Sound through ALSA, a 2D game, and a stack walk that cannot panic

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

include/dev/audio.h

@@ -79,6 +79,7 @@ int  audio_configure(struct audio_format *f);   // 0, or negative
 int  audio_write(const void *pcm, int bytes);   // bytes queued, or negative
 void audio_drain(void);
 void audio_stop(void);
+int  dsp_writable(void);                        // poll(): room for a write
 
 // --- what userspace uses ---------------------------------------------------
 //
@@ -92,6 +93,23 @@ void audio_stop(void);
 #define SNDCTL_DSP_SYNC       0x00005001   // wait for the queue to drain
 #define SNDCTL_DSP_GETOSPACE  0x8010500C   // struct audio_buf_info out
 #define SNDCTL_DSP_RESET      0x00005000   // stop, discard
+// The rest of what a buffer-and-pointer client needs -- ALSA's OSS plugin, which
+// is how SDL and everything else that speaks ALSA reaches this device.
+#define SNDCTL_DSP_SETFRAGMENT 0xC004500A  // int in/out: count<<16 | log2(size)
+#define SNDCTL_DSP_GETFMTS    0x8004500B   // int out: AFMT_* mask
+#define SNDCTL_DSP_GETCAPS    0x8004500F   // int out: DSP_CAP_*
+#define SNDCTL_DSP_SETTRIGGER 0x40045010   // int in: PCM_ENABLE_OUTPUT or 0
+#define SNDCTL_DSP_GETOPTR    0x800C5012   // struct count_info out
+#define SNDCTL_DSP_GETODELAY  0x80045017   // int out: bytes not yet heard
+
+#define DSP_CAP_TRIGGER       0x00001000
+#define PCM_ENABLE_OUTPUT     0x00000002
+
+struct count_info {
+  int bytes;          // played since the last reset
+  int blocks;         // fragments played since the last GETOPTR
+  int ptr;            // where playback is in a ring of fragsize * fragments
+};
 
 #define AFMT_S16_LE           0x00000010