SDL 3 has no OSS backend and SchOS has no ALSA kernel side, so sound now
goes SDL -> alsa-lib -> alsa-plugins' OSS plugin -> /dev/dsp. That path
found two bugs underneath it:
- ac97: resetting the PCM-out box also clears the descriptor list address,
and nothing set it again. Every stream after the first stop played a
"list" at physical address 0 -- silence -- and ALSA resets before each
stream. If the engine halts at the end of the queue, the run bit is set
again.
- /dev/dsp: SETFRAGMENT, GETFMTS, GETCAPS, SETTRIGGER, GETOPTR and GETODELAY
for the plugin's ring-and-pointer model, and poll(), writable once a whole
fragment fits (ALSA spun at 76% CPU otherwise). speaker-test now plays in
real time at 0% CPU, verified in a recording of QEMU's AC'97.
- /etc/asound.conf: the default device is a plug to the OSS device at 16-bit
stereo 48 kHz, what every SchOS audio driver plays.
Also:
- getstackpcs: each frame link must go up the same stack by less than
KSTACKSIZE. The chain ends at the syscall entry with the user's rbp,
which lbreakouthd uses as a register (-1024); the walk dereferenced it
under the syscall tracer and the kernel panicked inside acquire().
- fb: FBIOGET_VSCREENINFO reports nominal timings, as Linux's efifb does,
instead of zeros, which Xorg's fbdev turned into a 0 Hz RandR mode.
- procdump (Ctrl-P) shows where runnable threads are in user space.
- World: lbreakouthd instead of lbreakout2 (its stk.c passes vsnprintf a
size of 511 plus the prefix length, and Alpine's fortified headers trap on
that), plus curl, alsa-lib and alsa-plugins.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01S1giNvrHTrPMPw7V6x1kQ4
Sixty-five headers sat side by side in include/, so `#include "irq.h"`
said nothing about where irq.h was or what it belonged to, and errno.h,
stat.h and fcntl.h at the top shadowed the system's for any host tool
built with -I./include. They are in folders now -- arch/ core/ fs/
dev/ gfx/ net/ uapi/ (and ldls/ for the Linux driver layer) -- and
every include names the folder, as Linux's do: "core/proc.h".
Done mechanically and checked the hard way: the kernel built from this
tree disassembles identically to the one built from the tree before
it. The full image boots, and userland is the same set of programs.
Paths in docs and comments follow; docs/guides/layout.md has a table
of what is where, and tools/fix-includes.py rewrites old-style
includes from the folders in the tree, for code written before this.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01S1giNvrHTrPMPw7V6x1kQ4
A program that wants to make a sound opens /dev/dsp, writes PCM, and closes
it. That is what UNIX audio has always been and it now exists here:
tone 440 2
Three pieces, in the order a sample passes through them. **/dev/dsp** is a
character device where write() is audio and a handful of OSS ioctls say what
the samples are. **AudioEngine** is everything above a driver that is not
specific to a chip -- the format, whether there is room, and what a write does
when there is not. **A driver** is bound by class, exactly as a display driver
is: device_find_class(DEVCLASS_AUDIO).
OSS numbering rather than something new, because every UNIX audio program ever
written already knows it and the only thing a different set would achieve is
making ports harder. They are read/write ioctls, which is the part that
matters: a program asks for 44100, the device sets 48000 and writes 48000
back, the program resamples. One that ignores the answer plays 9% fast, and
that is its bug rather than the driver's.
**The engine is deliberately one stream.** Mixing several programs is a real
feature with a real design behind it, and building it before anything had
played a single sound would be exactly what docs/plan/1.0.md keeps a register
for: work that exists and is connected to nothing. The second program that
wants to play at the same time is what should cause it to be written.
**audio_wait() sleeps on the timer tick, not on the device.** That looks like
the lazy choice and is not: a ring that drains is a ring with room, so asking
again in ten milliseconds always makes progress, whereas sleeping on the card
makes audio depend on an interrupt arriving. On a machine whose line is
misrouted that is not a stutter, it is a writer that never wakes. The handler
still accounts for finished buffers -- deleting it would cost efficiency, not
correctness -- and it retires descriptors through the same `tail` pointer
queue() uses, so the two cannot disagree about what is in flight.
**AC'97 first because HDA is two unproven things at once.** It is the simplest
audio device that exists on real machines and in QEMU both. Intel HDA is what
the laptop actually has and is a much larger driver; writing it against an
interface nothing had ever played through would have meant debugging the
driver and the interface simultaneously. HDA arrives as a second
struct audio_ops and nothing above it changes -- which is the entire point of
the class.
Two things the wiring needed: PCI class 0x04 now maps to DEVCLASS_AUDIO (the
driver bound and /proc/devices showed it, but audio_get() found nothing,
because the class byte had never been translated), and include/x86.h grew
inw() -- AC'97's mixer is all 16-bit registers and reading one a byte at a
time returns the low half twice on some codecs.
**Tested by measuring what came out, not by not crashing.** QEMU's wav backend
writes the guest's audio to a file; `tone 440 2` produced 1.97 seconds of
44.1 kHz stereo with peak amplitude 14177 against the generator's table
maximum of 14178, and 880 zero crossings per second -- 440 Hz exactly. wav is
on every QEMU target rather than a host backend, because a build machine may
have no sound card and it is the only backend whose output can be checked.
Both supported configurations: one boot each, no traps, no panics, socktest
and memfdtest green.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
