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https://github.com/joel16/android_kernel_sony_msm8994.git
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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. -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.19 (GNU/Linux) iQIcBAABAgAGBQJQcpeWAAoJEGwxgFQ9KSmkpi4P/2etDDz5aEkEHNa1l4xEmFcm ymiGTgjaalqpUAVbM/gYx9G59EFMEbzUl1BHAqE5La4wO/v9lNPb+VrdUo+B+NZ7 WSxIPWcNqdinSuoSqyYPjoPMVnhs3EMtNOqmf4jm1JOvdqA+4rO29xQVAqK/5Gfu LpMOyPiRi5ODnbQ1BOIWwpKICioY/mLwGJudK3z0i/fYVA7gLub20f+w+sOjKIA4 wmwQAMTjAR798Cg/tVy4fQmf4SLw+c2nIgGe/PD+2gVlGXLNKBrJfMonHPTbmwKu lmJO/EtnijNOnpbn6up7ryUQ9cSoZAUZOfdIOgmAeQgQ/LWR0f+zf2IQehSPwrul g6hqOnQI2DNN7ugT3cYVbYnsh56TjyhnxhhxZgkapqh706QkqHGyKJNMRetzuXmP 1O//MnZJrFQWd6sOKLlTL2ZzRvnxEJcNVGaE6bbwZTfQMtPeo9l1842uIq1dLUtG VxZb/svKUkMXv4is1dwUYUkpDsKxsgMEmabmuovceGf2N7jj/irkXgqxf6LWkaY1 JQ7ZFWUJyDzEMXRaFfzdGO15T532CfB84wvFX5xoPMwMste2AA7QuybFBVstXhKu AtKNDgRJFUTlnLIxydpPBWdWH3UJdEaFwwsSfuNKI8OmmGKhWC/aP83k4hzueu9H KYLvY/0ObMSMqiwh/ndQ =uNqD -----END PGP SIGNATURE----- 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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