freebsd-skq/sys/i386/isa/sound/pas2_pcm.c
Steven Wallace 66e456c58e Revert to earlier code which contains FreeBSD snd[1-7] probe information,
$Id$ information, and other code to make sound driver compile and work
correctly with FreeBSD.

Integrate changes obtained from Sujal Patel.  These changes are:
 o  local.h:  reverse option logic from EXCLUDE_* to AUDIO_*
 o  pas2_mixer.c:  small addition
 o  ad1848.c:  minor change with macro names
 o  sequencer.c:  minor change with note check
 o  many spelling corrections in comments in about every other file
1995-03-05 22:11:57 +00:00

460 lines
11 KiB
C

#define _PAS2_PCM_C_
/*
* sound/pas2_pcm.c
*
* The low level driver for the Pro Audio Spectrum ADC/DAC.
*
* Copyright by Hannu Savolainen 1993
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
* met: 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer. 2.
* Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
* ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
* CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* pas2_pcm.c,v 1.7 1994/10/01 02:16:57 swallace Exp
*/
#include "sound_config.h"
#ifdef CONFIGURE_SOUNDCARD
#include "pas.h"
#if !defined(EXCLUDE_PAS) && !defined(EXCLUDE_AUDIO)
#define TRACE(WHAT) /*
* * * (WHAT) */
#define PAS_PCM_INTRBITS (0x08)
/*
* Sample buffer timer interrupt enable
*/
#define PCM_NON 0
#define PCM_DAC 1
#define PCM_ADC 2
static unsigned long pcm_speed = 0; /* sampling rate */
static unsigned char pcm_channels = 1; /* channels (1 or 2) */
static unsigned char pcm_bits = 8; /* bits/sample (8 or 16) */
static unsigned char pcm_filter = 0; /* filter FLAG */
static unsigned char pcm_mode = PCM_NON;
static unsigned long pcm_count = 0;
static unsigned short pcm_bitsok = 8; /* mask of OK bits */
static int my_devnum = 0;
int
pcm_set_speed (int arg)
{
int foo, tmp;
unsigned long flags;
if (arg > 44100)
arg = 44100;
if (arg < 5000)
arg = 5000;
foo = (1193180 + (arg / 2)) / arg;
arg = 1193180 / foo;
if (pcm_channels & 2)
foo = foo >> 1;
pcm_speed = arg;
tmp = pas_read (FILTER_FREQUENCY);
/*
* Set anti-aliasing filters according to sample rate. You really *NEED*
* to enable this feature for all normal recording unless you want to
* experiment with aliasing effects.
* These filters apply to the selected "recording" source.
* I (pfw) don't know the encoding of these 5 bits. The values shown
* come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/.
*/
#if !defined NO_AUTO_FILTER_SET
tmp &= 0xe0;
if (pcm_speed >= 2 * 17897)
tmp |= 0x21;
else if (pcm_speed >= 2 * 15909)
tmp |= 0x22;
else if (pcm_speed >= 2 * 11931)
tmp |= 0x29;
else if (pcm_speed >= 2 * 8948)
tmp |= 0x31;
else if (pcm_speed >= 2 * 5965)
tmp |= 0x39;
else if (pcm_speed >= 2 * 2982)
tmp |= 0x24;
pcm_filter = tmp;
#endif
DISABLE_INTR (flags);
pas_write (tmp & ~(F_F_PCM_RATE_COUNTER | F_F_PCM_BUFFER_COUNTER), FILTER_FREQUENCY);
pas_write (S_C_C_SAMPLE_RATE | S_C_C_LSB_THEN_MSB | S_C_C_SQUARE_WAVE, SAMPLE_COUNTER_CONTROL);
pas_write (foo & 0xff, SAMPLE_RATE_TIMER);
pas_write ((foo >> 8) & 0xff, SAMPLE_RATE_TIMER);
pas_write (tmp, FILTER_FREQUENCY);
RESTORE_INTR (flags);
return pcm_speed;
}
int
pcm_set_channels (int arg)
{
if ((arg != 1) && (arg != 2))
return pcm_channels;
if (arg != pcm_channels)
{
pas_write (pas_read (PCM_CONTROL) ^ P_C_PCM_MONO, PCM_CONTROL);
pcm_channels = arg;
