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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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bd7dd77c2a
Convert from snd_card_new() to the new snd_card_create() function in other sound subdirectories. Signed-off-by: Takashi Iwai <tiwai@suse.de>
1007 lines
27 KiB
C
1007 lines
27 KiB
C
/*
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* Sound driver for Silicon Graphics O2 Workstations A/V board audio.
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*
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* Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
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* Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
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* Mxier part taken from mace_audio.c:
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* Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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*/
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#include <linux/init.h>
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#include <linux/delay.h>
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#include <linux/spinlock.h>
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#include <linux/gfp.h>
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#include <linux/vmalloc.h>
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#include <linux/interrupt.h>
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#include <linux/dma-mapping.h>
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#include <linux/platform_device.h>
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#include <linux/io.h>
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#include <asm/ip32/ip32_ints.h>
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#include <asm/ip32/mace.h>
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#include <sound/core.h>
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#include <sound/control.h>
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#include <sound/pcm.h>
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#define SNDRV_GET_ID
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#include <sound/initval.h>
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#include <sound/ad1843.h>
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MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
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MODULE_DESCRIPTION("SGI O2 Audio");
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MODULE_LICENSE("GPL");
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MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
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static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
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static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
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module_param(index, int, 0444);
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MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
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module_param(id, charp, 0444);
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MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
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#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
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#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
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#define CODEC_CONTROL_WORD_SHIFT 0
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#define CODEC_CONTROL_READ BIT(16)
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#define CODEC_CONTROL_ADDRESS_SHIFT 17
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#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */
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#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */
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#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */
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#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */
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#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */
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#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */
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#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */
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#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
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#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */
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#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */
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#define CHANNEL_RING_SHIFT 12
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#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT)
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#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1)
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#define CHANNEL_LEFT_SHIFT 40
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#define CHANNEL_RIGHT_SHIFT 8
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struct snd_sgio2audio_chan {
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int idx;
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struct snd_pcm_substream *substream;
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int pos;
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snd_pcm_uframes_t size;
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spinlock_t lock;
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};
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/* definition of the chip-specific record */
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struct snd_sgio2audio {
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struct snd_card *card;
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/* codec */
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struct snd_ad1843 ad1843;
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spinlock_t ad1843_lock;
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/* channels */
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struct snd_sgio2audio_chan channel[3];
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/* resources */
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void *ring_base;
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dma_addr_t ring_base_dma;
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};
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/* AD1843 access */
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/*
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* read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
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*
