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
397 lines
11 KiB
C
397 lines
11 KiB
C
// AudioEngine: the audio class, and /dev/dsp above it.
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//
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// See include/dev/audio.h for what this is and what it deliberately is not. In
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// short: a program opens /dev/dsp, says what its samples are, writes them, and
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// stops. Everything here is either the class vtable's thin wrapper or the part
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// of "play this" that is not specific to a chip.
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#include "core/types.h"
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#include "core/defs.h"
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#include "core/param.h"
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#include "core/spinlock.h"
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#include "core/sleeplock.h"
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#include "fs/fs.h"
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#include "fs/file.h"
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#include "dev/device.h"
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#include "dev/audio.h"
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#include "core/proc.h"
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#include "core/uaccess.h"
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#include "uapi/errno.h"
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static struct spinlock audio_lock;
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static int audio_lock_init;
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// The format the device is currently in. Kept here rather than asked of the
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// driver on every write, because a caller that set 44100 and got 48000 needs
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// to be told once and then have it stay told.
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static struct audio_format current_fmt = { 48000, 2, AUDIO_FMT_S16LE };
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// The OSS view of the stream, for clients that track a play pointer in a ring
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// rather than just writing: bytes accepted by the driver since boot, where the
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// last reset left the count, the ring geometry SETFRAGMENT asked for, and the
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// played count GETOPTR last reported (for its fragment count).
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static uint64 queued_total;
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static uint64 reset_base;
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static uint64 last_optr;
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static int frag_size = 4096;
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static int frag_count = 16;
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static void
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lock_once(void)
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{
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if (!audio_lock_init) {
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initlock(&audio_lock, "audio");
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audio_lock_init = 1;
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}
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}
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// --- the class -------------------------------------------------------------
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struct device *
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audio_get(void)
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{
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return device_find_class(DEVCLASS_AUDIO);
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}
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static const struct audio_ops *
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ops_of(struct device *d)
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{
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return d && d->ops ? (const struct audio_ops *)d->ops : 0;
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}
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int
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audio_configure(struct audio_format *f)
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{
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struct device *d = audio_get();
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const struct audio_ops *o = ops_of(d);
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int r;
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if (!o || !o->configure)
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return -ENODEV;
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if (f->bits != AUDIO_FMT_S16LE)
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return -EINVAL;
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if (f->channels != 1 && f->channels != 2)
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return -EINVAL;
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if (f->rate < 4000 || f->rate > 192000)
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return -EINVAL;
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r = o->configure(d, f);
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if (r < 0)
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return r;
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lock_once();
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acquire(&audio_lock);
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current_fmt = *f; // what the driver settled on, not what was asked
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release(&audio_lock);
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return 0;
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}
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// Queue bytes, blocking while the hardware is full.
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//
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// The blocking is here rather than in a driver because it is the same in every
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// driver: there is a ring, it has room or it does not, and a writer that is
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// ahead of the speaker has nothing useful to do but wait. Sleeping on the
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// device and being woken by its interrupt is the whole of it.
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int
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audio_write(const void *pcm, int bytes)
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{
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struct device *d = audio_get();
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const struct audio_ops *o = ops_of(d);
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const char *p = pcm;
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int done = 0;
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if (!o || !o->queue)
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return -ENODEV;
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if (bytes <= 0)
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return 0;
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while (done < bytes) {
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int n = o->queue(d, p + done, bytes - done);
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if (n < 0)
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return done ? done : n;
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if (n == 0) {
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// Full. Start what is already queued -- a writer that fills the ring
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// and then waits for space would otherwise wait for a stream nobody
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// started -- and give the CPU up until some of it has been played.
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if (o->start)
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o->start(d);
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if (myproc() && myproc()->killed)
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return done ? done : -EINTR;
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audio_wait(d);
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continue;
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}
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done += n;
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lock_once();
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acquire(&audio_lock);
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queued_total += (uint64)n;
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release(&audio_lock);
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}
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if (o->start)
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o->start(d);
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return done;
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}
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// Wait for the hardware to make room. A driver signals this from its
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// interrupt handler with audio_wakeup().
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//
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// The timeout is not a nicety: a driver whose interrupt never arrives -- a
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// misrouted line, a chip that needs a kick -- would otherwise hang the writer
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// for ever with no way to tell what happened. Waking up anyway and asking the
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// hardware again turns that into audio that stutters, which is a bug report
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// somebody can make.
