mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-11-24 22:00:53 +07:00
f5a246eab9
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
183 lines
5.4 KiB
C
183 lines
5.4 KiB
C
#ifndef __SOUND_AD1816A_H
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#define __SOUND_AD1816A_H
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/*
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ad1816a.h - definitions for ADI SoundPort AD1816A chip.
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Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
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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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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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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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#include <sound/control.h>
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#include <sound/pcm.h>
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#include <sound/timer.h>
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#define AD1816A_REG(r) (chip->port + r)
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#define AD1816A_CHIP_STATUS 0x00
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#define AD1816A_INDIR_ADDR 0x00
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#define AD1816A_INTERRUPT_STATUS 0x01
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#define AD1816A_INDIR_DATA_LOW 0x02
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#define AD1816A_INDIR_DATA_HIGH 0x03
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#define AD1816A_PIO_DEBUG 0x04
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#define AD1816A_PIO_STATUS 0x05
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#define AD1816A_PIO_DATA 0x06
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#define AD1816A_RESERVED_7 0x07
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#define AD1816A_PLAYBACK_CONFIG 0x08
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#define AD1816A_CAPTURE_CONFIG 0x09
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#define AD1816A_RESERVED_10 0x0a
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#define AD1816A_RESERVED_11 0x0b
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#define AD1816A_JOYSTICK_RAW_DATA 0x0c
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#define AD1816A_JOYSTICK_CTRL 0x0d
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#define AD1816A_JOY_POS_DATA_LOW 0x0e
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#define AD1816A_JOY_POS_DATA_HIGH 0x0f
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#define AD1816A_LOW_BYTE_TMP 0x00
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#define AD1816A_INTERRUPT_ENABLE 0x01
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#define AD1816A_EXTERNAL_CTRL 0x01
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#define AD1816A_PLAYBACK_SAMPLE_RATE 0x02
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#define AD1816A_CAPTURE_SAMPLE_RATE 0x03
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#define AD1816A_VOICE_ATT 0x04
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#define AD1816A_FM_ATT 0x05
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#define AD1816A_I2S_1_ATT 0x06
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#define AD1816A_I2S_0_ATT 0x07
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#define AD1816A_PLAYBACK_BASE_COUNT 0x08
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#define AD1816A_PLAYBACK_CURR_COUNT 0x09
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#define AD1816A_CAPTURE_BASE_COUNT 0x0a
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#define AD1816A_CAPTURE_CURR_COUNT 0x0b
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#define AD1816A_TIMER_BASE_COUNT 0x0c
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#define AD1816A_TIMER_CURR_COUNT 0x0d
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#define AD1816A_MASTER_ATT 0x0e
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#define AD1816A_CD_GAIN_ATT 0x0f
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#define AD1816A_SYNTH_GAIN_ATT 0x10
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#define AD1816A_VID_GAIN_ATT 0x11
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#define AD1816A_LINE_GAIN_ATT 0x12
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#define AD1816A_MIC_GAIN_ATT 0x13
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#define AD1816A_PHONE_IN_GAIN_ATT 0x13
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#define AD1816A_ADC_SOURCE_SEL 0x14
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#define AD1816A_ADC_PGA 0x14
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#define AD1816A_CHIP_CONFIG 0x20
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#define AD1816A_DSP_CONFIG 0x21
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#define AD1816A_FM_SAMPLE_RATE 0x22
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#define AD1816A_I2S_1_SAMPLE_RATE 0x23
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#define AD1816A_I2S_0_SAMPLE_RATE 0x24
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#define AD1816A_RESERVED_37 0x25
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#define AD1816A_PROGRAM_CLOCK_RATE 0x26
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#define AD1816A_3D_PHAT_CTRL 0x27
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#define AD1816A_PHONE_OUT_ATT 0x27
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#define AD1816A_RESERVED_40 0x28
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#define AD1816A_HW_VOL_BUT 0x29
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#define AD1816A_DSP_MAILBOX_0 0x2a
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#define AD1816A_DSP_MAILBOX_1 0x2b
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#define AD1816A_POWERDOWN_CTRL 0x2c
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#define AD1816A_TIMER_CTRL 0x2c
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#define AD1816A_VERSION_ID 0x2d
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#define AD1816A_RESERVED_46 0x2e