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* Returns unsigned register value on success, -errno on failure.
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*/
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static int read_ad1843_reg(void *priv, int reg)
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{
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struct snd_sgio2audio *chip = priv;
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int val;
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unsigned long flags;
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spin_lock_irqsave(&chip->ad1843_lock, flags);
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writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
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CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
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wmb();
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val = readq(&mace->perif.audio.codec_control); /* flush bus */
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udelay(200);
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val = readq(&mace->perif.audio.codec_read);
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spin_unlock_irqrestore(&chip->ad1843_lock, flags);
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return val;
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}
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/*
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* write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
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*/
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static int write_ad1843_reg(void *priv, int reg, int word)
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{
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struct snd_sgio2audio *chip = priv;
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int val;
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unsigned long flags;
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spin_lock_irqsave(&chip->ad1843_lock, flags);
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writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
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(word << CODEC_CONTROL_WORD_SHIFT),
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&mace->perif.audio.codec_control);
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wmb();
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val = readq(&mace->perif.audio.codec_control); /* flush bus */
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udelay(200);
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spin_unlock_irqrestore(&chip->ad1843_lock, flags);
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return 0;
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}
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static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_info *uinfo)
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{
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struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
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uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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uinfo->count = 2;
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uinfo->value.integer.min = 0;
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uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
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(int)kcontrol->private_value);
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return 0;
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}
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static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
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int vol;
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vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
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ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
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ucontrol->value.integer.value[1] = vol & 0xFF;
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return 0;
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}
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static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
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int newvol, oldvol;
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oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
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newvol = (ucontrol->value.integer.value[0] << 8) |
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ucontrol->value.integer.value[1];
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newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
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newvol);
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return newvol != oldvol;
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}
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static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_info *uinfo)
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{
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static const char *texts[3] = {
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"Cam Mic", "Mic", "Line"
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};
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uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
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uinfo->count = 1;
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uinfo->value.enumerated.items = 3;
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if (uinfo->value.enumerated.item >= 3)
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uinfo->value.enumerated.item = 1;
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strcpy(uinfo->value.enumerated.name,
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texts[uinfo->value.enumerated.item]);
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return 0;
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}
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static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
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ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
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return 0;
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}
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static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
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int newsrc, oldsrc;
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oldsrc = ad1843_get_recsrc(&chip->ad1843);
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newsrc = ad1843_set_recsrc(&chip->ad1843,
