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Initial commit; kernel source import

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Nathan
2025-04-06 23:50:55 -05:00
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#ifndef __SOUND_AC97_CODEC_H
#define __SOUND_AC97_CODEC_H
/*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.1
* by Intel Corporation (http://developer.intel.com).
*
*
* 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 <linux/bitops.h>
#include <linux/device.h>
#include <linux/workqueue.h>
#include <sound/pcm.h>
#include <sound/control.h>
#include <sound/info.h>
/* maximum number of devices on the AC97 bus */
#define AC97_BUS_MAX_DEVICES 4
/*
* AC'97 codec registers
*/
#define AC97_RESET 0x00 /* Reset */
#define AC97_MASTER 0x02 /* Master Volume */
#define AC97_HEADPHONE 0x04 /* Headphone Volume (optional) */
#define AC97_MASTER_MONO 0x06 /* Master Volume Mono (optional) */
#define AC97_MASTER_TONE 0x08 /* Master Tone (Bass & Treble) (optional) */
#define AC97_PC_BEEP 0x0a /* PC Beep Volume (optinal) */
#define AC97_PHONE 0x0c /* Phone Volume (optional) */
#define AC97_MIC 0x0e /* MIC Volume */
#define AC97_LINE 0x10 /* Line In Volume */
#define AC97_CD 0x12 /* CD Volume */
#define AC97_VIDEO 0x14 /* Video Volume (optional) */
#define AC97_AUX 0x16 /* AUX Volume (optional) */
#define AC97_PCM 0x18 /* PCM Volume */
#define AC97_REC_SEL 0x1a /* Record Select */
#define AC97_REC_GAIN 0x1c /* Record Gain */
#define AC97_REC_GAIN_MIC 0x1e /* Record Gain MIC (optional) */
#define AC97_GENERAL_PURPOSE 0x20 /* General Purpose (optional) */
#define AC97_3D_CONTROL 0x22 /* 3D Control (optional) */
#define AC97_INT_PAGING 0x24 /* Audio Interrupt & Paging (AC'97 2.3) */
#define AC97_POWERDOWN 0x26 /* Powerdown control / status */
/* range 0x28-0x3a - AUDIO AC'97 2.0 extensions */
#define AC97_EXTENDED_ID 0x28 /* Extended Audio ID */
#define AC97_EXTENDED_STATUS 0x2a /* Extended Audio Status and Control */
#define AC97_PCM_FRONT_DAC_RATE 0x2c /* PCM Front DAC Rate */
#define AC97_PCM_SURR_DAC_RATE 0x2e /* PCM Surround DAC Rate */
#define AC97_PCM_LFE_DAC_RATE 0x30 /* PCM LFE DAC Rate */
#define AC97_PCM_LR_ADC_RATE 0x32 /* PCM LR ADC Rate */
#define AC97_PCM_MIC_ADC_RATE 0x34 /* PCM MIC ADC Rate */
#define AC97_CENTER_LFE_MASTER 0x36 /* Center + LFE Master Volume */
#define AC97_SURROUND_MASTER 0x38 /* Surround (Rear) Master Volume */
#define AC97_SPDIF 0x3a /* S/PDIF control */
/* range 0x3c-0x58 - MODEM */
#define AC97_EXTENDED_MID 0x3c /* Extended Modem ID */
#define AC97_EXTENDED_MSTATUS 0x3e /* Extended Modem Status and Control */
#define AC97_LINE1_RATE 0x40 /* Line1 DAC/ADC Rate */
#define AC97_LINE2_RATE 0x42 /* Line2 DAC/ADC Rate */
#define AC97_HANDSET_RATE 0x44 /* Handset DAC/ADC Rate */
#define AC97_LINE1_LEVEL 0x46 /* Line1 DAC/ADC Level */
#define AC97_LINE2_LEVEL 0x48 /* Line2 DAC/ADC Level */
#define AC97_HANDSET_LEVEL 0x4a /* Handset DAC/ADC Level */
#define AC97_GPIO_CFG 0x4c /* GPIO Configuration */
#define AC97_GPIO_POLARITY 0x4e /* GPIO Pin Polarity/Type, 0=low, 1=high active */
#define AC97_GPIO_STICKY 0x50 /* GPIO Pin Sticky, 0=not, 1=sticky */
#define AC97_GPIO_WAKEUP 0x52 /* GPIO Pin Wakeup, 0=no int, 1=yes int */
#define AC97_GPIO_STATUS 0x54 /* GPIO Pin Status, slot 12 */
#define AC97_MISC_AFE 0x56 /* Miscellaneous Modem AFE Status and Control */
/* range 0x5a-0x7b - Vendor Specific */
#define AC97_VENDOR_ID1 0x7c /* Vendor ID1 */
#define AC97_VENDOR_ID2 0x7e /* Vendor ID2 / revision */
/* range 0x60-0x6f (page 1) - extended codec registers */
#define AC97_CODEC_CLASS_REV 0x60 /* Codec Class/Revision */
#define AC97_PCI_SVID 0x62 /* PCI Subsystem Vendor ID */
#define AC97_PCI_SID 0x64 /* PCI Subsystem ID */
#define AC97_FUNC_SELECT 0x66 /* Function Select */
#define AC97_FUNC_INFO 0x68 /* Function Information */
#define AC97_SENSE_INFO 0x6a /* Sense Details */
/* volume controls */
#define AC97_MUTE_MASK_MONO 0x8000
#define AC97_MUTE_MASK_STEREO 0x8080
/* slot allocation */
#define AC97_SLOT_TAG 0
#define AC97_SLOT_CMD_ADDR 1
#define AC97_SLOT_CMD_DATA 2
#define AC97_SLOT_PCM_LEFT 3
#define AC97_SLOT_PCM_RIGHT 4
#define AC97_SLOT_MODEM_LINE1 5
#define AC97_SLOT_PCM_CENTER 6
#define AC97_SLOT_MIC 6 /* input */
#define AC97_SLOT_SPDIF_LEFT1 6
#define AC97_SLOT_PCM_SLEFT 7 /* surround left */
#define AC97_SLOT_PCM_LEFT_0 7 /* double rate operation */
#define AC97_SLOT_SPDIF_LEFT 7
#define AC97_SLOT_PCM_SRIGHT 8 /* surround right */
#define AC97_SLOT_PCM_RIGHT_0 8 /* double rate operation */
#define AC97_SLOT_SPDIF_RIGHT 8
#define AC97_SLOT_LFE 9
#define AC97_SLOT_SPDIF_RIGHT1 9
#define AC97_SLOT_MODEM_LINE2 10
#define AC97_SLOT_PCM_LEFT_1 10 /* double rate operation */
#define AC97_SLOT_SPDIF_LEFT2 10
#define AC97_SLOT_HANDSET 11 /* output */
#define AC97_SLOT_PCM_RIGHT_1 11 /* double rate operation */
#define AC97_SLOT_SPDIF_RIGHT2 11
#define AC97_SLOT_MODEM_GPIO 12 /* modem GPIO */
#define AC97_SLOT_PCM_CENTER_1 12 /* double rate operation */
/* basic capabilities (reset register) */
#define AC97_BC_DEDICATED_MIC 0x0001 /* Dedicated Mic PCM In Channel */
#define AC97_BC_RESERVED1 0x0002 /* Reserved (was Modem Line Codec support) */
#define AC97_BC_BASS_TREBLE 0x0004 /* Bass & Treble Control */
#define AC97_BC_SIM_STEREO 0x0008 /* Simulated stereo */
#define AC97_BC_HEADPHONE 0x0010 /* Headphone Out Support */
#define AC97_BC_LOUDNESS 0x0020 /* Loudness (bass boost) Support */
#define AC97_BC_16BIT_DAC 0x0000 /* 16-bit DAC resolution */
#define AC97_BC_18BIT_DAC 0x0040 /* 18-bit DAC resolution */
#define AC97_BC_20BIT_DAC 0x0080 /* 20-bit DAC resolution */
#define AC97_BC_DAC_MASK 0x00c0
#define AC97_BC_16BIT_ADC 0x0000 /* 16-bit ADC resolution */
#define AC97_BC_18BIT_ADC 0x0100 /* 18-bit ADC resolution */
#define AC97_BC_20BIT_ADC 0x0200 /* 20-bit ADC resolution */
#define AC97_BC_ADC_MASK 0x0300
#define AC97_BC_3D_TECH_ID_MASK 0x7c00 /* Per-vendor ID of 3D enhancement */
/* general purpose */
#define AC97_GP_DRSS_MASK 0x0c00 /* double rate slot select */
#define AC97_GP_DRSS_1011 0x0000 /* LR(C) 10+11(+12) */
#define AC97_GP_DRSS_78 0x0400 /* LR 7+8 */
/* powerdown bits */
#define AC97_PD_ADC_STATUS 0x0001 /* ADC status (RO) */
#define AC97_PD_DAC_STATUS 0x0002 /* DAC status (RO) */
#define AC97_PD_MIXER_STATUS 0x0004 /* Analog mixer status (RO) */
#define AC97_PD_VREF_STATUS 0x0008 /* Vref status (RO) */
#define AC97_PD_PR0 0x0100 /* Power down PCM ADCs and input MUX */
#define AC97_PD_PR1 0x0200 /* Power down PCM front DAC */
#define AC97_PD_PR2 0x0400 /* Power down Mixer (Vref still on) */
#define AC97_PD_PR3 0x0800 /* Power down Mixer (Vref off) */
#define AC97_PD_PR4 0x1000 /* Power down AC-Link */
#define AC97_PD_PR5 0x2000 /* Disable internal clock usage */
#define AC97_PD_PR6 0x4000 /* Headphone amplifier */
#define AC97_PD_EAPD 0x8000 /* External Amplifer Power Down (EAPD) */
/* extended audio ID bit defines */
#define AC97_EI_VRA 0x0001 /* Variable bit rate supported */
#define AC97_EI_DRA 0x0002 /* Double rate supported */
#define AC97_EI_SPDIF 0x0004 /* S/PDIF out supported */
#define AC97_EI_VRM 0x0008 /* Variable bit rate supported for MIC */
#define AC97_EI_DACS_SLOT_MASK 0x0030 /* DACs slot assignment */
#define AC97_EI_DACS_SLOT_SHIFT 4
#define AC97_EI_CDAC 0x0040 /* PCM Center DAC available */
#define AC97_EI_SDAC 0x0080 /* PCM Surround DACs available */
#define AC97_EI_LDAC 0x0100 /* PCM LFE DAC available */
#define AC97_EI_AMAP 0x0200 /* indicates optional slot/DAC mapping based on codec ID */
#define AC97_EI_REV_MASK 0x0c00 /* AC'97 revision mask */
#define AC97_EI_REV_22 0x0400 /* AC'97 revision 2.2 */
#define AC97_EI_REV_23 0x0800 /* AC'97 revision 2.3 */
#define AC97_EI_REV_SHIFT 10
#define AC97_EI_ADDR_MASK 0xc000 /* physical codec ID (address) */
#define AC97_EI_ADDR_SHIFT 14
/* extended audio status and control bit defines */
#define AC97_EA_VRA 0x0001 /* Variable bit rate enable bit */
#define AC97_EA_DRA 0x0002 /* Double-rate audio enable bit */
#define AC97_EA_SPDIF 0x0004 /* S/PDIF out enable bit */
#define AC97_EA_VRM 0x0008 /* Variable bit rate for MIC enable bit */
#define AC97_EA_SPSA_SLOT_MASK 0x0030 /* Mask for slot assignment bits */
#define AC97_EA_SPSA_SLOT_SHIFT 4
#define AC97_EA_SPSA_3_4 0x0000 /* Slot assigned to 3 & 4 */
#define AC97_EA_SPSA_7_8 0x0010 /* Slot assigned to 7 & 8 */
#define AC97_EA_SPSA_6_9 0x0020 /* Slot assigned to 6 & 9 */
#define AC97_EA_SPSA_10_11 0x0030 /* Slot assigned to 10 & 11 */
#define AC97_EA_CDAC 0x0040 /* PCM Center DAC is ready (Read only) */
#define AC97_EA_SDAC 0x0080 /* PCM Surround DACs are ready (Read only) */
#define AC97_EA_LDAC 0x0100 /* PCM LFE DAC is ready (Read only) */
#define AC97_EA_MDAC 0x0200 /* MIC ADC is ready (Read only) */
#define AC97_EA_SPCV 0x0400 /* S/PDIF configuration valid (Read only) */
#define AC97_EA_PRI 0x0800 /* Turns the PCM Center DAC off */
#define AC97_EA_PRJ 0x1000 /* Turns the PCM Surround DACs off */
#define AC97_EA_PRK 0x2000 /* Turns the PCM LFE DAC off */
#define AC97_EA_PRL 0x4000 /* Turns the MIC ADC off */
/* S/PDIF control bit defines */
#define AC97_SC_PRO 0x0001 /* Professional status */
#define AC97_SC_NAUDIO 0x0002 /* Non audio stream */
#define AC97_SC_COPY 0x0004 /* Copyright status */
#define AC97_SC_PRE 0x0008 /* Preemphasis status */
#define AC97_SC_CC_MASK 0x07f0 /* Category Code mask */
#define AC97_SC_CC_SHIFT 4
#define AC97_SC_L 0x0800 /* Generation Level status */
#define AC97_SC_SPSR_MASK 0x3000 /* S/PDIF Sample Rate bits */
#define AC97_SC_SPSR_SHIFT 12
#define AC97_SC_SPSR_44K 0x0000 /* Use 44.1kHz Sample rate */
#define AC97_SC_SPSR_48K 0x2000 /* Use 48kHz Sample rate */
#define AC97_SC_SPSR_32K 0x3000 /* Use 32kHz Sample rate */
#define AC97_SC_DRS 0x4000 /* Double Rate S/PDIF */
#define AC97_SC_V 0x8000 /* Validity status */
/* Interrupt and Paging bit defines (AC'97 2.3) */
#define AC97_PAGE_MASK 0x000f /* Page Selector */
#define AC97_PAGE_VENDOR 0 /* Vendor-specific registers */
#define AC97_PAGE_1 1 /* Extended Codec Registers page 1 */
#define AC97_INT_ENABLE 0x0800 /* Interrupt Enable */
#define AC97_INT_SENSE 0x1000 /* Sense Cycle */
#define AC97_INT_CAUSE_SENSE 0x2000 /* Sense Cycle Completed (RO) */
#define AC97_INT_CAUSE_GPIO 0x4000 /* GPIO bits changed (RO) */
#define AC97_INT_STATUS 0x8000 /* Interrupt Status */
/* extended modem ID bit defines */
#define AC97_MEI_LINE1 0x0001 /* Line1 present */
#define AC97_MEI_LINE2 0x0002 /* Line2 present */
#define AC97_MEI_HANDSET 0x0004 /* Handset present */
#define AC97_MEI_CID1 0x0008 /* caller ID decode for Line1 is supported */
#define AC97_MEI_CID2 0x0010 /* caller ID decode for Line2 is supported */
#define AC97_MEI_ADDR_MASK 0xc000 /* physical codec ID (address) */
#define AC97_MEI_ADDR_SHIFT 14
/* extended modem status and control bit defines */
#define AC97_MEA_GPIO 0x0001 /* GPIO is ready (ro) */
#define AC97_MEA_MREF 0x0002 /* Vref is up to nominal level (ro) */
#define AC97_MEA_ADC1 0x0004 /* ADC1 operational (ro) */
#define AC97_MEA_DAC1 0x0008 /* DAC1 operational (ro) */
#define AC97_MEA_ADC2 0x0010 /* ADC2 operational (ro) */
#define AC97_MEA_DAC2 0x0020 /* DAC2 operational (ro) */
#define AC97_MEA_HADC 0x0040 /* HADC operational (ro) */
#define AC97_MEA_HDAC 0x0080 /* HDAC operational (ro) */
#define AC97_MEA_PRA 0x0100 /* GPIO power down (high) */
#define AC97_MEA_PRB 0x0200 /* reserved */
#define AC97_MEA_PRC 0x0400 /* ADC1 power down (high) */
#define AC97_MEA_PRD 0x0800 /* DAC1 power down (high) */
#define AC97_MEA_PRE 0x1000 /* ADC2 power down (high) */
#define AC97_MEA_PRF 0x2000 /* DAC2 power down (high) */
#define AC97_MEA_PRG 0x4000 /* HADC power down (high) */
#define AC97_MEA_PRH 0x8000 /* HDAC power down (high) */
/* modem gpio status defines */
#define AC97_GPIO_LINE1_OH 0x0001 /* Off Hook Line1 */
#define AC97_GPIO_LINE1_RI 0x0002 /* Ring Detect Line1 */
#define AC97_GPIO_LINE1_CID 0x0004 /* Caller ID path enable Line1 */
#define AC97_GPIO_LINE1_LCS 0x0008 /* Loop Current Sense Line1 */
#define AC97_GPIO_LINE1_PULSE 0x0010 /* Opt./ Pulse Dial Line1 (out) */
#define AC97_GPIO_LINE1_HL1R 0x0020 /* Opt./ Handset to Line1 relay control (out) */
#define AC97_GPIO_LINE1_HOHD 0x0040 /* Opt./ Handset off hook detect Line1 (in) */
#define AC97_GPIO_LINE12_AC 0x0080 /* Opt./ Int.bit 1 / Line1/2 AC (out) */
#define AC97_GPIO_LINE12_DC 0x0100 /* Opt./ Int.bit 2 / Line1/2 DC (out) */
#define AC97_GPIO_LINE12_RS 0x0200 /* Opt./ Int.bit 3 / Line1/2 RS (out) */
#define AC97_GPIO_LINE2_OH 0x0400 /* Off Hook Line2 */
#define AC97_GPIO_LINE2_RI 0x0800 /* Ring Detect Line2 */
#define AC97_GPIO_LINE2_CID 0x1000 /* Caller ID path enable Line2 */
#define AC97_GPIO_LINE2_LCS 0x2000 /* Loop Current Sense Line2 */
#define AC97_GPIO_LINE2_PULSE 0x4000 /* Opt./ Pulse Dial Line2 (out) */
#define AC97_GPIO_LINE2_HL1R 0x8000 /* Opt./ Handset to Line2 relay control (out) */
/* specific - SigmaTel */
#define AC97_SIGMATEL_OUTSEL 0x64 /* Output Select, STAC9758 */
#define AC97_SIGMATEL_INSEL 0x66 /* Input Select, STAC9758 */
#define AC97_SIGMATEL_IOMISC 0x68 /* STAC9758 */
#define AC97_SIGMATEL_ANALOG 0x6c /* Analog Special */
#define AC97_SIGMATEL_DAC2INVERT 0x6e
#define AC97_SIGMATEL_BIAS1 0x70
#define AC97_SIGMATEL_BIAS2 0x72
#define AC97_SIGMATEL_VARIOUS 0x72 /* STAC9758 */
#define AC97_SIGMATEL_MULTICHN 0x74 /* Multi-Channel programming */
#define AC97_SIGMATEL_CIC1 0x76
#define AC97_SIGMATEL_CIC2 0x78
/* specific - Analog Devices */
#define AC97_AD_TEST 0x5a /* test register */
#define AC97_AD_TEST2 0x5c /* undocumented test register 2 */
#define AC97_AD_HPFD_SHIFT 12 /* High Pass Filter Disable */
#define AC97_AD_CODEC_CFG 0x70 /* codec configuration */
#define AC97_AD_JACK_SPDIF 0x72 /* Jack Sense & S/PDIF */
#define AC97_AD_SERIAL_CFG 0x74 /* Serial Configuration */
#define AC97_AD_MISC 0x76 /* Misc Control Bits */
#define AC97_AD_VREFD_SHIFT 2 /* V_REFOUT Disable (AD1888) */
/* specific - Cirrus Logic */
#define AC97_CSR_ACMODE 0x5e /* AC Mode Register */
#define AC97_CSR_MISC_CRYSTAL 0x60 /* Misc Crystal Control */
#define AC97_CSR_SPDIF 0x68 /* S/PDIF Register */
#define AC97_CSR_SERIAL 0x6a /* Serial Port Control */
#define AC97_CSR_SPECF_ADDR 0x6c /* Special Feature Address */
#define AC97_CSR_SPECF_DATA 0x6e /* Special Feature Data */
#define AC97_CSR_BDI_STATUS 0x7a /* BDI Status */
/* specific - Conexant */
#define AC97_CXR_AUDIO_MISC 0x5c
#define AC97_CXR_SPDIFEN (1<<3)
#define AC97_CXR_COPYRGT (1<<2)
#define AC97_CXR_SPDIF_MASK (3<<0)
#define AC97_CXR_SPDIF_PCM 0x0
#define AC97_CXR_SPDIF_AC3 0x2
/* specific - ALC */
#define AC97_ALC650_SPDIF_INPUT_STATUS1 0x60
/* S/PDIF input status 1 bit defines */
#define AC97_ALC650_PRO 0x0001 /* Professional status */
#define AC97_ALC650_NAUDIO 0x0002 /* Non audio stream */
#define AC97_ALC650_COPY 0x0004 /* Copyright status */
#define AC97_ALC650_PRE 0x0038 /* Preemphasis status */
#define AC97_ALC650_PRE_SHIFT 3
#define AC97_ALC650_MODE 0x00C0 /* Preemphasis status */
#define AC97_ALC650_MODE_SHIFT 6
#define AC97_ALC650_CC_MASK 0x7f00 /* Category Code mask */
#define AC97_ALC650_CC_SHIFT 8
#define AC97_ALC650_L 0x8000 /* Generation Level status */
#define AC97_ALC650_SPDIF_INPUT_STATUS2 0x62
/* S/PDIF input status 2 bit defines */
#define AC97_ALC650_SOUCE_MASK 0x000f /* Source number */
#define AC97_ALC650_CHANNEL_MASK 0x00f0 /* Channel number */
#define AC97_ALC650_CHANNEL_SHIFT 4
#define AC97_ALC650_SPSR_MASK 0x0f00 /* S/PDIF Sample Rate bits */
#define AC97_ALC650_SPSR_SHIFT 8
#define AC97_ALC650_SPSR_44K 0x0000 /* Use 44.1kHz Sample rate */
#define AC97_ALC650_SPSR_48K 0x0200 /* Use 48kHz Sample rate */
#define AC97_ALC650_SPSR_32K 0x0300 /* Use 32kHz Sample rate */
#define AC97_ALC650_CLOCK_ACCURACY 0x3000 /* Clock accuracy */
#define AC97_ALC650_CLOCK_SHIFT 12
#define AC97_ALC650_CLOCK_LOCK 0x4000 /* Clock locked status */
#define AC97_ALC650_V 0x8000 /* Validity status */
#define AC97_ALC650_SURR_DAC_VOL 0x64
#define AC97_ALC650_LFE_DAC_VOL 0x66
#define AC97_ALC650_UNKNOWN1 0x68
#define AC97_ALC650_MULTICH 0x6a
#define AC97_ALC650_UNKNOWN2 0x6c
#define AC97_ALC650_REVISION 0x6e
#define AC97_ALC650_UNKNOWN3 0x70
#define AC97_ALC650_UNKNOWN4 0x72
#define AC97_ALC650_MISC 0x74
#define AC97_ALC650_GPIO_SETUP 0x76
#define AC97_ALC650_GPIO_STATUS 0x78
#define AC97_ALC650_CLOCK 0x7a
/* specific - Yamaha YMF7x3 */
#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */
#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */
/* specific - C-Media */
#define AC97_CM9738_VENDOR_CTRL 0x5a
#define AC97_CM9739_MULTI_CHAN 0x64
#define AC97_CM9739_SPDIF_IN_STATUS 0x68 /* 32bit */
#define AC97_CM9739_SPDIF_CTRL 0x6c
/* specific - wolfson */
#define AC97_WM97XX_FMIXER_VOL 0x72
#define AC97_WM9704_RMIXER_VOL 0x74
#define AC97_WM9704_TEST 0x5a
#define AC97_WM9704_RPCM_VOL 0x70
#define AC97_WM9711_OUT3VOL 0x16
/* ac97->scaps */
#define AC97_SCAP_AUDIO (1<<0) /* audio codec 97 */
#define AC97_SCAP_MODEM (1<<1) /* modem codec 97 */
#define AC97_SCAP_SURROUND_DAC (1<<2) /* surround L&R DACs are present */
#define AC97_SCAP_CENTER_LFE_DAC (1<<3) /* center and LFE DACs are present */
#define AC97_SCAP_SKIP_AUDIO (1<<4) /* skip audio part of codec */
#define AC97_SCAP_SKIP_MODEM (1<<5) /* skip modem part of codec */
#define AC97_SCAP_INDEP_SDIN (1<<6) /* independent SDIN */
#define AC97_SCAP_INV_EAPD (1<<7) /* inverted EAPD */
#define AC97_SCAP_DETECT_BY_VENDOR (1<<8) /* use vendor registers for read tests */
#define AC97_SCAP_NO_SPDIF (1<<9) /* don't build SPDIF controls */
#define AC97_SCAP_EAPD_LED (1<<10) /* EAPD as mute LED */
#define AC97_SCAP_POWER_SAVE (1<<11) /* capable for aggressive power-saving */
/* ac97->flags */
#define AC97_HAS_PC_BEEP (1<<0) /* force PC Speaker usage */
#define AC97_AD_MULTI (1<<1) /* Analog Devices - multi codecs */
#define AC97_CS_SPDIF (1<<2) /* Cirrus Logic uses funky SPDIF */
#define AC97_CX_SPDIF (1<<3) /* Conexant's spdif interface */
#define AC97_STEREO_MUTES (1<<4) /* has stereo mute bits */
#define AC97_DOUBLE_RATE (1<<5) /* supports double rate playback */
#define AC97_HAS_NO_MASTER_VOL (1<<6) /* no Master volume */
#define AC97_HAS_NO_PCM_VOL (1<<7) /* no PCM volume */
#define AC97_DEFAULT_POWER_OFF (1<<8) /* no RESET write */
#define AC97_MODEM_PATCH (1<<9) /* modem patch */
#define AC97_HAS_NO_REC_GAIN (1<<10) /* no Record gain */
#define AC97_HAS_NO_PHONE (1<<11) /* no PHONE volume */
#define AC97_HAS_NO_PC_BEEP (1<<12) /* no PC Beep volume */
#define AC97_HAS_NO_VIDEO (1<<13) /* no Video volume */
#define AC97_HAS_NO_CD (1<<14) /* no CD volume */
#define AC97_HAS_NO_MIC (1<<15) /* no MIC volume */
#define AC97_HAS_NO_TONE (1<<16) /* no Tone volume */
#define AC97_HAS_NO_STD_PCM (1<<17) /* no standard AC97 PCM volume and mute */
#define AC97_HAS_NO_AUX (1<<18) /* no standard AC97 AUX volume and mute */
#define AC97_HAS_8CH (1<<19) /* supports 8-channel output */
/* rates indexes */
#define AC97_RATES_FRONT_DAC 0
#define AC97_RATES_SURR_DAC 1
#define AC97_RATES_LFE_DAC 2
#define AC97_RATES_ADC 3
#define AC97_RATES_MIC_ADC 4
#define AC97_RATES_SPDIF 5
/*
*
*/
struct snd_ac97;
struct snd_pcm_chmap;
struct snd_ac97_build_ops {
int (*build_3d) (struct snd_ac97 *ac97);
int (*build_specific) (struct snd_ac97 *ac97);
int (*build_spdif) (struct snd_ac97 *ac97);
int (*build_post_spdif) (struct snd_ac97 *ac97);
#ifdef CONFIG_PM
void (*suspend) (struct snd_ac97 *ac97);
void (*resume) (struct snd_ac97 *ac97);
#endif
void (*update_jacks) (struct snd_ac97 *ac97); /* for jack-sharing */
};
struct snd_ac97_bus_ops {
void (*reset) (struct snd_ac97 *ac97);
void (*warm_reset)(struct snd_ac97 *ac97);
void (*write) (struct snd_ac97 *ac97, unsigned short reg, unsigned short val);
unsigned short (*read) (struct snd_ac97 *ac97, unsigned short reg);
void (*wait) (struct snd_ac97 *ac97);
void (*init) (struct snd_ac97 *ac97);
};
struct snd_ac97_bus {
/* -- lowlevel (hardware) driver specific -- */
struct snd_ac97_bus_ops *ops;
void *private_data;
void (*private_free) (struct snd_ac97_bus *bus);
/* --- */
struct snd_card *card;
unsigned short num; /* bus number */
unsigned short no_vra: 1, /* bridge doesn't support VRA */
dra: 1, /* bridge supports double rate */
isdin: 1;/* independent SDIN */
unsigned int clock; /* AC'97 base clock (usually 48000Hz) */
spinlock_t bus_lock; /* used mainly for slot allocation */
unsigned short used_slots[2][4]; /* actually used PCM slots */
unsigned short pcms_count; /* count of PCMs */
struct ac97_pcm *pcms;
struct snd_ac97 *codec[4];
struct snd_info_entry *proc;
};
/* static resolution table */
struct snd_ac97_res_table {
unsigned short reg; /* register */
unsigned short bits; /* resolution bitmask */
};
struct snd_ac97_template {
void *private_data;
void (*private_free) (struct snd_ac97 *ac97);
struct pci_dev *pci; /* assigned PCI device - used for quirks */
unsigned short num; /* number of codec: 0 = primary, 1 = secondary */
unsigned short addr; /* physical address of codec [0-3] */
unsigned int scaps; /* driver capabilities */
const struct snd_ac97_res_table *res_table; /* static resolution */
};
struct snd_ac97 {
/* -- lowlevel (hardware) driver specific -- */
const struct snd_ac97_build_ops *build_ops;
void *private_data;
void (*private_free) (struct snd_ac97 *ac97);
/* --- */
struct snd_ac97_bus *bus;
struct pci_dev *pci; /* assigned PCI device - used for quirks */
struct snd_info_entry *proc;
struct snd_info_entry *proc_regs;
unsigned short subsystem_vendor;
unsigned short subsystem_device;
struct mutex reg_mutex;
struct mutex page_mutex; /* mutex for AD18xx multi-codecs and paging (2.3) */
unsigned short num; /* number of codec: 0 = primary, 1 = secondary */
unsigned short addr; /* physical address of codec [0-3] */
unsigned int id; /* identification of codec */
unsigned short caps; /* capabilities (register 0) */
unsigned short ext_id; /* extended feature identification (register 28) */
unsigned short ext_mid; /* extended modem ID (register 3C) */
const struct snd_ac97_res_table *res_table; /* static resolution */
unsigned int scaps; /* driver capabilities */
unsigned int flags; /* specific code */
unsigned int rates[6]; /* see AC97_RATES_* defines */
unsigned int spdif_status;
unsigned short regs[0x80]; /* register cache */
DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */
union { /* vendor specific code */
struct {
unsigned short unchained[3]; // 0 = C34, 1 = C79, 2 = C69
unsigned short chained[3]; // 0 = C34, 1 = C79, 2 = C69
unsigned short id[3]; // codec IDs (lower 16-bit word)
unsigned short pcmreg[3]; // PCM registers
unsigned short codec_cfg[3]; // CODEC_CFG bits
unsigned char swap_mic_linein; // AD1986/AD1986A only
unsigned char lo_as_master; /* LO as master */
} ad18xx;
unsigned int dev_flags; /* device specific */
} spec;
/* jack-sharing info */
unsigned char indep_surround;
unsigned char channel_mode;
#ifdef CONFIG_SND_AC97_POWER_SAVE
unsigned int power_up; /* power states */
struct delayed_work power_work;
#endif
struct device dev;
struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */
};
#define to_ac97_t(d) container_of(d, struct snd_ac97, dev)
/* conditions */
static inline int ac97_is_audio(struct snd_ac97 * ac97)
{
return (ac97->scaps & AC97_SCAP_AUDIO);
}
static inline int ac97_is_modem(struct snd_ac97 * ac97)
{
return (ac97->scaps & AC97_SCAP_MODEM);
}
static inline int ac97_is_rev22(struct snd_ac97 * ac97)
{
return (ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_22;
}
static inline int ac97_can_amap(struct snd_ac97 * ac97)
{
return (ac97->ext_id & AC97_EI_AMAP) != 0;
}
static inline int ac97_can_spdif(struct snd_ac97 * ac97)
{
return (ac97->ext_id & AC97_EI_SPDIF) != 0;
}
/* functions */
/* create new AC97 bus */
int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops,
void *private_data, struct snd_ac97_bus **rbus);
/* create mixer controls */
int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
struct snd_ac97 **rac97);
const char *snd_ac97_get_short_name(struct snd_ac97 *ac97);
void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value);
#ifdef CONFIG_SND_AC97_POWER_SAVE
int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup);
#else
static inline int snd_ac97_update_power(struct snd_ac97 *ac97, int reg,
int powerup)
{
return 0;
}
#endif
#ifdef CONFIG_PM
void snd_ac97_suspend(struct snd_ac97 *ac97);
void snd_ac97_resume(struct snd_ac97 *ac97);
#endif
/* quirk types */
enum {
AC97_TUNE_DEFAULT = -1, /* use default from quirk list (not valid in list) */
AC97_TUNE_NONE = 0, /* nothing extra to do */
AC97_TUNE_HP_ONLY, /* headphone (true line-out) control as master only */
AC97_TUNE_SWAP_HP, /* swap headphone and master controls */
AC97_TUNE_SWAP_SURROUND, /* swap master and surround controls */
AC97_TUNE_AD_SHARING, /* for AD1985, turn on OMS bit and use headphone */
AC97_TUNE_ALC_JACK, /* for Realtek, enable JACK detection */
AC97_TUNE_INV_EAPD, /* inverted EAPD implementation */
AC97_TUNE_MUTE_LED, /* EAPD bit works as mute LED */
AC97_TUNE_HP_MUTE_LED, /* EAPD bit works as mute LED, use headphone control as master */
};
struct ac97_quirk {
unsigned short subvendor; /* PCI subsystem vendor id */
unsigned short subdevice; /* PCI subsystem device id */
unsigned short mask; /* device id bit mask, 0 = accept all */
unsigned int codec_id; /* codec id (if any), 0 = accept all */
const char *name; /* name shown as info */
int type; /* quirk type above */
};
int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override);
int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate);
/*
* PCM allocation
*/
enum ac97_pcm_cfg {
AC97_PCM_CFG_FRONT = 2,
AC97_PCM_CFG_REAR = 10, /* alias surround */
AC97_PCM_CFG_LFE = 11, /* center + lfe */
AC97_PCM_CFG_40 = 4, /* front + rear */
AC97_PCM_CFG_51 = 6, /* front + rear + center/lfe */
AC97_PCM_CFG_SPDIF = 20
};
struct ac97_pcm {
struct snd_ac97_bus *bus;
unsigned int stream: 1, /* stream type: 1 = capture */
exclusive: 1, /* exclusive mode, don't override with other pcms */
copy_flag: 1, /* lowlevel driver must fill all entries */
spdif: 1; /* spdif pcm */
unsigned short aslots; /* active slots */
unsigned short cur_dbl; /* current double-rate state */
unsigned int rates; /* available rates */
struct {
unsigned short slots; /* driver input: requested AC97 slot numbers */
unsigned short rslots[4]; /* allocated slots per codecs */
unsigned char rate_table[4];
struct snd_ac97 *codec[4]; /* allocated codecs */
} r[2]; /* 0 = standard rates, 1 = double rates */
unsigned long private_value; /* used by the hardware driver */
};
int snd_ac97_pcm_assign(struct snd_ac97_bus *ac97,
unsigned short pcms_count,
const struct ac97_pcm *pcms);
int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
enum ac97_pcm_cfg cfg, unsigned short slots);
int snd_ac97_pcm_close(struct ac97_pcm *pcm);
int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
/* ad hoc AC97 device driver access */
extern struct bus_type ac97_bus_type;
/* AC97 platform_data adding function */
static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
{
ac97->dev.platform_data = data;
}
#endif /* __SOUND_AC97_CODEC_H */

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#ifndef _ACI_H_
#define _ACI_H_
#define ACI_REG_COMMAND 0 /* write register offset */
#define ACI_REG_STATUS 1 /* read register offset */
#define ACI_REG_BUSY 2 /* busy register offset */
#define ACI_REG_RDS 2 /* PCM20: RDS register offset */
#define ACI_MINTIME 500 /* ACI time out limit */
#define ACI_SET_MUTE 0x0d
#define ACI_SET_POWERAMP 0x0f
#define ACI_SET_TUNERMUTE 0xa3
#define ACI_SET_TUNERMONO 0xa4
#define ACI_SET_IDE 0xd0
#define ACI_SET_WSS 0xd1
#define ACI_SET_SOLOMODE 0xd2
#define ACI_SET_PREAMP 0x03
#define ACI_GET_PREAMP 0x21
#define ACI_WRITE_TUNE 0xa7
#define ACI_READ_TUNERSTEREO 0xa8
#define ACI_READ_TUNERSTATION 0xa9
#define ACI_READ_VERSION 0xf1
#define ACI_READ_IDCODE 0xf2
#define ACI_INIT 0xff
#define ACI_STATUS 0xf0
#define ACI_S_GENERAL 0x00
#define ACI_ERROR_OP 0xdf
/* ACI Mixer */
/* These are the values for the right channel GET registers.
Add an offset of 0x01 for the left channel register.
(left=right+0x01) */
#define ACI_GET_MASTER 0x03
#define ACI_GET_MIC 0x05
#define ACI_GET_LINE 0x07
#define ACI_GET_CD 0x09
#define ACI_GET_SYNTH 0x0b
#define ACI_GET_PCM 0x0d
#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */
#define ACI_GET_LINE2 0x12
#define ACI_GET_EQ1 0x22 /* from Bass ... */
#define ACI_GET_EQ2 0x24
#define ACI_GET_EQ3 0x26
#define ACI_GET_EQ4 0x28
#define ACI_GET_EQ5 0x2a
#define ACI_GET_EQ6 0x2c
#define ACI_GET_EQ7 0x2e /* ... to Treble */
/* And these are the values for the right channel SET registers.
For left channel access you have to add an offset of 0x08.
MASTER is an exception, which needs an offset of 0x01 */
#define ACI_SET_MASTER 0x00
#define ACI_SET_MIC 0x30
#define ACI_SET_LINE 0x31
#define ACI_SET_CD 0x34
#define ACI_SET_SYNTH 0x33
#define ACI_SET_PCM 0x32
#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */
#define ACI_SET_LINE2 0x36
#define ACI_SET_EQ1 0x40 /* from Bass ... */
#define ACI_SET_EQ2 0x41
#define ACI_SET_EQ3 0x42
#define ACI_SET_EQ4 0x43
#define ACI_SET_EQ5 0x44
#define ACI_SET_EQ6 0x45
#define ACI_SET_EQ7 0x46 /* ... to Treble */
struct snd_miro_aci {
unsigned long aci_port;
int aci_vendor;
int aci_product;
int aci_version;
int aci_amp;
int aci_preamp;
int aci_solomode;
struct mutex aci_mutex;
};
int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3);
struct snd_miro_aci *snd_aci_get_aci(void);
#endif /* _ACI_H_ */

