ca0106_main.c

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/*
 *  Copyright (c) 2004 James Courtier-Dutton <[e-mail]>
 *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
 *  Version: 0.0.21
 *
 *  FEATURES currently supported:
 *    Front, Rear and Center/LFE.
 *    Surround40 and Surround51.
 *    Capture from MIC an LINE IN input.
 *    SPDIF digital playback of PCM stereo and AC3/DTS works.
 *    (One can use a standard mono mini-jack to one RCA plugs cable.
 *     or one can use a standard stereo mini-jack to two RCA plugs cable.
 *     Plug one of the RCA plugs into the Coax input of the external decoder/receiver.)
 *    ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. )
 *    Notes on how to capture sound:
 *      The AC97 is used in the PLAYBACK direction.
 *      The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC.
 *      So, to record from the MIC, set the MIC Playback volume to max,
 *      unmute the MIC and turn up the MASTER Playback volume.
 *      So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume.
 *   
 *    The only playback controls that currently do anything are: -
 *    Analog Front
 *    Analog Rear
 *    Analog Center/LFE
 *    SPDIF Front
 *    SPDIF Rear
 *    SPDIF Center/LFE
 *   
 *    For capture from Mic in or Line in.
 *    Digital/Analog ( switch must be in Analog mode for CAPTURE. )
 * 
 *    CAPTURE feedback into PLAYBACK
 * 
 *  Changelog:
 *    Support interrupts per period.
 *    Removed noise from Center/LFE channel when in Analog mode.
 *    Rename and remove mixer controls.
 *  0.0.6
 *    Use separate card based DMA buffer for periods table list.
 *  0.0.7
 *    Change remove and rename ctrls into lists.
 *  0.0.8
 *    Try to fix capture sources.
 *  0.0.9
 *    Fix AC3 output.
 *    Enable S32_LE format support.
 *  0.0.10
 *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
 *  0.0.11
 *    Add Model name recognition.
 *  0.0.12
 *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
 *    Remove redundent "voice" handling.
 *  0.0.13
 *    Single trigger call for multi channels.
 *  0.0.14
 *    Set limits based on what the sound card hardware can do.
 *    playback periods_min=2, periods_max=8
 *    capture hw constraints require period_size = n * 64 bytes.
 *    playback hw constraints require period_size = n * 64 bytes.
 *  0.0.15
 *    Minor updates.
 *  0.0.16
 *    Implement 192000 sample rate.
 *  0.0.17
 *    Add support for SB0410 and SB0413.
 *  0.0.18
 *    Modified Copyright message.
 *  0.0.19
 *    Finally fix support for SB Live 24 bit. SB0410 and SB0413.
 *    The output codec needs resetting, otherwise all output is muted.
 *  0.0.20
 *    Merge "pci_disable_device(pci);" fixes.
 *  0.0.21
 *    Add 4 capture channels. (SPDIF only comes in on channel 0. )
 *    Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.)
 *
 *  BUGS:
 *    Some stability problems when unloading the snd-ca0106 kernel module.
 *    --
 *
 *  TODO:
 *    4 Capture channels, only one implemented so far.
 *    Other capture rates apart from 48khz not implemented.
 *    MIDI
 *    --
 *  GENERAL INFO:
 *    Model: SB0310
 *    P17 Chip: CA0106-DAT
 *    AC97 Codec: STAC 9721
 *    ADC: Philips 1361T (Stereo 24bit)
 *    DAC: WM8746EDS (6-channel, 24bit, 192Khz)
 *
 *  GENERAL INFO:
 *    Model: SB0410
 *    P17 Chip: CA0106-DAT
 *    AC97 Codec: None
 *    ADC: WM8775EDS (4 Channel)
 *    DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support)
 *    SPDIF Out control switches between Mic in and SPDIF out.
 *    No sound out or mic input working yet.
 * 
 *  GENERAL INFO:
 *    Model: SB0413
 *    P17 Chip: CA0106-DAT
 *    AC97 Codec: None.
 *    ADC: Unknown
 *    DAC: Unknown
 *    Trying to handle it like the SB0410.
 *
 *  This code was initally based on code from ALSA's emu10k1x.c which is:
 *  Copyright (c) by Francisco Moraes <[e-mail]>
 *
 *   This program is free software; you can redistribute it and/or modify
 *   it under the terms of the GNU General Public License as published by
 *   the Free Software Foundation; either version 2 of the License, or
 *   (at your option) any later version.
 *
 *   This program is distributed in the hope that it will be useful,
 *   but WITHOUT ANY WARRANTY; without even the implied warranty of
 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *   GNU General Public License for more details.
 *
 *   You should have received a copy of the GNU General Public License
 *   along with this program; if not, write to the Free Software
 *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
 *
 */
#include <sound/driver.h>
#include <linux/delay.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/slab.h>
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/info.h>