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ucontrol->value.enumerated.item[0]);
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return newsrc != oldsrc;
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}
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/* dac1/pcm0 mixer control */
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static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "PCM Playback Volume",
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.index = 0,
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.private_value = AD1843_GAIN_PCM_0,
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.info = sgio2audio_gain_info,
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.get = sgio2audio_gain_get,
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.put = sgio2audio_gain_put,
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};
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/* dac2/pcm1 mixer control */
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static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "PCM Playback Volume",
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.index = 1,
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.private_value = AD1843_GAIN_PCM_1,
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.info = sgio2audio_gain_info,
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.get = sgio2audio_gain_get,
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.put = sgio2audio_gain_put,
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};
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/* record level mixer control */
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static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "Capture Volume",
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.private_value = AD1843_GAIN_RECLEV,
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.info = sgio2audio_gain_info,
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.get = sgio2audio_gain_get,
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.put = sgio2audio_gain_put,
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};
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/* record level source control */
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static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "Capture Source",
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.info = sgio2audio_source_info,
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.get = sgio2audio_source_get,
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.put = sgio2audio_source_put,
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};
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/* line mixer control */
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static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "Line Playback Volume",
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.index = 0,
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.private_value = AD1843_GAIN_LINE,
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.info = sgio2audio_gain_info,
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.get = sgio2audio_gain_get,
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.put = sgio2audio_gain_put,
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};
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/* cd mixer control */
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static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "Line Playback Volume",
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.index = 1,
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.private_value = AD1843_GAIN_LINE_2,
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.info = sgio2audio_gain_info,
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.get = sgio2audio_gain_get,
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.put = sgio2audio_gain_put,
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};
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/* mic mixer control */
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static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.name = "Mic Playback Volume",
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
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.private_value = AD1843_GAIN_MIC,
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.info = sgio2audio_gain_info,
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.get = sgio2audio_gain_get,
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.put = sgio2audio_gain_put,
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};
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static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
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{
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int err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
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if (err < 0)
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return err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
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if (err < 0)
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return err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
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if (err < 0)
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return err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
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if (err < 0)
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return err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_line, chip));
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if (err < 0)
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return err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
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if (err < 0)
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return err;
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
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if (err < 0)
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return err;
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return 0;
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}
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/* low-level audio interface DMA */
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/* get data out of bounce buffer, count must be a multiple of 32 */
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/* returns 1 if a period has elapsed */
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static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
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unsigned int ch, unsigned int count)
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{
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int ret;
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unsigned long src_base, src_pos, dst_mask;
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unsigned char *dst_base;