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#ifndef __SOUND_AD1816A_H
#define __SOUND_AD1816A_H
/*
ad1816a.h - definitions for ADI SoundPort AD1816A chip.
Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
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/control.h>
#include <sound/pcm.h>
#include <sound/timer.h>
#define AD1816A_REG(r) (chip->port + r)
#define AD1816A_CHIP_STATUS 0x00
#define AD1816A_INDIR_ADDR 0x00
#define AD1816A_INTERRUPT_STATUS 0x01
#define AD1816A_INDIR_DATA_LOW 0x02
#define AD1816A_INDIR_DATA_HIGH 0x03
#define AD1816A_PIO_DEBUG 0x04
#define AD1816A_PIO_STATUS 0x05
#define AD1816A_PIO_DATA 0x06
#define AD1816A_RESERVED_7 0x07
#define AD1816A_PLAYBACK_CONFIG 0x08
#define AD1816A_CAPTURE_CONFIG 0x09
#define AD1816A_RESERVED_10 0x0a
#define AD1816A_RESERVED_11 0x0b
#define AD1816A_JOYSTICK_RAW_DATA 0x0c
#define AD1816A_JOYSTICK_CTRL 0x0d
#define AD1816A_JOY_POS_DATA_LOW 0x0e
#define AD1816A_JOY_POS_DATA_HIGH 0x0f
#define AD1816A_LOW_BYTE_TMP 0x00
#define AD1816A_INTERRUPT_ENABLE 0x01
#define AD1816A_EXTERNAL_CTRL 0x01
#define AD1816A_PLAYBACK_SAMPLE_RATE 0x02
#define AD1816A_CAPTURE_SAMPLE_RATE 0x03
#define AD1816A_VOICE_ATT 0x04
#define AD1816A_FM_ATT 0x05
#define AD1816A_I2S_1_ATT 0x06
#define AD1816A_I2S_0_ATT 0x07
#define AD1816A_PLAYBACK_BASE_COUNT 0x08
#define AD1816A_PLAYBACK_CURR_COUNT 0x09
#define AD1816A_CAPTURE_BASE_COUNT 0x0a
#define AD1816A_CAPTURE_CURR_COUNT 0x0b
#define AD1816A_TIMER_BASE_COUNT 0x0c
#define AD1816A_TIMER_CURR_COUNT 0x0d
#define AD1816A_MASTER_ATT 0x0e
#define AD1816A_CD_GAIN_ATT 0x0f
#define AD1816A_SYNTH_GAIN_ATT 0x10
#define AD1816A_VID_GAIN_ATT 0x11
#define AD1816A_LINE_GAIN_ATT 0x12
#define AD1816A_MIC_GAIN_ATT 0x13
#define AD1816A_PHONE_IN_GAIN_ATT 0x13
#define AD1816A_ADC_SOURCE_SEL 0x14
#define AD1816A_ADC_PGA 0x14
#define AD1816A_CHIP_CONFIG 0x20
#define AD1816A_DSP_CONFIG 0x21
#define AD1816A_FM_SAMPLE_RATE 0x22
#define AD1816A_I2S_1_SAMPLE_RATE 0x23
#define AD1816A_I2S_0_SAMPLE_RATE 0x24
#define AD1816A_RESERVED_37 0x25
#define AD1816A_PROGRAM_CLOCK_RATE 0x26
#define AD1816A_3D_PHAT_CTRL 0x27
#define AD1816A_PHONE_OUT_ATT 0x27
#define AD1816A_RESERVED_40 0x28
#define AD1816A_HW_VOL_BUT 0x29
#define AD1816A_DSP_MAILBOX_0 0x2a
#define AD1816A_DSP_MAILBOX_1 0x2b
#define AD1816A_POWERDOWN_CTRL 0x2c
#define AD1816A_TIMER_CTRL 0x2c
#define AD1816A_VERSION_ID 0x2d
#define AD1816A_RESERVED_46 0x2e
#define AD1816A_READY 0x80
#define AD1816A_PLAYBACK_IRQ_PENDING 0x80
#define AD1816A_CAPTURE_IRQ_PENDING 0x40
#define AD1816A_TIMER_IRQ_PENDING 0x20
#define AD1816A_PLAYBACK_ENABLE 0x01
#define AD1816A_PLAYBACK_PIO 0x02
#define AD1816A_CAPTURE_ENABLE 0x01
#define AD1816A_CAPTURE_PIO 0x02
#define AD1816A_FMT_LINEAR_8 0x00
#define AD1816A_FMT_ULAW_8 0x08
#define AD1816A_FMT_LINEAR_16_LIT 0x10
#define AD1816A_FMT_ALAW_8 0x18
#define AD1816A_FMT_LINEAR_16_BIG 0x30
#define AD1816A_FMT_ALL 0x38
#define AD1816A_FMT_STEREO 0x04
#define AD1816A_PLAYBACK_IRQ_ENABLE 0x8000
#define AD1816A_CAPTURE_IRQ_ENABLE 0x4000
#define AD1816A_TIMER_IRQ_ENABLE 0x2000
#define AD1816A_TIMER_ENABLE 0x0080
#define AD1816A_SRC_LINE 0x00
#define AD1816A_SRC_OUT 0x10
#define AD1816A_SRC_CD 0x20
#define AD1816A_SRC_SYNTH 0x30
#define AD1816A_SRC_VIDEO 0x40
#define AD1816A_SRC_MIC 0x50
#define AD1816A_SRC_MONO 0x50
#define AD1816A_SRC_PHONE_IN 0x60
#define AD1816A_SRC_MASK 0x70
#define AD1816A_CAPTURE_NOT_EQUAL 0x1000
#define AD1816A_WSS_ENABLE 0x8000
struct snd_ad1816a {
unsigned long port;
struct resource *res_port;
int irq;
int dma1;
int dma2;
unsigned short hardware;
unsigned short version;
spinlock_t lock;
unsigned short mode;
unsigned int clock_freq;
struct snd_card *card;
struct snd_pcm *pcm;
struct snd_pcm_substream *playback_substream;
struct snd_pcm_substream *capture_substream;
unsigned int p_dma_size;
unsigned int c_dma_size;
struct snd_timer *timer;
#ifdef CONFIG_PM
unsigned short image[48];
#endif
};
#define AD1816A_HW_AUTO 0
#define AD1816A_HW_AD1816A 1
#define AD1816A_HW_AD1815 2
#define AD1816A_HW_AD18MAX10 3
#define AD1816A_MODE_PLAYBACK 0x01
#define AD1816A_MODE_CAPTURE 0x02
#define AD1816A_MODE_TIMER 0x04
#define AD1816A_MODE_OPEN (AD1816A_MODE_PLAYBACK | \
AD1816A_MODE_CAPTURE | \
AD1816A_MODE_TIMER)
extern int snd_ad1816a_create(struct snd_card *card, unsigned long port,
int irq, int dma1, int dma2,
struct snd_ad1816a *chip);
extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm);
extern int snd_ad1816a_mixer(struct snd_ad1816a *chip);
extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device,
struct snd_timer **rtimer);
#ifdef CONFIG_PM
extern void snd_ad1816a_suspend(struct snd_ad1816a *chip);
extern void snd_ad1816a_resume(struct snd_ad1816a *chip);
#endif
#endif /* __SOUND_AD1816A_H */

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/*
* This file is subject to the terms and conditions of the GNU General Public
* License. See the file "COPYING" in the main directory of this archive
* for more details.
*
* Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
* Copyright 2008 Thomas Bogendoerfer <tsbogend@franken.de>
*/
#ifndef __SOUND_AD1843_H
#define __SOUND_AD1843_H
struct snd_ad1843 {
void *chip;
int (*read)(void *chip, int reg);
int (*write)(void *chip, int reg, int val);
};
#define AD1843_GAIN_RECLEV 0
#define AD1843_GAIN_LINE 1
#define AD1843_GAIN_LINE_2 2
#define AD1843_GAIN_MIC 3
#define AD1843_GAIN_PCM_0 4
#define AD1843_GAIN_PCM_1 5
#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1)
int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id);
int ad1843_get_gain(struct snd_ad1843 *ad1843, int id);
int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval);
int ad1843_get_recsrc(struct snd_ad1843 *ad1843);
int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc);
void ad1843_setup_dac(struct snd_ad1843 *ad1843,
unsigned int id,
unsigned int framerate,
snd_pcm_format_t fmt,
unsigned int channels);
void ad1843_shutdown_dac(struct snd_ad1843 *ad1843,
unsigned int id);
void ad1843_setup_adc(struct snd_ad1843 *ad1843,
unsigned int framerate,
snd_pcm_format_t fmt,
unsigned int channels);
void ad1843_shutdown_adc(struct snd_ad1843 *ad1843);
int ad1843_init(struct snd_ad1843 *ad1843);
#endif /* __SOUND_AD1843_H */

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/*
* Analog Devices ADAU1373 Audio Codec drive
*
* Copyright 2011 Analog Devices Inc.
* Author: Lars-Peter Clausen <lars@metafoo.de>
*
* Licensed under the GPL-2 or later.
*/
#ifndef __SOUND_ADAU1373_H__
#define __SOUND_ADAU1373_H__
enum adau1373_micbias_voltage {
ADAU1373_MICBIAS_2_9V = 0,
ADAU1373_MICBIAS_2_2V = 1,
ADAU1373_MICBIAS_2_6V = 2,
ADAU1373_MICBIAS_1_8V = 3,
};
#define ADAU1373_DRC_SIZE 13
struct adau1373_platform_data {
bool input_differential[4];
bool lineout_differential;
bool lineout_ground_sense;
unsigned int num_drc;
uint8_t drc_setting[3][ADAU1373_DRC_SIZE];
enum adau1373_micbias_voltage micbias1;
enum adau1373_micbias_voltage micbias2;
};
#endif

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/*
* AESS IP block reset
*
* Copyright (C) 2012 Texas Instruments, Inc.
* Paul Walmsley
*
* 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 version 2.
*
* This program is distributed "as is" WITHOUT ANY WARRANTY of any
* kind, whether express or implied; 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., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef __SOUND_AESS_H__
#define __SOUND_AESS_H__
#include <linux/kernel.h>
#include <linux/io.h>
/*
* AESS_AUTO_GATING_ENABLE_OFFSET: offset in bytes of the AESS IP
* block's AESS_AUTO_GATING_ENABLE__1 register from the IP block's
* base address
*/
#define AESS_AUTO_GATING_ENABLE_OFFSET 0x07c
/* Register bitfields in the AESS_AUTO_GATING_ENABLE__1 register */
#define AESS_AUTO_GATING_ENABLE_SHIFT 0
/**
* aess_enable_autogating - enable AESS internal autogating
* @oh: struct omap_hwmod *
*
* Enable internal autogating on the AESS. This allows the AESS to
* indicate that it is idle to the OMAP PRCM. Returns 0.
*/
static inline void aess_enable_autogating(void __iomem *base)
{
u32 v;
/* Set AESS_AUTO_GATING_ENABLE__1.ENABLE to allow idle entry */
v = 1 << AESS_AUTO_GATING_ENABLE_SHIFT;
writel(v, base + AESS_AUTO_GATING_ENABLE_OFFSET);
}
#endif /* __SOUND_AESS_H__ */

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#ifndef __SOUND_AK4113_H
#define __SOUND_AK4113_H
/*
* Routines for Asahi Kasei AK4113
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Copyright (c) by Pavel Hofman <pavel.hofman@ivitera.com>,
*
*
* 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
*
*/
/* AK4113 registers */
/* power down */
#define AK4113_REG_PWRDN 0x00
/* format control */
#define AK4113_REG_FORMAT 0x01
/* input/output control */
#define AK4113_REG_IO0 0x02
/* input/output control */
#define AK4113_REG_IO1 0x03
/* interrupt0 mask */
#define AK4113_REG_INT0_MASK 0x04
/* interrupt1 mask */
#define AK4113_REG_INT1_MASK 0x05
/* DAT mask & DTS select */
#define AK4113_REG_DATDTS 0x06
/* receiver status 0 */
#define AK4113_REG_RCS0 0x07
/* receiver status 1 */
#define AK4113_REG_RCS1 0x08
/* receiver status 2 */
#define AK4113_REG_RCS2 0x09
/* RX channel status byte 0 */
#define AK4113_REG_RXCSB0 0x0a
/* RX channel status byte 1 */
#define AK4113_REG_RXCSB1 0x0b
/* RX channel status byte 2 */
#define AK4113_REG_RXCSB2 0x0c
/* RX channel status byte 3 */
#define AK4113_REG_RXCSB3 0x0d
/* RX channel status byte 4 */
#define AK4113_REG_RXCSB4 0x0e
/* burst preamble Pc byte 0 */
#define AK4113_REG_Pc0 0x0f
/* burst preamble Pc byte 1 */
#define AK4113_REG_Pc1 0x10
/* burst preamble Pd byte 0 */
#define AK4113_REG_Pd0 0x11
/* burst preamble Pd byte 1 */
#define AK4113_REG_Pd1 0x12
/* Q-subcode address + control */
#define AK4113_REG_QSUB_ADDR 0x13
/* Q-subcode track */
#define AK4113_REG_QSUB_TRACK 0x14
/* Q-subcode index */
#define AK4113_REG_QSUB_INDEX 0x15
/* Q-subcode minute */
#define AK4113_REG_QSUB_MINUTE 0x16
/* Q-subcode second */
#define AK4113_REG_QSUB_SECOND 0x17
/* Q-subcode frame */
#define AK4113_REG_QSUB_FRAME 0x18
/* Q-subcode zero */
#define AK4113_REG_QSUB_ZERO 0x19
/* Q-subcode absolute minute */
#define AK4113_REG_QSUB_ABSMIN 0x1a
/* Q-subcode absolute second */
#define AK4113_REG_QSUB_ABSSEC 0x1b
/* Q-subcode absolute frame */
#define AK4113_REG_QSUB_ABSFRM 0x1c
/* sizes */
#define AK4113_REG_RXCSB_SIZE ((AK4113_REG_RXCSB4-AK4113_REG_RXCSB0)+1)
#define AK4113_REG_QSUB_SIZE ((AK4113_REG_QSUB_ABSFRM-AK4113_REG_QSUB_ADDR)\
+1)
#define AK4113_WRITABLE_REGS (AK4113_REG_DATDTS + 1)
/* AK4113_REG_PWRDN bits */
/* Channel Status Select */
#define AK4113_CS12 (1<<7)
/* Block Start & C/U Output Mode */
#define AK4113_BCU (1<<6)
/* Master Clock Operation Select */
#define AK4113_CM1 (1<<5)
/* Master Clock Operation Select */
#define AK4113_CM0 (1<<4)
/* Master Clock Frequency Select */
#define AK4113_OCKS1 (1<<3)
/* Master Clock Frequency Select */
#define AK4113_OCKS0 (1<<2)
/* 0 = power down, 1 = normal operation */
#define AK4113_PWN (1<<1)
/* 0 = reset & initialize (except thisregister), 1 = normal operation */
#define AK4113_RST (1<<0)
/* AK4113_REQ_FORMAT bits */
/* V/TX Output select: 0 = Validity Flag Output, 1 = TX */
#define AK4113_VTX (1<<7)
/* Audio Data Control */
#define AK4113_DIF2 (1<<6)
/* Audio Data Control */
#define AK4113_DIF1 (1<<5)
/* Audio Data Control */
#define AK4113_DIF0 (1<<4)
/* Deemphasis Autodetect Enable (1 = enable) */
#define AK4113_DEAU (1<<3)
/* 32kHz-48kHz Deemphasis Control */
#define AK4113_DEM1 (1<<2)
/* 32kHz-48kHz Deemphasis Control */
#define AK4113_DEM0 (1<<1)
#define AK4113_DEM_OFF (AK4113_DEM0)
#define AK4113_DEM_44KHZ (0)
#define AK4113_DEM_48KHZ (AK4113_DEM1)
#define AK4113_DEM_32KHZ (AK4113_DEM0|AK4113_DEM1)
/* STDO: 16-bit, right justified */
#define AK4113_DIF_16R (0)
/* STDO: 18-bit, right justified */
#define AK4113_DIF_18R (AK4113_DIF0)
/* STDO: 20-bit, right justified */
#define AK4113_DIF_20R (AK4113_DIF1)
/* STDO: 24-bit, right justified */
#define AK4113_DIF_24R (AK4113_DIF1|AK4113_DIF0)
/* STDO: 24-bit, left justified */
#define AK4113_DIF_24L (AK4113_DIF2)
/* STDO: I2S */
#define AK4113_DIF_24I2S (AK4113_DIF2|AK4113_DIF0)
/* STDO: 24-bit, left justified; LRCLK, BICK = Input */
#define AK4113_DIF_I24L (AK4113_DIF2|AK4113_DIF1)
/* STDO: I2S; LRCLK, BICK = Input */
#define AK4113_DIF_I24I2S (AK4113_DIF2|AK4113_DIF1|AK4113_DIF0)
/* AK4113_REG_IO0 */
/* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
#define AK4113_XTL1 (1<<6)
/* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
#define AK4113_XTL0 (1<<5)
/* Block Start Signal Output: 0 = U-bit, 1 = C-bit (req. BCU = 1) */
#define AK4113_UCE (1<<4)
/* TX Output Enable (1 = enable) */
#define AK4113_TXE (1<<3)
/* Output Through Data Selector for TX pin */
#define AK4113_OPS2 (1<<2)
/* Output Through Data Selector for TX pin */
#define AK4113_OPS1 (1<<1)
/* Output Through Data Selector for TX pin */
#define AK4113_OPS0 (1<<0)
/* 11.2896 MHz ref. Xtal freq. */
#define AK4113_XTL_11_2896M (0)
/* 12.288 MHz ref. Xtal freq. */
#define AK4113_XTL_12_288M (AK4113_XTL0)
/* 24.576 MHz ref. Xtal freq. */
#define AK4113_XTL_24_576M (AK4113_XTL1)
/* AK4113_REG_IO1 */
/* Interrupt 0 pin Hold */
#define AK4113_EFH1 (1<<7)
/* Interrupt 0 pin Hold */
#define AK4113_EFH0 (1<<6)
#define AK4113_EFH_512LRCLK (0)
#define AK4113_EFH_1024LRCLK (AK4113_EFH0)
#define AK4113_EFH_2048LRCLK (AK4113_EFH1)
#define AK4113_EFH_4096LRCLK (AK4113_EFH1|AK4113_EFH0)
/* PLL Lock Time: 0 = 384/fs, 1 = 1/fs */
#define AK4113_FAST (1<<5)
/* MCKO2 Output Select: 0 = CMx/OCKSx, 1 = Xtal */
#define AK4113_XMCK (1<<4)
/* MCKO2 Output Freq. Select: 0 = x1, 1 = x0.5 (req. XMCK = 1) */
#define AK4113_DIV (1<<3)
/* Input Recovery Data Select */
#define AK4113_IPS2 (1<<2)
/* Input Recovery Data Select */
#define AK4113_IPS1 (1<<1)
/* Input Recovery Data Select */
#define AK4113_IPS0 (1<<0)
#define AK4113_IPS(x) ((x)&7)
/* AK4113_REG_INT0_MASK && AK4113_REG_INT1_MASK*/
/* mask enable for QINT bit */
#define AK4113_MQI (1<<7)
/* mask enable for AUTO bit */
#define AK4113_MAUT (1<<6)
/* mask enable for CINT bit */
#define AK4113_MCIT (1<<5)
/* mask enable for UNLOCK bit */
#define AK4113_MULK (1<<4)
/* mask enable for V bit */
#define AK4113_V (1<<3)
/* mask enable for STC bit */
#define AK4113_STC (1<<2)
/* mask enable for AUDN bit */
#define AK4113_MAN (1<<1)
/* mask enable for PAR bit */
#define AK4113_MPR (1<<0)
/* AK4113_REG_DATDTS */
/* DAT Start ID Counter */
#define AK4113_DCNT (1<<4)
/* DTS-CD 16-bit Sync Word Detect */
#define AK4113_DTS16 (1<<3)
/* DTS-CD 14-bit Sync Word Detect */
#define AK4113_DTS14 (1<<2)
/* mask enable for DAT bit (if 1, no INT1 effect */
#define AK4113_MDAT1 (1<<1)
/* mask enable for DAT bit (if 1, no INT0 effect */
#define AK4113_MDAT0 (1<<0)
/* AK4113_REG_RCS0 */
/* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
#define AK4113_QINT (1<<7)
/* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
#define AK4113_AUTO (1<<6)
/* channel status buffer interrupt, 0 = no change, 1 = change */
#define AK4113_CINT (1<<5)
/* PLL lock status, 0 = lock, 1 = unlock */
#define AK4113_UNLCK (1<<4)
/* Validity bit, 0 = valid, 1 = invalid */
#define AK4113_V (1<<3)
/* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
#define AK4113_STC (1<<2)
/* audio bit output, 0 = audio, 1 = non-audio */
#define AK4113_AUDION (1<<1)
/* parity error or biphase error status, 0 = no error, 1 = error */
#define AK4113_PAR (1<<0)
/* AK4113_REG_RCS1 */
/* sampling frequency detection */
#define AK4113_FS3 (1<<7)
#define AK4113_FS2 (1<<6)
#define AK4113_FS1 (1<<5)
#define AK4113_FS0 (1<<4)
/* Pre-emphasis detect, 0 = OFF, 1 = ON */
#define AK4113_PEM (1<<3)
/* DAT Start ID Detect, 0 = no detect, 1 = detect */
#define AK4113_DAT (1<<2)
/* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
#define AK4113_DTSCD (1<<1)
/* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
#define AK4113_NPCM (1<<0)
#define AK4113_FS_8000HZ (AK4113_FS3|AK4113_FS0)
#define AK4113_FS_11025HZ (AK4113_FS2|AK4113_FS0)
#define AK4113_FS_16000HZ (AK4113_FS2|AK4113_FS1|AK4113_FS0)
#define AK4113_FS_22050HZ (AK4113_FS2)
#define AK4113_FS_24000HZ (AK4113_FS2|AK4113_FS1)
#define AK4113_FS_32000HZ (AK4113_FS1|AK4113_FS0)
#define AK4113_FS_44100HZ (0)
#define AK4113_FS_48000HZ (AK4113_FS1)
#define AK4113_FS_64000HZ (AK4113_FS3|AK4113_FS1|AK4113_FS0)
#define AK4113_FS_88200HZ (AK4113_FS3)
#define AK4113_FS_96000HZ (AK4113_FS3|AK4113_FS1)
#define AK4113_FS_176400HZ (AK4113_FS3|AK4113_FS2)
#define AK4113_FS_192000HZ (AK4113_FS3|AK4113_FS2|AK4113_FS1)
/* AK4113_REG_RCS2 */
/* CRC for Q-subcode, 0 = no error, 1 = error */
#define AK4113_QCRC (1<<1)
/* CRC for channel status, 0 = no error, 1 = error */
#define AK4113_CCRC (1<<0)
/* flags for snd_ak4113_check_rate_and_errors() */
#define AK4113_CHECK_NO_STAT (1<<0) /* no statistics */
#define AK4113_CHECK_NO_RATE (1<<1) /* no rate check */
#define AK4113_CONTROLS 13
typedef void (ak4113_write_t)(void *private_data, unsigned char addr,
unsigned char data);
typedef unsigned char (ak4113_read_t)(void *private_data, unsigned char addr);
struct ak4113 {
struct snd_card *card;
ak4113_write_t *write;
ak4113_read_t *read;
void *private_data;
unsigned int init:1;
spinlock_t lock;
unsigned char regmap[AK4113_WRITABLE_REGS];
struct snd_kcontrol *kctls[AK4113_CONTROLS];
struct snd_pcm_substream *substream;
unsigned long parity_errors;
unsigned long v_bit_errors;
unsigned long qcrc_errors;
unsigned long ccrc_errors;
unsigned char rcs0;
unsigned char rcs1;
unsigned char rcs2;
struct delayed_work work;
unsigned int check_flags;
void *change_callback_private;
void (*change_callback)(struct ak4113 *ak4113, unsigned char c0,
unsigned char c1);
};
int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
ak4113_write_t *write,
const unsigned char *pgm,
void *private_data, struct ak4113 **r_ak4113);
void snd_ak4113_reg_write(struct ak4113 *ak4113, unsigned char reg,
unsigned char mask, unsigned char val);
void snd_ak4113_reinit(struct ak4113 *ak4113);
int snd_ak4113_build(struct ak4113 *ak4113,
struct snd_pcm_substream *capture_substream);
int snd_ak4113_external_rate(struct ak4113 *ak4113);
int snd_ak4113_check_rate_and_errors(struct ak4113 *ak4113, unsigned int flags);
#endif /* __SOUND_AK4113_H */

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#ifndef __SOUND_AK4114_H
#define __SOUND_AK4114_H
/*
* Routines for Asahi Kasei AK4114
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* 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
*
*/
/* AK4114 registers */
#define AK4114_REG_PWRDN 0x00 /* power down */
#define AK4114_REG_FORMAT 0x01 /* format control */
#define AK4114_REG_IO0 0x02 /* input/output control */
#define AK4114_REG_IO1 0x03 /* input/output control */
#define AK4114_REG_INT0_MASK 0x04 /* interrupt0 mask */
#define AK4114_REG_INT1_MASK 0x05 /* interrupt1 mask */
#define AK4114_REG_RCS0 0x06 /* receiver status 0 */
#define AK4114_REG_RCS1 0x07 /* receiver status 1 */
#define AK4114_REG_RXCSB0 0x08 /* RX channel status byte 0 */
#define AK4114_REG_RXCSB1 0x09 /* RX channel status byte 1 */
#define AK4114_REG_RXCSB2 0x0a /* RX channel status byte 2 */
#define AK4114_REG_RXCSB3 0x0b /* RX channel status byte 3 */
#define AK4114_REG_RXCSB4 0x0c /* RX channel status byte 4 */
#define AK4114_REG_TXCSB0 0x0d /* TX channel status byte 0 */
#define AK4114_REG_TXCSB1 0x0e /* TX channel status byte 1 */
#define AK4114_REG_TXCSB2 0x0f /* TX channel status byte 2 */
#define AK4114_REG_TXCSB3 0x10 /* TX channel status byte 3 */
#define AK4114_REG_TXCSB4 0x11 /* TX channel status byte 4 */
#define AK4114_REG_Pc0 0x12 /* burst preamble Pc byte 0 */
#define AK4114_REG_Pc1 0x13 /* burst preamble Pc byte 1 */
#define AK4114_REG_Pd0 0x14 /* burst preamble Pd byte 0 */
#define AK4114_REG_Pd1 0x15 /* burst preamble Pd byte 1 */
#define AK4114_REG_QSUB_ADDR 0x16 /* Q-subcode address + control */
#define AK4114_REG_QSUB_TRACK 0x17 /* Q-subcode track */
#define AK4114_REG_QSUB_INDEX 0x18 /* Q-subcode index */
#define AK4114_REG_QSUB_MINUTE 0x19 /* Q-subcode minute */
#define AK4114_REG_QSUB_SECOND 0x1a /* Q-subcode second */
#define AK4114_REG_QSUB_FRAME 0x1b /* Q-subcode frame */
#define AK4114_REG_QSUB_ZERO 0x1c /* Q-subcode zero */
#define AK4114_REG_QSUB_ABSMIN 0x1d /* Q-subcode absolute minute */
#define AK4114_REG_QSUB_ABSSEC 0x1e /* Q-subcode absolute second */
#define AK4114_REG_QSUB_ABSFRM 0x1f /* Q-subcode absolute frame */
/* sizes */
#define AK4114_REG_RXCSB_SIZE ((AK4114_REG_RXCSB4-AK4114_REG_RXCSB0)+1)
#define AK4114_REG_TXCSB_SIZE ((AK4114_REG_TXCSB4-AK4114_REG_TXCSB0)+1)
#define AK4114_REG_QSUB_SIZE ((AK4114_REG_QSUB_ABSFRM-AK4114_REG_QSUB_ADDR)+1)
/* AK4117_REG_PWRDN bits */
#define AK4114_CS12 (1<<7) /* Channel Status Select */
#define AK4114_BCU (1<<6) /* Block Start & C/U Output Mode */
#define AK4114_CM1 (1<<5) /* Master Clock Operation Select */
#define AK4114_CM0 (1<<4) /* Master Clock Operation Select */
#define AK4114_OCKS1 (1<<3) /* Master Clock Frequency Select */
#define AK4114_OCKS0 (1<<2) /* Master Clock Frequency Select */
#define AK4114_PWN (1<<1) /* 0 = power down, 1 = normal operation */
#define AK4114_RST (1<<0) /* 0 = reset & initialize (except this register), 1 = normal operation */
/* AK4114_REQ_FORMAT bits */
#define AK4114_MONO (1<<7) /* Double Sampling Frequency Mode: 0 = stereo, 1 = mono */
#define AK4114_DIF2 (1<<6) /* Audio Data Control */
#define AK4114_DIF1 (1<<5) /* Audio Data Control */
#define AK4114_DIF0 (1<<4) /* Audio Data Control */
#define AK4114_DIF_16R (0) /* STDO: 16-bit, right justified */
#define AK4114_DIF_18R (AK4114_DIF0) /* STDO: 18-bit, right justified */
#define AK4114_DIF_20R (AK4114_DIF1) /* STDO: 20-bit, right justified */
#define AK4114_DIF_24R (AK4114_DIF1|AK4114_DIF0) /* STDO: 24-bit, right justified */
#define AK4114_DIF_24L (AK4114_DIF2) /* STDO: 24-bit, left justified */
#define AK4114_DIF_24I2S (AK4114_DIF2|AK4114_DIF0) /* STDO: I2S */
#define AK4114_DIF_I24L (AK4114_DIF2|AK4114_DIF1) /* STDO: 24-bit, left justified; LRCLK, BICK = Input */
#define AK4114_DIF_I24I2S (AK4114_DIF2|AK4114_DIF1|AK4114_DIF0) /* STDO: I2S; LRCLK, BICK = Input */
#define AK4114_DEAU (1<<3) /* Deemphasis Autodetect Enable (1 = enable) */
#define AK4114_DEM1 (1<<2) /* 32kHz-48kHz Deemphasis Control */
#define AK4114_DEM0 (1<<1) /* 32kHz-48kHz Deemphasis Control */
#define AK4114_DEM_44KHZ (0)
#define AK4114_DEM_48KHZ (AK4114_DEM1)
#define AK4114_DEM_32KHZ (AK4114_DEM0|AK4114_DEM1)
#define AK4114_DEM_96KHZ (AK4114_DEM1) /* DFS must be set */
#define AK4114_DFS (1<<0) /* 96kHz Deemphasis Control */
/* AK4114_REG_IO0 */
#define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */
#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */
#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */
#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */
#define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */
#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */
#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */
#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */
/* AK4114_REG_IO1 */
#define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */
#define AK4114_EFH0 (1<<6) /* Interrupt 0 pin Hold */
#define AK4114_EFH_512 (0)
#define AK4114_EFH_1024 (AK4114_EFH0)
#define AK4114_EFH_2048 (AK4114_EFH1)
#define AK4114_EFH_4096 (AK4114_EFH1|AK4114_EFH0)
#define AK4114_UDIT (1<<5) /* U-bit Control for DIT (0 = fixed '0', 1 = recovered) */
#define AK4114_TLR (1<<4) /* Double Sampling Frequency Select for DIT (0 = L channel, 1 = R channel) */
#define AK4114_DIT (1<<3) /* TX1 out: 0 = Through Data (RX data), 1 = Transmit Data (DAUX data) */
#define AK4114_IPS2 (1<<2) /* Input Recovery Data Select */
#define AK4114_IPS1 (1<<1) /* Input Recovery Data Select */
#define AK4114_IPS0 (1<<0) /* Input Recovery Data Select */
#define AK4114_IPS(x) ((x)&7)
/* AK4114_REG_INT0_MASK && AK4114_REG_INT1_MASK*/
#define AK4117_MQI (1<<7) /* mask enable for QINT bit */
#define AK4117_MAT (1<<6) /* mask enable for AUTO bit */
#define AK4117_MCI (1<<5) /* mask enable for CINT bit */
#define AK4117_MUL (1<<4) /* mask enable for UNLOCK bit */
#define AK4117_MDTS (1<<3) /* mask enable for DTSCD bit */
#define AK4117_MPE (1<<2) /* mask enable for PEM bit */
#define AK4117_MAN (1<<1) /* mask enable for AUDN bit */
#define AK4117_MPR (1<<0) /* mask enable for PAR bit */
/* AK4114_REG_RCS0 */
#define AK4114_QINT (1<<7) /* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
#define AK4114_AUTO (1<<6) /* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
#define AK4114_CINT (1<<5) /* channel status buffer interrupt, 0 = no change, 1 = change */
#define AK4114_UNLCK (1<<4) /* PLL lock status, 0 = lock, 1 = unlock */
#define AK4114_DTSCD (1<<3) /* DTS-CD Detect, 0 = No detect, 1 = Detect */
#define AK4114_PEM (1<<2) /* Pre-emphasis Detect, 0 = OFF, 1 = ON */
#define AK4114_AUDION (1<<1) /* audio bit output, 0 = audio, 1 = non-audio */
#define AK4114_PAR (1<<0) /* parity error or biphase error status, 0 = no error, 1 = error */
/* AK4114_REG_RCS1 */
#define AK4114_FS3 (1<<7) /* sampling frequency detection */
#define AK4114_FS2 (1<<6)
#define AK4114_FS1 (1<<5)
#define AK4114_FS0 (1<<4)
#define AK4114_FS_44100HZ (0)
#define AK4114_FS_48000HZ (AK4114_FS1)
#define AK4114_FS_32000HZ (AK4114_FS1|AK4114_FS0)
#define AK4114_FS_88200HZ (AK4114_FS3)
#define AK4114_FS_96000HZ (AK4114_FS3|AK4114_FS1)
#define AK4114_FS_176400HZ (AK4114_FS3|AK4114_FS2)
#define AK4114_FS_192000HZ (AK4114_FS3|AK4114_FS2|AK4114_FS1)
#define AK4114_V (1<<3) /* Validity of Channel Status, 0 = Valid, 1 = Invalid */
#define AK4114_QCRC (1<<1) /* CRC for Q-subcode, 0 = no error, 1 = error */
#define AK4114_CCRC (1<<0) /* CRC for channel status, 0 = no error, 1 = error */
/* flags for snd_ak4114_check_rate_and_errors() */
#define AK4114_CHECK_NO_STAT (1<<0) /* no statistics */
#define AK4114_CHECK_NO_RATE (1<<1) /* no rate check */
#define AK4114_CONTROLS 15
typedef void (ak4114_write_t)(void *private_data, unsigned char addr, unsigned char data);
typedef unsigned char (ak4114_read_t)(void *private_data, unsigned char addr);
struct ak4114 {
struct snd_card *card;
ak4114_write_t * write;
ak4114_read_t * read;
void * private_data;
unsigned int init: 1;
spinlock_t lock;
unsigned char regmap[7];
unsigned char txcsb[5];
struct snd_kcontrol *kctls[AK4114_CONTROLS];
struct snd_pcm_substream *playback_substream;
struct snd_pcm_substream *capture_substream;
unsigned long parity_errors;
unsigned long v_bit_errors;
unsigned long qcrc_errors;
unsigned long ccrc_errors;
unsigned char rcs0;
unsigned char rcs1;
struct delayed_work work;
unsigned int check_flags;
void *change_callback_private;
void (*change_callback)(struct ak4114 *ak4114, unsigned char c0, unsigned char c1);
};
int snd_ak4114_create(struct snd_card *card,
ak4114_read_t *read, ak4114_write_t *write,
const unsigned char pgm[7], const unsigned char txcsb[5],
void *private_data, struct ak4114 **r_ak4114);
void snd_ak4114_reg_write(struct ak4114 *ak4114, unsigned char reg, unsigned char mask, unsigned char val);
void snd_ak4114_reinit(struct ak4114 *ak4114);
int snd_ak4114_build(struct ak4114 *ak4114,
struct snd_pcm_substream *playback_substream,
struct snd_pcm_substream *capture_substream);
int snd_ak4114_external_rate(struct ak4114 *ak4114);
int snd_ak4114_check_rate_and_errors(struct ak4114 *ak4114, unsigned int flags);
#endif /* __SOUND_AK4114_H */

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#ifndef __SOUND_AK4117_H
#define __SOUND_AK4117_H
/*
* Routines for Asahi Kasei AK4117
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* 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
*
*/
#define AK4117_REG_PWRDN 0x00 /* power down */
#define AK4117_REG_CLOCK 0x01 /* clock control */
#define AK4117_REG_IO 0x02 /* input/output control */
#define AK4117_REG_INT0_MASK 0x03 /* interrupt0 mask */
#define AK4117_REG_INT1_MASK 0x04 /* interrupt1 mask */
#define AK4117_REG_RCS0 0x05 /* receiver status 0 */
#define AK4117_REG_RCS1 0x06 /* receiver status 1 */
#define AK4117_REG_RCS2 0x07 /* receiver status 2 */
#define AK4117_REG_RXCSB0 0x08 /* RX channel status byte 0 */
#define AK4117_REG_RXCSB1 0x09 /* RX channel status byte 1 */
#define AK4117_REG_RXCSB2 0x0a /* RX channel status byte 2 */
#define AK4117_REG_RXCSB3 0x0b /* RX channel status byte 3 */
#define AK4117_REG_RXCSB4 0x0c /* RX channel status byte 4 */
#define AK4117_REG_Pc0 0x0d /* burst preamble Pc byte 0 */
#define AK4117_REG_Pc1 0x0e /* burst preamble Pc byte 1 */
#define AK4117_REG_Pd0 0x0f /* burst preamble Pd byte 0 */
#define AK4117_REG_Pd1 0x10 /* burst preamble Pd byte 1 */
#define AK4117_REG_QSUB_ADDR 0x11 /* Q-subcode address + control */
#define AK4117_REG_QSUB_TRACK 0x12 /* Q-subcode track */
#define AK4117_REG_QSUB_INDEX 0x13 /* Q-subcode index */
#define AK4117_REG_QSUB_MINUTE 0x14 /* Q-subcode minute */
#define AK4117_REG_QSUB_SECOND 0x15 /* Q-subcode second */
#define AK4117_REG_QSUB_FRAME 0x16 /* Q-subcode frame */
#define AK4117_REG_QSUB_ZERO 0x17 /* Q-subcode zero */
#define AK4117_REG_QSUB_ABSMIN 0x18 /* Q-subcode absolute minute */
#define AK4117_REG_QSUB_ABSSEC 0x19 /* Q-subcode absolute second */
#define AK4117_REG_QSUB_ABSFRM 0x1a /* Q-subcode absolute frame */
/* sizes */
#define AK4117_REG_RXCSB_SIZE ((AK4117_REG_RXCSB4-AK4117_REG_RXCSB0)+1)
#define AK4117_REG_QSUB_SIZE ((AK4117_REG_QSUB_ABSFRM-AK4117_REG_QSUB_ADDR)+1)
/* AK4117_REG_PWRDN bits */
#define AK4117_EXCT (1<<4) /* 0 = X'tal mode, 1 = external clock mode */
#define AK4117_XTL1 (1<<3) /* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
#define AK4117_XTL0 (1<<2) /* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
#define AK4117_XTL_11_2896M (0)
#define AK4117_XTL_12_288M AK4117_XTL0
#define AK4117_XTL_24_576M AK4117_XTL1
#define AK4117_XTL_EXT (AK4117_XTL1|AK4117_XTL0)
#define AK4117_PWN (1<<1) /* 0 = power down, 1 = normal operation */
#define AK4117_RST (1<<0) /* 0 = reset & initialize (except this register), 1 = normal operation */
/* AK4117_REQ_CLOCK bits */
#define AK4117_LP (1<<7) /* 0 = normal mode, 1 = low power mode (Fs up to 48kHz only) */
#define AK4117_PKCS1 (1<<6) /* master clock frequency at PLL mode (when LP == 0) */
#define AK4117_PKCS0 (1<<5)
#define AK4117_PKCS_512fs (0)
#define AK4117_PKCS_256fs AK4117_PKCS0
#define AK4117_PKCS_128fs AK4117_PKCS1
#define AK4117_DIV (1<<4) /* 0 = MCKO == Fs, 1 = MCKO == Fs / 2; X'tal mode only */
#define AK4117_XCKS1 (1<<3) /* master clock frequency at X'tal mode */
#define AK4117_XCKS0 (1<<2)
#define AK4117_XCKS_128fs (0)
#define AK4117_XCKS_256fs AK4117_XCKS0
#define AK4117_XCKS_512fs AK4117_XCKS1
#define AK4117_XCKS_1024fs (AK4117_XCKS1|AK4117_XCKS0)
#define AK4117_CM1 (1<<1) /* MCKO operation mode select */
#define AK4117_CM0 (1<<0)
#define AK4117_CM_PLL (0) /* use RX input as master clock */
#define AK4117_CM_XTAL (AK4117_CM0) /* use X'tal as master clock */
#define AK4117_CM_PLL_XTAL (AK4117_CM1) /* use Rx input but X'tal when PLL loses lock */
#define AK4117_CM_MONITOR (AK4117_CM0|AK4117_CM1) /* use X'tal as master clock, but use PLL for monitoring */
/* AK4117_REG_IO */
#define AK4117_IPS (1<<7) /* Input Recovery Data Select, 0 = RX0, 1 = RX1 */
#define AK4117_UOUTE (1<<6) /* U-bit output enable to UOUT, 0 = disable, 1 = enable */
#define AK4117_CS12 (1<<5) /* channel status select, 0 = channel1, 1 = channel2 */
#define AK4117_EFH2 (1<<4) /* INT0 pin hold count select */
#define AK4117_EFH1 (1<<3)
#define AK4117_EFH_512LRCLK (0)
#define AK4117_EFH_1024LRCLK (AK4117_EFH1)
#define AK4117_EFH_2048LRCLK (AK4117_EFH2)
#define AK4117_EFH_4096LRCLK (AK4117_EFH1|AK4117_EFH2)
#define AK4117_DIF2 (1<<2) /* audio data format control */
#define AK4117_DIF1 (1<<1)
#define AK4117_DIF0 (1<<0)
#define AK4117_DIF_16R (0) /* STDO: 16-bit, right justified */
#define AK4117_DIF_18R (AK4117_DIF0) /* STDO: 18-bit, right justified */
#define AK4117_DIF_20R (AK4117_DIF1) /* STDO: 20-bit, right justified */
#define AK4117_DIF_24R (AK4117_DIF1|AK4117_DIF0) /* STDO: 24-bit, right justified */
#define AK4117_DIF_24L (AK4117_DIF2) /* STDO: 24-bit, left justified */
#define AK4117_DIF_24I2S (AK4117_DIF2|AK4117_DIF0) /* STDO: I2S */
/* AK4117_REG_INT0_MASK & AK4117_REG_INT1_MASK */
#define AK4117_MULK (1<<7) /* mask enable for UNLOCK bit */
#define AK4117_MPAR (1<<6) /* mask enable for PAR bit */
#define AK4117_MAUTO (1<<5) /* mask enable for AUTO bit */
#define AK4117_MV (1<<4) /* mask enable for V bit */
#define AK4117_MAUD (1<<3) /* mask enable for AUDION bit */
#define AK4117_MSTC (1<<2) /* mask enable for STC bit */
#define AK4117_MCIT (1<<1) /* mask enable for CINT bit */
#define AK4117_MQIT (1<<0) /* mask enable for QINT bit */
/* AK4117_REG_RCS0 */
#define AK4117_UNLCK (1<<7) /* PLL lock status, 0 = lock, 1 = unlock */
#define AK4117_PAR (1<<6) /* parity error or biphase error status, 0 = no error, 1 = error */
#define AK4117_AUTO (1<<5) /* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
#define AK4117_V (1<<4) /* Validity bit, 0 = valid, 1 = invalid */
#define AK4117_AUDION (1<<3) /* audio bit output, 0 = audio, 1 = non-audio */
#define AK4117_STC (1<<2) /* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
#define AK4117_CINT (1<<1) /* channel status buffer interrupt, 0 = no change, 1 = change */
#define AK4117_QINT (1<<0) /* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
/* AK4117_REG_RCS1 */
#define AK4117_DTSCD (1<<6) /* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
#define AK4117_NPCM (1<<5) /* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
#define AK4117_PEM (1<<4) /* Pre-emphasis detect, 0 = OFF, 1 = ON */
#define AK4117_FS3 (1<<3) /* sampling frequency detection */
#define AK4117_FS2 (1<<2)
#define AK4117_FS1 (1<<1)
#define AK4117_FS0 (1<<0)
#define AK4117_FS_44100HZ (0)
#define AK4117_FS_48000HZ (AK4117_FS1)
#define AK4117_FS_32000HZ (AK4117_FS1|AK4117_FS0)
#define AK4117_FS_88200HZ (AK4117_FS3)
#define AK4117_FS_96000HZ (AK4117_FS3|AK4117_FS1)
#define AK4117_FS_176400HZ (AK4117_FS3|AK4117_FS2)
#define AK4117_FS_192000HZ (AK4117_FS3|AK4117_FS2|AK4117_FS1)
/* AK4117_REG_RCS2 */
#define AK4117_CCRC (1<<1) /* CRC for channel status, 0 = no error, 1 = error */
#define AK4117_QCRC (1<<0) /* CRC for Q-subcode, 0 = no error, 1 = error */
/* flags for snd_ak4117_check_rate_and_errors() */
#define AK4117_CHECK_NO_STAT (1<<0) /* no statistics */
#define AK4117_CHECK_NO_RATE (1<<1) /* no rate check */
#define AK4117_CONTROLS 13
typedef void (ak4117_write_t)(void *private_data, unsigned char addr, unsigned char data);
typedef unsigned char (ak4117_read_t)(void *private_data, unsigned char addr);
struct ak4117 {
struct snd_card *card;
ak4117_write_t * write;
ak4117_read_t * read;
void * private_data;
unsigned int init: 1;
spinlock_t lock;
unsigned char regmap[5];
struct snd_kcontrol *kctls[AK4117_CONTROLS];
struct snd_pcm_substream *substream;
unsigned long parity_errors;
unsigned long v_bit_errors;
unsigned long qcrc_errors;
unsigned long ccrc_errors;
unsigned char rcs0;
unsigned char rcs1;
unsigned char rcs2;
struct timer_list timer; /* statistic timer */
void *change_callback_private;
void (*change_callback)(struct ak4117 *ak4117, unsigned char c0, unsigned char c1);
};
int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t *write,
const unsigned char pgm[5], void *private_data, struct ak4117 **r_ak4117);
void snd_ak4117_reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char mask, unsigned char val);
void snd_ak4117_reinit(struct ak4117 *ak4117);
int snd_ak4117_build(struct ak4117 *ak4117, struct snd_pcm_substream *capture_substream);
int snd_ak4117_external_rate(struct ak4117 *ak4117);
int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags);
#endif /* __SOUND_AK4117_H */