MODULE_AUTHOR("James Courtier-Dutton <[e-mail]>");
MODULE_DESCRIPTION("CA0106");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}");

// module parameters (see "Module Parameters")
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;

module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard.");
module_param_array(id, charp, NULL, 0444);
MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard.");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard.");

#include "ca0106.h"

typedef struct {
	u32 serial;
	char * name;
} ca0106_names_t;

static ca0106_names_t ca0106_chip_names[] = {
	 { 0x10021102, "AudigyLS [SB0310]"} , 
	 { 0x10051102, "AudigyLS [SB0310b]"} , /* Unknown AudigyLS that also says SB0310 on it */
	 { 0x10061102, "Live! 7.1 24bit [SB0410]"} , /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
	 { 0x10071102, "Live! 7.1 24bit [SB0413]"} , /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97.  */
	 { 0, "AudigyLS [Unknown]" }
};

/* hardware definition */
static snd_pcm_hardware_t snd_ca0106_playback_hw = {
	.info =			(SNDRV_PCM_INFO_MMAP | 
				 SNDRV_PCM_INFO_INTERLEAVED |
				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
				 SNDRV_PCM_INFO_MMAP_VALID),
	.formats =		SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
	.rates =		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
	.rate_min =		48000,
	.rate_max =		192000,
	.channels_min =		2,  //1,
	.channels_max =		2,  //6,
	.buffer_bytes_max =	(32*1024),
	.period_bytes_min =	64,
	.period_bytes_max =	(16*1024),
	.periods_min =		2,
	.periods_max =		8,
	.fifo_size =		0,
};

static snd_pcm_hardware_t snd_ca0106_capture_hw = {
	.info =			(SNDRV_PCM_INFO_MMAP | 
				 SNDRV_PCM_INFO_INTERLEAVED |
				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
				 SNDRV_PCM_INFO_MMAP_VALID),
	.formats =		SNDRV_PCM_FMTBIT_S16_LE,
	.rates =		SNDRV_PCM_RATE_48000,
	.rate_min =		48000,
	.rate_max =		48000,
	.channels_min =		2,
	.channels_max =		2,
	.buffer_bytes_max =	(32*1024),
	.period_bytes_min =	64,
	.period_bytes_max =	(16*1024),
	.periods_min =		2,
	.periods_max =		2,
	.fifo_size =		0,
};

unsigned int snd_ca0106_ptr_read(ca0106_t * emu, 
					  unsigned int reg, 
					  unsigned int chn)
{
	unsigned long flags;
	unsigned int regptr, val;
  
	regptr = (reg << 16) | chn;

	spin_lock_irqsave(&emu->emu_lock, flags);
	outl(regptr, emu->port + PTR);
	val = inl(emu->port + DATA);
	spin_unlock_irqrestore(&emu->emu_lock, flags);
	return val;
}

void snd_ca0106_ptr_write(ca0106_t *emu, 
				   unsigned int reg, 
				   unsigned int chn, 
				   unsigned int data)
{
	unsigned int regptr;
	unsigned long flags;

	regptr = (reg << 16) | chn;

	spin_lock_irqsave(&emu->emu_lock, flags);
	outl(regptr, emu->port + PTR);
	outl(data, emu->port + DATA);
	spin_unlock_irqrestore(&emu->emu_lock, flags);
}

static void snd_ca0106_intr_enable(ca0106_t *emu, unsigned int intrenb)
{
	unsigned long flags;
	unsigned int enable;
  
	spin_lock_irqsave(&emu->emu_lock, flags);
	enable = inl(emu->port + INTE) | intrenb;
	outl(enable, emu->port + INTE);
	spin_unlock_irqrestore(&emu->emu_lock, flags);
}

static void snd_ca0106_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
	ca0106_pcm_t *epcm = runtime->private_data;
  