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int dst_pos;
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u64 *src;
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s16 *dst;
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u64 x;
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unsigned long flags;
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struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
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spin_lock_irqsave(&chip->channel[ch].lock, flags);
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src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
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src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
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dst_base = runtime->dma_area;
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dst_pos = chip->channel[ch].pos;
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dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
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/* check if a period has elapsed */
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chip->channel[ch].size += (count >> 3); /* in frames */
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ret = chip->channel[ch].size >= runtime->period_size;
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chip->channel[ch].size %= runtime->period_size;
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while (count) {
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src = (u64 *)(src_base + src_pos);
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dst = (s16 *)(dst_base + dst_pos);
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x = *src;
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dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
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dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
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src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
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dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
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count -= sizeof(u64);
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}
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writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
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chip->channel[ch].pos = dst_pos;
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spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
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return ret;
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}
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/* put some DMA data in bounce buffer, count must be a multiple of 32 */
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/* returns 1 if a period has elapsed */
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static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
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unsigned int ch, unsigned int count)
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{
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int ret;
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s64 l, r;
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unsigned long dst_base, dst_pos, src_mask;
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unsigned char *src_base;
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int src_pos;
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u64 *dst;
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s16 *src;
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unsigned long flags;
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struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
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spin_lock_irqsave(&chip->channel[ch].lock, flags);
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dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
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dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
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src_base = runtime->dma_area;
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src_pos = chip->channel[ch].pos;
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src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
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/* check if a period has elapsed */
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chip->channel[ch].size += (count >> 3); /* in frames */
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ret = chip->channel[ch].size >= runtime->period_size;
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chip->channel[ch].size %= runtime->period_size;
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while (count) {
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src = (s16 *)(src_base + src_pos);
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dst = (u64 *)(dst_base + dst_pos);
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l = src[0]; /* sign extend */
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r = src[1]; /* sign extend */
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*dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
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((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
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dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
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src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
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count -= sizeof(u64);
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}
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writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
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chip->channel[ch].pos = src_pos;
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spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
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return ret;
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}
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static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
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{
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struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
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struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
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int ch = chan->idx;
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/* reset DMA channel */
|
|
writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
|
|
udelay(10);
|
|
writeq(0, &mace->perif.audio.chan[ch].control);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
/* push a full buffer */
|
|
snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
|
|
}
|
|
/* set DMA to wake on 50% empty and enable interrupt */
|
|
writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
|
|
&mace->perif.audio.chan[ch].control);
|
|
return 0;
|
|
}
|
|
|
|
static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
|
|
|
|
writeq(0, &mace->perif.audio.chan[chan->idx].control);
|
|
return 0;
|
|
}
|
|
|
|
static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
|
|
{
|
|
struct snd_sgio2audio_chan *chan = dev_id;
|
|
struct snd_pcm_substream *substream;
|
|
struct snd_sgio2audio *chip;
|
|
int count, ch;
|
|
|
|
substream = chan->substream;
|
|
chip = snd_pcm_substream_chip(substream);
|
|
ch = chan->idx;
|
|
|
|
/* empty the ring */
|
|
count = CHANNEL_RING_SIZE -
|
|
readq(&mace->perif.audio.chan[ch].depth) - 32;
|
|
if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
|
|
snd_pcm_period_elapsed(substream);
|
|
|
|
return IRQ_HANDLED;
|
|
}
|
|
|
|
static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
|
|
{
|
|
struct snd_sgio2audio_chan *chan = dev_id;
|
|
struct snd_pcm_substream *substream;
|
|
struct snd_sgio2audio *chip;
|
|
int count, ch;
|
|
|
|
substream = chan->substream;
|
|
chip = snd_pcm_substream_chip(substream);
|
|
ch = chan->idx;
|
|
/* fill the ring */
|
|
count = CHANNEL_RING_SIZE -
|
|
readq(&mace->perif.audio.chan[ch].depth) - 32;
|
|
if (snd_sgio2audio_dma_push_frag(chip, ch, count))
|
|
snd_pcm_period_elapsed(substream);
|
|
|
|
return IRQ_HANDLED;
|
|
}
|
|
|
|
static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
|
|
{
|
|
struct snd_sgio2audio_chan *chan = dev_id;
|
|
struct snd_pcm_substream *substream;
|
|
|
|
substream = chan->substream;
|
|
snd_sgio2audio_dma_stop(substream);
|
|
snd_sgio2audio_dma_start(substream);
|
|
return IRQ_HANDLED;
|
|
}
|
|
|
|
/* PCM part */
|
|
/* PCM hardware definition */
|
|
static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
|
|
.info = (SNDRV_PCM_INFO_MMAP |
|
|
SNDRV_PCM_INFO_MMAP_VALID |
|
|
SNDRV_PCM_INFO_INTERLEAVED |
|
|
SNDRV_PCM_INFO_BLOCK_TRANSFER),
|
|
.formats = SNDRV_PCM_FMTBIT_S16_BE,
|
|
.rates = SNDRV_PCM_RATE_8000_48000,
|
|
.rate_min = 8000,
|
|
.rate_max = 48000,
|
|
.channels_min = 2,
|
|
.channels_max = 2,
|
|
.buffer_bytes_max = 65536,
|
|
.period_bytes_min = 32768,
|
|
.period_bytes_max = 65536,
|
|
.periods_min = 1,
|
|
.periods_max = 1024,
|
|
};
|
|
|
|
/* PCM playback open callback */
|
|
static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
|
|
runtime->hw = snd_sgio2audio_pcm_hw;
|
|
runtime->private_data = &chip->channel[1];
|
|
return 0;
|
|
}
|
|
|
|
static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
|
|
runtime->hw = snd_sgio2audio_pcm_hw;
|
|
runtime->private_data = &chip->channel[2];
|
|
return 0;
|
|
}
|
|
|
|
/* PCM capture open callback */
|
|
static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
|
|
runtime->hw = snd_sgio2audio_pcm_hw;
|
|
runtime->private_data = &chip->channel[0];
|
|
return 0;
|
|
}
|
|
|
|
/* PCM close callback */
|
|
static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
|
|
runtime->private_data = NULL;
|
|
return 0;
|
|
}
|
|
|
|
|
|
/* hw_params callback */
|
|
static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
|
|
struct snd_pcm_hw_params *hw_params)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
int size = params_buffer_bytes(hw_params);
|
|
|
|
/* alloc virtual 'dma' area */
|
|
if (runtime->dma_area)
|
|
vfree(runtime->dma_area);
|
|
runtime->dma_area = vmalloc(size);
|
|
if (runtime->dma_area == NULL)
|
|
return -ENOMEM;
|
|
runtime->dma_bytes = size;
|
|
return 0;
|
|
}
|
|
|
|
/* hw_free callback */
|
|
static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
|
|
{
|
|
if (substream->runtime->dma_area)
|
|
vfree(substream->runtime->dma_area);
|
|
substream->runtime->dma_area = NULL;
|
|
return 0;
|
|
}
|
|
|
|
/* prepare callback */
|
|
static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
|
|
int ch = chan->idx;
|
|
unsigned long flags;
|
|
|
|
spin_lock_irqsave(&chip->channel[ch].lock, flags);
|
|
|
|
/* Setup the pseudo-dma transfer pointers. */
|
|
chip->channel[ch].pos = 0;
|
|
chip->channel[ch].size = 0;
|
|
chip->channel[ch].substream = substream;
|
|
|
|
/* set AD1843 format */
|
|
/* hardware format is always S16_LE */
|
|
switch (substream->stream) {
|
|
case SNDRV_PCM_STREAM_PLAYBACK:
|
|
ad1843_setup_dac(&chip->ad1843,
|
|
ch - 1,
|
|
runtime->rate,
|
|
SNDRV_PCM_FORMAT_S16_LE,
|
|
runtime->channels);
|
|
break;
|
|
case SNDRV_PCM_STREAM_CAPTURE:
|
|
ad1843_setup_adc(&chip->ad1843,
|
|
runtime->rate,
|
|
SNDRV_PCM_FORMAT_S16_LE,
|
|
runtime->channels);
|
|
break;
|
|
}
|
|
spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
|
|
return 0;
|
|
}
|
|
|
|
/* trigger callback */
|
|
static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
|
|
int cmd)
|
|
{
|
|
switch (cmd) {
|
|
case SNDRV_PCM_TRIGGER_START:
|
|
/* start the PCM engine */
|
|
snd_sgio2audio_dma_start(substream);
|
|
break;
|
|
case SNDRV_PCM_TRIGGER_STOP:
|
|
/* stop the PCM engine */
|
|
snd_sgio2audio_dma_stop(substream);
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* pointer callback */
|
|
static snd_pcm_uframes_t
|
|
snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
|
|
struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
|
|
|
|
/* get the current hardware pointer */
|
|
return bytes_to_frames(substream->runtime,
|
|
chip->channel[chan->idx].pos);
|
|
}
|
|
|
|
/* get the physical page pointer on the given offset */
|
|
static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
|
|
unsigned long offset)
|
|
{
|
|
return vmalloc_to_page(substream->runtime->dma_area + offset);
|
|
}
|
|
|
|
/* operators */
|
|
static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
|
|
.open = snd_sgio2audio_playback1_open,
|
|
.close = snd_sgio2audio_pcm_close,
|
|
.ioctl = snd_pcm_lib_ioctl,
|
|
.hw_params = snd_sgio2audio_pcm_hw_params,
|
|
.hw_free = snd_sgio2audio_pcm_hw_free,
|
|
.prepare = snd_sgio2audio_pcm_prepare,
|
|
.trigger = snd_sgio2audio_pcm_trigger,
|
|
.pointer = snd_sgio2audio_pcm_pointer,
|
|
.page = snd_sgio2audio_page,
|
|
};
|
|
|
|
static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
|
|
.open = snd_sgio2audio_playback2_open,
|
|
.close = snd_sgio2audio_pcm_close,
|
|
.ioctl = snd_pcm_lib_ioctl,
|
|
.hw_params = snd_sgio2audio_pcm_hw_params,
|
|
.hw_free = snd_sgio2audio_pcm_hw_free,
|
|
.prepare = snd_sgio2audio_pcm_prepare,
|
|
.trigger = snd_sgio2audio_pcm_trigger,
|
|
.pointer = snd_sgio2audio_pcm_pointer,
|
|
.page = snd_sgio2audio_page,
|
|
};
|
|
|
|
static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
|
|
.open = snd_sgio2audio_capture_open,
|
|
.close = snd_sgio2audio_pcm_close,
|
|
.ioctl = snd_pcm_lib_ioctl,
|
|
.hw_params = snd_sgio2audio_pcm_hw_params,
|
|
.hw_free = snd_sgio2audio_pcm_hw_free,
|
|
.prepare = snd_sgio2audio_pcm_prepare,
|
|
.trigger = snd_sgio2audio_pcm_trigger,
|
|
.pointer = snd_sgio2audio_pcm_pointer,
|
|
.page = snd_sgio2audio_page,
|
|
};
|
|
|
|
/*
|
|
* definitions of capture are omitted here...