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#ifndef __SOUND_AK4531_CODEC_H
#define __SOUND_AK4531_CODEC_H
/*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.1
* by Intel Corporation (http://developer.intel.com).
*
*
* 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/info.h>
#include <sound/control.h>
/*
* ASAHI KASEI - AK4531 codec
* - not really AC'97 codec, but it uses very similar interface as AC'97
*/
/*
* AK4531 codec registers
*/
#define AK4531_LMASTER 0x00 /* master volume left */
#define AK4531_RMASTER 0x01 /* master volume right */
#define AK4531_LVOICE 0x02 /* channel volume left */
#define AK4531_RVOICE 0x03 /* channel volume right */
#define AK4531_LFM 0x04 /* FM volume left */
#define AK4531_RFM 0x05 /* FM volume right */
#define AK4531_LCD 0x06 /* CD volume left */
#define AK4531_RCD 0x07 /* CD volume right */
#define AK4531_LLINE 0x08 /* LINE volume left */
#define AK4531_RLINE 0x09 /* LINE volume right */
#define AK4531_LAUXA 0x0a /* AUXA volume left */
#define AK4531_RAUXA 0x0b /* AUXA volume right */
#define AK4531_MONO1 0x0c /* MONO1 volume left */
#define AK4531_MONO2 0x0d /* MONO1 volume right */
#define AK4531_MIC 0x0e /* MIC volume */
#define AK4531_MONO_OUT 0x0f /* Mono-out volume */
#define AK4531_OUT_SW1 0x10 /* Output mixer switch 1 */
#define AK4531_OUT_SW2 0x11 /* Output mixer switch 2 */
#define AK4531_LIN_SW1 0x12 /* Input left mixer switch 1 */
#define AK4531_RIN_SW1 0x13 /* Input right mixer switch 1 */
#define AK4531_LIN_SW2 0x14 /* Input left mixer switch 2 */
#define AK4531_RIN_SW2 0x15 /* Input right mixer switch 2 */
#define AK4531_RESET 0x16 /* Reset & power down */
#define AK4531_CLOCK 0x17 /* Clock select */
#define AK4531_AD_IN 0x18 /* AD input select */
#define AK4531_MIC_GAIN 0x19 /* MIC amplified gain */
struct snd_ak4531 {
void (*write) (struct snd_ak4531 *ak4531, unsigned short reg,
unsigned short val);
void *private_data;
void (*private_free) (struct snd_ak4531 *ak4531);
/* --- */
unsigned char regs[0x20];
struct mutex reg_mutex;
};
int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531,
struct snd_ak4531 **rak4531);
#ifdef CONFIG_PM
void snd_ak4531_suspend(struct snd_ak4531 *ak4531);
void snd_ak4531_resume(struct snd_ak4531 *ak4531);
#endif
#endif /* __SOUND_AK4531_CODEC_H */

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/*
* AK4641 ALSA SoC Codec driver
*
* Copyright 2009 Philipp Zabel
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __AK4641_H
#define __AK4641_H
/**
* struct ak4641_platform_data - platform specific AK4641 configuration
* @gpio_power: GPIO to control external power to AK4641
* @gpio_npdn: GPIO connected to AK4641 nPDN pin
*
* Both GPIO parameters are optional.
*/
struct ak4641_platform_data {
int gpio_power;
int gpio_npdn;
};
#endif /* __AK4641_H */

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#ifndef __SOUND_AK4XXX_ADDA_H
#define __SOUND_AK4XXX_ADDA_H
/*
* ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4381
* AD and DA converters
*
* Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* 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
*
*/
#ifndef AK4XXX_MAX_CHIPS
#define AK4XXX_MAX_CHIPS 4
#endif
struct snd_akm4xxx;
struct snd_ak4xxx_ops {
void (*lock)(struct snd_akm4xxx *ak, int chip);
void (*unlock)(struct snd_akm4xxx *ak, int chip);
void (*write)(struct snd_akm4xxx *ak, int chip, unsigned char reg,
unsigned char val);
void (*set_rate_val)(struct snd_akm4xxx *ak, unsigned int rate);
};
#define AK4XXX_IMAGE_SIZE (AK4XXX_MAX_CHIPS * 16) /* 64 bytes */
/* DAC label and channels */
struct snd_akm4xxx_dac_channel {
char *name; /* mixer volume name */
unsigned int num_channels;
char *switch_name; /* mixer switch*/
};
/* ADC labels and channels */
struct snd_akm4xxx_adc_channel {
char *name; /* capture gain volume label */
char *switch_name; /* capture switch */
unsigned int num_channels;
char *selector_name; /* capture source select label */
const char **input_names; /* capture source names (NULL terminated) */
};
struct snd_akm4xxx {
struct snd_card *card;
unsigned int num_adcs; /* AK4524 or AK4528 ADCs */
unsigned int num_dacs; /* AK4524 or AK4528 DACs */
unsigned char images[AK4XXX_IMAGE_SIZE]; /* saved register image */
unsigned char volumes[AK4XXX_IMAGE_SIZE]; /* saved volume values */
unsigned long private_value[AK4XXX_MAX_CHIPS]; /* helper for driver */
void *private_data[AK4XXX_MAX_CHIPS]; /* helper for driver */
/* template should fill the following fields */
unsigned int idx_offset; /* control index offset */
enum {
SND_AK4524, SND_AK4528, SND_AK4529,
SND_AK4355, SND_AK4358, SND_AK4381,
SND_AK5365, SND_AK4620,
} type;
/* (array) information of combined codecs */
const struct snd_akm4xxx_dac_channel *dac_info;
const struct snd_akm4xxx_adc_channel *adc_info;
struct snd_ak4xxx_ops ops;
unsigned int num_chips;
unsigned int total_regs;
const char *name;
};
void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
unsigned char val);
void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state);
void snd_akm4xxx_init(struct snd_akm4xxx *ak);
int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak);
#define snd_akm4xxx_get(ak,chip,reg) \
(ak)->images[(chip) * 16 + (reg)]
#define snd_akm4xxx_set(ak,chip,reg,val) \
((ak)->images[(chip) * 16 + (reg)] = (val))
#define snd_akm4xxx_get_vol(ak,chip,reg) \
(ak)->volumes[(chip) * 16 + (reg)]
#define snd_akm4xxx_set_vol(ak,chip,reg,val) \
((ak)->volumes[(chip) * 16 + (reg)] = (val))
#endif /* __SOUND_AK4XXX_ADDA_H */

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#ifndef _INCLUDE_SOUND_ALC5623_H
#define _INCLUDE_SOUND_ALC5623_H
struct alc5623_platform_data {
/* configure : */
/* Lineout/Speaker Amps Vmid ratio control */
/* enable/disable adc/dac high pass filters */
unsigned int add_ctrl;
/* configure : */
/* output to enable when jack is low */
/* output to enable when jack is high */
/* jack detect (gpio/nc/jack detect [12] */
unsigned int jack_det_ctrl;
};
#endif

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/*
* Main header file for the ALSA sequencer
* Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
* (c) 1998-1999 by Jaroslav Kysela <perex@perex.cz>
*
*
* 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
*
*/
#ifndef __SOUND_ASEQUENCER_H
#define __SOUND_ASEQUENCER_H
#include <linux/ioctl.h>
#include <sound/asound.h>
#include <uapi/sound/asequencer.h>
/* helper macro */
#define snd_seq_event_bounce_ext_data(ev) ((void*)((char *)(ev)->data.ext.ptr + sizeof(struct snd_seq_event_bounce)))
/*
* type check macros
*/
/* result events: 0-4 */
#define snd_seq_ev_is_result_type(ev) ((ev)->type < 5)
/* channel specific events: 5-19 */
#define snd_seq_ev_is_channel_type(ev) ((ev)->type >= 5 && (ev)->type < 20)
/* note events: 5-9 */
#define snd_seq_ev_is_note_type(ev) ((ev)->type >= 5 && (ev)->type < 10)
/* control events: 10-19 */
#define snd_seq_ev_is_control_type(ev) ((ev)->type >= 10 && (ev)->type < 20)
/* queue control events: 30-39 */
#define snd_seq_ev_is_queue_type(ev) ((ev)->type >= 30 && (ev)->type < 40)
/* system status messages */
#define snd_seq_ev_is_message_type(ev) ((ev)->type >= 60 && (ev)->type < 69)
/* sample messages */
#define snd_seq_ev_is_sample_type(ev) ((ev)->type >= 70 && (ev)->type < 79)
/* user-defined messages */
#define snd_seq_ev_is_user_type(ev) ((ev)->type >= 90 && (ev)->type < 99)
/* fixed length events: 0-99 */
#define snd_seq_ev_is_fixed_type(ev) ((ev)->type < 100)
/* variable length events: 130-139 */
#define snd_seq_ev_is_variable_type(ev) ((ev)->type >= 130 && (ev)->type < 140)
/* reserved for kernel */
#define snd_seq_ev_is_reserved(ev) ((ev)->type >= 150)
/* direct dispatched events */
#define snd_seq_ev_is_direct(ev) ((ev)->queue == SNDRV_SEQ_QUEUE_DIRECT)
/*
* macros to check event flags
*/
/* prior events */
#define snd_seq_ev_is_prior(ev) (((ev)->flags & SNDRV_SEQ_PRIORITY_MASK) == SNDRV_SEQ_PRIORITY_HIGH)
/* event length type */
#define snd_seq_ev_length_type(ev) ((ev)->flags & SNDRV_SEQ_EVENT_LENGTH_MASK)
#define snd_seq_ev_is_fixed(ev) (snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_FIXED)
#define snd_seq_ev_is_variable(ev) (snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
#define snd_seq_ev_is_varusr(ev) (snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_VARUSR)
/* time-stamp type */
#define snd_seq_ev_timestamp_type(ev) ((ev)->flags & SNDRV_SEQ_TIME_STAMP_MASK)
#define snd_seq_ev_is_tick(ev) (snd_seq_ev_timestamp_type(ev) == SNDRV_SEQ_TIME_STAMP_TICK)
#define snd_seq_ev_is_real(ev) (snd_seq_ev_timestamp_type(ev) == SNDRV_SEQ_TIME_STAMP_REAL)
/* time-mode type */
#define snd_seq_ev_timemode_type(ev) ((ev)->flags & SNDRV_SEQ_TIME_MODE_MASK)
#define snd_seq_ev_is_abstime(ev) (snd_seq_ev_timemode_type(ev) == SNDRV_SEQ_TIME_MODE_ABS)
#define snd_seq_ev_is_reltime(ev) (snd_seq_ev_timemode_type(ev) == SNDRV_SEQ_TIME_MODE_REL)
/* queue sync port */
#define snd_seq_queue_sync_port(q) ((q) + 16)
#endif /* __SOUND_ASEQUENCER_H */

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/*
* Advanced Linux Sound Architecture - ALSA - Driver
* Copyright (c) 1994-2003 by Jaroslav Kysela <perex@perex.cz>,
* Abramo Bagnara <abramo@alsa-project.org>
*
*
* 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
*
*/
#ifndef __SOUND_ASOUND_H
#define __SOUND_ASOUND_H
#include <linux/ioctl.h>
#include <linux/time.h>
#include <asm/byteorder.h>
#ifdef __LITTLE_ENDIAN
#define SNDRV_LITTLE_ENDIAN
#else
#ifdef __BIG_ENDIAN
#define SNDRV_BIG_ENDIAN
#else
#error "Unsupported endian..."
#endif
#endif
#include <uapi/sound/asound.h>
#endif /* __SOUND_ASOUND_H */

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#ifndef __SOUND_ASOUNDEF_H
#define __SOUND_ASOUNDEF_H
/*
* Advanced Linux Sound Architecture - ALSA - Driver
* Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
*
*
* 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
*
*/
/****************************************************************************
* *
* Digital audio interface *
* *
****************************************************************************/
/* AES/IEC958 channel status bits */
#define IEC958_AES0_PROFESSIONAL (1<<0) /* 0 = consumer, 1 = professional */
#define IEC958_AES0_NONAUDIO (1<<1) /* 0 = audio, 1 = non-audio */
#define IEC958_AES0_PRO_EMPHASIS (7<<2) /* mask - emphasis */
#define IEC958_AES0_PRO_EMPHASIS_NOTID (0<<2) /* emphasis not indicated */
#define IEC958_AES0_PRO_EMPHASIS_NONE (1<<2) /* none emphasis */
#define IEC958_AES0_PRO_EMPHASIS_5015 (3<<2) /* 50/15us emphasis */
#define IEC958_AES0_PRO_EMPHASIS_CCITT (7<<2) /* CCITT J.17 emphasis */
#define IEC958_AES0_PRO_FREQ_UNLOCKED (1<<5) /* source sample frequency: 0 = locked, 1 = unlocked */
#define IEC958_AES0_PRO_FS (3<<6) /* mask - sample frequency */
#define IEC958_AES0_PRO_FS_NOTID (0<<6) /* fs not indicated */
#define IEC958_AES0_PRO_FS_44100 (1<<6) /* 44.1kHz */
#define IEC958_AES0_PRO_FS_48000 (2<<6) /* 48kHz */
#define IEC958_AES0_PRO_FS_32000 (3<<6) /* 32kHz */
#define IEC958_AES0_CON_NOT_COPYRIGHT (1<<2) /* 0 = copyright, 1 = not copyright */
#define IEC958_AES0_CON_EMPHASIS (7<<3) /* mask - emphasis */
#define IEC958_AES0_CON_EMPHASIS_NONE (0<<3) /* none emphasis */
#define IEC958_AES0_CON_EMPHASIS_5015 (1<<3) /* 50/15us emphasis */
#define IEC958_AES0_CON_MODE (3<<6) /* mask - mode */
#define IEC958_AES1_PRO_MODE (15<<0) /* mask - channel mode */
#define IEC958_AES1_PRO_MODE_NOTID (0<<0) /* not indicated */
#define IEC958_AES1_PRO_MODE_STEREOPHONIC (2<<0) /* stereophonic - ch A is left */
#define IEC958_AES1_PRO_MODE_SINGLE (4<<0) /* single channel */
#define IEC958_AES1_PRO_MODE_TWO (8<<0) /* two channels */
#define IEC958_AES1_PRO_MODE_PRIMARY (12<<0) /* primary/secondary */
#define IEC958_AES1_PRO_MODE_BYTE3 (15<<0) /* vector to byte 3 */
#define IEC958_AES1_PRO_USERBITS (15<<4) /* mask - user bits */
#define IEC958_AES1_PRO_USERBITS_NOTID (0<<4) /* not indicated */
#define IEC958_AES1_PRO_USERBITS_192 (8<<4) /* 192-bit structure */
#define IEC958_AES1_PRO_USERBITS_UDEF (12<<4) /* user defined application */
#define IEC958_AES1_CON_CATEGORY 0x7f
#define IEC958_AES1_CON_GENERAL 0x00
#define IEC958_AES1_CON_LASEROPT_MASK 0x07
#define IEC958_AES1_CON_LASEROPT_ID 0x01
#define IEC958_AES1_CON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x00)
#define IEC958_AES1_CON_NON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x08)
#define IEC958_AES1_CON_MINI_DISC (IEC958_AES1_CON_LASEROPT_ID|0x48)
#define IEC958_AES1_CON_DVD (IEC958_AES1_CON_LASEROPT_ID|0x18)
#define IEC958_AES1_CON_LASTEROPT_OTHER (IEC958_AES1_CON_LASEROPT_ID|0x78)
#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07
#define IEC958_AES1_CON_DIGDIGCONV_ID 0x02
#define IEC958_AES1_CON_PCM_CODER (IEC958_AES1_CON_DIGDIGCONV_ID|0x00)
#define IEC958_AES1_CON_MIXER (IEC958_AES1_CON_DIGDIGCONV_ID|0x10)
#define IEC958_AES1_CON_RATE_CONVERTER (IEC958_AES1_CON_DIGDIGCONV_ID|0x18)
#define IEC958_AES1_CON_SAMPLER (IEC958_AES1_CON_DIGDIGCONV_ID|0x20)
#define IEC958_AES1_CON_DSP (IEC958_AES1_CON_DIGDIGCONV_ID|0x28)
#define IEC958_AES1_CON_DIGDIGCONV_OTHER (IEC958_AES1_CON_DIGDIGCONV_ID|0x78)
#define IEC958_AES1_CON_MAGNETIC_MASK 0x07
#define IEC958_AES1_CON_MAGNETIC_ID 0x03
#define IEC958_AES1_CON_DAT (IEC958_AES1_CON_MAGNETIC_ID|0x00)
#define IEC958_AES1_CON_VCR (IEC958_AES1_CON_MAGNETIC_ID|0x08)
#define IEC958_AES1_CON_DCC (IEC958_AES1_CON_MAGNETIC_ID|0x40)
#define IEC958_AES1_CON_MAGNETIC_DISC (IEC958_AES1_CON_MAGNETIC_ID|0x18)
#define IEC958_AES1_CON_MAGNETIC_OTHER (IEC958_AES1_CON_MAGNETIC_ID|0x78)
#define IEC958_AES1_CON_BROADCAST1_MASK 0x07
#define IEC958_AES1_CON_BROADCAST1_ID 0x04
#define IEC958_AES1_CON_DAB_JAPAN (IEC958_AES1_CON_BROADCAST1_ID|0x00)
#define IEC958_AES1_CON_DAB_EUROPE (IEC958_AES1_CON_BROADCAST1_ID|0x08)
#define IEC958_AES1_CON_DAB_USA (IEC958_AES1_CON_BROADCAST1_ID|0x60)
#define IEC958_AES1_CON_SOFTWARE (IEC958_AES1_CON_BROADCAST1_ID|0x40)
#define IEC958_AES1_CON_IEC62105 (IEC958_AES1_CON_BROADCAST1_ID|0x20)
#define IEC958_AES1_CON_BROADCAST1_OTHER (IEC958_AES1_CON_BROADCAST1_ID|0x78)
#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f
#define IEC958_AES1_CON_BROADCAST2_ID 0x0e
#define IEC958_AES1_CON_MUSICAL_MASK 0x07
#define IEC958_AES1_CON_MUSICAL_ID 0x05
#define IEC958_AES1_CON_SYNTHESIZER (IEC958_AES1_CON_MUSICAL_ID|0x00)
#define IEC958_AES1_CON_MICROPHONE (IEC958_AES1_CON_MUSICAL_ID|0x08)
#define IEC958_AES1_CON_MUSICAL_OTHER (IEC958_AES1_CON_MUSICAL_ID|0x78)
#define IEC958_AES1_CON_ADC_MASK 0x1f
#define IEC958_AES1_CON_ADC_ID 0x06
#define IEC958_AES1_CON_ADC (IEC958_AES1_CON_ADC_ID|0x00)
#define IEC958_AES1_CON_ADC_OTHER (IEC958_AES1_CON_ADC_ID|0x60)
#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f
#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x16
#define IEC958_AES1_CON_ADC_COPYRIGHT (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x00)
#define IEC958_AES1_CON_ADC_COPYRIGHT_OTHER (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x60)
#define IEC958_AES1_CON_SOLIDMEM_MASK 0x0f
#define IEC958_AES1_CON_SOLIDMEM_ID 0x08
#define IEC958_AES1_CON_SOLIDMEM_DIGITAL_RECORDER_PLAYER (IEC958_AES1_CON_SOLIDMEM_ID|0x00)
#define IEC958_AES1_CON_SOLIDMEM_OTHER (IEC958_AES1_CON_SOLIDMEM_ID|0x70)
#define IEC958_AES1_CON_EXPERIMENTAL 0x40
#define IEC958_AES1_CON_ORIGINAL (1<<7) /* this bits depends on the category code */
#define IEC958_AES2_PRO_SBITS (7<<0) /* mask - sample bits */
#define IEC958_AES2_PRO_SBITS_20 (2<<0) /* 20-bit - coordination */
#define IEC958_AES2_PRO_SBITS_24 (4<<0) /* 24-bit - main audio */
#define IEC958_AES2_PRO_SBITS_UDEF (6<<0) /* user defined application */
#define IEC958_AES2_PRO_WORDLEN (7<<3) /* mask - source word length */
#define IEC958_AES2_PRO_WORDLEN_NOTID (0<<3) /* not indicated */
#define IEC958_AES2_PRO_WORDLEN_22_18 (2<<3) /* 22-bit or 18-bit */
#define IEC958_AES2_PRO_WORDLEN_23_19 (4<<3) /* 23-bit or 19-bit */
#define IEC958_AES2_PRO_WORDLEN_24_20 (5<<3) /* 24-bit or 20-bit */
#define IEC958_AES2_PRO_WORDLEN_20_16 (6<<3) /* 20-bit or 16-bit */
#define IEC958_AES2_CON_SOURCE (15<<0) /* mask - source number */
#define IEC958_AES2_CON_SOURCE_UNSPEC (0<<0) /* unspecified */
#define IEC958_AES2_CON_CHANNEL (15<<4) /* mask - channel number */
#define IEC958_AES2_CON_CHANNEL_UNSPEC (0<<4) /* unspecified */
#define IEC958_AES3_CON_FS (15<<0) /* mask - sample frequency */
#define IEC958_AES3_CON_FS_44100 (0<<0) /* 44.1kHz */
#define IEC958_AES3_CON_FS_NOTID (1<<0) /* non indicated */
#define IEC958_AES3_CON_FS_48000 (2<<0) /* 48kHz */
#define IEC958_AES3_CON_FS_32000 (3<<0) /* 32kHz */
#define IEC958_AES3_CON_FS_22050 (4<<0) /* 22.05kHz */
#define IEC958_AES3_CON_FS_24000 (6<<0) /* 24kHz */
#define IEC958_AES3_CON_FS_88200 (8<<0) /* 88.2kHz */
#define IEC958_AES3_CON_FS_768000 (9<<0) /* 768kHz */
#define IEC958_AES3_CON_FS_96000 (10<<0) /* 96kHz */
#define IEC958_AES3_CON_FS_176400 (12<<0) /* 176.4kHz */
#define IEC958_AES3_CON_FS_192000 (14<<0) /* 192kHz */
#define IEC958_AES3_CON_CLOCK (3<<4) /* mask - clock accuracy */
#define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /* 1000 ppm */
#define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /* 50 ppm */
#define IEC958_AES3_CON_CLOCK_VARIABLE (2<<4) /* variable pitch */
#define IEC958_AES4_CON_MAX_WORDLEN_24 (1<<0) /* 0 = 20-bit, 1 = 24-bit */
#define IEC958_AES4_CON_WORDLEN (7<<1) /* mask - sample word length */
#define IEC958_AES4_CON_WORDLEN_NOTID (0<<1) /* not indicated */
#define IEC958_AES4_CON_WORDLEN_20_16 (1<<1) /* 20-bit or 16-bit */
#define IEC958_AES4_CON_WORDLEN_22_18 (2<<1) /* 22-bit or 18-bit */
#define IEC958_AES4_CON_WORDLEN_23_19 (4<<1) /* 23-bit or 19-bit */
#define IEC958_AES4_CON_WORDLEN_24_20 (5<<1) /* 24-bit or 20-bit */
#define IEC958_AES4_CON_WORDLEN_21_17 (6<<1) /* 21-bit or 17-bit */
#define IEC958_AES4_CON_ORIGFS (15<<4) /* mask - original sample frequency */
#define IEC958_AES4_CON_ORIGFS_NOTID (0<<4) /* not indicated */
#define IEC958_AES4_CON_ORIGFS_192000 (1<<4) /* 192kHz */
#define IEC958_AES4_CON_ORIGFS_12000 (2<<4) /* 12kHz */
#define IEC958_AES4_CON_ORIGFS_176400 (3<<4) /* 176.4kHz */
#define IEC958_AES4_CON_ORIGFS_96000 (5<<4) /* 96kHz */
#define IEC958_AES4_CON_ORIGFS_8000 (6<<4) /* 8kHz */
#define IEC958_AES4_CON_ORIGFS_88200 (7<<4) /* 88.2kHz */
#define IEC958_AES4_CON_ORIGFS_16000 (8<<4) /* 16kHz */
#define IEC958_AES4_CON_ORIGFS_24000 (9<<4) /* 24kHz */
#define IEC958_AES4_CON_ORIGFS_11025 (10<<4) /* 11.025kHz */
#define IEC958_AES4_CON_ORIGFS_22050 (11<<4) /* 22.05kHz */
#define IEC958_AES4_CON_ORIGFS_32000 (12<<4) /* 32kHz */
#define IEC958_AES4_CON_ORIGFS_48000 (13<<4) /* 48kHz */
#define IEC958_AES4_CON_ORIGFS_44100 (15<<4) /* 44.1kHz */
#define IEC958_AES5_CON_CGMSA (3<<0) /* mask - CGMS-A */
#define IEC958_AES5_CON_CGMSA_COPYFREELY (0<<0) /* copying is permitted without restriction */
#define IEC958_AES5_CON_CGMSA_COPYONCE (1<<0) /* one generation of copies may be made */
#define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /* condition not be used */
#define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /* no copying is permitted */
/****************************************************************************
* *
* CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort *
* *
****************************************************************************/
#define CEA861_AUDIO_INFOFRAME_DB1CC (7<<0) /* mask - channel count */
#define CEA861_AUDIO_INFOFRAME_DB1CT (0xf<<4) /* mask - coding type */
#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */
#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958 (1<<4) /* IEC-60958 L-PCM */
#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3 (2<<4) /* AC-3 */
#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1 (3<<4) /* MPEG1 Layers 1 & 2 */
#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3 (4<<4) /* MPEG1 Layer 3 */
#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */
#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC (6<<4) /* AAC */
#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS (7<<4) /* DTS */
#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC (8<<4) /* ATRAC */
#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT (9<<4) /* One Bit Audio */
#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */
#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD (11<<4) /* DTS-HD */
#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT (12<<4) /* MAT (MLP) */
#define CEA861_AUDIO_INFOFRAME_DB1CT_DST (13<<4) /* DST */
#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO (14<<4) /* WMA Pro */
#define CEA861_AUDIO_INFOFRAME_DB2SF (7<<2) /* mask - sample frequency */
#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */
#define CEA861_AUDIO_INFOFRAME_DB2SF_32000 (1<<2) /* 32kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SF_44100 (2<<2) /* 44.1kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SF_48000 (3<<2) /* 48kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SF_88200 (4<<2) /* 88.2kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SF_96000 (5<<2) /* 96kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SF_176400 (6<<2) /* 176.4kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SF_192000 (7<<2) /* 192kHz */
#define CEA861_AUDIO_INFOFRAME_DB2SS (3<<0) /* mask - sample size */
#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */
#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT (1<<0) /* 16 bits */
#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT (2<<0) /* 20 bits */
#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT (3<<0) /* 24 bits */
#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH (1<<7) /* mask - inhibit downmixing */
#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */
#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */
#define CEA861_AUDIO_INFOFRAME_DB5_LSV (0xf<<3) /* mask - level-shift values */
/*****************************************************************************
* *
* MIDI v1.0 interface *
* *
*****************************************************************************/
#define MIDI_CHANNELS 16
#define MIDI_GM_DRUM_CHANNEL (10-1)
/*
* MIDI commands
*/
#define MIDI_CMD_NOTE_OFF 0x80
#define MIDI_CMD_NOTE_ON 0x90
#define MIDI_CMD_NOTE_PRESSURE 0xa0
#define MIDI_CMD_CONTROL 0xb0
#define MIDI_CMD_PGM_CHANGE 0xc0
#define MIDI_CMD_CHANNEL_PRESSURE 0xd0
#define MIDI_CMD_BENDER 0xe0
#define MIDI_CMD_COMMON_SYSEX 0xf0
#define MIDI_CMD_COMMON_MTC_QUARTER 0xf1
#define MIDI_CMD_COMMON_SONG_POS 0xf2
#define MIDI_CMD_COMMON_SONG_SELECT 0xf3
#define MIDI_CMD_COMMON_TUNE_REQUEST 0xf6
#define MIDI_CMD_COMMON_SYSEX_END 0xf7
#define MIDI_CMD_COMMON_CLOCK 0xf8
#define MIDI_CMD_COMMON_START 0xfa
#define MIDI_CMD_COMMON_CONTINUE 0xfb
#define MIDI_CMD_COMMON_STOP 0xfc
#define MIDI_CMD_COMMON_SENSING 0xfe
#define MIDI_CMD_COMMON_RESET 0xff
/*
* MIDI controllers
*/
#define MIDI_CTL_MSB_BANK 0x00
#define MIDI_CTL_MSB_MODWHEEL 0x01
#define MIDI_CTL_MSB_BREATH 0x02
#define MIDI_CTL_MSB_FOOT 0x04
#define MIDI_CTL_MSB_PORTAMENTO_TIME 0x05
#define MIDI_CTL_MSB_DATA_ENTRY 0x06
#define MIDI_CTL_MSB_MAIN_VOLUME 0x07
#define MIDI_CTL_MSB_BALANCE 0x08
#define MIDI_CTL_MSB_PAN 0x0a
#define MIDI_CTL_MSB_EXPRESSION 0x0b
#define MIDI_CTL_MSB_EFFECT1 0x0c
#define MIDI_CTL_MSB_EFFECT2 0x0d
#define MIDI_CTL_MSB_GENERAL_PURPOSE1 0x10
#define MIDI_CTL_MSB_GENERAL_PURPOSE2 0x11
#define MIDI_CTL_MSB_GENERAL_PURPOSE3 0x12
#define MIDI_CTL_MSB_GENERAL_PURPOSE4 0x13
#define MIDI_CTL_LSB_BANK 0x20
#define MIDI_CTL_LSB_MODWHEEL 0x21
#define MIDI_CTL_LSB_BREATH 0x22
#define MIDI_CTL_LSB_FOOT 0x24
#define MIDI_CTL_LSB_PORTAMENTO_TIME 0x25
#define MIDI_CTL_LSB_DATA_ENTRY 0x26
#define MIDI_CTL_LSB_MAIN_VOLUME 0x27
#define MIDI_CTL_LSB_BALANCE 0x28
#define MIDI_CTL_LSB_PAN 0x2a
#define MIDI_CTL_LSB_EXPRESSION 0x2b
#define MIDI_CTL_LSB_EFFECT1 0x2c
#define MIDI_CTL_LSB_EFFECT2 0x2d
#define MIDI_CTL_LSB_GENERAL_PURPOSE1 0x30
#define MIDI_CTL_LSB_GENERAL_PURPOSE2 0x31
#define MIDI_CTL_LSB_GENERAL_PURPOSE3 0x32
#define MIDI_CTL_LSB_GENERAL_PURPOSE4 0x33
#define MIDI_CTL_SUSTAIN 0x40
#define MIDI_CTL_PORTAMENTO 0x41
#define MIDI_CTL_SOSTENUTO 0x42
#define MIDI_CTL_SOFT_PEDAL 0x43
#define MIDI_CTL_LEGATO_FOOTSWITCH 0x44
#define MIDI_CTL_HOLD2 0x45
#define MIDI_CTL_SC1_SOUND_VARIATION 0x46
#define MIDI_CTL_SC2_TIMBRE 0x47
#define MIDI_CTL_SC3_RELEASE_TIME 0x48
#define MIDI_CTL_SC4_ATTACK_TIME 0x49
#define MIDI_CTL_SC5_BRIGHTNESS 0x4a
#define MIDI_CTL_SC6 0x4b
#define MIDI_CTL_SC7 0x4c
#define MIDI_CTL_SC8 0x4d
#define MIDI_CTL_SC9 0x4e
#define MIDI_CTL_SC10 0x4f
#define MIDI_CTL_GENERAL_PURPOSE5 0x50
#define MIDI_CTL_GENERAL_PURPOSE6 0x51
#define MIDI_CTL_GENERAL_PURPOSE7 0x52
#define MIDI_CTL_GENERAL_PURPOSE8 0x53
#define MIDI_CTL_PORTAMENTO_CONTROL 0x54
#define MIDI_CTL_E1_REVERB_DEPTH 0x5b
#define MIDI_CTL_E2_TREMOLO_DEPTH 0x5c
#define MIDI_CTL_E3_CHORUS_DEPTH 0x5d
#define MIDI_CTL_E4_DETUNE_DEPTH 0x5e
#define MIDI_CTL_E5_PHASER_DEPTH 0x5f
#define MIDI_CTL_DATA_INCREMENT 0x60
#define MIDI_CTL_DATA_DECREMENT 0x61
#define MIDI_CTL_NONREG_PARM_NUM_LSB 0x62
#define MIDI_CTL_NONREG_PARM_NUM_MSB 0x63
#define MIDI_CTL_REGIST_PARM_NUM_LSB 0x64
#define MIDI_CTL_REGIST_PARM_NUM_MSB 0x65
#define MIDI_CTL_ALL_SOUNDS_OFF 0x78
#define MIDI_CTL_RESET_CONTROLLERS 0x79
#define MIDI_CTL_LOCAL_CONTROL_SWITCH 0x7a
#define MIDI_CTL_ALL_NOTES_OFF 0x7b
#define MIDI_CTL_OMNI_OFF 0x7c
#define MIDI_CTL_OMNI_ON 0x7d
#define MIDI_CTL_MONO1 0x7e
#define MIDI_CTL_MONO2 0x7f
#endif /* __SOUND_ASOUNDEF_H */

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/*
* Driver for the Atmel Audio Bitstream DAC (ABDAC)
*
* Copyright (C) 2009 Atmel Corporation
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License version 2 as published
* by the Free Software Foundation.
*/
#ifndef __INCLUDE_SOUND_ATMEL_ABDAC_H
#define __INCLUDE_SOUND_ATMEL_ABDAC_H
#include <linux/dw_dmac.h>
/**
* struct atmel_abdac_pdata - board specific ABDAC configuration
* @dws: DMA slave interface to use for sound playback.
*/
struct atmel_abdac_pdata {
struct dw_dma_slave dws;
};
#endif /* __INCLUDE_SOUND_ATMEL_ABDAC_H */

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/*
* Driver for the Atmel AC97C controller
*
* Copyright (C) 2005-2009 Atmel Corporation
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License version 2 as published
* by the Free Software Foundation.
*/
#ifndef __INCLUDE_SOUND_ATMEL_AC97C_H
#define __INCLUDE_SOUND_ATMEL_AC97C_H
#include <linux/dw_dmac.h>
#define AC97C_CAPTURE 0x01
#define AC97C_PLAYBACK 0x02
#define AC97C_BOTH (AC97C_CAPTURE | AC97C_PLAYBACK)
/**
* struct atmel_ac97c_pdata - board specific AC97C configuration
* @rx_dws: DMA slave interface to use for sound capture.
* @tx_dws: DMA slave interface to use for sound playback.
* @reset_pin: GPIO pin wired to the reset input on the external AC97 codec,
* optional to use, set to -ENODEV if not in use. AC97 layer will
* try to do a software reset of the external codec anyway.
* @flags: Flags for which directions should be enabled.
*
* If the user do not want to use a DMA channel for playback or capture, i.e.
* only one feature is required on the board. The slave for playback or capture
* can be set to NULL. The AC97C driver will take use of this when setting up
* the sound streams.
*/
struct ac97c_platform_data {
struct dw_dma_slave rx_dws;
struct dw_dma_slave tx_dws;
unsigned int flags;
int reset_pin;
};
#endif /* __INCLUDE_SOUND_ATMEL_AC97C_H */

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/*
* compress_driver.h - compress offload driver definations
*
* Copyright (C) 2011 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
* Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* 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; version 2 of the License.
*
* 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.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
*/
#ifndef __COMPRESS_DRIVER_H
#define __COMPRESS_DRIVER_H
#include <linux/types.h>
#include <linux/sched.h>
#include <sound/compress_offload.h>
#include <sound/asound.h>
#include <sound/pcm.h>
struct snd_compr_ops;
/**
* struct snd_compr_runtime: runtime stream description
* @state: stream state
* @ops: pointer to DSP callbacks
* @buffer: pointer to kernel buffer, valid only when not in mmap mode or
* DSP doesn't implement copy
* @buffer_size: size of the above buffer
* @fragment_size: size of buffer fragment in bytes
* @fragments: number of such fragments
* @hw_pointer: offset of last location in buffer where DSP copied data
* @app_pointer: offset of last location in buffer where app wrote data
* @total_bytes_available: cumulative number of bytes made available in
* the ring buffer
* @total_bytes_transferred: cumulative bytes transferred by offload DSP
* @sleep: poll sleep
*/
struct snd_compr_runtime {
snd_pcm_state_t state;
struct snd_compr_ops *ops;
void *buffer;
u64 buffer_size;
u32 fragment_size;
u32 fragments;
u64 total_bytes_available;
u64 total_bytes_transferred;
wait_queue_head_t sleep;
void *private_data;
};
/**
* struct snd_compr_stream: compressed stream
* @name: device name
* @ops: pointer to DSP callbacks
* @runtime: pointer to runtime structure
* @device: device pointer
* @direction: stream direction, playback/recording
* @metadata_set: metadata set flag, true when set
* @next_track: has userspace signall next track transistion, true when set
* @private_data: pointer to DSP private data
*/
struct snd_compr_stream {
const char *name;
struct snd_compr_ops *ops;
struct snd_compr_runtime *runtime;
struct snd_compr *device;
enum snd_compr_direction direction;
bool metadata_set;
bool next_track;
void *private_data;
};
/**
* struct snd_compr_ops: compressed path DSP operations
* @open: Open the compressed stream
* This callback is mandatory and shall keep dsp ready to receive the stream
* parameter
* @free: Close the compressed stream, mandatory
* @set_params: Sets the compressed stream parameters, mandatory
* This can be called in during stream creation only to set codec params
* and the stream properties
* @get_params: retrieve the codec parameters, mandatory
* @trigger: Trigger operations like start, pause, resume, drain, stop.
* This callback is mandatory
* @pointer: Retrieve current h/w pointer information. Mandatory
* @copy: Copy the compressed data to/from userspace, Optional
* Can't be implemented if DSP supports mmap
* @mmap: DSP mmap method to mmap DSP memory
* @ack: Ack for DSP when data is written to audio buffer, Optional
* Not valid if copy is implemented
* @get_caps: Retrieve DSP capabilities, mandatory
* @get_codec_caps: Retrieve capabilities for a specific codec, mandatory
*/
struct snd_compr_ops {
int (*open)(struct snd_compr_stream *stream);
int (*free)(struct snd_compr_stream *stream);
int (*set_params)(struct snd_compr_stream *stream,
struct snd_compr_params *params);
int (*get_params)(struct snd_compr_stream *stream,
struct snd_codec *params);
int (*set_metadata)(struct snd_compr_stream *stream,
struct snd_compr_metadata *metadata);
int (*get_metadata)(struct snd_compr_stream *stream,
struct snd_compr_metadata *metadata);
int (*trigger)(struct snd_compr_stream *stream, int cmd);
int (*pointer)(struct snd_compr_stream *stream,
struct snd_compr_tstamp *tstamp);
int (*copy)(struct snd_compr_stream *stream, char __user *buf,
size_t count);
int (*mmap)(struct snd_compr_stream *stream,
struct vm_area_struct *vma);
int (*ack)(struct snd_compr_stream *stream, size_t bytes);
int (*get_caps) (struct snd_compr_stream *stream,
struct snd_compr_caps *caps);
int (*get_codec_caps) (struct snd_compr_stream *stream,
struct snd_compr_codec_caps *codec);
};
/**
* struct snd_compr: Compressed device
* @name: DSP device name
* @dev: Device pointer
* @ops: pointer to DSP callbacks
* @private_data: pointer to DSP pvt data
* @card: sound card pointer
* @direction: Playback or capture direction
* @lock: device lock
* @device: device id
*/
struct snd_compr {
const char *name;
struct device *dev;
struct snd_compr_ops *ops;
void *private_data;
struct snd_card *card;
unsigned int direction;
struct mutex lock;
int device;
};
/* compress device register APIs */
int snd_compress_register(struct snd_compr *device);
int snd_compress_deregister(struct snd_compr *device);
int snd_compress_new(struct snd_card *card, int device,
int type, struct snd_compr *compr);
/* dsp driver callback apis
* For playback: driver should call snd_compress_fragment_elapsed() to let the
* framework know that a fragment has been consumed from the ring buffer
*
* For recording: we want to know when a frame is available or when
* at least one frame is available so snd_compress_frame_elapsed()
* callback should be called when a encodeded frame is available
*/
static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream)
{
wake_up(&stream->runtime->sleep);
}
#endif

