dmasound_core.c

linux 内核源代码

/*
 *  linux/sound/oss/dmasound/dmasound_core.c
 *
 *
 *  OSS/Free compatible Atari TT/Falcon and Amiga DMA sound driver for
 *  Linux/m68k
 *  Extended to support Power Macintosh for Linux/ppc by Paul Mackerras
 *
 *  (c) 1995 by Michael Schlueter & Michael Marte
 *
 *  Michael Schlueter ([e-mail]) did the basic structure of the VFS
 *  interface and the u-law to signed byte conversion.
 *
 *  Michael Marte ([e-mail]) did the sound queue,
 *  /dev/mixer, /dev/sndstat and complemented the VFS interface. He would like
 *  to thank:
 *    - Michael Schlueter for initial ideas and documentation on the MFP and
 *	the DMA sound hardware.
 *    - Therapy? for their CD 'Troublegum' which really made me rock.
 *
 *  /dev/sndstat is based on code by Hannu Savolainen, the author of the
 *  VoxWare family of drivers.
 *
 *  This file is subject to the terms and conditions of the GNU General Public
 *  License.  See the file COPYING in the main directory of this archive
 *  for more details.
 *
 *  History:
 *
 *	1995/8/25	First release
 *
 *	1995/9/02	Roman Hodek:
 *			  - Fixed atari_stram_alloc() call, the timer
 *			    programming and several race conditions
 *	1995/9/14	Roman Hodek:
 *			  - After some discussion with Michael Schlueter,
 *			    revised the interrupt disabling
 *			  - Slightly speeded up U8->S8 translation by using
 *			    long operations where possible
 *			  - Added 4:3 interpolation for /dev/audio
 *
 *	1995/9/20	Torsten Scherer:
 *			  - Fixed a bug in sq_write and changed /dev/audio
 *			    converting to play at 12517Hz instead of 6258Hz.
 *
 *	1995/9/23	Torsten Scherer:
 *			  - Changed sq_interrupt() and sq_play() to pre-program
 *			    the DMA for another frame while there's still one
 *			    running. This allows the IRQ response to be
 *			    arbitrarily delayed and playing will still continue.
 *
 *	1995/10/14	Guenther Kelleter, Torsten Scherer:
 *			  - Better support for Falcon audio (the Falcon doesn't
 *			    raise an IRQ at the end of a frame, but at the
 *			    beginning instead!). uses 'if (codec_dma)' in lots
 *			    of places to simply switch between Falcon and TT
 *			    code.
 *
 *	1995/11/06	Torsten Scherer:
 *			  - Started introducing a hardware abstraction scheme
 *			    (may perhaps also serve for Amigas?)
 *			  - Can now play samples at almost all frequencies by
 *			    means of a more generalized expand routine
 *			  - Takes a good deal of care to cut data only at
 *			    sample sizes
 *			  - Buffer size is now a kernel runtime option
 *			  - Implemented fsync() & several minor improvements
 *			Guenther Kelleter:
 *			  - Useful hints and bug fixes
 *			  - Cross-checked it for Falcons
 *
 *	1996/3/9	Geert Uytterhoeven:
 *			  - Support added for Amiga, A-law, 16-bit little
 *			    endian.
 *			  - Unification to drivers/sound/dmasound.c.
 *
 *	1996/4/6	Martin Mitchell:
 *			  - Updated to 1.3 kernel.
 *
 *	1996/6/13       Topi Kanerva:
 *			  - Fixed things that were broken (mainly the amiga
 *			    14-bit routines)
 *			  - /dev/sndstat shows now the real hardware frequency
 *			  - The lowpass filter is disabled by default now
 *
 *	1996/9/25	Geert Uytterhoeven:
 *			  - Modularization
 *
 *	1998/6/10	Andreas Schwab:
 *			  - Converted to use sound_core
 *
 *	1999/12/28	Richard Zidlicky:
 *			  - Added support for Q40
 *
 *	2000/2/27	Geert Uytterhoeven:
 *			  - Clean up and split the code into 4 parts:
 *			      o dmasound_core: machine-independent code
 *			      o dmasound_atari: Atari TT and Falcon support
 *			      o dmasound_awacs: Apple PowerMac support
 *			      o dmasound_paula: Amiga support
 *
 *	2000/3/25	Geert Uytterhoeven:
 *			  - Integration of dmasound_q40
 *			  - Small clean ups
 *
 *	2001/01/26 [1.0] Iain Sandoe
 *			  - make /dev/sndstat show revision & edition info.
 *			  - since dmasound.mach.sq_setup() can fail on pmac
 *			    its type has been changed to int and the returns
 *			    are checked.
 *		   [1.1]  - stop missing translations from being called.
 *	2001/02/08 [1.2]  - remove unused translation tables & move machine-
 *			    specific tables to low-level.
 *			  - return correct info. for SNDCTL_DSP_GETFMTS.
 *		   [1.3]  - implement SNDCTL_DSP_GETCAPS fully.
 *		   [1.4]  - make /dev/sndstat text length usage deterministic.
 *			  - make /dev/sndstat call to low-level
 *			    dmasound.mach.state_info() pass max space to ll driver.
 *			  - tidy startup banners and output info.
 *		   [1.5]  - tidy up a little (removed some unused #defines in
 *			    dmasound.h)
 *			  - fix up HAS_RECORD conditionalisation.
 *			  - add record code in places it is missing...
 *			  - change buf-sizes to bytes to allow < 1kb for pmac
 *			    if user param entry is < 256 the value is taken to
 *			    be in kb > 256 is taken to be in bytes.
 *			  - make default buff/frag params conditional on
 *			    machine to allow smaller values for pmac.
 *			  - made the ioctls, read & write comply with the OSS
 *			    rules on setting params.
 *			  - added parsing of _setup() params for record.
 *	2001/04/04 [1.6]  - fix bug where sample rates higher than maximum were
 *			    being reported as OK.
 *			  - fix open() to return -EBUSY as per OSS doc. when
 *			    audio is in use - this is independent of O_NOBLOCK.
 *			  - fix bug where SNDCTL_DSP_POST was blocking.
 */

