dmasound_core.c

linux-2.6.15.6

/*
 *  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_PARM(dmasound_catchRadius, "i");

static unsigned int numWriteBufs = DEFAULT_N_BUFFERS;
MODULE_PARM(numWriteBufs, "i");
static unsigned int writeBufSize = DEFAULT_BUFF_SIZE ;	/* in bytes */
MODULE_PARM(writeBufSize, "i");

#ifdef HAS_RECORD
static unsigned int numReadBufs = DEFAULT_N_BUFFERS;
MODULE_PARM(numReadBufs, "i");
static unsigned int readBufSize = DEFAULT_BUFF_SIZE;	/* in bytes */
MODULE_PARM(readBufSize, "i");
#endif

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 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) ;
#ifdef HAS_RECORD
struct sound_queue dmasound_read_sq;
static void sq_reset_input(void) ;
#endif

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 ;
	}