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#ifndef __SOUND_CONTROL_H
#define __SOUND_CONTROL_H
/*
* Header file for control interface
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/asound.h>
#define snd_kcontrol_chip(kcontrol) ((kcontrol)->private_data)
struct snd_kcontrol;
typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_info * uinfo);
typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol,
int op_flag, /* 0=read,1=write,-1=command */
unsigned int size,
unsigned int __user *tlv);
struct snd_kcontrol_new {
snd_ctl_elem_iface_t iface; /* interface identifier */
unsigned int device; /* device/client number */
unsigned int subdevice; /* subdevice (substream) number */
const unsigned char *name; /* ASCII name of item */
unsigned int index; /* index of item */
unsigned int access; /* access rights */
unsigned int count; /* count of same elements */
snd_kcontrol_info_t *info;
snd_kcontrol_get_t *get;
snd_kcontrol_put_t *put;
union {
snd_kcontrol_tlv_rw_t *c;
const unsigned int *p;
} tlv;
unsigned long private_value;
};
struct snd_kcontrol_volatile {
struct snd_ctl_file *owner; /* locked */
unsigned int access; /* access rights */
};
struct snd_kcontrol {
struct list_head list; /* list of controls */
struct snd_ctl_elem_id id;
unsigned int count; /* count of same elements */
snd_kcontrol_info_t *info;
snd_kcontrol_get_t *get;
snd_kcontrol_put_t *put;
union {
snd_kcontrol_tlv_rw_t *c;
const unsigned int *p;
} tlv;
unsigned long private_value;
void *private_data;
void (*private_free)(struct snd_kcontrol *kcontrol);
struct snd_kcontrol_volatile vd[0]; /* volatile data */
};
#define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list)
struct snd_kctl_event {
struct list_head list; /* list of events */
struct snd_ctl_elem_id id;
unsigned int mask;
};
#define snd_kctl_event(n) list_entry(n, struct snd_kctl_event, list)
struct pid;
struct snd_ctl_file {
struct list_head list; /* list of all control files */
struct snd_card *card;
struct pid *pid;
int prefer_pcm_subdevice;
int prefer_rawmidi_subdevice;
wait_queue_head_t change_sleep;
spinlock_t read_lock;
struct fasync_struct *fasync;
int subscribed; /* read interface is activated */
struct list_head events; /* waiting events for read */
};
#define snd_ctl_file(n) list_entry(n, struct snd_ctl_file, list)
typedef int (*snd_kctl_ioctl_func_t) (struct snd_card * card,
struct snd_ctl_file * control,
unsigned int cmd, unsigned long arg);
void snd_ctl_notify(struct snd_card * card, unsigned int mask, struct snd_ctl_elem_id * id);
struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, void * private_data);
void snd_ctl_free_one(struct snd_kcontrol * kcontrol);
int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol);
int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol);
int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, bool add_on_replace);
int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id);
int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id);
int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
int active);
struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid);
struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id);
int snd_ctl_create(struct snd_card *card);
int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn);
int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn);
#ifdef CONFIG_COMPAT
int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn);
int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn);
#else
#define snd_ctl_register_ioctl_compat(fcn)
#define snd_ctl_unregister_ioctl_compat(fcn)
#endif
static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
{
return id->numid - kctl->id.numid;
}
static inline unsigned int snd_ctl_get_ioffidx(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
{
return id->index - kctl->id.index;
}
static inline unsigned int snd_ctl_get_ioff(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
{
if (id->numid) {
return snd_ctl_get_ioffnum(kctl, id);
} else {
return snd_ctl_get_ioffidx(kctl, id);
}
}
static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id *dst_id,
struct snd_kcontrol *src_kctl,
unsigned int offset)
{
*dst_id = src_kctl->id;
dst_id->index += offset;
dst_id->numid += offset;
return dst_id;
}
/*
* Frequently used control callbacks/helpers
*/
int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
unsigned int items, const char *const names[]);
/*
* virtual master control
*/
struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
const unsigned int *tlv);
int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
unsigned int flags);
/* optional flags for slave */
#define SND_CTL_SLAVE_NEED_UPDATE (1 << 0)
/**
* snd_ctl_add_slave - Add a virtual slave control
* @master: vmaster element
* @slave: slave element to add
*
* Add a virtual slave control to the given master element created via
* snd_ctl_create_virtual_master() beforehand.
*
* All slaves must be the same type (returning the same information
* via info callback). The function doesn't check it, so it's your
* responsibility.
*
* Also, some additional limitations:
* at most two channels,
* logarithmic volume control (dB level) thus no linear volume,
* master can only attenuate the volume without gain
*
* Return: Zero if successful or a negative error code.
*/
static inline int
snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
{
return _snd_ctl_add_slave(master, slave, 0);
}
/**
* snd_ctl_add_slave_uncached - Add a virtual slave control
* @master: vmaster element
* @slave: slave element to add
*
* Add a virtual slave control to the given master.
* Unlike snd_ctl_add_slave(), the element added via this function
* is supposed to have volatile values, and get callback is called
* at each time quried from the master.
*
* When the control peeks the hardware values directly and the value
* can be changed by other means than the put callback of the element,
* this function should be used to keep the value always up-to-date.
*
* Return: Zero if successful or a negative error code.
*/
static inline int
snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
struct snd_kcontrol *slave)
{
return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
}
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
void (*hook)(void *private_data, int),
void *private_data);
void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kctl);
/*
* Helper functions for jack-detection controls
*/
struct snd_kcontrol *
snd_kctl_jack_new(const char *name, int idx, void *private_data);
void snd_kctl_jack_report(struct snd_card *card,
struct snd_kcontrol *kctl, bool status);
#endif /* __SOUND_CONTROL_H */

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#ifndef __SOUND_CORE_H
#define __SOUND_CORE_H
/*
* Main header file for the ALSA driver
* Copyright (c) 1994-2001 by Jaroslav Kysela <perex@perex.cz>
*
*
* 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 <linux/sched.h> /* wake_up() */
#include <linux/mutex.h> /* struct mutex */
#include <linux/rwsem.h> /* struct rw_semaphore */
#include <linux/pm.h> /* pm_message_t */
#include <linux/stringify.h>
/* number of supported soundcards */
#ifdef CONFIG_SND_DYNAMIC_MINORS
#define SNDRV_CARDS 32
#else
#define SNDRV_CARDS 8 /* don't change - minor numbers */
#endif
#define CONFIG_SND_MAJOR 116 /* standard configuration */
/* forward declarations */
struct pci_dev;
struct module;
struct device;
struct device_attribute;
/* device allocation stuff */
#define SNDRV_DEV_TYPE_RANGE_SIZE 0x1000
typedef int __bitwise snd_device_type_t;
#define SNDRV_DEV_TOPLEVEL ((__force snd_device_type_t) 0)
#define SNDRV_DEV_CONTROL ((__force snd_device_type_t) 1)
#define SNDRV_DEV_LOWLEVEL_PRE ((__force snd_device_type_t) 2)
#define SNDRV_DEV_LOWLEVEL_NORMAL ((__force snd_device_type_t) 0x1000)
#define SNDRV_DEV_PCM ((__force snd_device_type_t) 0x1001)
#define SNDRV_DEV_RAWMIDI ((__force snd_device_type_t) 0x1002)
#define SNDRV_DEV_TIMER ((__force snd_device_type_t) 0x1003)
#define SNDRV_DEV_SEQUENCER ((__force snd_device_type_t) 0x1004)
#define SNDRV_DEV_HWDEP ((__force snd_device_type_t) 0x1005)
#define SNDRV_DEV_INFO ((__force snd_device_type_t) 0x1006)
#define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007)
#define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008)
#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009)
#define SNDRV_DEV_COMPRESS ((__force snd_device_type_t) 0x100A)
#define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000)
typedef int __bitwise snd_device_state_t;
#define SNDRV_DEV_BUILD ((__force snd_device_state_t) 0)
#define SNDRV_DEV_REGISTERED ((__force snd_device_state_t) 1)
#define SNDRV_DEV_DISCONNECTED ((__force snd_device_state_t) 2)
typedef int __bitwise snd_device_cmd_t;
#define SNDRV_DEV_CMD_PRE ((__force snd_device_cmd_t) 0)
#define SNDRV_DEV_CMD_NORMAL ((__force snd_device_cmd_t) 1)
#define SNDRV_DEV_CMD_POST ((__force snd_device_cmd_t) 2)
struct snd_device;
struct snd_device_ops {
int (*dev_free)(struct snd_device *dev);
int (*dev_register)(struct snd_device *dev);
int (*dev_disconnect)(struct snd_device *dev);
};
struct snd_device {
struct list_head list; /* list of registered devices */
struct snd_card *card; /* card which holds this device */
snd_device_state_t state; /* state of the device */
snd_device_type_t type; /* device type */
void *device_data; /* device structure */
struct snd_device_ops *ops; /* operations */
};
#define snd_device(n) list_entry(n, struct snd_device, list)
/* main structure for soundcard */
struct snd_card {
int number; /* number of soundcard (index to
snd_cards) */
char id[16]; /* id string of this card */
char driver[16]; /* driver name */
char shortname[32]; /* short name of this soundcard */
char longname[80]; /* name of this soundcard */
char mixername[80]; /* mixer name */
char components[128]; /* card components delimited with
space */
struct module *module; /* top-level module */
void *private_data; /* private data for soundcard */
void (*private_free) (struct snd_card *card); /* callback for freeing of
private data */
struct list_head devices; /* devices */
unsigned int last_numid; /* last used numeric ID */
struct rw_semaphore controls_rwsem; /* controls list lock */
rwlock_t ctl_files_rwlock; /* ctl_files list lock */
int controls_count; /* count of all controls */
int user_ctl_count; /* count of all user controls */
struct list_head controls; /* all controls for this card */
struct list_head ctl_files; /* active control files */
struct mutex user_ctl_lock; /* protects user controls against
concurrent access */
struct snd_info_entry *proc_root; /* root for soundcard specific files */
struct snd_info_entry *proc_id; /* the card id */
struct proc_dir_entry *proc_root_link; /* number link to real id */
struct list_head files_list; /* all files associated to this card */
struct snd_shutdown_f_ops *s_f_ops; /* file operations in the shutdown
state */
spinlock_t files_lock; /* lock the files for this card */
int shutdown; /* this card is going down */
int free_on_last_close; /* free in context of file_release */
wait_queue_head_t shutdown_sleep;
atomic_t refcount; /* refcount for disconnection */
struct device *dev; /* device assigned to this card */
struct device *card_dev; /* cardX object for sysfs */
#ifdef CONFIG_PM
unsigned int power_state; /* power state */
struct mutex power_lock; /* power lock */
wait_queue_head_t power_sleep;
#endif
#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
struct snd_mixer_oss *mixer_oss;
int mixer_oss_change_count;
#endif
};
#ifdef CONFIG_PM
static inline void snd_power_lock(struct snd_card *card)
{
mutex_lock(&card->power_lock);
}
static inline void snd_power_unlock(struct snd_card *card)
{
mutex_unlock(&card->power_lock);
}
static inline unsigned int snd_power_get_state(struct snd_card *card)
{
return card->power_state;
}
static inline void snd_power_change_state(struct snd_card *card, unsigned int state)
{
card->power_state = state;
wake_up(&card->power_sleep);
}
/* init.c */
int snd_power_wait(struct snd_card *card, unsigned int power_state);
#else /* ! CONFIG_PM */
#define snd_power_lock(card) do { (void)(card); } while (0)
#define snd_power_unlock(card) do { (void)(card); } while (0)
static inline int snd_power_wait(struct snd_card *card, unsigned int state) { return 0; }
#define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; })
#define snd_power_change_state(card, state) do { (void)(card); } while (0)
#endif /* CONFIG_PM */
struct snd_minor {
int type; /* SNDRV_DEVICE_TYPE_XXX */
int card; /* card number */
int device; /* device number */
const struct file_operations *f_ops; /* file operations */
void *private_data; /* private data for f_ops->open */
struct device *dev; /* device for sysfs */
struct snd_card *card_ptr; /* assigned card instance */
};
/* return a device pointer linked to each sound device as a parent */
static inline struct device *snd_card_get_device_link(struct snd_card *card)
{
return card ? card->card_dev : NULL;
}
/* sound.c */
extern int snd_major;
extern int snd_ecards_limit;
extern struct class *sound_class;
void snd_request_card(int card);
int snd_register_device_for_dev(int type, struct snd_card *card,
int dev,
const struct file_operations *f_ops,
void *private_data,
const char *name,
struct device *device);
/**
* snd_register_device - Register the ALSA device file for the card
* @type: the device type, SNDRV_DEVICE_TYPE_XXX
* @card: the card instance
* @dev: the device index
* @f_ops: the file operations
* @private_data: user pointer for f_ops->open()
* @name: the device file name
*
* Registers an ALSA device file for the given card.
* The operators have to be set in reg parameter.
*
* This function uses the card's device pointer to link to the
* correct &struct device.
*
* Return: Zero if successful, or a negative error code on failure.
*/
static inline int snd_register_device(int type, struct snd_card *card, int dev,
const struct file_operations *f_ops,
void *private_data,
const char *name)
{
return snd_register_device_for_dev(type, card, dev, f_ops,
private_data, name,
snd_card_get_device_link(card));
}
int snd_unregister_device(int type, struct snd_card *card, int dev);
void *snd_lookup_minor_data(unsigned int minor, int type);
int snd_add_device_sysfs_file(int type, struct snd_card *card, int dev,
struct device_attribute *attr);
#ifdef CONFIG_SND_OSSEMUL
int snd_register_oss_device(int type, struct snd_card *card, int dev,
const struct file_operations *f_ops, void *private_data,
const char *name);
int snd_unregister_oss_device(int type, struct snd_card *card, int dev);
void *snd_lookup_oss_minor_data(unsigned int minor, int type);
#endif
int snd_minor_info_init(void);
int snd_minor_info_done(void);
/* sound_oss.c */
#ifdef CONFIG_SND_OSSEMUL
int snd_minor_info_oss_init(void);
int snd_minor_info_oss_done(void);
#else
static inline int snd_minor_info_oss_init(void) { return 0; }
static inline int snd_minor_info_oss_done(void) { return 0; }
#endif
/* memory.c */
int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count);
int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count);
/* init.c */
extern struct snd_card *snd_cards[SNDRV_CARDS];
int snd_card_locked(int card);
#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
#define SND_MIXER_OSS_NOTIFY_REGISTER 0
#define SND_MIXER_OSS_NOTIFY_DISCONNECT 1
#define SND_MIXER_OSS_NOTIFY_FREE 2
extern int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int cmd);
#endif
int snd_card_create(int idx, const char *id,
struct module *module, int extra_size,
struct snd_card **card_ret);
int snd_card_disconnect(struct snd_card *card);
int snd_card_free(struct snd_card *card);
int snd_card_free_when_closed(struct snd_card *card);
void snd_card_set_id(struct snd_card *card, const char *id);
int snd_card_register(struct snd_card *card);
int snd_card_info_init(void);
int snd_card_info_done(void);
int snd_component_add(struct snd_card *card, const char *component);
int snd_card_file_add(struct snd_card *card, struct file *file);
int snd_card_file_remove(struct snd_card *card, struct file *file);
void snd_card_unref(struct snd_card *card);
#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr))
/* device.c */
int snd_device_new(struct snd_card *card, snd_device_type_t type,
void *device_data, struct snd_device_ops *ops);
int snd_device_register(struct snd_card *card, void *device_data);
int snd_device_register_all(struct snd_card *card);
int snd_device_disconnect(struct snd_card *card, void *device_data);
int snd_device_disconnect_all(struct snd_card *card);
int snd_device_free(struct snd_card *card, void *device_data);
int snd_device_free_all(struct snd_card *card, snd_device_cmd_t cmd);
/* isadma.c */
#ifdef CONFIG_ISA_DMA_API
#define DMA_MODE_NO_ENABLE 0x0100
void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode);
void snd_dma_disable(unsigned long dma);
unsigned int snd_dma_pointer(unsigned long dma, unsigned int size);
#endif
/* misc.c */
struct resource;
void release_and_free_resource(struct resource *res);
/* --- */
/* sound printk debug levels */
enum {
SND_PR_ALWAYS,
SND_PR_DEBUG,
SND_PR_VERBOSE,
};
#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
__printf(4, 5)
void __snd_printk(unsigned int level, const char *file, int line,
const char *format, ...);
#else
#define __snd_printk(level, file, line, format, args...) \
printk(format, ##args)
#endif
/**
* snd_printk - printk wrapper
* @fmt: format string
*
* Works like printk() but prints the file and the line of the caller
* when configured with CONFIG_SND_VERBOSE_PRINTK.
*/
#define snd_printk(fmt, args...) \
__snd_printk(0, __FILE__, __LINE__, fmt, ##args)
#ifdef CONFIG_SND_DEBUG
/**
* snd_printd - debug printk
* @fmt: format string
*
* Works like snd_printk() for debugging purposes.
* Ignored when CONFIG_SND_DEBUG is not set.
*/
#define snd_printd(fmt, args...) \
__snd_printk(1, __FILE__, __LINE__, fmt, ##args)
#define _snd_printd(level, fmt, args...) \
__snd_printk(level, __FILE__, __LINE__, fmt, ##args)
/**
* snd_BUG - give a BUG warning message and stack trace
*
* Calls WARN() if CONFIG_SND_DEBUG is set.
* Ignored when CONFIG_SND_DEBUG is not set.
*/
#define snd_BUG() WARN(1, "BUG?\n")
/**
* snd_BUG_ON - debugging check macro
* @cond: condition to evaluate
*
* Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set,
* otherwise just evaluates the conditional and returns the value.
*/
#define snd_BUG_ON(cond) WARN_ON((cond))
#else /* !CONFIG_SND_DEBUG */
__printf(1, 2)
static inline void snd_printd(const char *format, ...) {}
__printf(2, 3)
static inline void _snd_printd(int level, const char *format, ...) {}
#define snd_BUG() do { } while (0)
#define snd_BUG_ON(condition) ({ \
int __ret_warn_on = !!(condition); \
unlikely(__ret_warn_on); \
})
#endif /* CONFIG_SND_DEBUG */
#ifdef CONFIG_SND_DEBUG_VERBOSE
/**
* snd_printdd - debug printk
* @format: format string
*
* Works like snd_printk() for debugging purposes.
* Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
*/
#define snd_printdd(format, args...) \
__snd_printk(2, __FILE__, __LINE__, format, ##args)
#else
__printf(1, 2)
static inline void snd_printdd(const char *format, ...) {}
#endif
#define SNDRV_OSS_VERSION ((3<<16)|(8<<8)|(1<<4)|(0)) /* 3.8.1a */
/* for easier backward-porting */
#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE)
#define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev))
#define gameport_set_port_data(gp,r) ((gp)->port_data = (r))
#define gameport_get_port_data(gp) (gp)->port_data
#endif
/* PCI quirk list helper */
struct snd_pci_quirk {
unsigned short subvendor; /* PCI subvendor ID */
unsigned short subdevice; /* PCI subdevice ID */
unsigned short subdevice_mask; /* bitmask to match */
int value; /* value */
#ifdef CONFIG_SND_DEBUG_VERBOSE
const char *name; /* name of the device (optional) */
#endif
};
#define _SND_PCI_QUIRK_ID_MASK(vend, mask, dev) \
.subvendor = (vend), .subdevice = (dev), .subdevice_mask = (mask)
#define _SND_PCI_QUIRK_ID(vend, dev) \
_SND_PCI_QUIRK_ID_MASK(vend, 0xffff, dev)
#define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)}
#ifdef CONFIG_SND_DEBUG_VERBOSE
#define SND_PCI_QUIRK(vend,dev,xname,val) \
{_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)}
#define SND_PCI_QUIRK_VENDOR(vend, xname, val) \
{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val), .name = (xname)}
#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \
{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), \
.value = (val), .name = (xname)}
#define snd_pci_quirk_name(q) ((q)->name)
#else
#define SND_PCI_QUIRK(vend,dev,xname,val) \
{_SND_PCI_QUIRK_ID(vend, dev), .value = (val)}
#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \
{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)}
#define SND_PCI_QUIRK_VENDOR(vend, xname, val) \
{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)}
#define snd_pci_quirk_name(q) ""
#endif
#ifdef CONFIG_PCI
const struct snd_pci_quirk *
snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list);
const struct snd_pci_quirk *
snd_pci_quirk_lookup_id(u16 vendor, u16 device,
const struct snd_pci_quirk *list);
#else
static inline const struct snd_pci_quirk *
snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list)
{
return NULL;
}
static inline const struct snd_pci_quirk *
snd_pci_quirk_lookup_id(u16 vendor, u16 device,
const struct snd_pci_quirk *list)
{
return NULL;
}
#endif
#endif /* __SOUND_CORE_H */

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#ifndef __SOUND_CS4231_REGS_H
#define __SOUND_CS4231_REGS_H
/*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Definitions for CS4231 & InterWave chips & compatible chips registers
*
*
* 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
*
*/
/* IO ports */
#define CS4231P(x) (c_d_c_CS4231##x)
#define c_d_c_CS4231REGSEL 0
#define c_d_c_CS4231REG 1
#define c_d_c_CS4231STATUS 2
#define c_d_c_CS4231PIO 3
/* codec registers */
#define CS4231_LEFT_INPUT 0x00 /* left input control */
#define CS4231_RIGHT_INPUT 0x01 /* right input control */
#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */
#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */
#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */
#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */
#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */
#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */
#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */
#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */
#define CS4231_PIN_CTRL 0x0a /* pin control */
#define CS4231_TEST_INIT 0x0b /* test and initialization */
#define CS4231_MISC_INFO 0x0c /* miscellaneous information */
#define CS4231_LOOPBACK 0x0d /* loopback control */
#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */
#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */
#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */
#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */
#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */
#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */
#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */
#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */
#define CS4231_TIMER_LOW 0x14 /* timer low byte */
#define CS4231_TIMER_HIGH 0x15 /* timer high byte */
#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */
#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */
#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */
#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */
#define CS4236_EXT_REG 0x17 /* extended register access */
#define CS4231_IRQ_STATUS 0x18 /* irq status register */
#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */
#define CS4231_VERSION 0x19 /* CS4231(A) - version values */
#define CS4231_MONO_CTRL 0x1a /* mono input/output control */
#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */
#define AD1845_PWR_DOWN 0x1b /* power down control */
#define CS4235_LEFT_MASTER 0x1b /* left master output control */
#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */
#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */
#define CS4235_RIGHT_MASTER 0x1d /* right master output control */
#define CS4231_REC_UPR_CNT 0x1e /* record upper count */
#define CS4231_REC_LWR_CNT 0x1f /* record lower count */
/* definitions for codec register select port - CODECP( REGSEL ) */
#define CS4231_INIT 0x80 /* CODEC is initializing */
#define CS4231_MCE 0x40 /* mode change enable */
#define CS4231_TRD 0x20 /* transfer request disable */
/* definitions for codec status register - CODECP( STATUS ) */
#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */
/* definitions for codec irq status */
#define CS4231_PLAYBACK_IRQ 0x10
#define CS4231_RECORD_IRQ 0x20
#define CS4231_TIMER_IRQ 0x40
#define CS4231_ALL_IRQS 0x70
#define CS4231_REC_UNDERRUN 0x08
#define CS4231_REC_OVERRUN 0x04
#define CS4231_PLY_OVERRUN 0x02
#define CS4231_PLY_UNDERRUN 0x01
/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
#define CS4231_ENABLE_MIC_GAIN 0x20
#define CS4231_MIXS_LINE 0x00
#define CS4231_MIXS_AUX1 0x40
#define CS4231_MIXS_MIC 0x80
#define CS4231_MIXS_ALL 0xc0
/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */
#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */
#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */
#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */
#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */
#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */
#define CS4231_STEREO 0x10 /* stereo mode */
/* bits 3-1 define frequency divisor */
#define CS4231_XTAL1 0x00 /* 24.576 crystal */
#define CS4231_XTAL2 0x01 /* 16.9344 crystal */
/* definitions for interface control register - CS4231_IFACE_CTRL */
#define CS4231_RECORD_PIO 0x80 /* record PIO enable */
#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */
#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */
#define CS4231_AUTOCALIB 0x08 /* auto calibrate */
#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */
#define CS4231_RECORD_ENABLE 0x02 /* record enable */
#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */
/* definitions for pin control register - CS4231_PIN_CTRL */
#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */
#define CS4231_XCTL1 0x40 /* external control #1 */
#define CS4231_XCTL0 0x80 /* external control #0 */
/* definitions for test and init register - CS4231_TEST_INIT */
#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */
#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */
/* definitions for misc control register - CS4231_MISC_INFO */
#define CS4231_MODE2 0x40 /* MODE 2 */
#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */
#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */
/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
#define CS4231_DACZ 0x01 /* zero DAC when underrun */
#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */
#define CS4231_OLB 0x80 /* output level bit */
/* definitions for Extended Registers - CS4236+ */
#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */
#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */
#define CS4236_LEFT_MIC 0x28 /* left MIC volume */
#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */
#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */
#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */
#define CS4236_LEFT_FM 0x68 /* left FM volume */
#define CS4236_RIGHT_FM 0x78 /* right FM volume */
#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */
#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */
#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */
#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */
#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */
#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */
#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */
#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */
#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */
#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */
#define CS4236_VERSION 0x9c /* chip version and ID */
/* definitions for extended registers - OPTI93X */
#define OPTi931_AUX_LEFT_INPUT 0x10
#define OPTi931_AUX_RIGHT_INPUT 0x11
#define OPTi93X_MIC_LEFT_INPUT 0x14
#define OPTi93X_MIC_RIGHT_INPUT 0x15
#define OPTi93X_OUT_LEFT 0x16
#define OPTi93X_OUT_RIGHT 0x17
#endif /* __SOUND_CS4231_REGS_H */

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/*
* Definitions for CS4271 ASoC codec driver
*
* Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
*
* 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.
*/
#ifndef __CS4271_H
#define __CS4271_H
struct cs4271_platform_data {
int gpio_nreset; /* GPIO driving Reset pin, if any */
bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */
/*
* The CS4271 requires its LRCLK and MCLK to be stable before its RESET
* line is de-asserted. That also means that clocks cannot be changed
* without putting the chip back into hardware reset, which also requires
* a complete re-initialization of all registers.
*
* One (undocumented) workaround is to assert and de-assert the PDN bit
* in the MODE2 register. This workaround can be enabled with the
* following flag.
*
* Note that this is not needed in case the clocks are stable
* throughout the entire runtime of the codec.
*/
bool enable_soft_reset;
};
#endif /* __CS4271_H */

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/*
* linux/sound/cs42l52.h -- Platform data for CS42L52
*
* Copyright (c) 2012 Cirrus Logic Inc.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __CS42L52_H
#define __CS42L52_H
struct cs42l52_platform_data {
/* MICBIAS Level. Check datasheet Pg48 */
unsigned int micbias_lvl;
/* MICA mode selection 0=Single 1=Differential */
unsigned int mica_cfg;
/* MICB mode selection 0=Single 1=Differential */
unsigned int micb_cfg;
/* MICA Select 0=MIC1A 1=MIC2A */
unsigned int mica_sel;
/* MICB Select 0=MIC2A 1=MIC2B */
unsigned int micb_sel;
/* Charge Pump Freq. Check datasheet Pg73 */
unsigned int chgfreq;
};
#endif /* __CS42L52_H */

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#ifndef __SOUND_CS8403_H
#define __SOUND_CS8403_H
/*
* Routines for Cirrus Logic CS8403/CS8404A IEC958 (S/PDIF) Transmitter
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Takashi Iwai <tiwai@suse.de>
*
*
* 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
*
*/
#ifdef SND_CS8403
#ifndef SND_CS8403_DECL
#define SND_CS8403_DECL static
#endif
#ifndef SND_CS8403_DECODE
#define SND_CS8403_DECODE snd_cs8403_decode_spdif_bits
#endif
#ifndef SND_CS8403_ENCODE
#define SND_CS8403_ENCODE snd_cs8403_encode_spdif_bits
#endif
SND_CS8403_DECL void SND_CS8403_DECODE(struct snd_aes_iec958 *diga, unsigned char bits)
{
if (bits & 0x01) { /* consumer */
if (!(bits & 0x02))
diga->status[0] |= IEC958_AES0_NONAUDIO;
if (!(bits & 0x08))
diga->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
switch (bits & 0x10) {
case 0x10: diga->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; break;
case 0x00: diga->status[0] |= IEC958_AES0_CON_EMPHASIS_5015; break;
}
if (!(bits & 0x80))
diga->status[1] |= IEC958_AES1_CON_ORIGINAL;
switch (bits & 0x60) {
case 0x00: diga->status[1] |= IEC958_AES1_CON_MAGNETIC_ID; break;
case 0x20: diga->status[1] |= IEC958_AES1_CON_DIGDIGCONV_ID; break;
case 0x40: diga->status[1] |= IEC958_AES1_CON_LASEROPT_ID; break;
case 0x60: diga->status[1] |= IEC958_AES1_CON_GENERAL; break;
}
switch (bits & 0x06) {
case 0x00: diga->status[3] |= IEC958_AES3_CON_FS_44100; break;
case 0x02: diga->status[3] |= IEC958_AES3_CON_FS_48000; break;
case 0x04: diga->status[3] |= IEC958_AES3_CON_FS_32000; break;
}
} else {
diga->status[0] = IEC958_AES0_PROFESSIONAL;
switch (bits & 0x18) {
case 0x00: diga->status[0] |= IEC958_AES0_PRO_FS_32000; break;
case 0x10: diga->status[0] |= IEC958_AES0_PRO_FS_44100; break;
case 0x08: diga->status[0] |= IEC958_AES0_PRO_FS_48000; break;
case 0x18: diga->status[0] |= IEC958_AES0_PRO_FS_NOTID; break;
}
switch (bits & 0x60) {
case 0x20: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NONE; break;
case 0x40: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_5015; break;
case 0x00: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_CCITT; break;
case 0x60: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NOTID; break;
}
if (bits & 0x80)
diga->status[1] |= IEC958_AES1_PRO_MODE_STEREOPHONIC;
}
}
SND_CS8403_DECL unsigned char SND_CS8403_ENCODE(struct snd_aes_iec958 *diga)
{
unsigned char bits;
if (!(diga->status[0] & IEC958_AES0_PROFESSIONAL)) {
bits = 0x01; /* consumer mode */
if (diga->status[0] & IEC958_AES0_NONAUDIO)
bits &= ~0x02;
else
bits |= 0x02;
if (diga->status[0] & IEC958_AES0_CON_NOT_COPYRIGHT)
bits &= ~0x08;
else
bits |= 0x08;
switch (diga->status[0] & IEC958_AES0_CON_EMPHASIS) {
default:
case IEC958_AES0_CON_EMPHASIS_NONE: bits |= 0x10; break;
case IEC958_AES0_CON_EMPHASIS_5015: bits |= 0x00; break;
}
if (diga->status[1] & IEC958_AES1_CON_ORIGINAL)
bits &= ~0x80;
else
bits |= 0x80;
if ((diga->status[1] & IEC958_AES1_CON_CATEGORY) == IEC958_AES1_CON_GENERAL)
bits |= 0x60;
else {
switch(diga->status[1] & IEC958_AES1_CON_MAGNETIC_MASK) {
case IEC958_AES1_CON_MAGNETIC_ID:
bits |= 0x00; break;
case IEC958_AES1_CON_DIGDIGCONV_ID:
bits |= 0x20; break;
default:
case IEC958_AES1_CON_LASEROPT_ID:
bits |= 0x40; break;
}
}
switch (diga->status[3] & IEC958_AES3_CON_FS) {
default:
case IEC958_AES3_CON_FS_44100: bits |= 0x00; break;
case IEC958_AES3_CON_FS_48000: bits |= 0x02; break;
case IEC958_AES3_CON_FS_32000: bits |= 0x04; break;
}
} else {
bits = 0x00; /* professional mode */
if (diga->status[0] & IEC958_AES0_NONAUDIO)
bits &= ~0x02;
else
bits |= 0x02;
/* CHECKME: I'm not sure about the bit order in val here */
switch (diga->status[0] & IEC958_AES0_PRO_FS) {
case IEC958_AES0_PRO_FS_32000: bits |= 0x00; break;
case IEC958_AES0_PRO_FS_44100: bits |= 0x10; break; /* 44.1kHz */
case IEC958_AES0_PRO_FS_48000: bits |= 0x08; break; /* 48kHz */
default:
case IEC958_AES0_PRO_FS_NOTID: bits |= 0x18; break;
}
switch (diga->status[0] & IEC958_AES0_PRO_EMPHASIS) {
case IEC958_AES0_PRO_EMPHASIS_NONE: bits |= 0x20; break;
case IEC958_AES0_PRO_EMPHASIS_5015: bits |= 0x40; break;
case IEC958_AES0_PRO_EMPHASIS_CCITT: bits |= 0x00; break;
default:
case IEC958_AES0_PRO_EMPHASIS_NOTID: bits |= 0x60; break;
}
switch (diga->status[1] & IEC958_AES1_PRO_MODE) {
case IEC958_AES1_PRO_MODE_TWO:
case IEC958_AES1_PRO_MODE_STEREOPHONIC: bits |= 0x00; break;
default: bits |= 0x80; break;
}
}
return bits;
}
#endif /* SND_CS8403 */
#ifdef SND_CS8404
#ifndef SND_CS8404_DECL
#define SND_CS8404_DECL static
#endif
#ifndef SND_CS8404_DECODE
#define SND_CS8404_DECODE snd_cs8404_decode_spdif_bits
#endif
#ifndef SND_CS8404_ENCODE
#define SND_CS8404_ENCODE snd_cs8404_encode_spdif_bits
#endif
SND_CS8404_DECL void SND_CS8404_DECODE(struct snd_aes_iec958 *diga, unsigned char bits)
{
if (bits & 0x10) { /* consumer */
if (!(bits & 0x20))
diga->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
if (!(bits & 0x40))
diga->status[0] |= IEC958_AES0_CON_EMPHASIS_5015;
if (!(bits & 0x80))
diga->status[1] |= IEC958_AES1_CON_ORIGINAL;
switch (bits & 0x03) {
case 0x00: diga->status[1] |= IEC958_AES1_CON_DAT; break;
case 0x03: diga->status[1] |= IEC958_AES1_CON_GENERAL; break;
}
switch (bits & 0x06) {
case 0x02: diga->status[3] |= IEC958_AES3_CON_FS_32000; break;
case 0x04: diga->status[3] |= IEC958_AES3_CON_FS_48000; break;
case 0x06: diga->status[3] |= IEC958_AES3_CON_FS_44100; break;
}
} else {
diga->status[0] = IEC958_AES0_PROFESSIONAL;
if (!(bits & 0x04))
diga->status[0] |= IEC958_AES0_NONAUDIO;
switch (bits & 0x60) {
case 0x00: diga->status[0] |= IEC958_AES0_PRO_FS_32000; break;
case 0x40: diga->status[0] |= IEC958_AES0_PRO_FS_44100; break;
case 0x20: diga->status[0] |= IEC958_AES0_PRO_FS_48000; break;
case 0x60: diga->status[0] |= IEC958_AES0_PRO_FS_NOTID; break;
}
switch (bits & 0x03) {
case 0x02: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NONE; break;
case 0x01: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_5015; break;
case 0x00: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_CCITT; break;
case 0x03: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NOTID; break;
}
if (!(bits & 0x80))
diga->status[1] |= IEC958_AES1_PRO_MODE_STEREOPHONIC;
}
}
SND_CS8404_DECL unsigned char SND_CS8404_ENCODE(struct snd_aes_iec958 *diga)
{
unsigned char bits;
if (!(diga->status[0] & IEC958_AES0_PROFESSIONAL)) {
bits = 0x10; /* consumer mode */
if (!(diga->status[0] & IEC958_AES0_CON_NOT_COPYRIGHT))
bits |= 0x20;
if ((diga->status[0] & IEC958_AES0_CON_EMPHASIS) == IEC958_AES0_CON_EMPHASIS_NONE)
bits |= 0x40;
if (!(diga->status[1] & IEC958_AES1_CON_ORIGINAL))
bits |= 0x80;
if ((diga->status[1] & IEC958_AES1_CON_CATEGORY) == IEC958_AES1_CON_GENERAL)
bits |= 0x03;
switch (diga->status[3] & IEC958_AES3_CON_FS) {
default:
case IEC958_AES3_CON_FS_44100: bits |= 0x06; break;
case IEC958_AES3_CON_FS_48000: bits |= 0x04; break;
case IEC958_AES3_CON_FS_32000: bits |= 0x02; break;
}
} else {
bits = 0x00; /* professional mode */
if (!(diga->status[0] & IEC958_AES0_NONAUDIO))
bits |= 0x04;
switch (diga->status[0] & IEC958_AES0_PRO_FS) {
case IEC958_AES0_PRO_FS_32000: bits |= 0x00; break;
case IEC958_AES0_PRO_FS_44100: bits |= 0x40; break; /* 44.1kHz */
case IEC958_AES0_PRO_FS_48000: bits |= 0x20; break; /* 48kHz */
default:
case IEC958_AES0_PRO_FS_NOTID: bits |= 0x00; break;
}
switch (diga->status[0] & IEC958_AES0_PRO_EMPHASIS) {
case IEC958_AES0_PRO_EMPHASIS_NONE: bits |= 0x02; break;
case IEC958_AES0_PRO_EMPHASIS_5015: bits |= 0x01; break;
case IEC958_AES0_PRO_EMPHASIS_CCITT: bits |= 0x00; break;
default:
case IEC958_AES0_PRO_EMPHASIS_NOTID: bits |= 0x03; break;
}
switch (diga->status[1] & IEC958_AES1_PRO_MODE) {
case IEC958_AES1_PRO_MODE_TWO:
case IEC958_AES1_PRO_MODE_STEREOPHONIC: bits |= 0x00; break;
default: bits |= 0x80; break;
}
}
return bits;
}
#endif /* SND_CS8404 */
#endif /* __SOUND_CS8403_H */