 /* Record capability notes 30/01/2001:
  * At present these observations apply only to pmac LL driver (the only one
  * that can do record, at present).  However, if other LL drivers for machines
  * with record are added they may apply.
  *
  * The fragment parameters for the record and play channels are separate.
  * However, if the driver is opened O_RDWR there is no way (in the current OSS
  * API) to specify their values independently for the record and playback
  * channels.  Since the only common factor between the input & output is the
  * sample rate (on pmac) it should be possible to open /dev/dspX O_WRONLY and
  * /dev/dspY O_RDONLY.  The input & output channels could then have different
  * characteristics (other than the first that sets sample rate claiming the
  * right to set it for ever).  As it stands, the format, channels, number of
  * bits & sample rate are assumed to be common.  In the future perhaps these
  * should be the responsibility of the LL driver - and then if a card really
  * does not share items between record & playback they can be specified
  * separately.
*/

/* Thread-safeness of shared_resources notes: 31/01/2001
 * If the user opens O_RDWR and then splits record & play between two threads
 * both of which inherit the fd - and then starts changing things from both
 * - we will have difficulty telling.
 *
 * It's bad application coding - but ...
 * TODO: think about how to sort this out... without bogging everything down in
 * semaphores.
 *
 * Similarly, the OSS spec says "all changes to parameters must be between
 * open() and the first read() or write(). - and a bit later on (by
 * implication) "between SNDCTL_DSP_RESET and the first read() or write() after
 * it".  If the app is multi-threaded and this rule is broken between threads
 * we will have trouble spotting it - and the fault will be rather obscure :-(
 *
 * We will try and put out at least a kmsg if we see it happen... but I think
 * it will be quite hard to trap it with an -EXXX return... because we can't
 * see the fault until after the damage is done.
*/

#include <linux/module.h>
#include <linux/slab.h>
#include <linux/sound.h>
#include <linux/init.h>
#include <linux/soundcard.h>
#include <linux/poll.h>
#include <linux/smp_lock.h>

#include <asm/uaccess.h>

#include "dmasound.h"

#define DMASOUND_CORE_REVISION 1
#define DMASOUND_CORE_EDITION 6

    /*
     *  Declarations
     */

int dmasound_catchRadius = 0;
module_param(dmasound_catchRadius, int, 0);

static unsigned int numWriteBufs = DEFAULT_N_BUFFERS;
module_param(numWriteBufs, int, 0);
static unsigned int writeBufSize = DEFAULT_BUFF_SIZE ;	/* in bytes */
module_param(writeBufSize, int, 0);

MODULE_LICENSE("GPL");

#ifdef MODULE
static int sq_unit = -1;
static int mixer_unit = -1;
static int state_unit = -1;
static int irq_installed;
#endif /* MODULE */

/* software implemented recording volume! */
uint software_input_volume = SW_INPUT_VOLUME_SCALE * SW_INPUT_VOLUME_DEFAULT;
EXPORT_SYMBOL(software_input_volume);

/* control over who can modify resources shared between play/record */
static mode_t shared_resource_owner;
static int shared_resources_initialised;

