audio: /dev/dsp, AudioEngine, and an AC'97 under it
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>
This commit is contained in:
parent
3eb2e97f7b
commit
9fea18b18b
18 changed files with 1110 additions and 5 deletions
284
kernel/dev/audio.c
Normal file
284
kernel/dev/audio.c
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@ -0,0 +1,284 @@
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// AudioEngine: the audio class, and /dev/dsp above it.
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//
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// See include/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 "types.h"
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#include "defs.h"
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#include "param.h"
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#include "spinlock.h"
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#include "sleeplock.h"
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#include "fs.h"
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#include "file.h"
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#include "device.h"
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#include "audio.h"
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#include "proc.h"
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#include "uaccess.h"
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#include "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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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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}
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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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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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return 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 = 4096;
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bi.bytes = space;
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bi.fragments = space / bi.fragsize;
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bi.fragstotal = 0;
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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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@ -211,4 +211,8 @@ struct devsw devsw[] = {
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// reverse of major 9: that hands a device to a process, this lets a process
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// be one. See include/netuser.h.
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[11] = { netuser_read, netuser_write, netuser_ready },
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// /dev/dsp (major 12) -- UNIX audio, kernel/dev/audio.c. Write PCM to it.
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// The close hook plays out what is still queued: a program that writes a
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// sound and exits expects to hear the end of it.
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[12] = { dspread, dspwrite, 0, dspclose },
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};
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@ -40,6 +40,7 @@ static const char *class_names[DEVCLASS_NCLASS] = {
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[DEVCLASS_STORAGE] = "storage",
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[DEVCLASS_NET] = "net",
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[DEVCLASS_BUS] = "bus",
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[DEVCLASS_AUDIO] = "audio",
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};
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const char *
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@ -365,6 +365,7 @@ pci_class_of(const struct pci_dev *d)
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case PCI_CLASS_STORAGE: return DEVCLASS_STORAGE;
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case PCI_CLASS_NETWORK: return DEVCLASS_NET;
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case PCI_CLASS_SERIAL_BUS: return DEVCLASS_BUS;
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case PCI_CLASS_MULTIMEDIA: return DEVCLASS_AUDIO;
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default: return DEVCLASS_OTHER;
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}
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}
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