	if (epcm) {
		kfree(epcm);
	}
}

/* open_playback callback */
static int snd_ca0106_pcm_open_playback_channel(snd_pcm_substream_t *substream, int channel_id)
{
	ca0106_t *chip = snd_pcm_substream_chip(substream);
        ca0106_channel_t *channel = &(chip->playback_channels[channel_id]);
	ca0106_pcm_t *epcm;
	snd_pcm_runtime_t *runtime = substream->runtime;
	int err;

	epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL);

	if (epcm == NULL)
		return -ENOMEM;
	epcm->emu = chip;
	epcm->substream = substream;
        epcm->channel_id=channel_id;
  
	runtime->private_data = epcm;
	runtime->private_free = snd_ca0106_pcm_free_substream;
  
	runtime->hw = snd_ca0106_playback_hw;

        channel->emu = chip;
        channel->number = channel_id;

        channel->use=1;
        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
        //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
        channel->epcm=epcm;
	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
                return err;
	if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
                return err;
	return 0;
}

/* close callback */
static int snd_ca0106_pcm_close_playback(snd_pcm_substream_t *substream)
{
	ca0106_t *chip = snd_pcm_substream_chip(substream);
	snd_pcm_runtime_t *runtime = substream->runtime;
        ca0106_pcm_t *epcm = runtime->private_data;
        chip->playback_channels[epcm->channel_id].use=0;
/* FIXME: maybe zero others */
	return 0;
}

static int snd_ca0106_pcm_open_playback_front(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
}

static int snd_ca0106_pcm_open_playback_center_lfe(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL);
}

static int snd_ca0106_pcm_open_playback_unknown(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL);
}

static int snd_ca0106_pcm_open_playback_rear(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL);
}

/* open_capture callback */
static int snd_ca0106_pcm_open_capture_channel(snd_pcm_substream_t *substream, int channel_id)
{
	ca0106_t *chip = snd_pcm_substream_chip(substream);
        ca0106_channel_t *channel = &(chip->capture_channels[channel_id]);
	ca0106_pcm_t *epcm;
	snd_pcm_runtime_t *runtime = substream->runtime;
	int err;

	epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL);
	if (epcm == NULL) {
                snd_printk("open_capture_channel: failed epcm alloc\n");
		return -ENOMEM;
        }
	epcm->emu = chip;
	epcm->substream = substream;
        epcm->channel_id=channel_id;
  
	runtime->private_data = epcm;
	runtime->private_free = snd_ca0106_pcm_free_substream;
  
	runtime->hw = snd_ca0106_capture_hw;

        channel->emu = chip;
        channel->number = channel_id;

        channel->use=1;
        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
        //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
        channel->epcm=epcm;
	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
                return err;
	//snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes);
	if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
                return err;
	return 0;
}

/* close callback */
static int snd_ca0106_pcm_close_capture(snd_pcm_substream_t *substream)
{
	ca0106_t *chip = snd_pcm_substream_chip(substream);
	snd_pcm_runtime_t *runtime = substream->runtime;
        ca0106_pcm_t *epcm = runtime->private_data;
        chip->capture_channels[epcm->channel_id].use=0;
/* FIXME: maybe zero others */
	return 0;
}

static int snd_ca0106_pcm_open_0_capture(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_capture_channel(substream, 0);
}

static int snd_ca0106_pcm_open_1_capture(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_capture_channel(substream, 1);
}

static int snd_ca0106_pcm_open_2_capture(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_capture_channel(substream, 2);
}

static int snd_ca0106_pcm_open_3_capture(snd_pcm_substream_t *substream)
{
	return snd_ca0106_pcm_open_capture_channel(substream, 3);
}

/* hw_params callback */
static int snd_ca0106_pcm_hw_params_playback(snd_pcm_substream_t *substream,
				      snd_pcm_hw_params_t * hw_params)
{
	return snd_pcm_lib_malloc_pages(substream,
					params_buffer_bytes(hw_params));
}

/* hw_free callback */
static int snd_ca0106_pcm_hw_free_playback(snd_pcm_substream_t *substream)
{
	return snd_pcm_lib_free_pages(substream);
}

/* hw_params callback */
static int snd_ca0106_pcm_hw_params_capture(snd_pcm_substream_t *substream,
				      snd_pcm_hw_params_t * hw_params)
{
	return snd_pcm_lib_malloc_pages(substream,
					params_buffer_bytes(hw_params));
}

/* hw_free callback */
static int snd_ca0106_pcm_hw_free_capture(snd_pcm_substream_t *substream)