|
|
*/
|
|
|
|
/* create a pcm device */
|
|
static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
|
|
{
|
|
struct snd_pcm *pcm;
|
|
int err;
|
|
|
|
/* create first pcm device with one outputs and one input */
|
|
err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
|
|
if (err < 0)
|
|
return err;
|
|
|
|
pcm->private_data = chip;
|
|
strcpy(pcm->name, "SGI O2 DAC1");
|
|
|
|
/* set operators */
|
|
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
|
|
&snd_sgio2audio_playback1_ops);
|
|
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
|
|
&snd_sgio2audio_capture_ops);
|
|
|
|
/* create second pcm device with one outputs and no input */
|
|
err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
|
|
if (err < 0)
|
|
return err;
|
|
|
|
pcm->private_data = chip;
|
|
strcpy(pcm->name, "SGI O2 DAC2");
|
|
|
|
/* set operators */
|
|
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
|
|
&snd_sgio2audio_playback2_ops);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct {
|
|
int idx;
|
|
int irq;
|
|
irqreturn_t (*isr)(int, void *);
|
|
const char *desc;
|
|
} snd_sgio2_isr_table[] = {
|
|
{
|
|
.idx = 0,
|
|
.irq = MACEISA_AUDIO1_DMAT_IRQ,
|
|
.isr = snd_sgio2audio_dma_in_isr,
|
|
.desc = "Capture DMA Channel 0"
|
|
}, {
|
|
.idx = 0,
|
|
.irq = MACEISA_AUDIO1_OF_IRQ,
|
|
.isr = snd_sgio2audio_error_isr,
|
|
.desc = "Capture Overflow"
|
|
}, {
|
|
.idx = 1,
|
|
.irq = MACEISA_AUDIO2_DMAT_IRQ,
|
|
.isr = snd_sgio2audio_dma_out_isr,
|
|
.desc = "Playback DMA Channel 1"
|
|
}, {
|
|
.idx = 1,
|
|
.irq = MACEISA_AUDIO2_MERR_IRQ,
|
|
.isr = snd_sgio2audio_error_isr,
|
|
.desc = "Memory Error Channel 1"
|
|
}, {
|
|
.idx = 2,
|
|
.irq = MACEISA_AUDIO3_DMAT_IRQ,
|
|
.isr = snd_sgio2audio_dma_out_isr,
|
|
.desc = "Playback DMA Channel 2"
|
|
}, {
|
|
.idx = 2,
|
|
.irq = MACEISA_AUDIO3_MERR_IRQ,
|
|
.isr = snd_sgio2audio_error_isr,
|
|
.desc = "Memory Error Channel 2"
|
|
}
|
|
};
|
|
|
|
/* ALSA driver */
|
|
|
|
static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
|
|
{
|
|
int i;
|
|
|
|
/* reset interface */
|
|
writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
|
|
udelay(1);
|
|
writeq(0, &mace->perif.audio.control);
|
|
|
|
/* release IRQ's */
|
|
for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
|
|
free_irq(snd_sgio2_isr_table[i].irq,
|
|
&chip->channel[snd_sgio2_isr_table[i].idx]);
|
|
|
|
dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
|
|
chip->ring_base, chip->ring_base_dma);
|
|
|
|
/* release card data */
|
|
kfree(chip);
|
|
return 0;
|
|
}
|
|
|
|
static int snd_sgio2audio_dev_free(struct snd_device *device)
|
|
{
|
|
struct snd_sgio2audio *chip = device->device_data;
|
|
|
|
return snd_sgio2audio_free(chip);
|
|
}
|
|
|
|
static struct snd_device_ops ops = {
|
|
.dev_free = snd_sgio2audio_dev_free,
|
|
};
|
|
|
|
static int __devinit snd_sgio2audio_create(struct snd_card *card,
|
|
struct snd_sgio2audio **rchip)
|
|
{
|
|
struct snd_sgio2audio *chip;
|
|
int i, err;
|
|
|
|
*rchip = NULL;
|
|
|
|
/* check if a codec is attached to the interface */
|
|
/* (Audio or Audio/Video board present) */
|
|
if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
|
|
return -ENOENT;
|
|
|
|
chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
|
|
if (chip == NULL)
|
|
return -ENOMEM;
|
|
|
|
chip->card = card;
|
|
|
|
chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