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#define AD1816A_READY 0x80
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#define AD1816A_PLAYBACK_IRQ_PENDING 0x80
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#define AD1816A_CAPTURE_IRQ_PENDING 0x40
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#define AD1816A_TIMER_IRQ_PENDING 0x20
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#define AD1816A_PLAYBACK_ENABLE 0x01
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#define AD1816A_PLAYBACK_PIO 0x02
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#define AD1816A_CAPTURE_ENABLE 0x01
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#define AD1816A_CAPTURE_PIO 0x02
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#define AD1816A_FMT_LINEAR_8 0x00
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#define AD1816A_FMT_ULAW_8 0x08
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#define AD1816A_FMT_LINEAR_16_LIT 0x10
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#define AD1816A_FMT_ALAW_8 0x18
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#define AD1816A_FMT_LINEAR_16_BIG 0x30
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#define AD1816A_FMT_ALL 0x38
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#define AD1816A_FMT_STEREO 0x04
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#define AD1816A_PLAYBACK_IRQ_ENABLE 0x8000
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#define AD1816A_CAPTURE_IRQ_ENABLE 0x4000
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#define AD1816A_TIMER_IRQ_ENABLE 0x2000
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#define AD1816A_TIMER_ENABLE 0x0080
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#define AD1816A_SRC_LINE 0x00
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#define AD1816A_SRC_OUT 0x10
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#define AD1816A_SRC_CD 0x20
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#define AD1816A_SRC_SYNTH 0x30
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#define AD1816A_SRC_VIDEO 0x40
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#define AD1816A_SRC_MIC 0x50
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#define AD1816A_SRC_MONO 0x50
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#define AD1816A_SRC_PHONE_IN 0x60
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#define AD1816A_SRC_MASK 0x70
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#define AD1816A_CAPTURE_NOT_EQUAL 0x1000
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#define AD1816A_WSS_ENABLE 0x8000
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struct snd_ad1816a {
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unsigned long port;
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struct resource *res_port;
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int irq;
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int dma1;
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int dma2;
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unsigned short hardware;
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unsigned short version;
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spinlock_t lock;
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unsigned short mode;
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unsigned int clock_freq;
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struct snd_card *card;
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struct snd_pcm *pcm;
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struct snd_pcm_substream *playback_substream;
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struct snd_pcm_substream *capture_substream;
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unsigned int p_dma_size;
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unsigned int c_dma_size;
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struct snd_timer *timer;
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#ifdef CONFIG_PM
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unsigned short image[48];
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#endif
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};
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#define AD1816A_HW_AUTO 0
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#define AD1816A_HW_AD1816A 1
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#define AD1816A_HW_AD1815 2
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#define AD1816A_HW_AD18MAX10 3
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#define AD1816A_MODE_PLAYBACK 0x01
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#define AD1816A_MODE_CAPTURE 0x02
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#define AD1816A_MODE_TIMER 0x04
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#define AD1816A_MODE_OPEN (AD1816A_MODE_PLAYBACK | \
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AD1816A_MODE_CAPTURE | \
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AD1816A_MODE_TIMER)
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extern int snd_ad1816a_create(struct snd_card *card, unsigned long port,
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int irq, int dma1, int dma2,
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struct snd_ad1816a *chip);
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extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm);
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extern int snd_ad1816a_mixer(struct snd_ad1816a *chip);
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extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device,
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struct snd_timer **rtimer);
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#ifdef CONFIG_PM
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extern void snd_ad1816a_suspend(struct snd_ad1816a *chip);
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extern void snd_ad1816a_resume(struct snd_ad1816a *chip);
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#endif
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#endif /* __SOUND_AD1816A_H */
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