pcm_set_speed (pcm_speed);/*
* The speed must be reinitialized
*/
}
return pcm_channels;
}
int
pcm_set_bits (int arg)
{
if ((arg & pcm_bitsok) != arg)
return pcm_bits;
if (arg != pcm_bits)
{
pas_write (pas_read (SYSTEM_CONFIGURATION_2) ^ S_C_2_PCM_16_BIT, SYSTEM_CONFIGURATION_2);
pcm_bits = arg;
}
return pcm_bits;
}
static int
pas_pcm_ioctl (int dev, unsigned int cmd, unsigned int arg, int local)
{
TRACE (printk ("pas2_pcm.c: static int pas_pcm_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg));
switch (cmd)
{
case SOUND_PCM_WRITE_RATE:
if (local)
return pcm_set_speed (arg);
return IOCTL_OUT (arg, pcm_set_speed (IOCTL_IN (arg)));
break;
case SOUND_PCM_READ_RATE:
if (local)
return pcm_speed;
return IOCTL_OUT (arg, pcm_speed);
break;
case SNDCTL_DSP_STEREO:
if (local)
return pcm_set_channels (arg + 1) - 1;
return IOCTL_OUT (arg, pcm_set_channels (IOCTL_IN (arg) + 1) - 1);
break;
case SOUND_PCM_WRITE_CHANNELS:
if (local)
return pcm_set_channels (arg);
return IOCTL_OUT (arg, pcm_set_channels (IOCTL_IN (arg)));
break;
case SOUND_PCM_READ_CHANNELS:
if (local)
return pcm_channels;
return IOCTL_OUT (arg, pcm_channels);
break;
case SNDCTL_DSP_SETFMT:
if (local)
return pcm_set_bits (arg);
return IOCTL_OUT (arg, pcm_set_bits (IOCTL_IN (arg)));
break;
case SOUND_PCM_READ_BITS:
if (local)
return pcm_bits;
return IOCTL_OUT (arg, pcm_bits);
case SOUND_PCM_WRITE_FILTER: /*
* NOT YET IMPLEMENTED
*/
if (IOCTL_IN (arg) > 1)
return IOCTL_OUT (arg, RET_ERROR (EINVAL));
break;
pcm_filter = IOCTL_IN (arg);
case SOUND_PCM_READ_FILTER:
return IOCTL_OUT (arg, pcm_filter);
break;
default:
return RET_ERROR (EINVAL);
}
return RET_ERROR (EINVAL);
}
static void
pas_pcm_reset (int dev)
{
TRACE (printk ("pas2_pcm.c: static void pas_pcm_reset(void)\n"));
pas_write (pas_read (PCM_CONTROL) & ~P_C_PCM_ENABLE, PCM_CONTROL);
}
static int
pas_pcm_open (int dev, int mode)
{
int err;
TRACE (printk ("pas2_pcm.c: static int pas_pcm_open(int mode = %X)\n", mode));
if ((err = pas_set_intr (PAS_PCM_INTRBITS)) < 0)
return err;
if (DMAbuf_open_dma (dev) < 0)
{
pas_remove_intr (PAS_PCM_INTRBITS);
return RET_ERROR (EBUSY);
}
pcm_count = 0;
return 0;
}
static void
pas_pcm_close (int dev)
{
unsigned long flags;
TRACE (printk ("pas2_pcm.c: static void pas_pcm_close(void)\n"));
DISABLE_INTR (flags);
pas_pcm_reset (dev);
DMAbuf_close_dma (dev);
pas_remove_intr (PAS_PCM_INTRBITS);
pcm_mode = PCM_NON;
RESTORE_INTR (flags);
}
static void
pas_pcm_output_block (int dev, unsigned long buf, int count,
int intrflag, int restart_dma)
{
unsigned long flags, cnt;
TRACE (printk ("pas2_pcm.c: static void pas_pcm_output_block(char *buf = %P, int count = %X)\n", buf, count));
cnt = count;
if (audio_devs[dev]->dmachan > 3)
cnt >>= 1;
if (audio_devs[dev]->flags & DMA_AUTOMODE &&
intrflag &&
cnt == pcm_count)
return; /*
* Auto mode on. No need to react
*/
DISABLE_INTR (flags);
pas_write (pas_read (PCM_CONTROL) & ~P_C_PCM_ENABLE,
PCM_CONTROL);
if (restart_dma)
DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE);
if (audio_devs[dev]->dmachan > 3)
count >>= 1;
if (count != pcm_count)
{
pas_write (pas_read (FILTER_FREQUENCY) & ~F_F_PCM_BUFFER_COUNTER, FILTER_FREQUENCY);
pas_write (S_C_C_SAMPLE_BUFFER | S_C_C_LSB_THEN_MSB | S_C_C_SQUARE_WAVE, SAMPLE_COUNTER_CONTROL);
pas_write (count & 0xff, SAMPLE_BUFFER_COUNTER);
pas_write ((count >> 8) & 0xff, SAMPLE_BUFFER_COUNTER);