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void
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audio_wait(struct device *d)
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{
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uint t0;
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(void)d;
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// One timer tick, the same way kernel/sys/sys_poll.c waits: there is no
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// timed sleep() in this kernel, and adding one for a single caller would be
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// a scheduler change to avoid a ten-line loop.
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//
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// Sleeping on `ticks` rather than on the device means a driver interrupt is
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// not strictly required to make progress -- which is the point. A ring that
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// drains is a ring that has room, and asking again in ten milliseconds
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// costs one wakeup per buffer.
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acquire(&tickslock);
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t0 = ticks;
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while (ticks == t0 && !(myproc() && myproc()->killed))
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sleep(&ticks, &tickslock);
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release(&tickslock);
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}
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void
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audio_drain(void)
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{
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struct device *d = audio_get();
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const struct audio_ops *o = ops_of(d);
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int spins;
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if (!o || !o->pending)
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return;
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// Bounded, for the same reason the wait above is. Ten seconds of audio is
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// longer than any buffer this engine will hold, so a `pending` that has not
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// reached zero by then is a driver that is not playing rather than one that
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// is busy.
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for (spins = 0; spins < 1000 && o->pending(d) > 0; spins++) {
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if (myproc() && myproc()->killed)
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return;
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audio_wait(d);
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}
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}
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void
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audio_stop(void)
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{
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struct device *d = audio_get();
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const struct audio_ops *o = ops_of(d);
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if (o && o->stop)
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o->stop(d);
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}
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// --- /dev/dsp ---------------------------------------------------------------
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int
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dspwrite(struct inode *ip, uint64 off, char *src, int n)
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{
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(void)ip; (void)off;
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return audio_write(src, n);
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}
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// Recording is not implemented, and says so. Returning zero would look like
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// end-of-file and a recorder would write an empty file and call it a success.
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int
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dspread(struct inode *ip, uint64 off, char *dst, int n)
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{
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(void)ip; (void)off; (void)dst; (void)n;
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return -ENOSYS;
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}
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// The last descriptor is closing. Anything still queued is played out first:
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// a program that writes a sound and exits expects to hear the end of it, and
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// dropping the tail is the one behaviour that would make this device feel
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// broken while technically working.
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void
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dspclose(struct inode *ip)
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{
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(void)ip;
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audio_drain();
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audio_stop();
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}
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// Bytes the speaker has played since boot: everything queued, less what is
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// still waiting. A reset discards the waiting part, which then counts as played
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// -- the count only goes forward, and GETOPTR measures from the reset.
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static uint64
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played_total(struct device *d, const struct audio_ops *o)
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{
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uint64 q, p;
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int pend = o->pending ? o->pending(d) : 0;
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lock_once();
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acquire(&audio_lock);
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q = queued_total;
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release(&audio_lock);
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p = (pend > 0 && (uint64)pend <= q) ? q - (uint64)pend : q;
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return p;
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}
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// Writable once a whole fragment fits, as OSS defines it. ALSA's plugin sets
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// the fragment to its period and waits in poll() for one to be free; reporting
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// writable at the first free byte had it wake, find less than a period of
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// room, and poll again -- 76% of a CPU to play a sine wave.
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int
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dsp_writable(void)
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{
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struct device *d = audio_get();
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const struct audio_ops *o = ops_of(d);
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int want;
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if (!o || !o->space)
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return 0;
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lock_once();
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acquire(&audio_lock);
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want = frag_size;
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release(&audio_lock);
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if (want > 65536)
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want = 65536; // never more than a driver's ring can free at once
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return o->space(d) >= want;
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}
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int
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dsp_ioctl(int request, uint64 argp)
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{
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struct device *d = audio_get();
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const struct audio_ops *o = ops_of(d);
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struct audio_format f;
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int val;
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if (!o)
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return -ENODEV;
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lock_once();
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acquire(&audio_lock);
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f = current_fmt;
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release(&audio_lock);
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switch (request) {
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case SNDCTL_DSP_SPEED:
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case SNDCTL_DSP_CHANNELS:
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case SNDCTL_DSP_SETFMT: {
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if (copy_from_user(&val, argp, sizeof(val)) < 0)
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return -EFAULT;
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if (request == SNDCTL_DSP_SPEED) f.rate = (unsigned)val;
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else if (request == SNDCTL_DSP_CHANNELS) f.channels = (unsigned)val;
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else {
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if (val != AFMT_S16_LE)
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return -EINVAL;
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f.bits = AUDIO_FMT_S16LE;
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}
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int r = audio_configure(&f);
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if (r < 0)
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return r;
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// OSS writes back what was actually set, which is the entire reason these
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// are read/write ioctls: a program asks for 44100, is told 48000, and
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// resamples. One that ignores the answer gets a chipmunk, and that is its
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// bug rather than ours.