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#ifndef __SOUND_CS8427_H
#define __SOUND_CS8427_H
/*
* Routines for Cirrus Logic CS8427
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* 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/i2c.h>
#define CS8427_BASE_ADDR 0x10 /* base I2C address */
#define CS8427_REG_AUTOINC 0x80 /* flag - autoincrement */
#define CS8427_REG_CONTROL1 0x01
#define CS8427_REG_CONTROL2 0x02
#define CS8427_REG_DATAFLOW 0x03
#define CS8427_REG_CLOCKSOURCE 0x04
#define CS8427_REG_SERIALINPUT 0x05
#define CS8427_REG_SERIALOUTPUT 0x06
#define CS8427_REG_INT1STATUS 0x07
#define CS8427_REG_INT2STATUS 0x08
#define CS8427_REG_INT1MASK 0x09
#define CS8427_REG_INT1MODEMSB 0x0a
#define CS8427_REG_INT1MODELSB 0x0b
#define CS8427_REG_INT2MASK 0x0c
#define CS8427_REG_INT2MODEMSB 0x0d
#define CS8427_REG_INT2MODELSB 0x0e
#define CS8427_REG_RECVCSDATA 0x0f
#define CS8427_REG_RECVERRORS 0x10
#define CS8427_REG_RECVERRMASK 0x11
#define CS8427_REG_CSDATABUF 0x12
#define CS8427_REG_UDATABUF 0x13
#define CS8427_REG_QSUBCODE 0x14 /* 0x14-0x1d (10 bytes) */
#define CS8427_REG_OMCKRMCKRATIO 0x1e
#define CS8427_REG_CORU_DATABUF 0x20 /* 24 byte buffer area */
#define CS8427_REG_ID_AND_VER 0x7f
/* CS8427_REG_CONTROL1 bits */
#define CS8427_SWCLK (1<<7) /* 0 = RMCK default, 1 = OMCK output on RMCK pin */
#define CS8427_VSET (1<<6) /* 0 = valid PCM data, 1 = invalid PCM data */
#define CS8427_MUTESAO (1<<5) /* mute control for the serial audio output port, 0 = disabled, 1 = enabled */
#define CS8427_MUTEAES (1<<4) /* mute control for the AES transmitter output, 0 = disabled, 1 = enabled */
#define CS8427_INTMASK (3<<1) /* interrupt output pin setup mask */
#define CS8427_INTACTHIGH (0<<1) /* active high */
#define CS8427_INTACTLOW (1<<1) /* active low */
#define CS8427_INTOPENDRAIN (2<<1) /* open drain, active low */
#define CS8427_TCBLDIR (1<<0) /* 0 = TCBL is an input, 1 = TCBL is an output */
/* CS8427_REQ_CONTROL2 bits */
#define CS8427_HOLDMASK (3<<5) /* action when a receiver error occurs */
#define CS8427_HOLDLASTSAMPLE (0<<5) /* hold the last valid sample */
#define CS8427_HOLDZERO (1<<5) /* replace the current audio sample with zero (mute) */
#define CS8427_HOLDNOCHANGE (2<<5) /* do not change the received audio sample */
#define CS8427_RMCKF (1<<4) /* 0 = 256*Fsi, 1 = 128*Fsi */
#define CS8427_MMR (1<<3) /* AES3 receiver operation, 0 = stereo, 1 = mono */
#define CS8427_MMT (1<<2) /* AES3 transmitter operation, 0 = stereo, 1 = mono */
#define CS8427_MMTCS (1<<1) /* 0 = use A + B CS data, 1 = use MMTLR CS data */
#define CS8427_MMTLR (1<<0) /* 0 = use A CS data, 1 = use B CS data */
/* CS8427_REG_DATAFLOW */
#define CS8427_TXOFF (1<<6) /* AES3 transmitter Output, 0 = normal operation, 1 = off (0V) */
#define CS8427_AESBP (1<<5) /* AES3 hardware bypass mode, 0 = normal, 1 = bypass (RX->TX) */
#define CS8427_TXDMASK (3<<3) /* AES3 Transmitter Data Source Mask */
#define CS8427_TXDSERIAL (1<<3) /* TXD - serial audio input port */
#define CS8427_TXAES3DRECEIVER (2<<3) /* TXD - AES3 receiver */
#define CS8427_SPDMASK (3<<1) /* Serial Audio Output Port Data Source Mask */
#define CS8427_SPDSERIAL (1<<1) /* SPD - serial audio input port */
#define CS8427_SPDAES3RECEIVER (2<<1) /* SPD - AES3 receiver */
/* CS8427_REG_CLOCKSOURCE */
#define CS8427_RUN (1<<6) /* 0 = clock off, 1 = clock on */
#define CS8427_CLKMASK (3<<4) /* OMCK frequency mask */
#define CS8427_CLK256 (0<<4) /* 256*Fso */
#define CS8427_CLK384 (1<<4) /* 384*Fso */
#define CS8427_CLK512 (2<<4) /* 512*Fso */
#define CS8427_OUTC (1<<3) /* Output Time Base, 0 = OMCK, 1 = recovered input clock */
#define CS8427_INC (1<<2) /* Input Time Base Clock Source, 0 = recoverd input clock, 1 = OMCK input pin */
#define CS8427_RXDMASK (3<<0) /* Recovered Input Clock Source Mask */
#define CS8427_RXDILRCK (0<<0) /* 256*Fsi from ILRCK pin */
#define CS8427_RXDAES3INPUT (1<<0) /* 256*Fsi from AES3 input */
#define CS8427_EXTCLOCKRESET (2<<0) /* bypass PLL, 256*Fsi clock, synchronous reset */
#define CS8427_EXTCLOCK (3<<0) /* bypass PLL, 256*Fsi clock */
/* CS8427_REG_SERIALINPUT */
#define CS8427_SIMS (1<<7) /* 0 = slave, 1 = master mode */
#define CS8427_SISF (1<<6) /* ISCLK freq, 0 = 64*Fsi, 1 = 128*Fsi */
#define CS8427_SIRESMASK (3<<4) /* Resolution of the input data for right justified formats */
#define CS8427_SIRES24 (0<<4) /* SIRES 24-bit */
#define CS8427_SIRES20 (1<<4) /* SIRES 20-bit */
#define CS8427_SIRES16 (2<<4) /* SIRES 16-bit */
#define CS8427_SIJUST (1<<3) /* Justification of SDIN data relative to ILRCK, 0 = left-justified, 1 = right-justified */
#define CS8427_SIDEL (1<<2) /* Delay of SDIN data relative to ILRCK for left-justified data formats, 0 = first ISCLK period, 1 = second ISCLK period */
#define CS8427_SISPOL (1<<1) /* ICLK clock polarity, 0 = rising edge of ISCLK, 1 = falling edge of ISCLK */
#define CS8427_SILRPOL (1<<0) /* ILRCK clock polarity, 0 = SDIN data left channel when ILRCK is high, 1 = SDIN right when ILRCK is high */
/* CS8427_REG_SERIALOUTPUT */
#define CS8427_SOMS (1<<7) /* 0 = slave, 1 = master mode */
#define CS8427_SOSF (1<<6) /* OSCLK freq, 0 = 64*Fso, 1 = 128*Fso */
#define CS8427_SORESMASK (3<<4) /* Resolution of the output data on SDOUT and AES3 output */
#define CS8427_SORES24 (0<<4) /* SIRES 24-bit */
#define CS8427_SORES20 (1<<4) /* SIRES 20-bit */
#define CS8427_SORES16 (2<<4) /* SIRES 16-bit */
#define CS8427_SORESDIRECT (2<<4) /* SIRES direct copy from AES3 receiver */
#define CS8427_SOJUST (1<<3) /* Justification of SDOUT data relative to OLRCK, 0 = left-justified, 1 = right-justified */
#define CS8427_SODEL (1<<2) /* Delay of SDOUT data relative to OLRCK for left-justified data formats, 0 = first OSCLK period, 1 = second OSCLK period */
#define CS8427_SOSPOL (1<<1) /* OSCLK clock polarity, 0 = rising edge of ISCLK, 1 = falling edge of ISCLK */
#define CS8427_SOLRPOL (1<<0) /* OLRCK clock polarity, 0 = SDOUT data left channel when OLRCK is high, 1 = SDOUT right when OLRCK is high */
/* CS8427_REG_INT1STATUS */
#define CS8427_TSLIP (1<<7) /* AES3 transmitter source data slip interrupt */
#define CS8427_OSLIP (1<<6) /* Serial audio output port data slip interrupt */
#define CS8427_DETC (1<<2) /* D to E C-buffer transfer interrupt */
#define CS8427_EFTC (1<<1) /* E to F C-buffer transfer interrupt */
#define CS8427_RERR (1<<0) /* A receiver error has occurred */
/* CS8427_REG_INT2STATUS */
#define CS8427_DETU (1<<3) /* D to E U-buffer transfer interrupt */
#define CS8427_EFTU (1<<2) /* E to F U-buffer transfer interrupt */
#define CS8427_QCH (1<<1) /* A new block of Q-subcode data is available for reading */
/* CS8427_REG_INT1MODEMSB && CS8427_REG_INT1MODELSB */
/* bits are defined in CS8427_REG_INT1STATUS */
/* CS8427_REG_INT2MODEMSB && CS8427_REG_INT2MODELSB */
/* bits are defined in CS8427_REG_INT2STATUS */
#define CS8427_INTMODERISINGMSB 0
#define CS8427_INTMODERESINGLSB 0
#define CS8427_INTMODEFALLINGMSB 0
#define CS8427_INTMODEFALLINGLSB 1
#define CS8427_INTMODELEVELMSB 1
#define CS8427_INTMODELEVELLSB 0
/* CS8427_REG_RECVCSDATA */
#define CS8427_AUXMASK (15<<4) /* auxiliary data field width */
#define CS8427_AUXSHIFT 4
#define CS8427_PRO (1<<3) /* Channel status block format indicator */
#define CS8427_AUDIO (1<<2) /* Audio indicator (0 = audio, 1 = nonaudio */
#define CS8427_COPY (1<<1) /* 0 = copyright asserted, 1 = copyright not asserted */
#define CS8427_ORIG (1<<0) /* SCMS generation indicator, 0 = 1st generation or highter, 1 = original */
/* CS8427_REG_RECVERRORS */
/* CS8427_REG_RECVERRMASK for CS8427_RERR */
#define CS8427_QCRC (1<<6) /* Q-subcode data CRC error indicator */
#define CS8427_CCRC (1<<5) /* Chancnel Status Block Cyclick Redundancy Check Bit */
#define CS8427_UNLOCK (1<<4) /* PLL lock status bit */
#define CS8427_V (1<<3) /* 0 = valid data */
#define CS8427_CONF (1<<2) /* Confidence bit */
#define CS8427_BIP (1<<1) /* Bi-phase error bit */
#define CS8427_PAR (1<<0) /* Parity error */
/* CS8427_REG_CSDATABUF */
#define CS8427_BSEL (1<<5) /* 0 = CS data, 1 = U data */
#define CS8427_CBMR (1<<4) /* 0 = overwrite first 5 bytes for CS D to E buffer, 1 = prevent */
#define CS8427_DETCI (1<<3) /* D to E CS data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
#define CS8427_EFTCI (1<<2) /* E to F CS data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
#define CS8427_CAM (1<<1) /* CS data buffer control port access mode bit, 0 = one byte, 1 = two byte */
#define CS8427_CHS (1<<0) /* Channel select bit, 0 = Channel A, 1 = Channel B */
/* CS8427_REG_UDATABUF */
#define CS8427_UD (1<<4) /* User data pin (U) direction, 0 = input, 1 = output */
#define CS8427_UBMMASK (3<<2) /* Operating mode of the AES3 U bit manager */
#define CS8427_UBMZEROS (0<<2) /* transmit all zeros mode */
#define CS8427_UBMBLOCK (1<<2) /* block mode */
#define CS8427_DETUI (1<<1) /* D to E U-data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
#define CS8427_EFTUI (1<<1) /* E to F U-data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
/* CS8427_REG_ID_AND_VER */
#define CS8427_IDMASK (15<<4)
#define CS8427_IDSHIFT 4
#define CS8427_VERMASK (15<<0)
#define CS8427_VERSHIFT 0
#define CS8427_VER8427A 0x71
struct snd_pcm_substream;
int snd_cs8427_create(struct snd_i2c_bus *bus, unsigned char addr,
unsigned int reset_timeout, struct snd_i2c_device **r_cs8427);
int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg,
unsigned char val);
int snd_cs8427_iec958_build(struct snd_i2c_device *cs8427,
struct snd_pcm_substream *playback_substream,
struct snd_pcm_substream *capture_substream);
int snd_cs8427_iec958_active(struct snd_i2c_device *cs8427, int active);
int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate);
#endif /* __SOUND_CS8427_H */

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/*
* da7213.h - DA7213 ASoC Codec Driver Platform Data
*
* Copyright (c) 2013 Dialog Semiconductor
*
* Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef _DA7213_PDATA_H
#define _DA7213_PDATA_H
enum da7213_micbias_voltage {
DA7213_MICBIAS_1_6V = 0,
DA7213_MICBIAS_2_2V = 1,
DA7213_MICBIAS_2_5V = 2,
DA7213_MICBIAS_3_0V = 3,
};
enum da7213_dmic_data_sel {
DA7213_DMIC_DATA_LRISE_RFALL = 0,
DA7213_DMIC_DATA_LFALL_RRISE = 1,
};
enum da7213_dmic_samplephase {
DA7213_DMIC_SAMPLE_ON_CLKEDGE = 0,
DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE = 1,
};
enum da7213_dmic_clk_rate {
DA7213_DMIC_CLK_3_0MHZ = 0,
DA7213_DMIC_CLK_1_5MHZ = 1,
};
struct da7213_platform_data {
/* Mic Bias voltage */
enum da7213_micbias_voltage micbias1_lvl;
enum da7213_micbias_voltage micbias2_lvl;
/* DMIC config */
enum da7213_dmic_data_sel dmic_data_sel;
enum da7213_dmic_samplephase dmic_samplephase;
enum da7213_dmic_clk_rate dmic_clk_rate;
/* MCLK squaring config */
bool mclk_squaring;
};
#endif /* _DA7213_PDATA_H */

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/*
* DA9055 ALSA Soc codec driver
*
* Copyright (c) 2012 Dialog Semiconductor
*
* Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C
* Written by David Chen <david.chen@diasemi.com> and
* Ashish Chavan <ashish.chavan@kpitcummins.com>
*
* 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.
*/
#ifndef __SOUND_DA9055_H__
#define __SOUND_DA9055_H__
enum da9055_micbias_voltage {
DA9055_MICBIAS_1_6V = 0,
DA9055_MICBIAS_1_8V = 1,
DA9055_MICBIAS_2_1V = 2,
DA9055_MICBIAS_2_2V = 3,
};
struct da9055_platform_data {
/* Selects which of the two MicBias pins acts as the bias source */
bool micbias_source;
/* Selects the micbias voltage */
enum da9055_micbias_voltage micbias;
};
#endif

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/*
* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com)
*
* 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
*
*/
#ifndef __SOUND_DESIGNWARE_I2S_H
#define __SOUND_DESIGNWARE_I2S_H
#include <linux/dmaengine.h>
#include <linux/types.h>
/*
* struct i2s_clk_config_data - represent i2s clk configuration data
* @chan_nr: number of channel
* @data_width: number of bits per sample (8/16/24/32 bit)
* @sample_rate: sampling frequency (8Khz, 16Khz, 32Khz, 44Khz, 48Khz)
*/
struct i2s_clk_config_data {
int chan_nr;
u32 data_width;
u32 sample_rate;
};
struct i2s_platform_data {
#define DWC_I2S_PLAY (1 << 0)
#define DWC_I2S_RECORD (1 << 1)
unsigned int cap;
int channel;
u32 snd_fmts;
u32 snd_rates;
void *play_dma_data;
void *capture_dma_data;
bool (*filter)(struct dma_chan *chan, void *slave);
int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
};
struct i2s_dma_data {
void *data;
dma_addr_t addr;
u32 max_burst;
enum dma_slave_buswidth addr_width;
bool (*filter)(struct dma_chan *chan, void *slave);
};
/* I2S DMA registers */
#define I2S_RXDMA 0x01C0
#define I2S_TXDMA 0x01C8
#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */
#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */
#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */
#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */
#endif /* __SOUND_DESIGNWARE_I2S_H */

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/*
* Copyright (C) 2012, Analog Devices Inc.
* Author: Lars-Peter Clausen <lars@metafoo.de>
*
* 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.
*
* 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.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#ifndef __SOUND_DMAENGINE_PCM_H__
#define __SOUND_DMAENGINE_PCM_H__
#include <sound/pcm.h>
#include <sound/soc.h>
#include <linux/dmaengine.h>
/**
* snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM
* substream
* @substream: PCM substream
*/
static inline enum dma_transfer_direction
snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
{
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return DMA_MEM_TO_DEV;
else
return DMA_DEV_TO_MEM;
}
int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream);
snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream);
int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
struct dma_chan *chan);
int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
dma_filter_fn filter_fn, void *filter_data);
int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream);
struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn,
void *filter_data);
struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
/**
* struct snd_dmaengine_dai_dma_data - DAI DMA configuration data
* @addr: Address of the DAI data source or destination register.
* @addr_width: Width of the DAI data source or destination register.
* @maxburst: Maximum number of words(note: words, as in units of the
* src_addr_width member, not bytes) that can be send to or received from the
* DAI in one burst.
* @slave_id: Slave requester id for the DMA channel.
* @filter_data: Custom DMA channel filter data, this will usually be used when
* requesting the DMA channel.
*/
struct snd_dmaengine_dai_dma_data {
dma_addr_t addr;
enum dma_slave_buswidth addr_width;
u32 maxburst;
unsigned int slave_id;
void *filter_data;
};
void snd_dmaengine_pcm_set_config_from_dai_data(
const struct snd_pcm_substream *substream,
const struct snd_dmaengine_dai_dma_data *dma_data,
struct dma_slave_config *config);
/*
* Try to request the DMA channel using compat_request_channel or
* compat_filter_fn if it couldn't be requested through devicetree.
*/
#define SND_DMAENGINE_PCM_FLAG_COMPAT BIT(0)
/*
* Don't try to request the DMA channels through devicetree. This flag only
* makes sense if SND_DMAENGINE_PCM_FLAG_COMPAT is set as well.
*/
#define SND_DMAENGINE_PCM_FLAG_NO_DT BIT(1)
/*
* The platforms dmaengine driver does not support reporting the amount of
* bytes that are still left to transfer.
*/
#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2)
/*
* The PCM is half duplex and the DMA channel is shared between capture and
* playback.
*/
#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
/**
* struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM
* @prepare_slave_config: Callback used to fill in the DMA slave_config for a
* PCM substream. Will be called from the PCM drivers hwparams callback.
* @compat_request_channel: Callback to request a DMA channel for platforms
* which do not use devicetree.
* @compat_filter_fn: Will be used as the filter function when requesting a
* channel for platforms which do not use devicetree. The filter parameter
* will be the DAI's DMA data.
* @pcm_hardware: snd_pcm_hardware struct to be used for the PCM.
* @prealloc_buffer_size: Size of the preallocated audio buffer.
*
* Note: If both compat_request_channel and compat_filter_fn are set
* compat_request_channel will be used to request the channel and
* compat_filter_fn will be ignored. Otherwise the channel will be requested
* using dma_request_channel with compat_filter_fn as the filter function.
*/
struct snd_dmaengine_pcm_config {
int (*prepare_slave_config)(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct dma_slave_config *slave_config);
struct dma_chan *(*compat_request_channel)(
struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_substream *substream);
dma_filter_fn compat_filter_fn;
const struct snd_pcm_hardware *pcm_hardware;
unsigned int prealloc_buffer_size;
};
int snd_dmaengine_pcm_register(struct device *dev,
const struct snd_dmaengine_pcm_config *config,
unsigned int flags);
void snd_dmaengine_pcm_unregister(struct device *dev);
int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct dma_slave_config *slave_config);
#endif

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#ifndef __EMU10K1_SYNTH_H
#define __EMU10K1_SYNTH_H
/*
* Defines for the Emu10k1 WaveTable synth
*
* Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
*
* 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/emu10k1.h>
#include <sound/emux_synth.h>
/* sequencer device id */
#define SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH "emu10k1-synth"
/* argument for snd_seq_device_new */
struct snd_emu10k1_synth_arg {
struct snd_emu10k1 *hwptr; /* chip */
int index; /* sequencer client index */
int seq_ports; /* number of sequencer ports to be created */
int max_voices; /* maximum number of voices for wavetable */
};
#define EMU10K1_MAX_MEMSIZE (32 * 1024 * 1024) /* 32MB */
#endif

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#ifndef __SOUND_EMU8000_H
#define __SOUND_EMU8000_H
/*
* Defines for the emu8000 (AWE32/64)
*
* Copyright (C) 1999 Steve Ratcliffe
* Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
*
* 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/emux_synth.h>
#include <sound/seq_kernel.h>
/*
* Hardware parameters.
*/
#define EMU8000_MAX_DRAM (28 * 1024 * 1024) /* Max on-board mem is 28Mb ???*/
#define EMU8000_DRAM_OFFSET 0x200000 /* Beginning of on board ram */
#define EMU8000_CHANNELS 32 /* Number of hardware channels */
#define EMU8000_DRAM_VOICES 30 /* number of normal voices */
/* Flags to set a dma channel to read or write */
#define EMU8000_RAM_READ 0
#define EMU8000_RAM_WRITE 1
#define EMU8000_RAM_CLOSE 2
#define EMU8000_RAM_MODE_MASK 0x03
#define EMU8000_RAM_RIGHT 0x10 /* use 'right' DMA channel */
enum {
EMU8000_CONTROL_BASS = 0,
EMU8000_CONTROL_TREBLE,
EMU8000_CONTROL_CHORUS_MODE,
EMU8000_CONTROL_REVERB_MODE,
EMU8000_CONTROL_FM_CHORUS_DEPTH,
EMU8000_CONTROL_FM_REVERB_DEPTH,
EMU8000_NUM_CONTROLS,
};
/*
* Structure to hold all state information for the emu8000 driver.
*
* Note 1: The chip supports 32 channels in hardware this is max_channels
* some of the channels may be used for other things so max_channels is
* the number in use for wave voices.
*/
struct snd_emu8000 {
struct snd_emux *emu;
int index; /* sequencer client index */
int seq_ports; /* number of sequencer ports */
int fm_chorus_depth; /* FM OPL3 chorus depth */
int fm_reverb_depth; /* FM OPL3 reverb depth */
int mem_size; /* memory size */
unsigned long port1; /* Port usually base+0 */
unsigned long port2; /* Port usually at base+0x400 */
unsigned long port3; /* Port usually at base+0x800 */
struct resource *res_port1;
struct resource *res_port2;
struct resource *res_port3;
unsigned short last_reg;/* Last register command */
spinlock_t reg_lock;
int dram_checked;
struct snd_card *card; /* The card that this belongs to */
int chorus_mode;
int reverb_mode;
int bass_level;
int treble_level;
struct snd_util_memhdr *memhdr;
spinlock_t control_lock;
struct snd_kcontrol *controls[EMU8000_NUM_CONTROLS];
struct snd_pcm *pcm; /* pcm on emu8000 wavetable */
};
/* sequencer device id */
#define SNDRV_SEQ_DEV_ID_EMU8000 "emu8000-synth"
/* exported functions */
int snd_emu8000_new(struct snd_card *card, int device, long port, int seq_ports,
struct snd_seq_device **ret);
void snd_emu8000_poke(struct snd_emu8000 *emu, unsigned int port, unsigned int reg,
unsigned int val);
unsigned short snd_emu8000_peek(struct snd_emu8000 *emu, unsigned int port,
unsigned int reg);
void snd_emu8000_poke_dw(struct snd_emu8000 *emu, unsigned int port, unsigned int reg,
unsigned int val);
unsigned int snd_emu8000_peek_dw(struct snd_emu8000 *emu, unsigned int port,
unsigned int reg);
void snd_emu8000_dma_chan(struct snd_emu8000 *emu, int ch, int mode);
void snd_emu8000_init_fm(struct snd_emu8000 *emu);
void snd_emu8000_update_chorus_mode(struct snd_emu8000 *emu);
void snd_emu8000_update_reverb_mode(struct snd_emu8000 *emu);
void snd_emu8000_update_equalizer(struct snd_emu8000 *emu);
int snd_emu8000_load_chorus_fx(struct snd_emu8000 *emu, int mode, const void __user *buf, long len);
int snd_emu8000_load_reverb_fx(struct snd_emu8000 *emu, int mode, const void __user *buf, long len);
#endif /* __SOUND_EMU8000_H */

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#ifndef __SOUND_EMU8000_REG_H
#define __SOUND_EMU8000_REG_H
/*
* Register operations for the EMU8000
*
* Copyright (C) 1999 Steve Ratcliffe
*
* Based on awe_wave.c by Takashi Iwai
*
* 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
*
*/
/*
* Data port addresses relative to the EMU base.
*/
#define EMU8000_DATA0(e) ((e)->port1)
#define EMU8000_DATA1(e) ((e)->port2)
#define EMU8000_DATA2(e) ((e)->port2+2)
#define EMU8000_DATA3(e) ((e)->port3)
#define EMU8000_PTR(e) ((e)->port3+2)
/*
* Make a command from a register and channel.
*/
#define EMU8000_CMD(reg, chan) ((reg)<<5 | (chan))
/*
* Commands to read and write the EMU8000 registers.
* These macros should be used for all register accesses.
*/
#define EMU8000_CPF_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(0, (chan)))
#define EMU8000_PTRX_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(1, (chan)))
#define EMU8000_CVCF_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(2, (chan)))
#define EMU8000_VTFT_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(3, (chan)))
#define EMU8000_PSST_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(6, (chan)))
#define EMU8000_CSL_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(7, (chan)))
#define EMU8000_CCCA_READ(emu, chan) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(0, (chan)))
#define EMU8000_HWCF4_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 9))
#define EMU8000_HWCF5_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 10))
#define EMU8000_HWCF6_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 13))
#define EMU8000_SMALR_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 20))
#define EMU8000_SMARR_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 21))
#define EMU8000_SMALW_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 22))
#define EMU8000_SMARW_READ(emu) \
snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 23))
#define EMU8000_SMLD_READ(emu) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 26))
#define EMU8000_SMRD_READ(emu) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 26))
#define EMU8000_WC_READ(emu) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 27))
#define EMU8000_HWCF1_READ(emu) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 29))
#define EMU8000_HWCF2_READ(emu) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 30))
#define EMU8000_HWCF3_READ(emu) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 31))
#define EMU8000_INIT1_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(2, (chan)))
#define EMU8000_INIT2_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(2, (chan)))
#define EMU8000_INIT3_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(3, (chan)))
#define EMU8000_INIT4_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(3, (chan)))
#define EMU8000_ENVVOL_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(4, (chan)))
#define EMU8000_DCYSUSV_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(5, (chan)))
#define EMU8000_ENVVAL_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(6, (chan)))
#define EMU8000_DCYSUS_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(7, (chan)))
#define EMU8000_ATKHLDV_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(4, (chan)))
#define EMU8000_LFO1VAL_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(5, (chan)))
#define EMU8000_ATKHLD_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(6, (chan)))
#define EMU8000_LFO2VAL_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(7, (chan)))
#define EMU8000_IP_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(0, (chan)))
#define EMU8000_IFATN_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(1, (chan)))
#define EMU8000_PEFE_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(2, (chan)))
#define EMU8000_FMMOD_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(3, (chan)))
#define EMU8000_TREMFRQ_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(4, (chan)))
#define EMU8000_FM2FRQ2_READ(emu, chan) \
snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(5, (chan)))
#define EMU8000_CPF_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(0, (chan)), (val))
#define EMU8000_PTRX_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(1, (chan)), (val))
#define EMU8000_CVCF_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(2, (chan)), (val))
#define EMU8000_VTFT_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(3, (chan)), (val))
#define EMU8000_PSST_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(6, (chan)), (val))
#define EMU8000_CSL_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(7, (chan)), (val))
#define EMU8000_CCCA_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(0, (chan)), (val))
#define EMU8000_HWCF4_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 9), (val))
#define EMU8000_HWCF5_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 10), (val))
#define EMU8000_HWCF6_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 13), (val))
/* this register is not documented */
#define EMU8000_HWCF7_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 14), (val))
#define EMU8000_SMALR_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 20), (val))
#define EMU8000_SMARR_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 21), (val))
#define EMU8000_SMALW_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 22), (val))
#define EMU8000_SMARW_WRITE(emu, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 23), (val))
#define EMU8000_SMLD_WRITE(emu, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 26), (val))
#define EMU8000_SMRD_WRITE(emu, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 26), (val))
#define EMU8000_WC_WRITE(emu, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 27), (val))
#define EMU8000_HWCF1_WRITE(emu, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 29), (val))
#define EMU8000_HWCF2_WRITE(emu, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 30), (val))
#define EMU8000_HWCF3_WRITE(emu, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 31), (val))
#define EMU8000_INIT1_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(2, (chan)), (val))
#define EMU8000_INIT2_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(2, (chan)), (val))
#define EMU8000_INIT3_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(3, (chan)), (val))
#define EMU8000_INIT4_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(3, (chan)), (val))
#define EMU8000_ENVVOL_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(4, (chan)), (val))
#define EMU8000_DCYSUSV_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(5, (chan)), (val))
#define EMU8000_ENVVAL_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(6, (chan)), (val))
#define EMU8000_DCYSUS_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(7, (chan)), (val))
#define EMU8000_ATKHLDV_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(4, (chan)), (val))
#define EMU8000_LFO1VAL_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(5, (chan)), (val))
#define EMU8000_ATKHLD_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(6, (chan)), (val))
#define EMU8000_LFO2VAL_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(7, (chan)), (val))
#define EMU8000_IP_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(0, (chan)), (val))
#define EMU8000_IFATN_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(1, (chan)), (val))
#define EMU8000_PEFE_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(2, (chan)), (val))
#define EMU8000_FMMOD_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(3, (chan)), (val))
#define EMU8000_TREMFRQ_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(4, (chan)), (val))
#define EMU8000_FM2FRQ2_WRITE(emu, chan, val) \
snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(5, (chan)), (val))
#define EMU8000_0080_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(4, (chan)), (val))
#define EMU8000_00A0_WRITE(emu, chan, val) \
snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(5, (chan)), (val))
#endif /* __SOUND_EMU8000_REG_H */

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#ifndef __SOUND_EMUX_LEGACY_H
#define __SOUND_EMUX_LEGACY_H
/*
* Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
*
* Definitions of OSS compatible headers for Emu8000 device informations
*
* 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/seq_oss_legacy.h>
/*
* awe hardware controls
*/
#define _EMUX_OSS_DEBUG_MODE 0x00
#define _EMUX_OSS_REVERB_MODE 0x01
#define _EMUX_OSS_CHORUS_MODE 0x02
#define _EMUX_OSS_REMOVE_LAST_SAMPLES 0x03
#define _EMUX_OSS_INITIALIZE_CHIP 0x04
#define _EMUX_OSS_SEND_EFFECT 0x05
#define _EMUX_OSS_TERMINATE_CHANNEL 0x06
#define _EMUX_OSS_TERMINATE_ALL 0x07
#define _EMUX_OSS_INITIAL_VOLUME 0x08
#define _EMUX_OSS_INITIAL_ATTEN _EMUX_OSS_INITIAL_VOLUME
#define _EMUX_OSS_RESET_CHANNEL 0x09
#define _EMUX_OSS_CHANNEL_MODE 0x0a
#define _EMUX_OSS_DRUM_CHANNELS 0x0b
#define _EMUX_OSS_MISC_MODE 0x0c
#define _EMUX_OSS_RELEASE_ALL 0x0d
#define _EMUX_OSS_NOTEOFF_ALL 0x0e
#define _EMUX_OSS_CHN_PRESSURE 0x0f
#define _EMUX_OSS_EQUALIZER 0x11
#define _EMUX_OSS_MODE_FLAG 0x80
#define _EMUX_OSS_COOKED_FLAG 0x40 /* not supported */
#define _EMUX_OSS_MODE_VALUE_MASK 0x3F
/*
* mode type definitions
*/
enum {
/* 0*/ EMUX_MD_EXCLUSIVE_OFF, /* obsolete */
/* 1*/ EMUX_MD_EXCLUSIVE_ON, /* obsolete */
/* 2*/ EMUX_MD_VERSION, /* read only */
/* 3*/ EMUX_MD_EXCLUSIVE_SOUND, /* 0/1: exclusive note on (default=1) */
/* 4*/ EMUX_MD_REALTIME_PAN, /* 0/1: do realtime pan change (default=1) */
/* 5*/ EMUX_MD_GUS_BANK, /* bank number for GUS patches (default=0) */
/* 6*/ EMUX_MD_KEEP_EFFECT, /* 0/1: keep effect values, (default=0) */
/* 7*/ EMUX_MD_ZERO_ATTEN, /* attenuation of max volume (default=32) */
/* 8*/ EMUX_MD_CHN_PRIOR, /* 0/1: set MIDI channel priority mode (default=1) */
/* 9*/ EMUX_MD_MOD_SENSE, /* integer: modwheel sensitivity (def=18) */
/*10*/ EMUX_MD_DEF_PRESET, /* integer: default preset number (def=0) */
/*11*/ EMUX_MD_DEF_BANK, /* integer: default bank number (def=0) */
/*12*/ EMUX_MD_DEF_DRUM, /* integer: default drumset number (def=0) */
/*13*/ EMUX_MD_TOGGLE_DRUM_BANK, /* 0/1: toggle drum flag with bank# (def=0) */
/*14*/ EMUX_MD_NEW_VOLUME_CALC, /* 0/1: volume calculation mode (def=1) */
/*15*/ EMUX_MD_CHORUS_MODE, /* integer: chorus mode (def=2) */
/*16*/ EMUX_MD_REVERB_MODE, /* integer: chorus mode (def=4) */
/*17*/ EMUX_MD_BASS_LEVEL, /* integer: bass level (def=5) */
/*18*/ EMUX_MD_TREBLE_LEVEL, /* integer: treble level (def=9) */
/*19*/ EMUX_MD_DEBUG_MODE, /* integer: debug level (def=0) */
/*20*/ EMUX_MD_PAN_EXCHANGE, /* 0/1: exchange panning direction (def=0) */
EMUX_MD_END,
};
/*
* effect parameters
*/
enum {
/* modulation envelope parameters */
/* 0*/ EMUX_FX_ENV1_DELAY, /* WORD: ENVVAL */
/* 1*/ EMUX_FX_ENV1_ATTACK, /* BYTE: up ATKHLD */
/* 2*/ EMUX_FX_ENV1_HOLD, /* BYTE: lw ATKHLD */
/* 3*/ EMUX_FX_ENV1_DECAY, /* BYTE: lw DCYSUS */
/* 4*/ EMUX_FX_ENV1_RELEASE, /* BYTE: lw DCYSUS */
/* 5*/ EMUX_FX_ENV1_SUSTAIN, /* BYTE: up DCYSUS */
/* 6*/ EMUX_FX_ENV1_PITCH, /* BYTE: up PEFE */
/* 7*/ EMUX_FX_ENV1_CUTOFF, /* BYTE: lw PEFE */
/* volume envelope parameters */
/* 8*/ EMUX_FX_ENV2_DELAY, /* WORD: ENVVOL */
/* 9*/ EMUX_FX_ENV2_ATTACK, /* BYTE: up ATKHLDV */
/*10*/ EMUX_FX_ENV2_HOLD, /* BYTE: lw ATKHLDV */
/*11*/ EMUX_FX_ENV2_DECAY, /* BYTE: lw DCYSUSV */
/*12*/ EMUX_FX_ENV2_RELEASE, /* BYTE: lw DCYSUSV */
/*13*/ EMUX_FX_ENV2_SUSTAIN, /* BYTE: up DCYSUSV */
/* LFO1 (tremolo & vibrato) parameters */
/*14*/ EMUX_FX_LFO1_DELAY, /* WORD: LFO1VAL */
/*15*/ EMUX_FX_LFO1_FREQ, /* BYTE: lo TREMFRQ */
/*16*/ EMUX_FX_LFO1_VOLUME, /* BYTE: up TREMFRQ */
/*17*/ EMUX_FX_LFO1_PITCH, /* BYTE: up FMMOD */
/*18*/ EMUX_FX_LFO1_CUTOFF, /* BYTE: lo FMMOD */
/* LFO2 (vibrato) parameters */
/*19*/ EMUX_FX_LFO2_DELAY, /* WORD: LFO2VAL */
/*20*/ EMUX_FX_LFO2_FREQ, /* BYTE: lo FM2FRQ2 */
/*21*/ EMUX_FX_LFO2_PITCH, /* BYTE: up FM2FRQ2 */
/* Other overall effect parameters */
/*22*/ EMUX_FX_INIT_PITCH, /* SHORT: pitch offset */
/*23*/ EMUX_FX_CHORUS, /* BYTE: chorus effects send (0-255) */
/*24*/ EMUX_FX_REVERB, /* BYTE: reverb effects send (0-255) */
/*25*/ EMUX_FX_CUTOFF, /* BYTE: up IFATN */
/*26*/ EMUX_FX_FILTERQ, /* BYTE: up CCCA */
/* Sample / loop offset changes */
/*27*/ EMUX_FX_SAMPLE_START, /* SHORT: offset */
/*28*/ EMUX_FX_LOOP_START, /* SHORT: offset */
/*29*/ EMUX_FX_LOOP_END, /* SHORT: offset */
/*30*/ EMUX_FX_COARSE_SAMPLE_START, /* SHORT: upper word offset */
/*31*/ EMUX_FX_COARSE_LOOP_START, /* SHORT: upper word offset */
/*32*/ EMUX_FX_COARSE_LOOP_END, /* SHORT: upper word offset */
/*33*/ EMUX_FX_ATTEN, /* BYTE: lo IFATN */
EMUX_FX_END,
};
/* number of effects */
#define EMUX_NUM_EFFECTS EMUX_FX_END
/* effect flag values */
#define EMUX_FX_FLAG_OFF 0
#define EMUX_FX_FLAG_SET 1
#define EMUX_FX_FLAG_ADD 2
#endif /* __SOUND_EMUX_LEGACY_H */

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#ifndef __SOUND_EMUX_SYNTH_H
#define __SOUND_EMUX_SYNTH_H
/*
* Defines for the Emu-series WaveTable chip
*
* Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
*
* 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/seq_kernel.h>
#include <sound/seq_device.h>
#include <sound/soundfont.h>
#include <sound/seq_midi_emul.h>
#ifdef CONFIG_SND_SEQUENCER_OSS
#include <sound/seq_oss.h>
#endif
#include <sound/emux_legacy.h>
#include <sound/seq_virmidi.h>
/*
* compile flags
*/
#define SNDRV_EMUX_USE_RAW_EFFECT
struct snd_emux;
struct snd_emux_port;
struct snd_emux_voice;
struct snd_emux_effect_table;
/*
* operators
*/
struct snd_emux_operators {
struct module *owner;
struct snd_emux_voice *(*get_voice)(struct snd_emux *emu,
struct snd_emux_port *port);
int (*prepare)(struct snd_emux_voice *vp);
void (*trigger)(struct snd_emux_voice *vp);
void (*release)(struct snd_emux_voice *vp);
void (*update)(struct snd_emux_voice *vp, int update);
void (*terminate)(struct snd_emux_voice *vp);
void (*free_voice)(struct snd_emux_voice *vp);
void (*reset)(struct snd_emux *emu, int ch);
/* the first parameters are struct snd_emux */
int (*sample_new)(struct snd_emux *emu, struct snd_sf_sample *sp,
struct snd_util_memhdr *hdr,
const void __user *data, long count);
int (*sample_free)(struct snd_emux *emu, struct snd_sf_sample *sp,
struct snd_util_memhdr *hdr);
void (*sample_reset)(struct snd_emux *emu);
int (*load_fx)(struct snd_emux *emu, int type, int arg,
const void __user *data, long count);
void (*sysex)(struct snd_emux *emu, char *buf, int len, int parsed,
struct snd_midi_channel_set *chset);
#ifdef CONFIG_SND_SEQUENCER_OSS
int (*oss_ioctl)(struct snd_emux *emu, int cmd, int p1, int p2);
#endif
};
/*
* constant values
*/
#define SNDRV_EMUX_MAX_PORTS 32 /* max # of sequencer ports */
#define SNDRV_EMUX_MAX_VOICES 64 /* max # of voices */
#define SNDRV_EMUX_MAX_MULTI_VOICES 16 /* max # of playable voices
* simultineously
*/
/*
* flags
*/
#define SNDRV_EMUX_ACCEPT_ROM (1<<0)
/*
* emuX wavetable
*/
struct snd_emux {
struct snd_card *card; /* assigned card */
/* following should be initialized before registration */
int max_voices; /* Number of voices */
int mem_size; /* memory size (in byte) */
int num_ports; /* number of ports to be created */
int pitch_shift; /* pitch shift value (for Emu10k1) */
struct snd_emux_operators ops; /* operators */
void *hw; /* hardware */
unsigned long flags; /* other conditions */
int midi_ports; /* number of virtual midi devices */
int midi_devidx; /* device offset of virtual midi */
unsigned int linear_panning: 1; /* panning is linear (sbawe = 1, emu10k1 = 0) */
int hwdep_idx; /* hwdep device index */
struct snd_hwdep *hwdep; /* hwdep device */
/* private */
int num_voices; /* current number of voices */
struct snd_sf_list *sflist; /* root of SoundFont list */
struct snd_emux_voice *voices; /* Voices (EMU 'channel') */
int use_time; /* allocation counter */
spinlock_t voice_lock; /* Lock for voice access */
struct mutex register_mutex;
int client; /* For the sequencer client */
int ports[SNDRV_EMUX_MAX_PORTS]; /* The ports for this device */
struct snd_emux_port *portptrs[SNDRV_EMUX_MAX_PORTS];
int used; /* use counter */
char *name; /* name of the device (internal) */
struct snd_rawmidi **vmidi;
struct timer_list tlist; /* for pending note-offs */
int timer_active;
struct snd_util_memhdr *memhdr; /* memory chunk information */
#ifdef CONFIG_PROC_FS
struct snd_info_entry *proc;
#endif
#ifdef CONFIG_SND_SEQUENCER_OSS
struct snd_seq_device *oss_synth;
#endif
};
/*
* sequencer port information
*/
struct snd_emux_port {
struct snd_midi_channel_set chset;
struct snd_emux *emu;
char port_mode; /* operation mode */
int volume_atten; /* emuX raw attenuation */
unsigned long drum_flags; /* drum bitmaps */
int ctrls[EMUX_MD_END]; /* control parameters */
#ifdef SNDRV_EMUX_USE_RAW_EFFECT
struct snd_emux_effect_table *effect;
#endif
#ifdef CONFIG_SND_SEQUENCER_OSS
struct snd_seq_oss_arg *oss_arg;
#endif
};
/* port_mode */
#define SNDRV_EMUX_PORT_MODE_MIDI 0 /* normal MIDI port */
#define SNDRV_EMUX_PORT_MODE_OSS_SYNTH 1 /* OSS synth port */
#define SNDRV_EMUX_PORT_MODE_OSS_MIDI 2 /* OSS multi channel synth port */
/*
* A structure to keep track of each hardware voice
*/
struct snd_emux_voice {
int ch; /* Hardware channel number */
int state; /* status */
#define SNDRV_EMUX_ST_OFF 0x00 /* Not playing, and inactive */
#define SNDRV_EMUX_ST_ON 0x01 /* Note on */
#define SNDRV_EMUX_ST_RELEASED (0x02|SNDRV_EMUX_ST_ON) /* Note released */
#define SNDRV_EMUX_ST_SUSTAINED (0x04|SNDRV_EMUX_ST_ON) /* Note sustained */
#define SNDRV_EMUX_ST_STANDBY (0x08|SNDRV_EMUX_ST_ON) /* Waiting to be triggered */
#define SNDRV_EMUX_ST_PENDING (0x10|SNDRV_EMUX_ST_ON) /* Note will be released */
#define SNDRV_EMUX_ST_LOCKED 0x100 /* Not accessible */
unsigned int time; /* An allocation time */
unsigned char note; /* Note currently assigned to this voice */
unsigned char key;
unsigned char velocity; /* Velocity of current note */
struct snd_sf_zone *zone; /* Zone assigned to this note */
void *block; /* sample block pointer (optional) */
struct snd_midi_channel *chan; /* Midi channel for this note */
struct snd_emux_port *port; /* associated port */
struct snd_emux *emu; /* assigned root info */
void *hw; /* hardware pointer (emu8000 or emu10k1) */
unsigned long ontime; /* jiffies at note triggered */
/* Emu8k/Emu10k1 registers */
struct soundfont_voice_info reg;
/* additional registers */
int avol; /* volume attenuation */
int acutoff; /* cutoff target */
int apitch; /* pitch offset */
int apan; /* pan/aux pair */
int aaux;
int ptarget; /* pitch target */
int vtarget; /* volume target */
int ftarget; /* filter target */
};
/*
* update flags (can be combined)
*/
#define SNDRV_EMUX_UPDATE_VOLUME (1<<0)
#define SNDRV_EMUX_UPDATE_PITCH (1<<1)
#define SNDRV_EMUX_UPDATE_PAN (1<<2)
#define SNDRV_EMUX_UPDATE_FMMOD (1<<3)
#define SNDRV_EMUX_UPDATE_TREMFREQ (1<<4)
#define SNDRV_EMUX_UPDATE_FM2FRQ2 (1<<5)
#define SNDRV_EMUX_UPDATE_Q (1<<6)
#ifdef SNDRV_EMUX_USE_RAW_EFFECT
/*
* effect table
*/
struct snd_emux_effect_table {
/* Emu8000 specific effects */
short val[EMUX_NUM_EFFECTS];
unsigned char flag[EMUX_NUM_EFFECTS];
};
#endif /* SNDRV_EMUX_USE_RAW_EFFECT */
/*
* prototypes - interface to Emu10k1 and Emu8k routines
*/
int snd_emux_new(struct snd_emux **remu);
int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, char *name);
int snd_emux_free(struct snd_emux *emu);
/*
* exported functions
*/
void snd_emux_terminate_all(struct snd_emux *emu);
void snd_emux_lock_voice(struct snd_emux *emu, int voice);
void snd_emux_unlock_voice(struct snd_emux *emu, int voice);
#endif /* __SOUND_EMUX_SYNTH_H */