pas_write (pas_read (FILTER_FREQUENCY) | F_F_PCM_BUFFER_COUNTER, FILTER_FREQUENCY);
pcm_count = count;
}
pas_write (pas_read (FILTER_FREQUENCY) | F_F_PCM_BUFFER_COUNTER | F_F_PCM_RATE_COUNTER, FILTER_FREQUENCY);
pas_write (pas_read (PCM_CONTROL) | P_C_PCM_ENABLE | P_C_PCM_DAC_MODE, PCM_CONTROL);
pcm_mode = PCM_DAC;
RESTORE_INTR (flags);
}
static void
pas_pcm_start_input (int dev, unsigned long buf, int count,
int intrflag, int restart_dma)
{
unsigned long flags;
int cnt;
TRACE (printk ("pas2_pcm.c: static void pas_pcm_start_input(char *buf = %P, int count = %X)\n", buf, count));
cnt = count;
if (audio_devs[dev]->dmachan > 3)
cnt >>= 1;
if (audio_devs[my_devnum]->flags & DMA_AUTOMODE &&
intrflag &&
cnt == pcm_count)
return; /*
* Auto mode on. No need to react
*/
DISABLE_INTR (flags);
if (restart_dma)
DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ);
if (audio_devs[dev]->dmachan > 3)
count >>= 1;
if (count != pcm_count)
{
pas_write (pas_read (FILTER_FREQUENCY) & ~F_F_PCM_BUFFER_COUNTER, FILTER_FREQUENCY);
pas_write (S_C_C_SAMPLE_BUFFER | S_C_C_LSB_THEN_MSB | S_C_C_SQUARE_WAVE, SAMPLE_COUNTER_CONTROL);
pas_write (count & 0xff, SAMPLE_BUFFER_COUNTER);
pas_write ((count >> 8) & 0xff, SAMPLE_BUFFER_COUNTER);
pas_write (pas_read (FILTER_FREQUENCY) | F_F_PCM_BUFFER_COUNTER, FILTER_FREQUENCY);
pcm_count = count;
}
pas_write (pas_read (FILTER_FREQUENCY) | F_F_PCM_BUFFER_COUNTER | F_F_PCM_RATE_COUNTER, FILTER_FREQUENCY);
pas_write ((pas_read (PCM_CONTROL) | P_C_PCM_ENABLE) & ~P_C_PCM_DAC_MODE, PCM_CONTROL);
pcm_mode = PCM_ADC;
RESTORE_INTR (flags);
}
static int
pas_pcm_prepare_for_input (int dev, int bsize, int bcount)
{
return 0;
}
static int
pas_pcm_prepare_for_output (int dev, int bsize, int bcount)
{
return 0;
}
static struct audio_operations pas_pcm_operations =
{
"Pro Audio Spectrum",
DMA_AUTOMODE,
AFMT_U8 | AFMT_S16_LE,
NULL,
pas_pcm_open,
pas_pcm_close,
pas_pcm_output_block,
pas_pcm_start_input,
pas_pcm_ioctl,
pas_pcm_prepare_for_input,
pas_pcm_prepare_for_output,
pas_pcm_reset,
pas_pcm_reset,
NULL,
NULL
};
long
pas_pcm_init (long mem_start, struct address_info *hw_config)
{
TRACE (printk ("pas2_pcm.c: long pas_pcm_init(long mem_start = %X)\n", mem_start));
pcm_bitsok = 8;
if (pas_read (OPERATION_MODE_1) & O_M_1_PCM_TYPE)
pcm_bitsok |= 16;
pcm_set_speed (DSP_DEFAULT_SPEED);
if (num_audiodevs < MAX_AUDIO_DEV)
{
audio_devs[my_devnum = num_audiodevs++] = &pas_pcm_operations;
audio_devs[my_devnum]->dmachan = hw_config->dma;
#ifndef NO_AUTODMA
audio_devs[my_devnum]->buffcount = 1;
#else
audio_devs[my_devnum]->flags &= ~DMA_AUTOMODE;
audio_devs[my_devnum]->buffcount = DSP_BUFFCOUNT;
#endif
audio_devs[my_devnum]->buffsize = 2 * DSP_BUFFSIZE;
}
else
printk ("PAS2: Too many PCM devices available\n");
return mem_start;
}
void
pas_pcm_interrupt (unsigned char status, int cause)
{
if (cause == 1) /*
* PCM buffer done
*/
{
/*
* Halt the PCM first. Otherwise we don't have time to start a new
* block before the PCM chip proceeds to the next sample
*/
if (!(audio_devs[my_devnum]->flags & DMA_AUTOMODE))
{
pas_write (pas_read (PCM_CONTROL) & ~P_C_PCM_ENABLE,
PCM_CONTROL);
}
switch (pcm_mode)
{
case PCM_DAC:
DMAbuf_outputintr (my_devnum, 1);
break;
case PCM_ADC:
DMAbuf_inputintr (my_devnum);
break;
default:
printk ("PAS: Unexpected PCM interrupt\n");
}
}
}
#endif
#endif