|
|
&chip->ring_base_dma, GFP_USER);
|
|
if (chip->ring_base == NULL) {
|
|
printk(KERN_ERR
|
|
"sgio2audio: could not allocate ring buffers\n");
|
|
kfree(chip);
|
|
return -ENOMEM;
|
|
}
|
|
|
|
spin_lock_init(&chip->ad1843_lock);
|
|
|
|
/* initialize channels */
|
|
for (i = 0; i < 3; i++) {
|
|
spin_lock_init(&chip->channel[i].lock);
|
|
chip->channel[i].idx = i;
|
|
}
|
|
|
|
/* allocate IRQs */
|
|
for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
|
|
if (request_irq(snd_sgio2_isr_table[i].irq,
|
|
snd_sgio2_isr_table[i].isr,
|
|
0,
|
|
snd_sgio2_isr_table[i].desc,
|
|
&chip->channel[snd_sgio2_isr_table[i].idx])) {
|
|
snd_sgio2audio_free(chip);
|
|
printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
|
|
snd_sgio2_isr_table[i].irq);
|
|
return -EBUSY;
|
|
}
|
|
}
|
|
|
|
/* reset the interface */
|
|
writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
|
|
udelay(1);
|
|
writeq(0, &mace->perif.audio.control);
|
|
msleep_interruptible(1); /* give time to recover */
|
|
|
|
/* set ring base */
|
|
writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
|
|
|
|
/* attach the AD1843 codec */
|
|
chip->ad1843.read = read_ad1843_reg;
|
|
chip->ad1843.write = write_ad1843_reg;
|
|
chip->ad1843.chip = chip;
|
|
|
|
/* initialize the AD1843 codec */
|
|
err = ad1843_init(&chip->ad1843);
|
|
if (err < 0) {
|
|
snd_sgio2audio_free(chip);
|
|
return err;
|
|
}
|
|
|
|
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
|
|
if (err < 0) {
|
|
snd_sgio2audio_free(chip);
|
|
return err;
|
|
}
|
|
*rchip = chip;
|
|
return 0;
|
|
}
|
|
|
|
static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
|
|
{
|
|
struct snd_card *card;
|
|
struct snd_sgio2audio *chip;
|
|
int err;
|
|
|
|
err = snd_card_create(index, id, THIS_MODULE, 0, &card);
|
|
if (err < 0)
|
|
return err;
|
|
|
|
err = snd_sgio2audio_create(card, &chip);
|
|
if (err < 0) {
|
|
snd_card_free(card);
|
|
return err;
|
|
}
|
|
snd_card_set_dev(card, &pdev->dev);
|
|
|
|
err = snd_sgio2audio_new_pcm(chip);
|
|
if (err < 0) {
|
|
snd_card_free(card);
|
|
return err;
|
|
}
|
|
err = snd_sgio2audio_new_mixer(chip);
|
|
if (err < 0) {
|
|
snd_card_free(card);
|
|
return err;
|
|
}
|
|
|
|
strcpy(card->driver, "SGI O2 Audio");
|
|
strcpy(card->shortname, "SGI O2 Audio");
|
|
sprintf(card->longname, "%s irq %i-%i",
|
|
card->shortname,
|
|
MACEISA_AUDIO1_DMAT_IRQ,
|
|
MACEISA_AUDIO3_MERR_IRQ);
|
|
|
|
err = snd_card_register(card);
|
|
if (err < 0) {
|
|
snd_card_free(card);
|
|
return err;
|
|
}
|
|
platform_set_drvdata(pdev, card);
|
|
return 0;
|
|
}
|
|
|
|
static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
|
|
{
|
|
struct snd_card *card = platform_get_drvdata(pdev);
|
|
|
|
snd_card_free(card);
|
|
platform_set_drvdata(pdev, NULL);
|
|
return 0;
|
|
}
|
|
|
|
static struct platform_driver sgio2audio_driver = {
|
|
.probe = snd_sgio2audio_probe,
|
|
.remove = __devexit_p(snd_sgio2audio_remove),
|
|
.driver = {
|
|
.name = "sgio2audio",
|
|
.owner = THIS_MODULE,
|
|
}
|
|
};
|
|
|
|
static int __init alsa_card_sgio2audio_init(void)
|
|
{
|
|
return platform_driver_register(&sgio2audio_driver);
|
|
}
|
|
|
|
static void __exit alsa_card_sgio2audio_exit(void)
|
|
{
|
|
platform_driver_unregister(&sgio2audio_driver);
|
|
}
|
|
|
|
module_init(alsa_card_sgio2audio_init)
|
|
module_exit(alsa_card_sgio2audio_exit)
|