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#ifndef __SOUND_ES1688_H
#define __SOUND_ES1688_H
/*
* Header file for ES488/ES1688
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/control.h>
#include <sound/pcm.h>
#include <linux/interrupt.h>
#define ES1688_HW_AUTO 0x0000
#define ES1688_HW_688 0x0001
#define ES1688_HW_1688 0x0002
#define ES1688_HW_UNDEF 0x0003
struct snd_es1688 {
unsigned long port; /* port of ESS chip */
struct resource *res_port;
unsigned long mpu_port; /* MPU-401 port of ESS chip */
int irq; /* IRQ number of ESS chip */
int mpu_irq; /* MPU IRQ */
int dma8; /* 8-bit DMA */
unsigned short version; /* version of ESS chip */
unsigned short hardware; /* see to ES1688_HW_XXXX */
unsigned short trigger_value;
unsigned char pad;
unsigned int dma_size;
struct snd_pcm *pcm;
struct snd_pcm_substream *playback_substream;
struct snd_pcm_substream *capture_substream;
spinlock_t reg_lock;
spinlock_t mixer_lock;
};
/* I/O ports */
#define ES1688P(codec, x) ((codec)->port + e_s_s_ESS1688##x)
#define e_s_s_ESS1688RESET 0x6
#define e_s_s_ESS1688READ 0xa
#define e_s_s_ESS1688WRITE 0xc
#define e_s_s_ESS1688COMMAND 0xc
#define e_s_s_ESS1688STATUS 0xc
#define e_s_s_ESS1688DATA_AVAIL 0xe
#define e_s_s_ESS1688DATA_AVAIL_16 0xf
#define e_s_s_ESS1688MIXER_ADDR 0x4
#define e_s_s_ESS1688MIXER_DATA 0x5
#define e_s_s_ESS1688OPL3_LEFT 0x0
#define e_s_s_ESS1688OPL3_RIGHT 0x2
#define e_s_s_ESS1688OPL3_BOTH 0x8
#define e_s_s_ESS1688ENABLE0 0x0
#define e_s_s_ESS1688ENABLE1 0x9
#define e_s_s_ESS1688ENABLE2 0xb
#define e_s_s_ESS1688INIT1 0x7
#define ES1688_DSP_CMD_DMAOFF 0xd0
#define ES1688_DSP_CMD_SPKON 0xd1
#define ES1688_DSP_CMD_SPKOFF 0xd3
#define ES1688_DSP_CMD_DMAON 0xd4
#define ES1688_PCM_DEV 0x14
#define ES1688_MIC_DEV 0x1a
#define ES1688_REC_DEV 0x1c
#define ES1688_MASTER_DEV 0x32
#define ES1688_FM_DEV 0x36
#define ES1688_CD_DEV 0x38
#define ES1688_AUX_DEV 0x3a
#define ES1688_SPEAKER_DEV 0x3c
#define ES1688_LINE_DEV 0x3e
#define ES1688_RECLEV_DEV 0xb4
#define ES1688_MIXS_MASK 0x17
#define ES1688_MIXS_MIC 0x00
#define ES1688_MIXS_MIC_MASTER 0x01
#define ES1688_MIXS_CD 0x02
#define ES1688_MIXS_AOUT 0x03
#define ES1688_MIXS_MIC1 0x04
#define ES1688_MIXS_REC_MIX 0x05
#define ES1688_MIXS_LINE 0x06
#define ES1688_MIXS_MASTER 0x07
#define ES1688_MIXS_MUTE 0x10
/*
*/
void snd_es1688_mixer_write(struct snd_es1688 *chip, unsigned char reg, unsigned char data);
int snd_es1688_create(struct snd_card *card,
struct snd_es1688 *chip,
unsigned long port,
unsigned long mpu_port,
int irq,
int mpu_irq,
int dma8,
unsigned short hardware);
int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, int device,
struct snd_pcm **rpcm);
int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip);
int snd_es1688_reset(struct snd_es1688 *chip);
#endif /* __SOUND_ES1688_H */

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#ifndef __SOUND_GUS_H
#define __SOUND_GUS_H
/*
* Global structures used for GUS part of ALSA driver
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/pcm.h>
#include <sound/rawmidi.h>
#include <sound/timer.h>
#include <sound/seq_midi_emul.h>
#include <sound/seq_device.h>
#include <asm/io.h>
/* IO ports */
#define GUSP(gus, x) ((gus)->gf1.port + SNDRV_g_u_s_##x)
#define SNDRV_g_u_s_MIDICTRL (0x320-0x220)
#define SNDRV_g_u_s_MIDISTAT (0x320-0x220)
#define SNDRV_g_u_s_MIDIDATA (0x321-0x220)
#define SNDRV_g_u_s_GF1PAGE (0x322-0x220)
#define SNDRV_g_u_s_GF1REGSEL (0x323-0x220)
#define SNDRV_g_u_s_GF1DATALOW (0x324-0x220)
#define SNDRV_g_u_s_GF1DATAHIGH (0x325-0x220)
#define SNDRV_g_u_s_IRQSTAT (0x226-0x220)
#define SNDRV_g_u_s_TIMERCNTRL (0x228-0x220)
#define SNDRV_g_u_s_TIMERDATA (0x229-0x220)
#define SNDRV_g_u_s_DRAM (0x327-0x220)
#define SNDRV_g_u_s_MIXCNTRLREG (0x220-0x220)
#define SNDRV_g_u_s_IRQDMACNTRLREG (0x22b-0x220)
#define SNDRV_g_u_s_REGCNTRLS (0x22f-0x220)
#define SNDRV_g_u_s_BOARDVERSION (0x726-0x220)
#define SNDRV_g_u_s_MIXCNTRLPORT (0x726-0x220)
#define SNDRV_g_u_s_IVER (0x325-0x220)
#define SNDRV_g_u_s_MIXDATAPORT (0x326-0x220)
#define SNDRV_g_u_s_MAXCNTRLPORT (0x326-0x220)
/* GF1 registers */
/* global registers */
#define SNDRV_GF1_GB_ACTIVE_VOICES 0x0e
#define SNDRV_GF1_GB_VOICES_IRQ 0x0f
#define SNDRV_GF1_GB_GLOBAL_MODE 0x19
#define SNDRV_GF1_GW_LFO_BASE 0x1a
#define SNDRV_GF1_GB_VOICES_IRQ_READ 0x1f
#define SNDRV_GF1_GB_DRAM_DMA_CONTROL 0x41
#define SNDRV_GF1_GW_DRAM_DMA_LOW 0x42
#define SNDRV_GF1_GW_DRAM_IO_LOW 0x43
#define SNDRV_GF1_GB_DRAM_IO_HIGH 0x44
#define SNDRV_GF1_GB_SOUND_BLASTER_CONTROL 0x45
#define SNDRV_GF1_GB_ADLIB_TIMER_1 0x46
#define SNDRV_GF1_GB_ADLIB_TIMER_2 0x47
#define SNDRV_GF1_GB_RECORD_RATE 0x48
#define SNDRV_GF1_GB_REC_DMA_CONTROL 0x49
#define SNDRV_GF1_GB_JOYSTICK_DAC_LEVEL 0x4b
#define SNDRV_GF1_GB_RESET 0x4c
#define SNDRV_GF1_GB_DRAM_DMA_HIGH 0x50
#define SNDRV_GF1_GW_DRAM_IO16 0x51
#define SNDRV_GF1_GW_MEMORY_CONFIG 0x52
#define SNDRV_GF1_GB_MEMORY_CONTROL 0x53
#define SNDRV_GF1_GW_FIFO_RECORD_BASE_ADDR 0x54
#define SNDRV_GF1_GW_FIFO_PLAY_BASE_ADDR 0x55
#define SNDRV_GF1_GW_FIFO_SIZE 0x56
#define SNDRV_GF1_GW_INTERLEAVE 0x57
#define SNDRV_GF1_GB_COMPATIBILITY 0x59
#define SNDRV_GF1_GB_DECODE_CONTROL 0x5a
#define SNDRV_GF1_GB_VERSION_NUMBER 0x5b
#define SNDRV_GF1_GB_MPU401_CONTROL_A 0x5c
#define SNDRV_GF1_GB_MPU401_CONTROL_B 0x5d
#define SNDRV_GF1_GB_EMULATION_IRQ 0x60
/* voice specific registers */
#define SNDRV_GF1_VB_ADDRESS_CONTROL 0x00
#define SNDRV_GF1_VW_FREQUENCY 0x01
#define SNDRV_GF1_VW_START_HIGH 0x02
#define SNDRV_GF1_VW_START_LOW 0x03
#define SNDRV_GF1_VA_START SNDRV_GF1_VW_START_HIGH
#define SNDRV_GF1_VW_END_HIGH 0x04
#define SNDRV_GF1_VW_END_LOW 0x05
#define SNDRV_GF1_VA_END SNDRV_GF1_VW_END_HIGH
#define SNDRV_GF1_VB_VOLUME_RATE 0x06
#define SNDRV_GF1_VB_VOLUME_START 0x07
#define SNDRV_GF1_VB_VOLUME_END 0x08
#define SNDRV_GF1_VW_VOLUME 0x09
#define SNDRV_GF1_VW_CURRENT_HIGH 0x0a
#define SNDRV_GF1_VW_CURRENT_LOW 0x0b
#define SNDRV_GF1_VA_CURRENT SNDRV_GF1_VW_CURRENT_HIGH
#define SNDRV_GF1_VB_PAN 0x0c
#define SNDRV_GF1_VW_OFFSET_RIGHT 0x0c
#define SNDRV_GF1_VB_VOLUME_CONTROL 0x0d
#define SNDRV_GF1_VB_UPPER_ADDRESS 0x10
#define SNDRV_GF1_VW_EFFECT_HIGH 0x11
#define SNDRV_GF1_VW_EFFECT_LOW 0x12
#define SNDRV_GF1_VA_EFFECT SNDRV_GF1_VW_EFFECT_HIGH
#define SNDRV_GF1_VW_OFFSET_LEFT 0x13
#define SNDRV_GF1_VB_ACCUMULATOR 0x14
#define SNDRV_GF1_VB_MODE 0x15
#define SNDRV_GF1_VW_EFFECT_VOLUME 0x16
#define SNDRV_GF1_VB_FREQUENCY_LFO 0x17
#define SNDRV_GF1_VB_VOLUME_LFO 0x18
#define SNDRV_GF1_VW_OFFSET_RIGHT_FINAL 0x1b
#define SNDRV_GF1_VW_OFFSET_LEFT_FINAL 0x1c
#define SNDRV_GF1_VW_EFFECT_VOLUME_FINAL 0x1d
/* ICS registers */
#define SNDRV_ICS_MIC_DEV 0
#define SNDRV_ICS_LINE_DEV 1
#define SNDRV_ICS_CD_DEV 2
#define SNDRV_ICS_GF1_DEV 3
#define SNDRV_ICS_NONE_DEV 4
#define SNDRV_ICS_MASTER_DEV 5
/* LFO */
#define SNDRV_LFO_TREMOLO 0
#define SNDRV_LFO_VIBRATO 1
/* misc */
#define SNDRV_GF1_DMA_UNSIGNED 0x80
#define SNDRV_GF1_DMA_16BIT 0x40
#define SNDRV_GF1_DMA_IRQ 0x20
#define SNDRV_GF1_DMA_WIDTH16 0x04
#define SNDRV_GF1_DMA_READ 0x02 /* read from GUS's DRAM */
#define SNDRV_GF1_DMA_ENABLE 0x01
/* ramp ranges */
#define SNDRV_GF1_ATTEN(x) (snd_gf1_atten_table[x])
#define SNDRV_GF1_MIN_VOLUME 1800
#define SNDRV_GF1_MAX_VOLUME 4095
#define SNDRV_GF1_MIN_OFFSET (SNDRV_GF1_MIN_VOLUME>>4)
#define SNDRV_GF1_MAX_OFFSET 255
#define SNDRV_GF1_MAX_TDEPTH 90
/* defines for memory manager */
#define SNDRV_GF1_MEM_BLOCK_16BIT 0x0001
#define SNDRV_GF1_MEM_OWNER_DRIVER 0x0001
#define SNDRV_GF1_MEM_OWNER_WAVE_SIMPLE 0x0002
#define SNDRV_GF1_MEM_OWNER_WAVE_GF1 0x0003
#define SNDRV_GF1_MEM_OWNER_WAVE_IWFFFF 0x0004
/* constants for interrupt handlers */
#define SNDRV_GF1_HANDLER_MIDI_OUT 0x00010000
#define SNDRV_GF1_HANDLER_MIDI_IN 0x00020000
#define SNDRV_GF1_HANDLER_TIMER1 0x00040000
#define SNDRV_GF1_HANDLER_TIMER2 0x00080000
#define SNDRV_GF1_HANDLER_VOICE 0x00100000
#define SNDRV_GF1_HANDLER_DMA_WRITE 0x00200000
#define SNDRV_GF1_HANDLER_DMA_READ 0x00400000
#define SNDRV_GF1_HANDLER_ALL (0xffff0000&~SNDRV_GF1_HANDLER_VOICE)
/* constants for DMA flags */
#define SNDRV_GF1_DMA_TRIGGER 1
/* --- */
struct snd_gus_card;
/* GF1 specific structure */
struct snd_gf1_bank_info {
unsigned int address;
unsigned int size;
};
struct snd_gf1_mem_block {
unsigned short flags; /* flags - SNDRV_GF1_MEM_BLOCK_XXXX */
unsigned short owner; /* owner - SNDRV_GF1_MEM_OWNER_XXXX */
unsigned int share; /* share count */
unsigned int share_id[4]; /* share ID */
unsigned int ptr;
unsigned int size;
char *name;
struct snd_gf1_mem_block *next;
struct snd_gf1_mem_block *prev;
};
struct snd_gf1_mem {
struct snd_gf1_bank_info banks_8[4];
struct snd_gf1_bank_info banks_16[4];
struct snd_gf1_mem_block *first;
struct snd_gf1_mem_block *last;
struct mutex memory_mutex;
};
struct snd_gf1_dma_block {
void *buffer; /* buffer in computer's RAM */
unsigned long buf_addr; /* buffer address */
unsigned int addr; /* address in onboard memory */
unsigned int count; /* count in bytes */
unsigned int cmd; /* DMA command (format) */
void (*ack)(struct snd_gus_card * gus, void *private_data);
void *private_data;
struct snd_gf1_dma_block *next;
};
struct snd_gus_port {
struct snd_midi_channel_set * chset;
struct snd_gus_card * gus;
int mode; /* operation mode */
int client; /* sequencer client number */
int port; /* sequencer port number */
unsigned int midi_has_voices: 1;
};
struct snd_gus_voice;
#define SNDRV_GF1_VOICE_TYPE_PCM 0
#define SNDRV_GF1_VOICE_TYPE_SYNTH 1
#define SNDRV_GF1_VOICE_TYPE_MIDI 2
#define SNDRV_GF1_VFLG_RUNNING (1<<0)
#define SNDRV_GF1_VFLG_EFFECT_TIMER1 (1<<1)
#define SNDRV_GF1_VFLG_PAN (1<<2)
enum snd_gus_volume_state {
VENV_BEFORE,
VENV_ATTACK,
VENV_SUSTAIN,
VENV_RELEASE,
VENV_DONE,
VENV_VOLUME
};
struct snd_gus_voice {
int number;
unsigned int use: 1,
pcm: 1,
synth:1,
midi: 1;
unsigned int flags;
unsigned char client;
unsigned char port;
unsigned char index;
unsigned char pad;
#ifdef CONFIG_SND_DEBUG
unsigned int interrupt_stat_wave;
unsigned int interrupt_stat_volume;
#endif
void (*handler_wave) (struct snd_gus_card * gus, struct snd_gus_voice * voice);
void (*handler_volume) (struct snd_gus_card * gus, struct snd_gus_voice * voice);
void (*handler_effect) (struct snd_gus_card * gus, struct snd_gus_voice * voice);
void (*volume_change) (struct snd_gus_card * gus);
struct snd_gus_sample_ops *sample_ops;
/* running status / registers */
unsigned short fc_register;
unsigned short fc_lfo;
unsigned short gf1_volume;
unsigned char control;
unsigned char mode;
unsigned char gf1_pan;
unsigned char effect_accumulator;
unsigned char volume_control;
unsigned char venv_value_next;
enum snd_gus_volume_state venv_state;
enum snd_gus_volume_state venv_state_prev;
unsigned short vlo;
unsigned short vro;
unsigned short gf1_effect_volume;
/* --- */
void *private_data;
void (*private_free)(struct snd_gus_voice *voice);
};
struct snd_gf1 {
unsigned int enh_mode:1, /* enhanced mode (GFA1) */
hw_lfo:1, /* use hardware LFO */
sw_lfo:1, /* use software LFO */
effect:1; /* use effect voices */
unsigned long port; /* port of GF1 chip */
struct resource *res_port1;
struct resource *res_port2;
int irq; /* IRQ number */
int dma1; /* DMA1 number */
int dma2; /* DMA2 number */
unsigned int memory; /* GUS's DRAM size in bytes */
unsigned int rom_memory; /* GUS's ROM size in bytes */
unsigned int rom_present; /* bitmask */
unsigned int rom_banks; /* GUS's ROM banks */
struct snd_gf1_mem mem_alloc;
/* registers */
unsigned short reg_page;
unsigned short reg_regsel;
unsigned short reg_data8;
unsigned short reg_data16;
unsigned short reg_irqstat;
unsigned short reg_dram;
unsigned short reg_timerctrl;
unsigned short reg_timerdata;
unsigned char ics_regs[6][2];
/* --------- */
unsigned char active_voices; /* active voices */
unsigned char active_voice; /* selected voice (GF1PAGE register) */
struct snd_gus_voice voices[32]; /* GF1 voices */
unsigned int default_voice_address;
unsigned short playback_freq; /* GF1 playback (mixing) frequency */
unsigned short mode; /* see to SNDRV_GF1_MODE_XXXX */
unsigned char volume_ramp;
unsigned char smooth_pan;
unsigned char full_range_pan;
unsigned char pad0;
unsigned char *lfos;
/* interrupt handlers */
void (*interrupt_handler_midi_out) (struct snd_gus_card * gus);
void (*interrupt_handler_midi_in) (struct snd_gus_card * gus);
void (*interrupt_handler_timer1) (struct snd_gus_card * gus);
void (*interrupt_handler_timer2) (struct snd_gus_card * gus);
void (*interrupt_handler_dma_write) (struct snd_gus_card * gus);
void (*interrupt_handler_dma_read) (struct snd_gus_card * gus);
#ifdef CONFIG_SND_DEBUG
unsigned int interrupt_stat_midi_out;
unsigned int interrupt_stat_midi_in;
unsigned int interrupt_stat_timer1;
unsigned int interrupt_stat_timer2;
unsigned int interrupt_stat_dma_write;
unsigned int interrupt_stat_dma_read;
unsigned int interrupt_stat_voice_lost;
#endif
/* synthesizer */
int seq_client;
struct snd_gus_port seq_ports[4];
/* timer */
unsigned short timer_enabled;
struct snd_timer *timer1;
struct snd_timer *timer2;
/* midi */
unsigned short uart_cmd;
unsigned int uart_framing;
unsigned int uart_overrun;
/* dma operations */
unsigned int dma_flags;
unsigned int dma_shared;
struct snd_gf1_dma_block *dma_data_pcm;
struct snd_gf1_dma_block *dma_data_pcm_last;
struct snd_gf1_dma_block *dma_data_synth;
struct snd_gf1_dma_block *dma_data_synth_last;
void (*dma_ack)(struct snd_gus_card * gus, void *private_data);
void *dma_private_data;
/* pcm */
int pcm_channels;
int pcm_alloc_voices;
unsigned short pcm_volume_level_left;
unsigned short pcm_volume_level_right;
unsigned short pcm_volume_level_left1;
unsigned short pcm_volume_level_right1;
unsigned char pcm_rcntrl_reg;
unsigned char pad_end;
};
/* main structure for GUS card */
struct snd_gus_card {
struct snd_card *card;
unsigned int
initialized: 1, /* resources were initialized */
equal_irq:1, /* GF1 and CODEC shares IRQ (GUS MAX only) */
equal_dma:1, /* if dma channels are equal (not valid for daughter board) */
ics_flag:1, /* have we ICS mixer chip */
ics_flipped:1, /* ICS mixer have flipped some channels? */
codec_flag:1, /* have we CODEC chip? */
max_flag:1, /* have we GUS MAX card? */
max_ctrl_flag:1, /* have we original GUS MAX card? */
daughter_flag:1, /* have we daughter board? */
interwave:1, /* hey - we have InterWave card */
ess_flag:1, /* ESS chip found... GUS Extreme */
ace_flag:1, /* GUS ACE detected */
uart_enable:1; /* enable MIDI UART */
unsigned short revision; /* revision of chip */
unsigned short max_cntrl_val; /* GUS MAX control value */
unsigned short mix_cntrl_reg; /* mixer control register */
unsigned short joystick_dac; /* joystick DAC level */
int timer_dev; /* timer device */
struct snd_gf1 gf1; /* gf1 specific variables */
struct snd_pcm *pcm;
struct snd_pcm_substream *pcm_cap_substream;
unsigned int c_dma_size;
unsigned int c_period_size;
unsigned int c_pos;
struct snd_rawmidi *midi_uart;
struct snd_rawmidi_substream *midi_substream_output;
struct snd_rawmidi_substream *midi_substream_input;
spinlock_t reg_lock;
spinlock_t voice_alloc;
spinlock_t active_voice_lock;
spinlock_t event_lock;
spinlock_t dma_lock;
spinlock_t pcm_volume_level_lock;
spinlock_t uart_cmd_lock;
struct mutex dma_mutex;
struct mutex register_mutex;
};
/* I/O functions for GF1/InterWave chip - gus_io.c */
static inline void snd_gf1_select_voice(struct snd_gus_card * gus, int voice)
{
unsigned long flags;
spin_lock_irqsave(&gus->active_voice_lock, flags);
if (voice != gus->gf1.active_voice) {
gus->gf1.active_voice = voice;
outb(voice, GUSP(gus, GF1PAGE));
}
spin_unlock_irqrestore(&gus->active_voice_lock, flags);
}
static inline void snd_gf1_uart_cmd(struct snd_gus_card * gus, unsigned char b)
{
outb(gus->gf1.uart_cmd = b, GUSP(gus, MIDICTRL));
}
static inline unsigned char snd_gf1_uart_stat(struct snd_gus_card * gus)
{
return inb(GUSP(gus, MIDISTAT));
}
static inline void snd_gf1_uart_put(struct snd_gus_card * gus, unsigned char b)
{
outb(b, GUSP(gus, MIDIDATA));
}
static inline unsigned char snd_gf1_uart_get(struct snd_gus_card * gus)
{
return inb(GUSP(gus, MIDIDATA));
}
extern void snd_gf1_delay(struct snd_gus_card * gus);
extern void snd_gf1_ctrl_stop(struct snd_gus_card * gus, unsigned char reg);
extern void snd_gf1_write8(struct snd_gus_card * gus, unsigned char reg, unsigned char data);
extern unsigned char snd_gf1_look8(struct snd_gus_card * gus, unsigned char reg);
static inline unsigned char snd_gf1_read8(struct snd_gus_card * gus, unsigned char reg)
{
return snd_gf1_look8(gus, reg | 0x80);
}
extern void snd_gf1_write16(struct snd_gus_card * gus, unsigned char reg, unsigned int data);
extern unsigned short snd_gf1_look16(struct snd_gus_card * gus, unsigned char reg);
static inline unsigned short snd_gf1_read16(struct snd_gus_card * gus, unsigned char reg)
{
return snd_gf1_look16(gus, reg | 0x80);
}
extern void snd_gf1_adlib_write(struct snd_gus_card * gus, unsigned char reg, unsigned char data);
extern void snd_gf1_dram_addr(struct snd_gus_card * gus, unsigned int addr);
extern void snd_gf1_poke(struct snd_gus_card * gus, unsigned int addr, unsigned char data);
extern unsigned char snd_gf1_peek(struct snd_gus_card * gus, unsigned int addr);
extern void snd_gf1_write_addr(struct snd_gus_card * gus, unsigned char reg, unsigned int addr, short w_16bit);
extern unsigned int snd_gf1_read_addr(struct snd_gus_card * gus, unsigned char reg, short w_16bit);
extern void snd_gf1_i_ctrl_stop(struct snd_gus_card * gus, unsigned char reg);
extern void snd_gf1_i_write8(struct snd_gus_card * gus, unsigned char reg, unsigned char data);
extern unsigned char snd_gf1_i_look8(struct snd_gus_card * gus, unsigned char reg);
extern void snd_gf1_i_write16(struct snd_gus_card * gus, unsigned char reg, unsigned int data);
static inline unsigned char snd_gf1_i_read8(struct snd_gus_card * gus, unsigned char reg)
{
return snd_gf1_i_look8(gus, reg | 0x80);
}
extern unsigned short snd_gf1_i_look16(struct snd_gus_card * gus, unsigned char reg);
static inline unsigned short snd_gf1_i_read16(struct snd_gus_card * gus, unsigned char reg)
{
return snd_gf1_i_look16(gus, reg | 0x80);
}
extern void snd_gf1_select_active_voices(struct snd_gus_card * gus);
/* gus_lfo.c */
struct _SND_IW_LFO_PROGRAM {
unsigned short freq_and_control;
unsigned char depth_final;
unsigned char depth_inc;
unsigned short twave;
unsigned short depth;
};
#if 0
extern irqreturn_t snd_gf1_lfo_effect_interrupt(struct snd_gus_card * gus, snd_gf1_voice_t * voice);
#endif
extern void snd_gf1_lfo_init(struct snd_gus_card * gus);
extern void snd_gf1_lfo_done(struct snd_gus_card * gus);
extern void snd_gf1_lfo_program(struct snd_gus_card * gus, int voice, int lfo_type, struct _SND_IW_LFO_PROGRAM *program);
extern void snd_gf1_lfo_enable(struct snd_gus_card * gus, int voice, int lfo_type);
extern void snd_gf1_lfo_disable(struct snd_gus_card * gus, int voice, int lfo_type);
extern void snd_gf1_lfo_change_freq(struct snd_gus_card * gus, int voice, int lfo_type, int freq);
extern void snd_gf1_lfo_change_depth(struct snd_gus_card * gus, int voice, int lfo_type, int depth);
extern void snd_gf1_lfo_setup(struct snd_gus_card * gus, int voice, int lfo_type, int freq, int current_depth, int depth, int sweep, int shape);
extern void snd_gf1_lfo_shutdown(struct snd_gus_card * gus, int voice, int lfo_type);
#if 0
extern void snd_gf1_lfo_command(struct snd_gus_card * gus, int voice, unsigned char *command);
#endif
/* gus_mem.c */
void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup);
int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * block);
struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owner,
char *name, int size, int w_16,
int align, unsigned int *share_id);
int snd_gf1_mem_free(struct snd_gf1_mem * alloc, unsigned int address);
int snd_gf1_mem_free_owner(struct snd_gf1_mem * alloc, int owner);
int snd_gf1_mem_init(struct snd_gus_card * gus);
int snd_gf1_mem_done(struct snd_gus_card * gus);
/* gus_mem_proc.c */
int snd_gf1_mem_proc_init(struct snd_gus_card * gus);
/* gus_dma.c */
int snd_gf1_dma_init(struct snd_gus_card * gus);
int snd_gf1_dma_done(struct snd_gus_card * gus);
int snd_gf1_dma_transfer_block(struct snd_gus_card * gus,
struct snd_gf1_dma_block * block,
int atomic,
int synth);
/* gus_volume.c */
unsigned short snd_gf1_lvol_to_gvol_raw(unsigned int vol);
unsigned short snd_gf1_translate_freq(struct snd_gus_card * gus, unsigned int freq2);
/* gus_reset.c */
void snd_gf1_set_default_handlers(struct snd_gus_card * gus, unsigned int what);
void snd_gf1_smart_stop_voice(struct snd_gus_card * gus, unsigned short voice);
void snd_gf1_stop_voice(struct snd_gus_card * gus, unsigned short voice);
void snd_gf1_stop_voices(struct snd_gus_card * gus, unsigned short v_min, unsigned short v_max);
struct snd_gus_voice *snd_gf1_alloc_voice(struct snd_gus_card * gus, int type, int client, int port);
void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice);
int snd_gf1_start(struct snd_gus_card * gus);
int snd_gf1_stop(struct snd_gus_card * gus);
/* gus_mixer.c */
int snd_gf1_new_mixer(struct snd_gus_card * gus);
/* gus_pcm.c */
int snd_gf1_pcm_new(struct snd_gus_card * gus, int pcm_dev, int control_index, struct snd_pcm ** rpcm);
#ifdef CONFIG_SND_DEBUG
extern void snd_gf1_print_voice_registers(struct snd_gus_card * gus);
#endif
/* gus.c */
int snd_gus_use_inc(struct snd_gus_card * gus);
void snd_gus_use_dec(struct snd_gus_card * gus);
int snd_gus_create(struct snd_card *card,
unsigned long port,
int irq, int dma1, int dma2,
int timer_dev,
int voices,
int pcm_channels,
int effect,
struct snd_gus_card ** rgus);
int snd_gus_initialize(struct snd_gus_card * gus);
/* gus_irq.c */
irqreturn_t snd_gus_interrupt(int irq, void *dev_id);
#ifdef CONFIG_SND_DEBUG
void snd_gus_irq_profile_init(struct snd_gus_card *gus);
#endif
/* gus_uart.c */
int snd_gf1_rawmidi_new(struct snd_gus_card * gus, int device, struct snd_rawmidi **rrawmidi);
/* gus_dram.c */
int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
unsigned int addr, unsigned int size);
int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
unsigned int addr, unsigned int size, int rom);
#endif /* __SOUND_GUS_H */

44
include/sound/hda_hwdep.h Normal file
View File

@@ -0,0 +1,44 @@
/*
* HWDEP Interface for HD-audio codec
*
* Copyright (c) 2007 Takashi Iwai <tiwai@suse.de>
*
* This driver 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 driver 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
*/
#ifndef __SOUND_HDA_HWDEP_H
#define __SOUND_HDA_HWDEP_H
#define HDA_HWDEP_VERSION ((1 << 16) | (0 << 8) | (0 << 0)) /* 1.0.0 */
/* verb */
#define HDA_REG_NID_SHIFT 24
#define HDA_REG_VERB_SHIFT 8
#define HDA_REG_VAL_SHIFT 0
#define HDA_VERB(nid,verb,param) ((nid)<<24 | (verb)<<8 | (param))
struct hda_verb_ioctl {
u32 verb; /* HDA_VERB() */
u32 res; /* response */
};
/*
* ioctls
*/
#define HDA_IOCTL_PVERSION _IOR('H', 0x10, int)
#define HDA_IOCTL_VERB_WRITE _IOWR('H', 0x11, struct hda_verb_ioctl)
#define HDA_IOCTL_GET_WCAP _IOWR('H', 0x12, struct hda_verb_ioctl)
#endif

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#ifndef __SOUND_HWDEP_H
#define __SOUND_HWDEP_H
/*
* Hardware dependent layer
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/asound.h>
#include <linux/poll.h>
struct snd_hwdep;
/* hwdep file ops; all ops can be NULL */
struct snd_hwdep_ops {
long long (*llseek)(struct snd_hwdep *hw, struct file *file,
long long offset, int orig);
long (*read)(struct snd_hwdep *hw, char __user *buf,
long count, loff_t *offset);
long (*write)(struct snd_hwdep *hw, const char __user *buf,
long count, loff_t *offset);
int (*open)(struct snd_hwdep *hw, struct file * file);
int (*release)(struct snd_hwdep *hw, struct file * file);
unsigned int (*poll)(struct snd_hwdep *hw, struct file *file,
poll_table *wait);
int (*ioctl)(struct snd_hwdep *hw, struct file *file,
unsigned int cmd, unsigned long arg);
int (*ioctl_compat)(struct snd_hwdep *hw, struct file *file,
unsigned int cmd, unsigned long arg);
int (*mmap)(struct snd_hwdep *hw, struct file *file,
struct vm_area_struct *vma);
int (*dsp_status)(struct snd_hwdep *hw,
struct snd_hwdep_dsp_status *status);
int (*dsp_load)(struct snd_hwdep *hw,
struct snd_hwdep_dsp_image *image);
};
struct snd_hwdep {
struct snd_card *card;
struct list_head list;
int device;
char id[32];
char name[80];
int iface;
#ifdef CONFIG_SND_OSSEMUL
char oss_dev[32];
int oss_type;
int ossreg;
#endif
struct snd_hwdep_ops ops;
wait_queue_head_t open_wait;
void *private_data;
void (*private_free) (struct snd_hwdep *hwdep);
struct mutex open_mutex;
int used; /* reference counter */
unsigned int dsp_loaded; /* bit fields of loaded dsp indices */
unsigned int exclusive:1; /* exclusive access mode */
};
extern int snd_hwdep_new(struct snd_card *card, char *id, int device,
struct snd_hwdep **rhwdep);
#endif /* __SOUND_HWDEP_H */

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#ifndef __SOUND_I2C_H
#define __SOUND_I2C_H
/*
*
*
* 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
*
*
*/
#define SND_I2C_DEVICE_ADDRTEN (1<<0) /* 10-bit I2C address */
struct snd_i2c_device {
struct list_head list;
struct snd_i2c_bus *bus; /* I2C bus */
char name[32]; /* some useful device name */
unsigned short flags; /* device flags */
unsigned short addr; /* device address (might be 10-bit) */
unsigned long private_value;
void *private_data;
void (*private_free)(struct snd_i2c_device *device);
};
#define snd_i2c_device(n) list_entry(n, struct snd_i2c_device, list)
struct snd_i2c_bit_ops {
void (*start)(struct snd_i2c_bus *bus); /* transfer start */
void (*stop)(struct snd_i2c_bus *bus); /* transfer stop */
void (*direction)(struct snd_i2c_bus *bus, int clock, int data); /* set line direction (0 = write, 1 = read) */
void (*setlines)(struct snd_i2c_bus *bus, int clock, int data);
int (*getclock)(struct snd_i2c_bus *bus);
int (*getdata)(struct snd_i2c_bus *bus, int ack);
};
struct snd_i2c_ops {
int (*sendbytes)(struct snd_i2c_device *device, unsigned char *bytes, int count);
int (*readbytes)(struct snd_i2c_device *device, unsigned char *bytes, int count);
int (*probeaddr)(struct snd_i2c_bus *bus, unsigned short addr);
};
struct snd_i2c_bus {
struct snd_card *card; /* card which I2C belongs to */
char name[32]; /* some useful label */
struct mutex lock_mutex;
struct snd_i2c_bus *master; /* master bus when SCK/SCL is shared */
struct list_head buses; /* master: slave buses sharing SCK/SCL, slave: link list */
struct list_head devices; /* attached devices to this bus */
union {
struct snd_i2c_bit_ops *bit;
void *ops;
} hw_ops; /* lowlevel operations */
struct snd_i2c_ops *ops; /* midlevel operations */
unsigned long private_value;
void *private_data;
void (*private_free)(struct snd_i2c_bus *bus);
};
#define snd_i2c_slave_bus(n) list_entry(n, struct snd_i2c_bus, buses)
int snd_i2c_bus_create(struct snd_card *card, const char *name,
struct snd_i2c_bus *master, struct snd_i2c_bus **ri2c);
int snd_i2c_device_create(struct snd_i2c_bus *bus, const char *name,
unsigned char addr, struct snd_i2c_device **rdevice);
int snd_i2c_device_free(struct snd_i2c_device *device);
static inline void snd_i2c_lock(struct snd_i2c_bus *bus)
{
if (bus->master)
mutex_lock(&bus->master->lock_mutex);
else
mutex_lock(&bus->lock_mutex);
}
static inline void snd_i2c_unlock(struct snd_i2c_bus *bus)
{
if (bus->master)
mutex_unlock(&bus->master->lock_mutex);
else
mutex_unlock(&bus->lock_mutex);
}
int snd_i2c_sendbytes(struct snd_i2c_device *device, unsigned char *bytes, int count);
int snd_i2c_readbytes(struct snd_i2c_device *device, unsigned char *bytes, int count);
int snd_i2c_probeaddr(struct snd_i2c_bus *bus, unsigned short addr);
#endif /* __SOUND_I2C_H */

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#ifndef __SOUND_INFO_H
#define __SOUND_INFO_H
/*
* Header file for info interface
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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 <linux/poll.h>
/* buffer for information */
struct snd_info_buffer {
char *buffer; /* pointer to begin of buffer */
unsigned int curr; /* current position in buffer */
unsigned int size; /* current size */
unsigned int len; /* total length of buffer */
int stop; /* stop flag */
int error; /* error code */
};
#define SNDRV_INFO_CONTENT_TEXT 0
#define SNDRV_INFO_CONTENT_DATA 1
struct snd_info_entry;
struct snd_info_entry_text {
void (*read)(struct snd_info_entry *entry,
struct snd_info_buffer *buffer);
void (*write)(struct snd_info_entry *entry,
struct snd_info_buffer *buffer);
};
struct snd_info_entry_ops {
int (*open)(struct snd_info_entry *entry,
unsigned short mode, void **file_private_data);
int (*release)(struct snd_info_entry *entry,
unsigned short mode, void *file_private_data);
ssize_t (*read)(struct snd_info_entry *entry, void *file_private_data,
struct file *file, char __user *buf,
size_t count, loff_t pos);
ssize_t (*write)(struct snd_info_entry *entry, void *file_private_data,
struct file *file, const char __user *buf,
size_t count, loff_t pos);
loff_t (*llseek)(struct snd_info_entry *entry,
void *file_private_data, struct file *file,
loff_t offset, int orig);
unsigned int (*poll)(struct snd_info_entry *entry,
void *file_private_data, struct file *file,
poll_table *wait);
int (*ioctl)(struct snd_info_entry *entry, void *file_private_data,
struct file *file, unsigned int cmd, unsigned long arg);
int (*mmap)(struct snd_info_entry *entry, void *file_private_data,
struct inode *inode, struct file *file,
struct vm_area_struct *vma);
};
struct snd_info_entry {
const char *name;
umode_t mode;
long size;
unsigned short content;
union {
struct snd_info_entry_text text;
struct snd_info_entry_ops *ops;
} c;
struct snd_info_entry *parent;
struct snd_card *card;
struct module *module;
void *private_data;
void (*private_free)(struct snd_info_entry *entry);
struct proc_dir_entry *p;
struct mutex access;
struct list_head children;
struct list_head list;
};
#if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS)
int snd_info_minor_register(void);
int snd_info_minor_unregister(void);
#else
#define snd_info_minor_register() /* NOP */
#define snd_info_minor_unregister() /* NOP */
#endif
#ifdef CONFIG_PROC_FS
extern struct snd_info_entry *snd_seq_root;
#ifdef CONFIG_SND_OSSEMUL
extern struct snd_info_entry *snd_oss_root;
void snd_card_info_read_oss(struct snd_info_buffer *buffer);
#else
#define snd_oss_root NULL
static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
#endif
__printf(2, 3)
int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...);
int snd_info_init(void);
int snd_info_done(void);
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
const char *snd_info_get_str(char *dest, const char *src, int len);
struct snd_info_entry *snd_info_create_module_entry(struct module *module,
const char *name,
struct snd_info_entry *parent);
struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
const char *name,
struct snd_info_entry *parent);
void snd_info_free_entry(struct snd_info_entry *entry);
int snd_info_store_text(struct snd_info_entry *entry);
int snd_info_restore_text(struct snd_info_entry *entry);
int snd_info_card_create(struct snd_card *card);
int snd_info_card_register(struct snd_card *card);
int snd_info_card_free(struct snd_card *card);
void snd_info_card_disconnect(struct snd_card *card);
void snd_info_card_id_change(struct snd_card *card);
int snd_info_register(struct snd_info_entry *entry);
/* for card drivers */
int snd_card_proc_new(struct snd_card *card, const char *name,
struct snd_info_entry **entryp);
static inline void snd_info_set_text_ops(struct snd_info_entry *entry,
void *private_data,
void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
{
entry->private_data = private_data;
entry->c.text.read = read;
}
int snd_info_check_reserved_words(const char *str);
#else
#define snd_seq_root NULL
#define snd_oss_root NULL
static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; }
static inline int snd_info_init(void) { return 0; }
static inline int snd_info_done(void) { return 0; }
static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; }
static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; }
static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; }
static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; }
static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; }
static inline int snd_info_card_create(struct snd_card *card) { return 0; }
static inline int snd_info_card_register(struct snd_card *card) { return 0; }
static inline int snd_info_card_free(struct snd_card *card) { return 0; }
static inline void snd_info_card_disconnect(struct snd_card *card) { }
static inline void snd_info_card_id_change(struct snd_card *card) { }
static inline int snd_info_register(struct snd_info_entry *entry) { return 0; }
static inline int snd_card_proc_new(struct snd_card *card, const char *name,
struct snd_info_entry **entryp) { return -EINVAL; }
static inline void snd_info_set_text_ops(struct snd_info_entry *entry __attribute__((unused)),
void *private_data,
void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) {}
static inline int snd_info_check_reserved_words(const char *str) { return 1; }
#endif
/*
* OSS info part
*/
#if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS)
#define SNDRV_OSS_INFO_DEV_AUDIO 0
#define SNDRV_OSS_INFO_DEV_SYNTH 1
#define SNDRV_OSS_INFO_DEV_MIDI 2
#define SNDRV_OSS_INFO_DEV_TIMERS 4
#define SNDRV_OSS_INFO_DEV_MIXERS 5
#define SNDRV_OSS_INFO_DEV_COUNT 6
int snd_oss_info_register(int dev, int num, char *string);
#define snd_oss_info_unregister(dev, num) snd_oss_info_register(dev, num, NULL)
#endif /* CONFIG_SND_OSSEMUL && CONFIG_PROC_FS */
#endif /* __SOUND_INFO_H */