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val = (request == SNDCTL_DSP_SPEED) ? (int)f.rate
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: (request == SNDCTL_DSP_CHANNELS) ? (int)f.channels
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: AFMT_S16_LE;
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if (copy_to_user(argp, &val, sizeof(val)) < 0)
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return -EFAULT;
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return 0;
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}
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case SNDCTL_DSP_SYNC:
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audio_drain();
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return 0;
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case SNDCTL_DSP_RESET: {
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audio_stop();
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uint64 p = played_total(d, o);
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acquire(&audio_lock);
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reset_base = last_optr = p;
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release(&audio_lock);
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return 0;
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}
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// The ring the client thinks in. The driver's own ring is whatever it is;
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// these only shape the numbers GETOPTR and GETOSPACE report, which is all a
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// client does with them. Fragments of 256 bytes to 64 KiB, 2 to 1024 of them.
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case SNDCTL_DSP_SETFRAGMENT: {
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if (copy_from_user(&val, argp, sizeof(val)) < 0)
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return -EFAULT;
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int shift = val & 0xffff, count = (val >> 16) & 0xffff;
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if (shift < 8) shift = 8;
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if (shift > 16) shift = 16;
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if (count < 2) count = 2;
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if (count > 1024 || count == 0x7fff) count = 1024;
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acquire(&audio_lock);
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frag_size = 1 << shift;
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frag_count = count;
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release(&audio_lock);
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val = (count << 16) | shift;
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return copy_to_user(argp, &val, sizeof(val)) < 0 ? -EFAULT : 0;
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}
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case SNDCTL_DSP_GETFMTS:
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val = AFMT_S16_LE;
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return copy_to_user(argp, &val, sizeof(val)) < 0 ? -EFAULT : 0;
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case SNDCTL_DSP_GETCAPS:
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val = DSP_CAP_TRIGGER;
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return copy_to_user(argp, &val, sizeof(val)) < 0 ? -EFAULT : 0;
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// Enabling output starts whatever is queued. Disabling is accepted and does
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// nothing: the stream stops by itself when the queue runs dry, and stopping
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// it here would throw away what is queued, which a pause must not.
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case SNDCTL_DSP_SETTRIGGER:
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if (copy_from_user(&val, argp, sizeof(val)) < 0)
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return -EFAULT;
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if ((val & PCM_ENABLE_OUTPUT) && o->start)
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o->start(d);
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return 0;
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case SNDCTL_DSP_GETOPTR: {
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struct count_info ci;
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uint64 p = played_total(d, o);
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acquire(&audio_lock);
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uint64 since = p - reset_base;
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uint64 ring = (uint64)frag_size * (uint64)frag_count;
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ci.bytes = (int)since;
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ci.blocks = (int)((p - last_optr) / (uint64)frag_size);
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ci.ptr = (int)(since % ring);
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last_optr = p;
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release(&audio_lock);
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return copy_to_user(argp, &ci, sizeof(ci)) < 0 ? -EFAULT : 0;
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}
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case SNDCTL_DSP_GETODELAY:
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val = o->pending ? o->pending(d) : 0;
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return copy_to_user(argp, &val, sizeof(val)) < 0 ? -EFAULT : 0;
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case SNDCTL_DSP_GETOSPACE: {
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struct audio_buf_info bi;
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int space = o->space ? o->space(d) : 0;
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bi.fragsize = frag_size;
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bi.bytes = space;
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bi.fragments = space / bi.fragsize;
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bi.fragstotal = frag_count;
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if (copy_to_user(argp, &bi, sizeof(bi)) < 0)
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return -EFAULT;
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return 0;
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}
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}
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return -ENOTTY;
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}
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