a buffer has finished playing.
That allows us to implement a blocking variant of play().
Use all of this in aplay to play WAV files chunk-at-a-time.
This is definitely not perfect and it's a little glitchy and skippy,
but I think it's a step in the right direction.
The center of this is now an ABuffer class in LibAudio.
ABuffer contains ASample, which has two channels (left/right) in
floating point for mixing purposes, in 44100hz.
This means that the loaders (AWavLoader in this case) needs to do some
manipulation to get things in the right format, but that we don't need
to care after format loading is done.
While we're at it, do some correctness fixes. PCM data is unsigned if
it's 8 bit, but 16 bit is signed. And /dev/audio also wants signed 16
bit audio, so give it what it wants.
On top of this, AudioServer now accepts requests to play a buffer.
The IPC mechanism here is pretty much a 1:1 copy-paste from
LibGUI/WindowServer. It can be generalized more in the future, but for
now I want to get AudioServer working decently first :)
Additionally, add a little "aplay" tool to load and play a WAV file. It
will break with large WAVs (run out of memory, heh...) but it's a start.
Future work needs to make AudioServer block buffer submission from
clients until it has played the buffer they are requesting to play.