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#ifndef __SOUND_INITVAL_H
#define __SOUND_INITVAL_H
/*
* Init values for soundcard modules
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* 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
*
*/
#define SNDRV_AUTO_PORT 1
#define SNDRV_AUTO_IRQ 0xffff
#define SNDRV_AUTO_DMA 0xffff
#define SNDRV_AUTO_DMA_SIZE (0x7fffffff)
#define SNDRV_DEFAULT_IDX1 (-1)
#define SNDRV_DEFAULT_STR1 NULL
#define SNDRV_DEFAULT_ENABLE1 1
#define SNDRV_DEFAULT_PORT1 SNDRV_AUTO_PORT
#define SNDRV_DEFAULT_IRQ1 SNDRV_AUTO_IRQ
#define SNDRV_DEFAULT_DMA1 SNDRV_AUTO_DMA
#define SNDRV_DEFAULT_DMA_SIZE1 SNDRV_AUTO_DMA_SIZE
#define SNDRV_DEFAULT_PTR1 SNDRV_DEFAULT_STR1
#define SNDRV_DEFAULT_IDX { [0 ... (SNDRV_CARDS-1)] = -1 }
#define SNDRV_DEFAULT_STR { [0 ... (SNDRV_CARDS-1)] = NULL }
#define SNDRV_DEFAULT_ENABLE { 1, [1 ... (SNDRV_CARDS-1)] = 0 }
#define SNDRV_DEFAULT_ENABLE_PNP { [0 ... (SNDRV_CARDS-1)] = 1 }
#ifdef CONFIG_PNP
#define SNDRV_DEFAULT_ENABLE_ISAPNP SNDRV_DEFAULT_ENABLE_PNP
#else
#define SNDRV_DEFAULT_ENABLE_ISAPNP SNDRV_DEFAULT_ENABLE
#endif
#define SNDRV_DEFAULT_PORT { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_PORT }
#define SNDRV_DEFAULT_IRQ { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_IRQ }
#define SNDRV_DEFAULT_DMA { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA }
#define SNDRV_DEFAULT_DMA_SIZE { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA_SIZE }
#define SNDRV_DEFAULT_PTR SNDRV_DEFAULT_STR
#ifdef SNDRV_LEGACY_FIND_FREE_IOPORT
static long snd_legacy_find_free_ioport(long *port_table, long size)
{
while (*port_table != -1) {
if (request_region(*port_table, size, "ALSA test")) {
release_region(*port_table, size);
return *port_table;
}
port_table++;
}
return -1;
}
#endif
#ifdef SNDRV_LEGACY_FIND_FREE_IRQ
#include <linux/interrupt.h>
static irqreturn_t snd_legacy_empty_irq_handler(int irq, void *dev_id)
{
return IRQ_HANDLED;
}
static int snd_legacy_find_free_irq(int *irq_table)
{
while (*irq_table != -1) {
if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
IRQF_PROBE_SHARED, "ALSA Test IRQ",
(void *) irq_table)) {
free_irq(*irq_table, (void *) irq_table);
return *irq_table;
}
irq_table++;
}
return -1;
}
#endif
#ifdef SNDRV_LEGACY_FIND_FREE_DMA
static int snd_legacy_find_free_dma(int *dma_table)
{
while (*dma_table != -1) {
if (!request_dma(*dma_table, "ALSA Test DMA")) {
free_dma(*dma_table);
return *dma_table;
}
dma_table++;
}
return -1;
}
#endif
#endif /* __SOUND_INITVAL_H */

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#ifndef __SOUND_JACK_H
#define __SOUND_JACK_H
/*
* Jack abstraction layer
*
* Copyright 2008 Wolfson Microelectronics plc
*
*
* 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/core.h>
struct input_dev;
/**
* Jack types which can be reported. These values are used as a
* bitmask.
*
* Note that this must be kept in sync with the lookup table in
* sound/core/jack.c.
*/
enum snd_jack_types {
SND_JACK_HEADPHONE = 0x0001,
SND_JACK_MICROPHONE = 0x0002,
SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
SND_JACK_LINEOUT = 0x0004,
SND_JACK_MECHANICAL = 0x0008, /* If detected separately */
SND_JACK_VIDEOOUT = 0x0010,
SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT,
SND_JACK_LINEIN = 0x0020,
/* Kept separate from switches to facilitate implementation */
SND_JACK_BTN_0 = 0x4000,
SND_JACK_BTN_1 = 0x2000,
SND_JACK_BTN_2 = 0x1000,
SND_JACK_BTN_3 = 0x0800,
SND_JACK_BTN_4 = 0x0400,
SND_JACK_BTN_5 = 0x0200,
};
/* Keep in sync with definitions above */
#define SND_JACK_SWITCH_TYPES 6
struct snd_jack {
struct input_dev *input_dev;
int registered;
int type;
const char *id;
char name[100];
unsigned int key[6]; /* Keep in sync with definitions above */
void *private_data;
void (*private_free)(struct snd_jack *);
};
#ifdef CONFIG_SND_JACK
int snd_jack_new(struct snd_card *card, const char *id, int type,
struct snd_jack **jack);
void snd_jack_set_parent(struct snd_jack *jack, struct device *parent);
int snd_jack_set_key(struct snd_jack *jack, enum snd_jack_types type,
int keytype);
void snd_jack_report(struct snd_jack *jack, int status);
#else
static inline int snd_jack_new(struct snd_card *card, const char *id, int type,
struct snd_jack **jack)
{
return 0;
}
static inline void snd_jack_set_parent(struct snd_jack *jack,
struct device *parent)
{
}
static inline void snd_jack_report(struct snd_jack *jack, int status)
{
}
#endif
#endif

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#ifndef _L3_H_
#define _L3_H_ 1
struct l3_pins {
void (*setdat)(int);
void (*setclk)(int);
void (*setmode)(int);
int data_hold;
int data_setup;
int clock_high;
int mode_hold;
int mode;
int mode_setup;
};
int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
#endif

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/*
* Platform data for MAX9768
* Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K.
* same licence as the driver
*/
#ifndef __SOUND_MAX9768_PDATA_H__
#define __SOUND_MAX9768_PDATA_H__
/**
* struct max9768_pdata - optional platform specific MAX9768 configuration
* @shdn_gpio: GPIO to SHDN pin. If not valid, pin must be hardwired HIGH
* @mute_gpio: GPIO to MUTE pin. If not valid, control for mute won't be added
* @flags: configuration flags, e.g. set classic PWM mode (check datasheet
* regarding "filterless modulation" which is default).
*/
struct max9768_pdata {
int shdn_gpio;
int mute_gpio;
unsigned flags;
#define MAX9768_FLAG_CLASSIC_PWM (1 << 0)
};
#endif /* __SOUND_MAX9768_PDATA_H__*/

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/*
* Platform data for MAX98088
*
* Copyright 2010 Maxim Integrated Products
*
* 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.
*
*/
#ifndef __SOUND_MAX98088_PDATA_H__
#define __SOUND_MAX98088_PDATA_H__
/* Equalizer filter response configuration */
struct max98088_eq_cfg {
const char *name;
unsigned int rate;
u16 band1[5];
u16 band2[5];
u16 band3[5];
u16 band4[5];
u16 band5[5];
};
/* codec platform data */
struct max98088_pdata {
/* Equalizers for DAI1 and DAI2 */
struct max98088_eq_cfg *eq_cfg;
unsigned int eq_cfgcnt;
/* Receiver output can be configured as power amplifier or LINE out */
/* Set receiver_mode to:
* 0 = amplifier output, or
* 1 = LINE level output
*/
unsigned int receiver_mode:1;
/* Analog/digital microphone configuration:
* 0 = analog microphone input (normal setting)
* 1 = digital microphone input
*/
unsigned int digmic_left_mode:1;
unsigned int digmic_right_mode:1;
};
#endif

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/*
* Platform data for MAX98090
*
* Copyright 2011-2012 Maxim Integrated Products
*
* 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.
*
*/
#ifndef __SOUND_MAX98090_PDATA_H__
#define __SOUND_MAX98090_PDATA_H__
/* codec platform data */
struct max98090_pdata {
/* Analog/digital microphone configuration:
* 0 = analog microphone input (normal setting)
* 1 = digital microphone input
*/
unsigned int digmic_left_mode:1;
unsigned int digmic_right_mode:1;
unsigned int digmic_3_mode:1;
unsigned int digmic_4_mode:1;
};
#endif

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/*
* Platform data for MAX98095
*
* Copyright 2011 Maxim Integrated Products
*
* 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.
*
*/
#ifndef __SOUND_MAX98095_PDATA_H__
#define __SOUND_MAX98095_PDATA_H__
/* Equalizer filter response configuration */
struct max98095_eq_cfg {
const char *name;
unsigned int rate;
u16 band1[5];
u16 band2[5];
u16 band3[5];
u16 band4[5];
u16 band5[5];
};
/* Biquad filter response configuration */
struct max98095_biquad_cfg {
const char *name;
unsigned int rate;
u16 band1[5];
u16 band2[5];
};
/* codec platform data */
struct max98095_pdata {
/* Equalizers for DAI1 and DAI2 */
struct max98095_eq_cfg *eq_cfg;
unsigned int eq_cfgcnt;
/* Biquad filter for DAI1 and DAI2 */
struct max98095_biquad_cfg *bq_cfg;
unsigned int bq_cfgcnt;
/* Analog/digital microphone configuration:
* 0 = analog microphone input (normal setting)
* 1 = digital microphone input
*/
unsigned int digmic_left_mode:1;
unsigned int digmic_right_mode:1;
/* Pin5 is the mechanical method of sensing jack insertion
* but it is something that might not be supported.
* 0 = PIN5 not supported
* 1 = PIN5 supported
*/
unsigned int jack_detect_pin5en:1;
/* Slew amount for jack detection. Calculated as 4 * (delay + 1).
* Default delay is 24 to get a time of 100ms.
*/
unsigned int jack_detect_delay;
};
#endif

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/*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Takashi Iwai <tiwai@suse.de>
*
* Generic memory allocators
*
*
* 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
*
*/
#ifndef __SOUND_MEMALLOC_H
#define __SOUND_MEMALLOC_H
struct device;
/*
* buffer device info
*/
struct snd_dma_device {
int type; /* SNDRV_DMA_TYPE_XXX */
struct device *dev; /* generic device */
};
#ifndef snd_dma_pci_data
#define snd_dma_pci_data(pci) (&(pci)->dev)
#define snd_dma_isa_data() NULL
#define snd_dma_continuous_data(x) ((struct device *)(__force unsigned long)(x))
#endif
/*
* buffer types
*/
#define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */
#define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */
#define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */
#ifdef CONFIG_SND_DMA_SGBUF
#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */
#else
#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */
#endif
/*
* info for buffer allocation
*/
struct snd_dma_buffer {
struct snd_dma_device dev; /* device type */
unsigned char *area; /* virtual pointer */
dma_addr_t addr; /* physical address */
size_t bytes; /* buffer size in bytes */
void *private_data; /* private for allocator; don't touch */
};
#ifdef CONFIG_SND_DMA_SGBUF
/*
* Scatter-Gather generic device pages
*/
void *snd_malloc_sgbuf_pages(struct device *device,
size_t size, struct snd_dma_buffer *dmab,
size_t *res_size);
int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab);
struct snd_sg_page {
void *buf;
dma_addr_t addr;
};
struct snd_sg_buf {
int size; /* allocated byte size */
int pages; /* allocated pages */
int tblsize; /* allocated table size */
struct snd_sg_page *table; /* address table */
struct page **page_table; /* page table (for vmap/vunmap) */
struct device *dev;
};
/*
* return the pages matching with the given byte size
*/
static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
{
return (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
}
/*
* return the physical address at the corresponding offset
*/
static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
addr &= PAGE_MASK;
return addr + offset % PAGE_SIZE;
}
/*
* return the virtual address at the corresponding offset
*/
static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
}
unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
unsigned int ofs, unsigned int size);
#else
/* non-SG versions */
static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
size_t offset)
{
return dmab->addr + offset;
}
static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
size_t offset)
{
return dmab->area + offset;
}
#define snd_sgbuf_get_chunk_size(dmab, ofs, size) (size)
#endif /* CONFIG_SND_DMA_SGBUF */
/* allocate/release a buffer */
int snd_dma_alloc_pages(int type, struct device *dev, size_t size,
struct snd_dma_buffer *dmab);
int snd_dma_alloc_pages_fallback(int type, struct device *dev, size_t size,
struct snd_dma_buffer *dmab);
void snd_dma_free_pages(struct snd_dma_buffer *dmab);
/* buffer-preservation managements */
#define snd_dma_pci_buf_id(pci) (((unsigned int)(pci)->vendor << 16) | (pci)->device)
size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id);
int snd_dma_reserve_buf(struct snd_dma_buffer *dmab, unsigned int id);
/* basic memory allocation functions */
void *snd_malloc_pages(size_t size, gfp_t gfp_flags);
void snd_free_pages(void *ptr, size_t size);
#endif /* __SOUND_MEMALLOC_H */

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#ifndef __SOUND_MINORS_H
#define __SOUND_MINORS_H
/*
* MINOR numbers
*
*
* 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
*
*/
#define SNDRV_OS_MINORS 256
#define SNDRV_MINOR_DEVICES 32
#define SNDRV_MINOR_CARD(minor) ((minor) >> 5)
#define SNDRV_MINOR_DEVICE(minor) ((minor) & 0x001f)
#define SNDRV_MINOR(card, dev) (((card) << 5) | (dev))
/* these minors can still be used for autoloading devices (/dev/aload*) */
#define SNDRV_MINOR_CONTROL 0 /* 0 */
#define SNDRV_MINOR_GLOBAL 1 /* 1 */
#define SNDRV_MINOR_SEQUENCER 1 /* SNDRV_MINOR_GLOBAL + 0 * 32 */
#define SNDRV_MINOR_TIMER 33 /* SNDRV_MINOR_GLOBAL + 1 * 32 */
#ifndef CONFIG_SND_DYNAMIC_MINORS
#define SNDRV_MINOR_COMPRESS 2 /* 2 - 3 */
#define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */
#define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */
#define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */
#define SNDRV_MINOR_PCM_CAPTURE 24 /* 24 - 31 */
/* same as first respective minor number to make minor allocation easier */
#define SNDRV_DEVICE_TYPE_CONTROL SNDRV_MINOR_CONTROL
#define SNDRV_DEVICE_TYPE_HWDEP SNDRV_MINOR_HWDEP
#define SNDRV_DEVICE_TYPE_RAWMIDI SNDRV_MINOR_RAWMIDI
#define SNDRV_DEVICE_TYPE_PCM_PLAYBACK SNDRV_MINOR_PCM_PLAYBACK
#define SNDRV_DEVICE_TYPE_PCM_CAPTURE SNDRV_MINOR_PCM_CAPTURE
#define SNDRV_DEVICE_TYPE_SEQUENCER SNDRV_MINOR_SEQUENCER
#define SNDRV_DEVICE_TYPE_TIMER SNDRV_MINOR_TIMER
#define SNDRV_DEVICE_TYPE_COMPRESS SNDRV_MINOR_COMPRESS
#else /* CONFIG_SND_DYNAMIC_MINORS */
enum {
SNDRV_DEVICE_TYPE_CONTROL,
SNDRV_DEVICE_TYPE_SEQUENCER,
SNDRV_DEVICE_TYPE_TIMER,
SNDRV_DEVICE_TYPE_HWDEP,
SNDRV_DEVICE_TYPE_RAWMIDI,
SNDRV_DEVICE_TYPE_PCM_PLAYBACK,
SNDRV_DEVICE_TYPE_PCM_CAPTURE,
SNDRV_DEVICE_TYPE_COMPRESS,
};
#endif /* CONFIG_SND_DYNAMIC_MINORS */
#define SNDRV_MINOR_HWDEPS 4
#define SNDRV_MINOR_RAWMIDIS 8
#define SNDRV_MINOR_PCMS 8
#ifdef CONFIG_SND_OSSEMUL
#define SNDRV_MINOR_OSS_DEVICES 16
#define SNDRV_MINOR_OSS_CARD(minor) ((minor) >> 4)
#define SNDRV_MINOR_OSS_DEVICE(minor) ((minor) & 0x000f)
#define SNDRV_MINOR_OSS(card, dev) (((card) << 4) | (dev))
#define SNDRV_MINOR_OSS_MIXER 0 /* /dev/mixer - OSS 3.XX compatible */
#define SNDRV_MINOR_OSS_SEQUENCER 1 /* /dev/sequencer - OSS 3.XX compatible */
#define SNDRV_MINOR_OSS_MIDI 2 /* /dev/midi - native midi interface - OSS 3.XX compatible - UART */
#define SNDRV_MINOR_OSS_PCM 3 /* alias */
#define SNDRV_MINOR_OSS_PCM_8 3 /* /dev/dsp - 8bit PCM - OSS 3.XX compatible */
#define SNDRV_MINOR_OSS_AUDIO 4 /* /dev/audio - SunSparc compatible */
#define SNDRV_MINOR_OSS_PCM_16 5 /* /dev/dsp16 - 16bit PCM - OSS 3.XX compatible */
#define SNDRV_MINOR_OSS_SNDSTAT 6 /* /dev/sndstat - for compatibility with OSS */
#define SNDRV_MINOR_OSS_RESERVED7 7 /* reserved for future use */
#define SNDRV_MINOR_OSS_MUSIC 8 /* /dev/music - OSS 3.XX compatible */
#define SNDRV_MINOR_OSS_DMMIDI 9 /* /dev/dmmidi0 - this device can have another minor # with OSS */
#define SNDRV_MINOR_OSS_DMFM 10 /* /dev/dmfm0 - this device can have another minor # with OSS */
#define SNDRV_MINOR_OSS_MIXER1 11 /* alternate mixer */
#define SNDRV_MINOR_OSS_PCM1 12 /* alternate PCM (GF-A-1) */
#define SNDRV_MINOR_OSS_MIDI1 13 /* alternate midi - SYNTH */
#define SNDRV_MINOR_OSS_DMMIDI1 14 /* alternate dmmidi - SYNTH */
#define SNDRV_MINOR_OSS_RESERVED15 15 /* reserved for future use */
#define SNDRV_OSS_DEVICE_TYPE_MIXER 0
#define SNDRV_OSS_DEVICE_TYPE_SEQUENCER 1
#define SNDRV_OSS_DEVICE_TYPE_PCM 2
#define SNDRV_OSS_DEVICE_TYPE_MIDI 3
#define SNDRV_OSS_DEVICE_TYPE_DMFM 4
#define SNDRV_OSS_DEVICE_TYPE_SNDSTAT 5
#define SNDRV_OSS_DEVICE_TYPE_MUSIC 6
#define MODULE_ALIAS_SNDRV_MINOR(type) \
MODULE_ALIAS("sound-service-?-" __stringify(type))
#endif
#endif /* __SOUND_MINORS_H */

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#ifndef __SOUND_MIXER_OSS_H
#define __SOUND_MIXER_OSS_H
/*
* OSS MIXER API
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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
*
*/
#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
#define SNDRV_OSS_MAX_MIXERS 32
struct snd_mixer_oss_file;
struct snd_mixer_oss_slot {
int number;
unsigned int stereo: 1;
int (*get_volume)(struct snd_mixer_oss_file *fmixer,
struct snd_mixer_oss_slot *chn,
int *left, int *right);
int (*put_volume)(struct snd_mixer_oss_file *fmixer,
struct snd_mixer_oss_slot *chn,
int left, int right);
int (*get_recsrc)(struct snd_mixer_oss_file *fmixer,
struct snd_mixer_oss_slot *chn,
int *active);
int (*put_recsrc)(struct snd_mixer_oss_file *fmixer,
struct snd_mixer_oss_slot *chn,
int active);
unsigned long private_value;
void *private_data;
void (*private_free)(struct snd_mixer_oss_slot *slot);
int volume[2];
};
struct snd_mixer_oss {
struct snd_card *card;
char id[16];
char name[32];
struct snd_mixer_oss_slot slots[SNDRV_OSS_MAX_MIXERS]; /* OSS mixer slots */
unsigned int mask_recsrc; /* exclusive recsrc mask */
int (*get_recsrc)(struct snd_mixer_oss_file *fmixer,
unsigned int *active_index);
int (*put_recsrc)(struct snd_mixer_oss_file *fmixer,
unsigned int active_index);
void *private_data_recsrc;
void (*private_free_recsrc)(struct snd_mixer_oss *mixer);
struct mutex reg_mutex;
struct snd_info_entry *proc_entry;
int oss_dev_alloc;
/* --- */
int oss_recsrc;
};
struct snd_mixer_oss_file {
struct snd_card *card;
struct snd_mixer_oss *mixer;
};
int snd_mixer_oss_ioctl_card(struct snd_card *card,
unsigned int cmd, unsigned long arg);
#endif /* CONFIG_SND_MIXER_OSS */
#endif /* __SOUND_MIXER_OSS_H */

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#ifndef __SOUND_MPU401_H
#define __SOUND_MPU401_H
/*
* Header file for MPU-401 and compatible cards
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/rawmidi.h>
#include <linux/interrupt.h>
#define MPU401_HW_MPU401 1 /* native MPU401 */
#define MPU401_HW_SB 2 /* SoundBlaster MPU-401 UART */
#define MPU401_HW_ES1688 3 /* AudioDrive ES1688 MPU-401 UART */
#define MPU401_HW_OPL3SA2 4 /* Yamaha OPL3-SA2 */
#define MPU401_HW_SONICVIBES 5 /* S3 SonicVibes */
#define MPU401_HW_CS4232 6 /* CS4232 */
#define MPU401_HW_ES18XX 7 /* AudioDrive ES18XX MPU-401 UART */
#define MPU401_HW_FM801 8 /* ForteMedia FM801 */
#define MPU401_HW_TRID4DWAVE 9 /* Trident 4DWave */
#define MPU401_HW_AZT2320 10 /* Aztech AZT2320 */
#define MPU401_HW_ALS100 11 /* Avance Logic ALS100 */
#define MPU401_HW_ICE1712 12 /* Envy24 */
#define MPU401_HW_VIA686A 13 /* VIA 82C686A */
#define MPU401_HW_YMFPCI 14 /* YMF DS-XG PCI */
#define MPU401_HW_CMIPCI 15 /* CMIPCI MPU-401 UART */
#define MPU401_HW_ALS4000 16 /* Avance Logic ALS4000 */
#define MPU401_HW_INTEL8X0 17 /* Intel8x0 driver */
#define MPU401_HW_PC98II 18 /* Roland PC98II */
#define MPU401_HW_AUREAL 19 /* Aureal Vortex */
#define MPU401_INFO_INPUT (1 << 0) /* input stream */
#define MPU401_INFO_OUTPUT (1 << 1) /* output stream */
#define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */
#define MPU401_INFO_MMIO (1 << 3) /* MMIO access */
#define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */
#define MPU401_INFO_IRQ_HOOK (1 << 5) /* mpu401 irq handler is called
from driver irq handler */
#define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */
#define MPU401_INFO_USE_TIMER (1 << 15) /* internal */
#define MPU401_MODE_BIT_INPUT 0
#define MPU401_MODE_BIT_OUTPUT 1
#define MPU401_MODE_BIT_INPUT_TRIGGER 2
#define MPU401_MODE_BIT_OUTPUT_TRIGGER 3
#define MPU401_MODE_INPUT (1<<MPU401_MODE_BIT_INPUT)
#define MPU401_MODE_OUTPUT (1<<MPU401_MODE_BIT_OUTPUT)
#define MPU401_MODE_INPUT_TRIGGER (1<<MPU401_MODE_BIT_INPUT_TRIGGER)
#define MPU401_MODE_OUTPUT_TRIGGER (1<<MPU401_MODE_BIT_OUTPUT_TRIGGER)
#define MPU401_MODE_INPUT_TIMER (1<<0)
#define MPU401_MODE_OUTPUT_TIMER (1<<1)
struct snd_mpu401 {
struct snd_rawmidi *rmidi;
unsigned short hardware; /* MPU401_HW_XXXX */
unsigned int info_flags; /* MPU401_INFO_XXX */
unsigned long port; /* base port of MPU-401 chip */
unsigned long cport; /* port + 1 (usually) */
struct resource *res; /* port resource */
int irq; /* IRQ number of MPU-401 chip */
unsigned long mode; /* MPU401_MODE_XXXX */
int timer_invoked;
int (*open_input) (struct snd_mpu401 * mpu);
void (*close_input) (struct snd_mpu401 * mpu);
int (*open_output) (struct snd_mpu401 * mpu);
void (*close_output) (struct snd_mpu401 * mpu);
void *private_data;
struct snd_rawmidi_substream *substream_input;
struct snd_rawmidi_substream *substream_output;
spinlock_t input_lock;
spinlock_t output_lock;
spinlock_t timer_lock;
struct timer_list timer;
void (*write) (struct snd_mpu401 * mpu, unsigned char data, unsigned long addr);
unsigned char (*read) (struct snd_mpu401 *mpu, unsigned long addr);
};
/* I/O ports */
#define MPU401C(mpu) (mpu)->cport
#define MPU401D(mpu) (mpu)->port
/*
* control register bits
*/
/* read MPU401C() */
#define MPU401_RX_EMPTY 0x80
#define MPU401_TX_FULL 0x40
/* write MPU401C() */
#define MPU401_RESET 0xff
#define MPU401_ENTER_UART 0x3f
/* read MPU401D() */
#define MPU401_ACK 0xfe
/*
*/
irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id);
irqreturn_t snd_mpu401_uart_interrupt_tx(int irq, void *dev_id);
int snd_mpu401_uart_new(struct snd_card *card,
int device,
unsigned short hardware,
unsigned long port,
unsigned int info_flags,
int irq,
struct snd_rawmidi ** rrawmidi);
#endif /* __SOUND_MPU401_H */

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#ifndef __SOUND_OPL3_H
#define __SOUND_OPL3_H
/*
* Definitions of the OPL-3 registers.
*
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Hannu Savolainen 1993-1996
*
*
* The OPL-3 mode is switched on by writing 0x01, to the offset 5
* of the right side.
*
* Another special register at the right side is at offset 4. It contains
* a bit mask defining which voices are used as 4 OP voices.
*
* The percussive mode is implemented in the left side only.
*
* With the above exceptions the both sides can be operated independently.
*
* A 4 OP voice can be created by setting the corresponding
* bit at offset 4 of the right side.
*
* For example setting the rightmost bit (0x01) changes the
* first voice on the right side to the 4 OP mode. The fourth
* voice is made inaccessible.
*
* If a voice is set to the 2 OP mode, it works like 2 OP modes
* of the original YM3812 (AdLib). In addition the voice can
* be connected the left, right or both stereo channels. It can
* even be left unconnected. This works with 4 OP voices also.
*
* The stereo connection bits are located in the FEEDBACK_CONNECTION
* register of the voice (0xC0-0xC8). In 4 OP voices these bits are
* in the second half of the voice.
*
*
* 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/core.h>
#include <sound/hwdep.h>
#include <sound/timer.h>
#include <sound/seq_midi_emul.h>
#ifdef CONFIG_SND_SEQUENCER_OSS
#include <sound/seq_oss.h>
#include <sound/seq_oss_legacy.h>
#endif
#include <sound/seq_device.h>
#include <sound/asound_fm.h>
/*
* Register numbers for the global registers
*/
#define OPL3_REG_TEST 0x01
#define OPL3_ENABLE_WAVE_SELECT 0x20
#define OPL3_REG_TIMER1 0x02
#define OPL3_REG_TIMER2 0x03
#define OPL3_REG_TIMER_CONTROL 0x04 /* Left side */
#define OPL3_IRQ_RESET 0x80
#define OPL3_TIMER1_MASK 0x40
#define OPL3_TIMER2_MASK 0x20
#define OPL3_TIMER1_START 0x01
#define OPL3_TIMER2_START 0x02
#define OPL3_REG_CONNECTION_SELECT 0x04 /* Right side */
#define OPL3_LEFT_4OP_0 0x01
#define OPL3_LEFT_4OP_1 0x02
#define OPL3_LEFT_4OP_2 0x04
#define OPL3_RIGHT_4OP_0 0x08
#define OPL3_RIGHT_4OP_1 0x10
#define OPL3_RIGHT_4OP_2 0x20
#define OPL3_REG_MODE 0x05 /* Right side */
#define OPL3_OPL3_ENABLE 0x01 /* OPL3 mode */
#define OPL3_OPL4_ENABLE 0x02 /* OPL4 mode */
#define OPL3_REG_KBD_SPLIT 0x08 /* Left side */
#define OPL3_COMPOSITE_SINE_WAVE_MODE 0x80 /* Don't use with OPL-3? */
#define OPL3_KEYBOARD_SPLIT 0x40
#define OPL3_REG_PERCUSSION 0xbd /* Left side only */
#define OPL3_TREMOLO_DEPTH 0x80
#define OPL3_VIBRATO_DEPTH 0x40
#define OPL3_PERCUSSION_ENABLE 0x20
#define OPL3_BASSDRUM_ON 0x10
#define OPL3_SNAREDRUM_ON 0x08
#define OPL3_TOMTOM_ON 0x04
#define OPL3_CYMBAL_ON 0x02
#define OPL3_HIHAT_ON 0x01
/*
* Offsets to the register banks for operators. To get the
* register number just add the operator offset to the bank offset
*
* AM/VIB/EG/KSR/Multiple (0x20 to 0x35)
*/
#define OPL3_REG_AM_VIB 0x20
#define OPL3_TREMOLO_ON 0x80
#define OPL3_VIBRATO_ON 0x40
#define OPL3_SUSTAIN_ON 0x20
#define OPL3_KSR 0x10 /* Key scaling rate */
#define OPL3_MULTIPLE_MASK 0x0f /* Frequency multiplier */
/*
* KSL/Total level (0x40 to 0x55)
*/
#define OPL3_REG_KSL_LEVEL 0x40
#define OPL3_KSL_MASK 0xc0 /* Envelope scaling bits */
#define OPL3_TOTAL_LEVEL_MASK 0x3f /* Strength (volume) of OP */
/*
* Attack / Decay rate (0x60 to 0x75)
*/
#define OPL3_REG_ATTACK_DECAY 0x60
#define OPL3_ATTACK_MASK 0xf0
#define OPL3_DECAY_MASK 0x0f
/*
* Sustain level / Release rate (0x80 to 0x95)
*/
#define OPL3_REG_SUSTAIN_RELEASE 0x80
#define OPL3_SUSTAIN_MASK 0xf0
#define OPL3_RELEASE_MASK 0x0f
/*
* Wave select (0xE0 to 0xF5)
*/
#define OPL3_REG_WAVE_SELECT 0xe0
#define OPL3_WAVE_SELECT_MASK 0x07
/*
* Offsets to the register banks for voices. Just add to the
* voice number to get the register number.
*
* F-Number low bits (0xA0 to 0xA8).
*/
#define OPL3_REG_FNUM_LOW 0xa0
/*
* F-number high bits / Key on / Block (octave) (0xB0 to 0xB8)
*/
#define OPL3_REG_KEYON_BLOCK 0xb0
#define OPL3_KEYON_BIT 0x20
#define OPL3_BLOCKNUM_MASK 0x1c
#define OPL3_FNUM_HIGH_MASK 0x03
/*
* Feedback / Connection (0xc0 to 0xc8)
*
* These registers have two new bits when the OPL-3 mode
* is selected. These bits controls connecting the voice
* to the stereo channels. For 4 OP voices this bit is
* defined in the second half of the voice (add 3 to the
* register offset).
*
* For 4 OP voices the connection bit is used in the
* both halves (gives 4 ways to connect the operators).
*/
#define OPL3_REG_FEEDBACK_CONNECTION 0xc0
#define OPL3_FEEDBACK_MASK 0x0e /* Valid just for 1st OP of a voice */
#define OPL3_CONNECTION_BIT 0x01
/*
* In the 4 OP mode there is four possible configurations how the
* operators can be connected together (in 2 OP modes there is just
* AM or FM). The 4 OP connection mode is defined by the rightmost
* bit of the FEEDBACK_CONNECTION (0xC0-0xC8) on the both halves.
*
* First half Second half Mode
*
* +---+
* v |
* 0 0 >+-1-+--2--3--4-->
*
*
*
* +---+
* | |
* 0 1 >+-1-+--2-+
* |->
* >--3----4-+
*
* +---+
* | |
* 1 0 >+-1-+-----+
* |->
* >--2--3--4-+
*
* +---+
* | |
* 1 1 >+-1-+--+
* |
* >--2--3-+->
* |
* >--4----+
*/
#define OPL3_STEREO_BITS 0x30 /* OPL-3 only */
#define OPL3_VOICE_TO_LEFT 0x10
#define OPL3_VOICE_TO_RIGHT 0x20
/*
*/
#define OPL3_LEFT 0x0000
#define OPL3_RIGHT 0x0100
#define OPL3_HW_AUTO 0x0000
#define OPL3_HW_OPL2 0x0200
#define OPL3_HW_OPL3 0x0300
#define OPL3_HW_OPL3_SV 0x0301 /* S3 SonicVibes */
#define OPL3_HW_OPL3_CS 0x0302 /* CS4232/CS4236+ */
#define OPL3_HW_OPL3_FM801 0x0303 /* FM801 */
#define OPL3_HW_OPL3_CS4281 0x0304 /* CS4281 */
#define OPL3_HW_OPL4 0x0400 /* YMF278B/YMF295 */
#define OPL3_HW_OPL4_ML 0x0401 /* YMF704/YMF721 */
#define OPL3_HW_MASK 0xff00
#define MAX_OPL2_VOICES 9
#define MAX_OPL3_VOICES 18
struct snd_opl3;
/*
* Instrument record, aka "Patch"
*/
/* FM operator */
struct fm_operator {
unsigned char am_vib;
unsigned char ksl_level;
unsigned char attack_decay;
unsigned char sustain_release;
unsigned char wave_select;
} __attribute__((packed));
/* Instrument data */
struct fm_instrument {
struct fm_operator op[4];
unsigned char feedback_connection[2];
unsigned char echo_delay;
unsigned char echo_atten;
unsigned char chorus_spread;
unsigned char trnsps;
unsigned char fix_dur;
unsigned char modes;
unsigned char fix_key;
};
/* type */
#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */
#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */
/* Instrument record */
struct fm_patch {
unsigned char prog;
unsigned char bank;
unsigned char type;
struct fm_instrument inst;
char name[24];
struct fm_patch *next;
};
/*
* A structure to keep track of each hardware voice
*/
struct snd_opl3_voice {
int state; /* status */
#define SNDRV_OPL3_ST_OFF 0 /* Not playing */
#define SNDRV_OPL3_ST_ON_2OP 1 /* 2op voice is allocated */
#define SNDRV_OPL3_ST_ON_4OP 2 /* 4op voice is allocated */
#define SNDRV_OPL3_ST_NOT_AVAIL -1 /* voice is not available */
unsigned int time; /* An allocation time */
unsigned char note; /* Note currently assigned to this voice */
unsigned long note_off; /* note-off time */
int note_off_check; /* check note-off time */
unsigned char keyon_reg; /* KON register shadow */
struct snd_midi_channel *chan; /* Midi channel for this note */
};
struct snd_opl3 {
unsigned long l_port;
unsigned long r_port;
struct resource *res_l_port;
struct resource *res_r_port;
unsigned short hardware;
/* hardware access */
void (*command) (struct snd_opl3 * opl3, unsigned short cmd, unsigned char val);
unsigned short timer_enable;
int seq_dev_num; /* sequencer device number */
struct snd_timer *timer1;
struct snd_timer *timer2;
spinlock_t timer_lock;
void *private_data;
void (*private_free)(struct snd_opl3 *);
struct snd_hwdep *hwdep;
spinlock_t reg_lock;
struct snd_card *card; /* The card that this belongs to */
unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */
unsigned char rhythm; /* percussion mode flag */
unsigned char max_voices; /* max number of voices */
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
#define SNDRV_OPL3_MODE_SYNTH 0 /* OSS - voices allocated by application */
#define SNDRV_OPL3_MODE_SEQ 1 /* ALSA - driver handles voice allocation */
int synth_mode; /* synth mode */
int seq_client;
struct snd_seq_device *seq_dev; /* sequencer device */
struct snd_midi_channel_set * chset;
#ifdef CONFIG_SND_SEQUENCER_OSS
struct snd_seq_device *oss_seq_dev; /* OSS sequencer device */
struct snd_midi_channel_set * oss_chset;
#endif
#define OPL3_PATCH_HASH_SIZE 32
struct fm_patch *patch_table[OPL3_PATCH_HASH_SIZE];
struct snd_opl3_voice voices[MAX_OPL3_VOICES]; /* Voices (OPL3 'channel') */
int use_time; /* allocation counter */
unsigned short connection_reg; /* connection reg shadow */
unsigned char drum_reg; /* percussion reg shadow */
spinlock_t voice_lock; /* Lock for voice access */
struct timer_list tlist; /* timer for note-offs and effects */
int sys_timer_status; /* system timer run status */
spinlock_t sys_timer_lock; /* Lock for system timer access */
#endif
};
/* opl3.c */
void snd_opl3_interrupt(struct snd_hwdep * hw);
int snd_opl3_new(struct snd_card *card, unsigned short hardware,
struct snd_opl3 **ropl3);
int snd_opl3_init(struct snd_opl3 *opl3);
int snd_opl3_create(struct snd_card *card,
unsigned long l_port, unsigned long r_port,
unsigned short hardware,
int integrated,
struct snd_opl3 ** opl3);
int snd_opl3_timer_new(struct snd_opl3 * opl3, int timer1_dev, int timer2_dev);
int snd_opl3_hwdep_new(struct snd_opl3 * opl3, int device, int seq_device,
struct snd_hwdep ** rhwdep);
/* opl3_synth */
int snd_opl3_open(struct snd_hwdep * hw, struct file *file);
int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
unsigned int cmd, unsigned long arg);
int snd_opl3_release(struct snd_hwdep * hw, struct file *file);
void snd_opl3_reset(struct snd_opl3 * opl3);
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count,
loff_t *offset);
int snd_opl3_load_patch(struct snd_opl3 *opl3,
int prog, int bank, int type,
const char *name,
const unsigned char *ext,
const unsigned char *data);
struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank,
int create_patch);
void snd_opl3_clear_patches(struct snd_opl3 *opl3);
#else
#define snd_opl3_write NULL
static inline void snd_opl3_clear_patches(struct snd_opl3 *opl3) {}
#endif
#endif /* __SOUND_OPL3_H */

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#ifndef __SOUND_OPL4_H
#define __SOUND_OPL4_H
/*
* Global definitions for the OPL4 driver
* Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
*
* 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/opl3.h>
struct snd_opl4;
extern int snd_opl4_create(struct snd_card *card,
unsigned long fm_port, unsigned long pcm_port,
int seq_device,
struct snd_opl3 **opl3, struct snd_opl4 **opl4);
#endif /* __SOUND_OPL4_H */