    /*
     *  Mid level stuff
     */

struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED };

static inline void sound_silence(void)
{
	dmasound.mach.silence(); /* _MUST_ stop DMA */
}

static inline int sound_set_format(int format)
{
	return dmasound.mach.setFormat(format);
}


static int sound_set_speed(int speed)
{
	if (speed < 0)
		return dmasound.soft.speed;

	/* trap out-of-range speed settings.
	   at present we allow (arbitrarily) low rates - using soft
	   up-conversion - but we can't allow > max because there is
	   no soft down-conversion.
	*/
	if (dmasound.mach.max_dsp_speed &&
	   (speed > dmasound.mach.max_dsp_speed))
		speed = dmasound.mach.max_dsp_speed ;

	dmasound.soft.speed = speed;

	if (dmasound.minDev == SND_DEV_DSP)
		dmasound.dsp.speed = dmasound.soft.speed;

	return dmasound.soft.speed;
}

static int sound_set_stereo(int stereo)
{
	if (stereo < 0)
		return dmasound.soft.stereo;

	stereo = !!stereo;    /* should be 0 or 1 now */

	dmasound.soft.stereo = stereo;
	if (dmasound.minDev == SND_DEV_DSP)
		dmasound.dsp.stereo = stereo;

	return stereo;
}

static ssize_t sound_copy_translate(TRANS *trans, const u_char __user *userPtr,
				    size_t userCount, u_char frame[],
				    ssize_t *frameUsed, ssize_t frameLeft)
{
	ssize_t (*ct_func)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);

	switch (dmasound.soft.format) {
	    case AFMT_MU_LAW:
		ct_func = trans->ct_ulaw;
		break;
	    case AFMT_A_LAW:
		ct_func = trans->ct_alaw;
		break;
	    case AFMT_S8:
		ct_func = trans->ct_s8;
		break;
	    case AFMT_U8:
		ct_func = trans->ct_u8;
		break;
	    case AFMT_S16_BE:
		ct_func = trans->ct_s16be;
		break;
	    case AFMT_U16_BE:
		ct_func = trans->ct_u16be;
		break;
	    case AFMT_S16_LE:
		ct_func = trans->ct_s16le;
		break;
	    case AFMT_U16_LE:
		ct_func = trans->ct_u16le;
		break;
	    default:
		return 0;
	}
	/* if the user has requested a non-existent translation don't try
	   to call it but just return 0 bytes moved
	*/
	if (ct_func)
		return ct_func(userPtr, userCount, frame, frameUsed, frameLeft);
	return 0;
}

    /*
     *  /dev/mixer abstraction
     */

static struct {
    int busy;
    int modify_counter;
} mixer;

static int mixer_open(struct inode *inode, struct file *file)
{
	if (!try_module_get(dmasound.mach.owner))
		return -ENODEV;
	mixer.busy = 1;
	return 0;
}

static int mixer_release(struct inode *inode, struct file *file)
{
	lock_kernel();
	mixer.busy = 0;
	module_put(dmasound.mach.owner);
	unlock_kernel();
	return 0;
}
static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd,
		       u_long arg)
{
	if (_SIOC_DIR(cmd) & _SIOC_WRITE)
	    mixer.modify_counter++;
	switch (cmd) {
	    case OSS_GETVERSION:
		return IOCTL_OUT(arg, SOUND_VERSION);
	    case SOUND_MIXER_INFO:
		{
		    mixer_info info;
		    memset(&info, 0, sizeof(info));
		    strlcpy(info.id, dmasound.mach.name2, sizeof(info.id));
		    strlcpy(info.name, dmasound.mach.name2, sizeof(info.name));
		    info.modify_counter = mixer.modify_counter;
		    if (copy_to_user((void __user *)arg, &info, sizeof(info)))
			    return -EFAULT;
		    return 0;
		}
	}
	if (dmasound.mach.mixer_ioctl)
	    return dmasound.mach.mixer_ioctl(cmd, arg);
	return -EINVAL;
}

static const struct file_operations mixer_fops =
{
	.owner		= THIS_MODULE,
	.llseek		= no_llseek,
	.ioctl		= mixer_ioctl,
	.open		= mixer_open,
	.release	= mixer_release,
};

static void mixer_init(void)
{
#ifndef MODULE
	int mixer_unit;
#endif
	mixer_unit = register_sound_mixer(&mixer_fops, -1);
	if (mixer_unit < 0)
		return;

	mixer.busy = 0;
	dmasound.treble = 0;
	dmasound.bass = 0;
	if (dmasound.mach.mixer_init)
	    dmasound.mach.mixer_init();
}


    /*
     *  Sound queue stuff, the heart of the driver
     */

struct sound_queue dmasound_write_sq;
static void sq_reset_output(void) ;

static int sq_allocate_buffers(struct sound_queue *sq, int num, int size)
{
	int i;

	if (sq->buffers)
		return 0;
	sq->numBufs = num;
	sq->bufSize = size;
	sq->buffers = kmalloc (num * sizeof(char *), GFP_KERNEL);
	if (!sq->buffers)
		return -ENOMEM;
	for (i = 0; i < num; i++) {
		sq->buffers[i] = dmasound.mach.dma_alloc(size, GFP_KERNEL);
		if (!sq->buffers[i]) {
			while (i--)
				dmasound.mach.dma_free(sq->buffers[i], size);
			kfree(sq->buffers);
			sq->buffers = NULL;
			return -ENOMEM;
		}
	}
	return 0;
}

static void sq_release_buffers(struct sound_queue *sq)
{
	int i;

	if (sq->buffers) {
		for (i = 0; i < sq->numBufs; i++)
			dmasound.mach.dma_free(sq->buffers[i], sq->bufSize);
		kfree(sq->buffers);
		sq->buffers = NULL;
	}
}


static int sq_setup(struct sound_queue *sq)
{
	int (*setup_func)(void) = NULL;
	int hard_frame ;

	if (sq->locked) { /* are we already set? - and not changeable */
#ifdef DEBUG_DMASOUND
printk("dmasound_core: tried to sq_setup a locked queue\n") ;
#endif
		return -EINVAL ;
	}
	sq->locked = 1 ; /* don't think we have a race prob. here _check_ */

	/* make sure that the parameters are set up
	   This should have been done already...
	*/

	dmasound.mach.init();

	/* OK.  If the user has set fragment parameters explicitly, then we
	   should leave them alone... as long as they are valid.
	   Invalid user fragment params can occur if we allow the whole buffer
	   to be used when the user requests the fragments sizes (with no soft
	   x-lation) and then the user subsequently sets a soft x-lation that
	   requires increased internal buffering.

	   Othwerwise (if the user did not set them) OSS says that we should
	   select frag params on the basis of 0.5 s output & 0.1 s input
	   latency. (TODO.  For now we will copy in the defaults.)
	*/

	if (sq->user_frags <= 0) {
		sq->max_count = sq->numBufs ;
		sq->max_active = sq->numBufs ;
		sq->block_size = sq->bufSize;
		/* set up the user info */
		sq->user_frags = sq->numBufs ;
		sq->user_frag_size = sq->bufSize ;
		sq->user_frag_size *=
			(dmasound.soft.size * (dmasound.soft.stereo+1) ) ;
		sq->user_frag_size /=
			(dmasound.hard.size * (dmasound.hard.stereo+1) ) ;
	} else {
		/* work out requested block size */
		sq->block_size = sq->user_frag_size ;
		sq->block_size *=
			(dmasound.hard.size * (dmasound.hard.stereo+1) ) ;
		sq->block_size /=
			(dmasound.soft.size * (dmasound.soft.stereo+1) ) ;
		/* the user wants to write frag-size chunks */
		sq->block_size *= dmasound.hard.speed ;
		sq->block_size /= dmasound.soft.speed ;
		/* this only works for size values which are powers of 2 */
		hard_frame =
			(dmasound.hard.size * (dmasound.hard.stereo+1))/8 ;
		sq->block_size +=  (hard_frame - 1) ;
		sq->block_size &= ~(hard_frame - 1) ; /* make sure we are aligned */
		/* let's just check for obvious mistakes */
		if ( sq->block_size <= 0 || sq->block_size > sq->bufSize) {
#ifdef DEBUG_DMASOUND
printk("dmasound_core: invalid frag size (user set %d)\n", sq->user_frag_size) ;
#endif
			sq->block_size = sq->bufSize ;
		}
		if ( sq->user_frags <= sq->numBufs ) {
			sq->max_count = sq->user_frags ;
			/* if user has set max_active - then use it */
			sq->max_active = (sq->max_active <= sq->max_count) ?
				sq->max_active : sq->max_count ;
		} else {
#ifdef DEBUG_DMASOUND
printk("dmasound_core: invalid frag count (user set %d)\n", sq->user_frags) ;
#endif
			sq->max_count =
			sq->max_active = sq->numBufs ;
		}
	}
	sq->front = sq->count = sq->rear_size = 0;
	sq->syncing = 0;
	sq->active = 0;

	if (sq == &write_sq) {
	    sq->rear = -1;