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/*
* Helper functions for indirect PCM data transfer
*
* Copyright (c) by Takashi Iwai <tiwai@suse.de>
* Jaroslav Kysela <perex@perex.cz>
*
* 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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __SOUND_PCM_INDIRECT_H
#define __SOUND_PCM_INDIRECT_H
#include <sound/pcm.h>
struct snd_pcm_indirect {
unsigned int hw_buffer_size; /* Byte size of hardware buffer */
unsigned int hw_queue_size; /* Max queue size of hw buffer (0 = buffer size) */
unsigned int hw_data; /* Offset to next dst (or src) in hw ring buffer */
unsigned int hw_io; /* Ring buffer hw pointer */
int hw_ready; /* Bytes ready for play (or captured) in hw ring buffer */
unsigned int sw_buffer_size; /* Byte size of software buffer */
unsigned int sw_data; /* Offset to next dst (or src) in sw ring buffer */
unsigned int sw_io; /* Current software pointer in bytes */
int sw_ready; /* Bytes ready to be transferred to/from hw */
snd_pcm_uframes_t appl_ptr; /* Last seen appl_ptr */
};
typedef void (*snd_pcm_indirect_copy_t)(struct snd_pcm_substream *substream,
struct snd_pcm_indirect *rec, size_t bytes);
/*
* helper function for playback ack callback
*/
static inline void
snd_pcm_indirect_playback_transfer(struct snd_pcm_substream *substream,
struct snd_pcm_indirect *rec,
snd_pcm_indirect_copy_t copy)
{
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr;
snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr;
int qsize;
if (diff) {
if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
diff += runtime->boundary;
rec->sw_ready += (int)frames_to_bytes(runtime, diff);
rec->appl_ptr = appl_ptr;
}
qsize = rec->hw_queue_size ? rec->hw_queue_size : rec->hw_buffer_size;
while (rec->hw_ready < qsize && rec->sw_ready > 0) {
unsigned int hw_to_end = rec->hw_buffer_size - rec->hw_data;
unsigned int sw_to_end = rec->sw_buffer_size - rec->sw_data;
unsigned int bytes = qsize - rec->hw_ready;
if (rec->sw_ready < (int)bytes)
bytes = rec->sw_ready;
if (hw_to_end < bytes)
bytes = hw_to_end;
if (sw_to_end < bytes)
bytes = sw_to_end;
if (! bytes)
break;
copy(substream, rec, bytes);
rec->hw_data += bytes;
if (rec->hw_data == rec->hw_buffer_size)
rec->hw_data = 0;
rec->sw_data += bytes;
if (rec->sw_data == rec->sw_buffer_size)
rec->sw_data = 0;
rec->hw_ready += bytes;
rec->sw_ready -= bytes;
}
}
/*
* helper function for playback pointer callback
* ptr = current byte pointer
*/
static inline snd_pcm_uframes_t
snd_pcm_indirect_playback_pointer(struct snd_pcm_substream *substream,
struct snd_pcm_indirect *rec, unsigned int ptr)
{
int bytes = ptr - rec->hw_io;
if (bytes < 0)
bytes += rec->hw_buffer_size;
rec->hw_io = ptr;
rec->hw_ready -= bytes;
rec->sw_io += bytes;
if (rec->sw_io >= rec->sw_buffer_size)
rec->sw_io -= rec->sw_buffer_size;
if (substream->ops->ack)
substream->ops->ack(substream);
return bytes_to_frames(substream->runtime, rec->sw_io);
}
/*
* helper function for capture ack callback
*/
static inline void
snd_pcm_indirect_capture_transfer(struct snd_pcm_substream *substream,
struct snd_pcm_indirect *rec,
snd_pcm_indirect_copy_t copy)
{
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr;
snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr;
if (diff) {
if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
diff += runtime->boundary;
rec->sw_ready -= frames_to_bytes(runtime, diff);
rec->appl_ptr = appl_ptr;
}
while (rec->hw_ready > 0 &&
rec->sw_ready < (int)rec->sw_buffer_size) {
size_t hw_to_end = rec->hw_buffer_size - rec->hw_data;
size_t sw_to_end = rec->sw_buffer_size - rec->sw_data;
size_t bytes = rec->sw_buffer_size - rec->sw_ready;
if (rec->hw_ready < (int)bytes)
bytes = rec->hw_ready;
if (hw_to_end < bytes)
bytes = hw_to_end;
if (sw_to_end < bytes)
bytes = sw_to_end;
if (! bytes)
break;
copy(substream, rec, bytes);
rec->hw_data += bytes;
if ((int)rec->hw_data == rec->hw_buffer_size)
rec->hw_data = 0;
rec->sw_data += bytes;
if (rec->sw_data == rec->sw_buffer_size)
rec->sw_data = 0;
rec->hw_ready -= bytes;
rec->sw_ready += bytes;
}
}
/*
* helper function for capture pointer callback,
* ptr = current byte pointer
*/
static inline snd_pcm_uframes_t
snd_pcm_indirect_capture_pointer(struct snd_pcm_substream *substream,
struct snd_pcm_indirect *rec, unsigned int ptr)
{
int qsize;
int bytes = ptr - rec->hw_io;
if (bytes < 0)
bytes += rec->hw_buffer_size;
rec->hw_io = ptr;
rec->hw_ready += bytes;
qsize = rec->hw_queue_size ? rec->hw_queue_size : rec->hw_buffer_size;
if (rec->hw_ready > qsize)
return SNDRV_PCM_POS_XRUN;
rec->sw_io += bytes;
if (rec->sw_io >= rec->sw_buffer_size)
rec->sw_io -= rec->sw_buffer_size;
if (substream->ops->ack)
substream->ops->ack(substream);
return bytes_to_frames(substream->runtime, rec->sw_io);
}
#endif /* __SOUND_PCM_INDIRECT_H */

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#ifndef __SOUND_PCM_OSS_H
#define __SOUND_PCM_OSS_H
/*
* Digital Audio (PCM) - OSS compatibility abstract layer
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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
*
*/
struct snd_pcm_oss_setup {
char *task_name;
unsigned int disable:1,
direct:1,
block:1,
nonblock:1,
partialfrag:1,
nosilence:1,
buggyptr:1;
unsigned int periods;
unsigned int period_size;
struct snd_pcm_oss_setup *next;
};
struct snd_pcm_oss_runtime {
unsigned params: 1, /* format/parameter change */
prepare: 1, /* need to prepare the operation */
trigger: 1, /* trigger flag */
sync_trigger: 1; /* sync trigger flag */
int rate; /* requested rate */
int format; /* requested OSS format */
unsigned int channels; /* requested channels */
unsigned int fragshift;
unsigned int maxfrags;
unsigned int subdivision; /* requested subdivision */
size_t period_bytes; /* requested period size */
size_t period_frames; /* period frames for poll */
size_t period_ptr; /* actual write pointer to period */
unsigned int periods;
size_t buffer_bytes; /* requested buffer size */
size_t bytes; /* total # bytes processed */
size_t mmap_bytes;
char *buffer; /* vmallocated period */
size_t buffer_used; /* used length from period buffer */
struct mutex params_lock;
#ifdef CONFIG_SND_PCM_OSS_PLUGINS
struct snd_pcm_plugin *plugin_first;
struct snd_pcm_plugin *plugin_last;
#endif
unsigned int prev_hw_ptr_period;
};
struct snd_pcm_oss_file {
struct snd_pcm_substream *streams[2];
};
struct snd_pcm_oss_substream {
unsigned oss: 1; /* oss mode */
struct snd_pcm_oss_setup setup; /* active setup */
};
struct snd_pcm_oss_stream {
struct snd_pcm_oss_setup *setup_list; /* setup list */
struct mutex setup_mutex;
#ifdef CONFIG_SND_VERBOSE_PROCFS
struct snd_info_entry *proc_entry;
#endif
};
struct snd_pcm_oss {
int reg;
unsigned int reg_mask;
};
#endif /* __SOUND_PCM_OSS_H */

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#ifndef __SOUND_PCM_PARAMS_H
#define __SOUND_PCM_PARAMS_H
/*
* PCM params helpers
* Copyright (c) by Abramo Bagnara <abramo@alsa-project.org>
*
*
* 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/pcm.h>
int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm,
struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var, int *dir);
int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var, int *dir);
int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var, int *dir);
#define SNDRV_MASK_BITS 64 /* we use so far 64bits only */
#define SNDRV_MASK_SIZE (SNDRV_MASK_BITS / 32)
#define MASK_OFS(i) ((i) >> 5)
#define MASK_BIT(i) (1U << ((i) & 31))
static inline unsigned int ld2(u_int32_t v)
{
unsigned r = 0;
if (v >= 0x10000) {
v >>= 16;
r += 16;
}
if (v >= 0x100) {
v >>= 8;
r += 8;
}
if (v >= 0x10) {
v >>= 4;
r += 4;
}
if (v >= 4) {
v >>= 2;
r += 2;
}
if (v >= 2)
r++;
return r;
}
static inline size_t snd_mask_sizeof(void)
{
return sizeof(struct snd_mask);
}
static inline void snd_mask_none(struct snd_mask *mask)
{
memset(mask, 0, sizeof(*mask));
}
static inline void snd_mask_any(struct snd_mask *mask)
{
memset(mask, 0xff, SNDRV_MASK_SIZE * sizeof(u_int32_t));
}
static inline int snd_mask_empty(const struct snd_mask *mask)
{
int i;
for (i = 0; i < SNDRV_MASK_SIZE; i++)
if (mask->bits[i])
return 0;
return 1;
}
static inline unsigned int snd_mask_min(const struct snd_mask *mask)
{
int i;
for (i = 0; i < SNDRV_MASK_SIZE; i++) {
if (mask->bits[i])
return ffs(mask->bits[i]) - 1 + (i << 5);
}
return 0;
}
static inline unsigned int snd_mask_max(const struct snd_mask *mask)
{
int i;
for (i = SNDRV_MASK_SIZE - 1; i >= 0; i--) {
if (mask->bits[i])
return ld2(mask->bits[i]) + (i << 5);
}
return 0;
}
static inline void snd_mask_set(struct snd_mask *mask, unsigned int val)
{
mask->bits[MASK_OFS(val)] |= MASK_BIT(val);
}
static inline void snd_mask_reset(struct snd_mask *mask, unsigned int val)
{
mask->bits[MASK_OFS(val)] &= ~MASK_BIT(val);
}
static inline void snd_mask_set_range(struct snd_mask *mask,
unsigned int from, unsigned int to)
{
unsigned int i;
for (i = from; i <= to; i++)
mask->bits[MASK_OFS(i)] |= MASK_BIT(i);
}
static inline void snd_mask_reset_range(struct snd_mask *mask,
unsigned int from, unsigned int to)
{
unsigned int i;
for (i = from; i <= to; i++)
mask->bits[MASK_OFS(i)] &= ~MASK_BIT(i);
}
static inline void snd_mask_leave(struct snd_mask *mask, unsigned int val)
{
unsigned int v;
v = mask->bits[MASK_OFS(val)] & MASK_BIT(val);
snd_mask_none(mask);
mask->bits[MASK_OFS(val)] = v;
}
static inline void snd_mask_intersect(struct snd_mask *mask,
const struct snd_mask *v)
{
int i;
for (i = 0; i < SNDRV_MASK_SIZE; i++)
mask->bits[i] &= v->bits[i];
}
static inline int snd_mask_eq(const struct snd_mask *mask,
const struct snd_mask *v)
{
return ! memcmp(mask, v, SNDRV_MASK_SIZE * sizeof(u_int32_t));
}
static inline void snd_mask_copy(struct snd_mask *mask,
const struct snd_mask *v)
{
*mask = *v;
}
static inline int snd_mask_test(const struct snd_mask *mask, unsigned int val)
{
return mask->bits[MASK_OFS(val)] & MASK_BIT(val);
}
static inline int snd_mask_single(const struct snd_mask *mask)
{
int i, c = 0;
for (i = 0; i < SNDRV_MASK_SIZE; i++) {
if (! mask->bits[i])
continue;
if (mask->bits[i] & (mask->bits[i] - 1))
return 0;
if (c)
return 0;
c++;
}
return 1;
}
static inline int snd_mask_refine(struct snd_mask *mask,
const struct snd_mask *v)
{
struct snd_mask old;
snd_mask_copy(&old, mask);
snd_mask_intersect(mask, v);
if (snd_mask_empty(mask))
return -EINVAL;
return !snd_mask_eq(mask, &old);
}
static inline int snd_mask_refine_first(struct snd_mask *mask)
{
if (snd_mask_single(mask))
return 0;
snd_mask_leave(mask, snd_mask_min(mask));
return 1;
}
static inline int snd_mask_refine_last(struct snd_mask *mask)
{
if (snd_mask_single(mask))
return 0;
snd_mask_leave(mask, snd_mask_max(mask));
return 1;
}
static inline int snd_mask_refine_min(struct snd_mask *mask, unsigned int val)
{
if (snd_mask_min(mask) >= val)
return 0;
snd_mask_reset_range(mask, 0, val - 1);
if (snd_mask_empty(mask))
return -EINVAL;
return 1;
}
static inline int snd_mask_refine_max(struct snd_mask *mask, unsigned int val)
{
if (snd_mask_max(mask) <= val)
return 0;
snd_mask_reset_range(mask, val + 1, SNDRV_MASK_BITS);
if (snd_mask_empty(mask))
return -EINVAL;
return 1;
}
static inline int snd_mask_refine_set(struct snd_mask *mask, unsigned int val)
{
int changed;
changed = !snd_mask_single(mask);
snd_mask_leave(mask, val);
if (snd_mask_empty(mask))
return -EINVAL;
return changed;
}
static inline int snd_mask_value(const struct snd_mask *mask)
{
return snd_mask_min(mask);
}
static inline void snd_interval_any(struct snd_interval *i)
{
i->min = 0;
i->openmin = 0;
i->max = UINT_MAX;
i->openmax = 0;
i->integer = 0;
i->empty = 0;
}
static inline void snd_interval_none(struct snd_interval *i)
{
i->empty = 1;
}
static inline int snd_interval_checkempty(const struct snd_interval *i)
{
return (i->min > i->max ||
(i->min == i->max && (i->openmin || i->openmax)));
}
static inline int snd_interval_empty(const struct snd_interval *i)
{
return i->empty;
}
static inline int snd_interval_single(const struct snd_interval *i)
{
return (i->min == i->max ||
(i->min + 1 == i->max && i->openmax));
}
static inline int snd_interval_value(const struct snd_interval *i)
{
return i->min;
}
static inline int snd_interval_min(const struct snd_interval *i)
{
return i->min;
}
static inline int snd_interval_max(const struct snd_interval *i)
{
unsigned int v;
v = i->max;
if (i->openmax)
v--;
return v;
}
static inline int snd_interval_test(const struct snd_interval *i, unsigned int val)
{
return !((i->min > val || (i->min == val && i->openmin) ||
i->max < val || (i->max == val && i->openmax)));
}
static inline void snd_interval_copy(struct snd_interval *d, const struct snd_interval *s)
{
*d = *s;
}
static inline int snd_interval_setinteger(struct snd_interval *i)
{
if (i->integer)
return 0;
if (i->openmin && i->openmax && i->min == i->max)
return -EINVAL;
i->integer = 1;
return 1;
}
static inline int snd_interval_eq(const struct snd_interval *i1, const struct snd_interval *i2)
{
if (i1->empty)
return i2->empty;
if (i2->empty)
return i1->empty;
return i1->min == i2->min && i1->openmin == i2->openmin &&
i1->max == i2->max && i1->openmax == i2->openmax;
}
static inline unsigned int add(unsigned int a, unsigned int b)
{
if (a >= UINT_MAX - b)
return UINT_MAX;
return a + b;
}
static inline unsigned int sub(unsigned int a, unsigned int b)
{
if (a > b)
return a - b;
return 0;
}
#define params_access(p) ((__force snd_pcm_access_t)\
snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_ACCESS)))
#define params_format(p) ((__force snd_pcm_format_t)\
snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_FORMAT)))
#define params_subformat(p) \
snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_SUBFORMAT))
static inline unsigned int
params_period_bytes(const struct snd_pcm_hw_params *p)
{
return (params_period_size(p) *
snd_pcm_format_physical_width(params_format(p)) *
params_channels(p)) / 8;
}
#endif /* __SOUND_PCM_PARAMS_H */

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/*
* ALSA Driver for the PT2258 volume controller.
*
* Copyright (c) 2006 Jochen Voss <voss@seehuhn.de>
*
* 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
*
*/
#ifndef __SOUND_PT2258_H
#define __SOUND_PT2258_H
struct snd_pt2258 {
struct snd_card *card;
struct snd_i2c_bus *i2c_bus;
struct snd_i2c_device *i2c_dev;
unsigned char volume[6];
int mute;
};
extern int snd_pt2258_reset(struct snd_pt2258 *pt);
extern int snd_pt2258_build_controls(struct snd_pt2258 *pt);
#endif /* __SOUND_PT2258_H */

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#ifndef PXA2XX_LIB_H
#define PXA2XX_LIB_H
#include <linux/platform_device.h>
#include <sound/ac97_codec.h>
/* PCM */
struct pxa2xx_pcm_dma_params {
char *name; /* stream identifier */
u32 dcmd; /* DMA descriptor dcmd field */
volatile u32 *drcmr; /* the DMA request channel to use */
u32 dev_addr; /* device physical address for DMA */
};
extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params);
extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);
extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream);
extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id);
extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream);
extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream);
extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma);
extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream);
extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm);
/* AC97 */
extern unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
extern void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val);
extern bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97);
extern bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97);
extern void pxa2xx_ac97_finish_reset(struct snd_ac97 *ac97);
extern int pxa2xx_ac97_hw_suspend(void);
extern int pxa2xx_ac97_hw_resume(void);
extern int pxa2xx_ac97_hw_probe(struct platform_device *dev);
extern void pxa2xx_ac97_hw_remove(struct platform_device *dev);
#endif

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#ifndef __SOUND_RAWMIDI_H
#define __SOUND_RAWMIDI_H
/*
* Abstract layer for MIDI v1.0 stream
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/asound.h>
#include <linux/interrupt.h>
#include <linux/spinlock.h>
#include <linux/wait.h>
#include <linux/mutex.h>
#include <linux/workqueue.h>
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
#include <sound/seq_device.h>
#endif
/*
* Raw MIDI interface
*/
#define SNDRV_RAWMIDI_DEVICES 8
#define SNDRV_RAWMIDI_LFLG_OUTPUT (1<<0)
#define SNDRV_RAWMIDI_LFLG_INPUT (1<<1)
#define SNDRV_RAWMIDI_LFLG_OPEN (3<<0)
#define SNDRV_RAWMIDI_LFLG_APPEND (1<<2)
struct snd_rawmidi;
struct snd_rawmidi_substream;
struct snd_seq_port_info;
struct pid;
struct snd_rawmidi_ops {
int (*open) (struct snd_rawmidi_substream * substream);
int (*close) (struct snd_rawmidi_substream * substream);
void (*trigger) (struct snd_rawmidi_substream * substream, int up);
void (*drain) (struct snd_rawmidi_substream * substream);
};
struct snd_rawmidi_global_ops {
int (*dev_register) (struct snd_rawmidi * rmidi);
int (*dev_unregister) (struct snd_rawmidi * rmidi);
void (*get_port_info)(struct snd_rawmidi *rmidi, int number,
struct snd_seq_port_info *info);
};
struct snd_rawmidi_runtime {
struct snd_rawmidi_substream *substream;
unsigned int drain: 1, /* drain stage */
oss: 1; /* OSS compatible mode */
/* midi stream buffer */
unsigned char *buffer; /* buffer for MIDI data */
size_t buffer_size; /* size of buffer */
size_t appl_ptr; /* application pointer */
size_t hw_ptr; /* hardware pointer */
size_t avail_min; /* min avail for wakeup */
size_t avail; /* max used buffer for wakeup */
size_t xruns; /* over/underruns counter */
/* misc */
spinlock_t lock;
wait_queue_head_t sleep;
/* event handler (new bytes, input only) */
void (*event)(struct snd_rawmidi_substream *substream);
/* defers calls to event [input] or ops->trigger [output] */
struct work_struct event_work;
/* private data */
void *private_data;
void (*private_free)(struct snd_rawmidi_substream *substream);
};
struct snd_rawmidi_substream {
struct list_head list; /* list of all substream for given stream */
int stream; /* direction */
int number; /* substream number */
unsigned int opened: 1, /* open flag */
append: 1, /* append flag (merge more streams) */
active_sensing: 1; /* send active sensing when close */
int use_count; /* use counter (for output) */
size_t bytes;
struct snd_rawmidi *rmidi;
struct snd_rawmidi_str *pstr;
char name[32];
struct snd_rawmidi_runtime *runtime;
struct pid *pid;
/* hardware layer */
struct snd_rawmidi_ops *ops;
};
struct snd_rawmidi_file {
struct snd_rawmidi *rmidi;
struct snd_rawmidi_substream *input;
struct snd_rawmidi_substream *output;
};
struct snd_rawmidi_str {
unsigned int substream_count;
unsigned int substream_opened;
struct list_head substreams;
};
struct snd_rawmidi {
struct snd_card *card;
struct list_head list;
unsigned int device; /* device number */
unsigned int info_flags; /* SNDRV_RAWMIDI_INFO_XXXX */
char id[64];
char name[80];
#ifdef CONFIG_SND_OSSEMUL
int ossreg;
#endif
struct snd_rawmidi_global_ops *ops;
struct snd_rawmidi_str streams[2];
void *private_data;
void (*private_free) (struct snd_rawmidi *rmidi);
struct mutex open_mutex;
wait_queue_head_t open_wait;
struct snd_info_entry *dev;
struct snd_info_entry *proc_entry;
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
struct snd_seq_device *seq_dev;
#endif
};
/* main rawmidi functions */
int snd_rawmidi_new(struct snd_card *card, char *id, int device,
int output_count, int input_count,
struct snd_rawmidi **rmidi);
void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream,
struct snd_rawmidi_ops *ops);
/* callbacks */
void snd_rawmidi_receive_reset(struct snd_rawmidi_substream *substream);
int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
const unsigned char *buffer, int count);
void snd_rawmidi_transmit_reset(struct snd_rawmidi_substream *substream);
int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream);
int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
unsigned char *buffer, int count);
int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count);
int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
unsigned char *buffer, int count);
/* main midi functions */
int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info);
int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice,
int mode, struct snd_rawmidi_file *rfile);
int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile);
int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
struct snd_rawmidi_params *params);
int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
struct snd_rawmidi_params *params);
int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream);
int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream);
int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream);
long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream,
unsigned char *buf, long count);
long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream,
const unsigned char *buf, long count);
#endif /* __SOUND_RAWMIDI_H */

22
include/sound/rt5640.h Normal file
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/*
* linux/sound/rt5640.h -- Platform data for RT5640
*
* Copyright 2011 Realtek Microelectronics
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __LINUX_SND_RT5640_H
#define __LINUX_SND_RT5640_H
struct rt5640_platform_data {
/* IN1 & IN2 can optionally be differential */
bool in1_diff;
bool in2_diff;
int ldo1_en; /* GPIO for LDO1_EN */
};
#endif

14
include/sound/s3c24xx_uda134x.h Normal file
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#ifndef _S3C24XX_UDA134X_H_
#define _S3C24XX_UDA134X_H_ 1
#include <sound/uda134x.h>
struct s3c24xx_uda134x_platform_data {
int l3_clk;
int l3_mode;
int l3_data;
void (*power) (int);
int model;
};
#endif

375
include/sound/sb.h Normal file
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#ifndef __SOUND_SB_H
#define __SOUND_SB_H
/*
* Header file for SoundBlaster cards
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* 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/pcm.h>
#include <sound/rawmidi.h>
#include <linux/interrupt.h>
#include <asm/io.h>
enum sb_hw_type {
SB_HW_AUTO,
SB_HW_10,
SB_HW_20,
SB_HW_201,
SB_HW_PRO,
SB_HW_JAZZ16, /* Media Vision Jazz16 */
SB_HW_16,
SB_HW_16CSP, /* SB16 with CSP chip */
SB_HW_ALS100, /* Avance Logic ALS100 chip */
SB_HW_ALS4000, /* Avance Logic ALS4000 chip */
SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */
SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */
};
#define SB_OPEN_PCM 0x01
#define SB_OPEN_MIDI_INPUT 0x02
#define SB_OPEN_MIDI_OUTPUT 0x04
#define SB_OPEN_MIDI_INPUT_TRIGGER 0x08
#define SB_OPEN_MIDI_OUTPUT_TRIGGER 0x10
#define SB_MODE_HALT 0x00
#define SB_MODE_PLAYBACK_8 0x01
#define SB_MODE_PLAYBACK_16 0x02
#define SB_MODE_PLAYBACK (SB_MODE_PLAYBACK_8 | SB_MODE_PLAYBACK_16)
#define SB_MODE_CAPTURE_8 0x04
#define SB_MODE_CAPTURE_16 0x08
#define SB_MODE_CAPTURE (SB_MODE_CAPTURE_8 | SB_MODE_CAPTURE_16)
#define SB_RATE_LOCK_PLAYBACK 0x10
#define SB_RATE_LOCK_CAPTURE 0x20
#define SB_RATE_LOCK (SB_RATE_LOCK_PLAYBACK | SB_RATE_LOCK_CAPTURE)
#define SB_MPU_INPUT 1
struct snd_sb {
unsigned long port; /* base port of DSP chip */
struct resource *res_port;
unsigned long mpu_port; /* MPU port for SB DSP 4.0+ */
int irq; /* IRQ number of DSP chip */
int dma8; /* 8-bit DMA */
int dma16; /* 16-bit DMA */
unsigned short version; /* version of DSP chip */
enum sb_hw_type hardware; /* see to SB_HW_XXXX */
unsigned long alt_port; /* alternate port (ALS4000) */
struct pci_dev *pci; /* ALS4000 */
unsigned int open; /* see to SB_OPEN_XXXX for sb8 */
/* also SNDRV_SB_CSP_MODE_XXX for sb16_csp */
unsigned int mode; /* current mode of stream */
unsigned int force_mode16; /* force 16-bit mode of streams */
unsigned int locked_rate; /* sb16 duplex */
unsigned int playback_format;
unsigned int capture_format;
struct timer_list midi_timer;
unsigned int p_dma_size;
unsigned int p_period_size;
unsigned int c_dma_size;
unsigned int c_period_size;
spinlock_t mixer_lock;
char name[32];
void *csp; /* used only when CONFIG_SND_SB16_CSP is set */
struct snd_card *card;
struct snd_pcm *pcm;
struct snd_pcm_substream *playback_substream;
struct snd_pcm_substream *capture_substream;
struct snd_rawmidi *rmidi;
struct snd_rawmidi_substream *midi_substream_input;
struct snd_rawmidi_substream *midi_substream_output;
irq_handler_t rmidi_callback;
spinlock_t reg_lock;
spinlock_t open_lock;
spinlock_t midi_input_lock;
struct snd_info_entry *proc_entry;
#ifdef CONFIG_PM
unsigned char saved_regs[0x20];
#endif
};
/* I/O ports */
#define SBP(chip, x) ((chip)->port + s_b_SB_##x)
#define SBP1(port, x) ((port) + s_b_SB_##x)
#define s_b_SB_RESET 0x6
#define s_b_SB_READ 0xa
#define s_b_SB_WRITE 0xc
#define s_b_SB_COMMAND 0xc
#define s_b_SB_STATUS 0xc
#define s_b_SB_DATA_AVAIL 0xe
#define s_b_SB_DATA_AVAIL_16 0xf
#define s_b_SB_MIXER_ADDR 0x4
#define s_b_SB_MIXER_DATA 0x5
#define s_b_SB_OPL3_LEFT 0x0
#define s_b_SB_OPL3_RIGHT 0x2
#define s_b_SB_OPL3_BOTH 0x8
#define SB_DSP_OUTPUT 0x14
#define SB_DSP_INPUT 0x24
#define SB_DSP_BLOCK_SIZE 0x48
#define SB_DSP_HI_OUTPUT 0x91
#define SB_DSP_HI_INPUT 0x99
#define SB_DSP_LO_OUTPUT_AUTO 0x1c
#define SB_DSP_LO_INPUT_AUTO 0x2c
#define SB_DSP_HI_OUTPUT_AUTO 0x90
#define SB_DSP_HI_INPUT_AUTO 0x98
#define SB_DSP_IMMED_INT 0xf2
#define SB_DSP_GET_VERSION 0xe1
#define SB_DSP_SPEAKER_ON 0xd1
#define SB_DSP_SPEAKER_OFF 0xd3
#define SB_DSP_DMA8_OFF 0xd0
#define SB_DSP_DMA8_ON 0xd4
#define SB_DSP_DMA8_EXIT 0xda
#define SB_DSP_DMA16_OFF 0xd5
#define SB_DSP_DMA16_ON 0xd6
#define SB_DSP_DMA16_EXIT 0xd9
#define SB_DSP_SAMPLE_RATE 0x40
#define SB_DSP_SAMPLE_RATE_OUT 0x41
#define SB_DSP_SAMPLE_RATE_IN 0x42
#define SB_DSP_MONO_8BIT 0xa0
#define SB_DSP_MONO_16BIT 0xa4
#define SB_DSP_STEREO_8BIT 0xa8
#define SB_DSP_STEREO_16BIT 0xac
#define SB_DSP_MIDI_INPUT_IRQ 0x31
#define SB_DSP_MIDI_UART_IRQ 0x35
#define SB_DSP_MIDI_OUTPUT 0x38
#define SB_DSP4_OUT8_AI 0xc6
#define SB_DSP4_IN8_AI 0xce
#define SB_DSP4_OUT16_AI 0xb6
#define SB_DSP4_IN16_AI 0xbe
#define SB_DSP4_MODE_UNS_MONO 0x00
#define SB_DSP4_MODE_SIGN_MONO 0x10
#define SB_DSP4_MODE_UNS_STEREO 0x20
#define SB_DSP4_MODE_SIGN_STEREO 0x30
#define SB_DSP4_OUTPUT 0x3c
#define SB_DSP4_INPUT_LEFT 0x3d
#define SB_DSP4_INPUT_RIGHT 0x3e
/* registers for SB 2.0 mixer */
#define SB_DSP20_MASTER_DEV 0x02
#define SB_DSP20_PCM_DEV 0x0A
#define SB_DSP20_CD_DEV 0x08
#define SB_DSP20_FM_DEV 0x06
/* registers for SB PRO mixer */
#define SB_DSP_MASTER_DEV 0x22
#define SB_DSP_PCM_DEV 0x04
#define SB_DSP_LINE_DEV 0x2e
#define SB_DSP_CD_DEV 0x28
#define SB_DSP_FM_DEV 0x26
#define SB_DSP_MIC_DEV 0x0a
#define SB_DSP_CAPTURE_SOURCE 0x0c
#define SB_DSP_CAPTURE_FILT 0x0c
#define SB_DSP_PLAYBACK_FILT 0x0e
#define SB_DSP_STEREO_SW 0x0e
#define SB_DSP_MIXS_MIC0 0x00 /* same as MIC */
#define SB_DSP_MIXS_CD 0x01
#define SB_DSP_MIXS_MIC 0x02
#define SB_DSP_MIXS_LINE 0x03
/* registers (only for left channel) for SB 16 mixer */
#define SB_DSP4_MASTER_DEV 0x30
#define SB_DSP4_BASS_DEV 0x46
#define SB_DSP4_TREBLE_DEV 0x44
#define SB_DSP4_SYNTH_DEV 0x34
#define SB_DSP4_PCM_DEV 0x32
#define SB_DSP4_SPEAKER_DEV 0x3b
#define SB_DSP4_LINE_DEV 0x38
#define SB_DSP4_MIC_DEV 0x3a
#define SB_DSP4_OUTPUT_SW 0x3c
#define SB_DSP4_CD_DEV 0x36
#define SB_DSP4_IGAIN_DEV 0x3f
#define SB_DSP4_OGAIN_DEV 0x41
#define SB_DSP4_MIC_AGC 0x43
/* additional registers for SB 16 mixer */
#define SB_DSP4_IRQSETUP 0x80
#define SB_DSP4_DMASETUP 0x81
#define SB_DSP4_IRQSTATUS 0x82
#define SB_DSP4_MPUSETUP 0x84
#define SB_DSP4_3DSE 0x90
/* Registers for DT-019x / ALS-007 mixer */
#define SB_DT019X_MASTER_DEV 0x62
#define SB_DT019X_PCM_DEV 0x64
#define SB_DT019X_SYNTH_DEV 0x66
#define SB_DT019X_CD_DEV 0x68
#define SB_DT019X_MIC_DEV 0x6a
#define SB_DT019X_SPKR_DEV 0x6a
#define SB_DT019X_LINE_DEV 0x6e
#define SB_DT019X_OUTPUT_SW2 0x4c
#define SB_DT019X_CAPTURE_SW 0x6c
#define SB_DT019X_CAP_CD 0x02
#define SB_DT019X_CAP_MIC 0x04
#define SB_DT019X_CAP_LINE 0x06
#define SB_DT019X_CAP_SYNTH 0x07
#define SB_DT019X_CAP_MAIN 0x07
#define SB_ALS4000_MONO_IO_CTRL 0x4b
#define SB_ALS4000_OUT_MIXER_CTRL_2 0x4c
#define SB_ALS4000_MIC_IN_GAIN 0x4d
#define SB_ALS4000_ANALOG_REFRNC_VOLT_CTRL 0x4e
#define SB_ALS4000_FMDAC 0x4f
#define SB_ALS4000_3D_SND_FX 0x50
#define SB_ALS4000_3D_TIME_DELAY 0x51
#define SB_ALS4000_3D_AUTO_MUTE 0x52
#define SB_ALS4000_ANALOG_BLOCK_CTRL 0x53
#define SB_ALS4000_3D_DELAYLINE_PATTERN 0x54
#define SB_ALS4000_CR3_CONFIGURATION 0xc3 /* bit 7 is Digital Loop Enable */
#define SB_ALS4000_QSOUND 0xdb
/* IRQ setting bitmap */
#define SB_IRQSETUP_IRQ9 0x01
#define SB_IRQSETUP_IRQ5 0x02
#define SB_IRQSETUP_IRQ7 0x04
#define SB_IRQSETUP_IRQ10 0x08
/* IRQ types */
#define SB_IRQTYPE_8BIT 0x01
#define SB_IRQTYPE_16BIT 0x02
#define SB_IRQTYPE_MPUIN 0x04
#define ALS4K_IRQTYPE_CR1E_DMA 0x20
/* DMA setting bitmap */
#define SB_DMASETUP_DMA0 0x01
#define SB_DMASETUP_DMA1 0x02
#define SB_DMASETUP_DMA3 0x08
#define SB_DMASETUP_DMA5 0x20
#define SB_DMASETUP_DMA6 0x40
#define SB_DMASETUP_DMA7 0x80
/*
*
*/
static inline void snd_sb_ack_8bit(struct snd_sb *chip)
{
inb(SBP(chip, DATA_AVAIL));
}
static inline void snd_sb_ack_16bit(struct snd_sb *chip)
{
inb(SBP(chip, DATA_AVAIL_16));
}
/* sb_common.c */
int snd_sbdsp_command(struct snd_sb *chip, unsigned char val);
int snd_sbdsp_get_byte(struct snd_sb *chip);
int snd_sbdsp_reset(struct snd_sb *chip);
int snd_sbdsp_create(struct snd_card *card,
unsigned long port,
int irq,
irq_handler_t irq_handler,
int dma8, int dma16,
unsigned short hardware,
struct snd_sb **r_chip);
/* sb_mixer.c */
void snd_sbmixer_write(struct snd_sb *chip, unsigned char reg, unsigned char data);
unsigned char snd_sbmixer_read(struct snd_sb *chip, unsigned char reg);
int snd_sbmixer_new(struct snd_sb *chip);
#ifdef CONFIG_PM
void snd_sbmixer_suspend(struct snd_sb *chip);
void snd_sbmixer_resume(struct snd_sb *chip);
#endif
/* sb8_init.c */
int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm);
/* sb8.c */
irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip);
int snd_sb8_playback_open(struct snd_pcm_substream *substream);
int snd_sb8_capture_open(struct snd_pcm_substream *substream);
int snd_sb8_playback_close(struct snd_pcm_substream *substream);
int snd_sb8_capture_close(struct snd_pcm_substream *substream);
/* midi8.c */
irqreturn_t snd_sb8dsp_midi_interrupt(struct snd_sb *chip);
int snd_sb8dsp_midi(struct snd_sb *chip, int device, struct snd_rawmidi ** rrawmidi);
/* sb16_init.c */
int snd_sb16dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm);
const struct snd_pcm_ops *snd_sb16dsp_get_pcm_ops(int direction);
int snd_sb16dsp_configure(struct snd_sb *chip);
/* sb16.c */
irqreturn_t snd_sb16dsp_interrupt(int irq, void *dev_id);
/* exported mixer stuffs */
enum {
SB_MIX_SINGLE,
SB_MIX_DOUBLE,
SB_MIX_INPUT_SW,
SB_MIX_CAPTURE_PRO,
SB_MIX_CAPTURE_DT019X,
SB_MIX_MONO_CAPTURE_ALS4K
};
#define SB_MIXVAL_DOUBLE(left_reg, right_reg, left_shift, right_shift, mask) \
((left_reg) | ((right_reg) << 8) | ((left_shift) << 16) | ((right_shift) << 19) | ((mask) << 24))
#define SB_MIXVAL_SINGLE(reg, shift, mask) \
((reg) | ((shift) << 16) | ((mask) << 24))
#define SB_MIXVAL_INPUT_SW(reg1, reg2, left_shift, right_shift) \
((reg1) | ((reg2) << 8) | ((left_shift) << 16) | ((right_shift) << 24))
int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int type, unsigned long value);
/* for ease of use */
struct sbmix_elem {
const char *name;
int type;
unsigned long private_value;
};
#define SB_SINGLE(xname, reg, shift, mask) \
{ .name = xname, \
.type = SB_MIX_SINGLE, \
.private_value = SB_MIXVAL_SINGLE(reg, shift, mask) }
#define SB_DOUBLE(xname, left_reg, right_reg, left_shift, right_shift, mask) \
{ .name = xname, \
.type = SB_MIX_DOUBLE, \
.private_value = SB_MIXVAL_DOUBLE(left_reg, right_reg, left_shift, right_shift, mask) }
#define SB16_INPUT_SW(xname, reg1, reg2, left_shift, right_shift) \
{ .name = xname, \
.type = SB_MIX_INPUT_SW, \
.private_value = SB_MIXVAL_INPUT_SW(reg1, reg2, left_shift, right_shift) }
static inline int snd_sbmixer_add_ctl_elem(struct snd_sb *chip, const struct sbmix_elem *c)
{
return snd_sbmixer_add_ctl(chip, c->name, 0, c->type, c->private_value);
}
#endif /* __SOUND_SB_H */

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/*
* Copyright (c) 1999 by Uros Bizjak <uros@kss-loka.si>
* Takashi Iwai <tiwai@suse.de>
*
* SB16ASP/AWE32 CSP control
*
* 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
*
*/
#ifndef __SOUND_SB16_CSP_H
#define __SOUND_SB16_CSP_H
#include <sound/sb.h>
#include <sound/hwdep.h>
#include <linux/firmware.h>
#include <uapi/sound/sb16_csp.h>
struct snd_sb_csp;
/* indices for the known CSP programs */
enum {
CSP_PROGRAM_MULAW,
CSP_PROGRAM_ALAW,
CSP_PROGRAM_ADPCM_INIT,
CSP_PROGRAM_ADPCM_PLAYBACK,
CSP_PROGRAM_ADPCM_CAPTURE,
CSP_PROGRAM_COUNT
};
/*
* CSP operators
*/
struct snd_sb_csp_ops {
int (*csp_use) (struct snd_sb_csp * p);
int (*csp_unuse) (struct snd_sb_csp * p);
int (*csp_autoload) (struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode);
int (*csp_start) (struct snd_sb_csp * p, int sample_width, int channels);
int (*csp_stop) (struct snd_sb_csp * p);
int (*csp_qsound_transfer) (struct snd_sb_csp * p);
};
/*
* CSP private data
*/
struct snd_sb_csp {
struct snd_sb *chip; /* SB16 DSP */
int used; /* usage flag - exclusive */
char codec_name[16]; /* name of codec */
unsigned short func_nr; /* function number */
unsigned int acc_format; /* accepted PCM formats */
int acc_channels; /* accepted channels */
int acc_width; /* accepted sample width */
int acc_rates; /* accepted sample rates */
int mode; /* MODE */
int run_channels; /* current CSP channels */
int run_width; /* current sample width */
int version; /* CSP version (0x10 - 0x1f) */
int running; /* running state */
struct snd_sb_csp_ops ops; /* operators */
spinlock_t q_lock; /* locking */
int q_enabled; /* enabled flag */
int qpos_left; /* left position */
int qpos_right; /* right position */
int qpos_changed; /* position changed flag */
struct snd_kcontrol *qsound_switch;
struct snd_kcontrol *qsound_space;
struct mutex access_mutex; /* locking */
const struct firmware *csp_programs[CSP_PROGRAM_COUNT];
};
int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep);
#endif /* __SOUND_SB16_CSP */

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#ifndef __SOUND_SEQ_DEVICE_H
#define __SOUND_SEQ_DEVICE_H
/*
* ALSA sequencer device management
* Copyright (c) 1999 by Takashi Iwai <tiwai@suse.de>
*
* 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
*
*/
/*
* registered device information
*/
#define ID_LEN 32
/* status flag */
#define SNDRV_SEQ_DEVICE_FREE 0
#define SNDRV_SEQ_DEVICE_REGISTERED 1
struct snd_seq_device {
/* device info */
struct snd_card *card; /* sound card */
int device; /* device number */
char id[ID_LEN]; /* driver id */
char name[80]; /* device name */
int argsize; /* size of the argument */
void *driver_data; /* private data for driver */
int status; /* flag - read only */
void *private_data; /* private data for the caller */
void (*private_free)(struct snd_seq_device *device);
struct list_head list; /* link to next device */
};
/* driver operators
* init_device:
* Initialize the device with given parameters.
* Typically,
* 1. call snd_hwdep_new
* 2. allocate private data and initialize it
* 3. call snd_hwdep_register
* 4. store the instance to dev->driver_data pointer.
*
* free_device:
* Release the private data.
* Typically, call snd_device_free(dev->card, dev->driver_data)
*/
struct snd_seq_dev_ops {
int (*init_device)(struct snd_seq_device *dev);
int (*free_device)(struct snd_seq_device *dev);
};
/*
* prototypes
*/
void snd_seq_device_load_drivers(void);
int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result);
int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize);
int snd_seq_device_unregister_driver(char *id);
#define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device))
/*
* id strings for generic devices
*/
#define SNDRV_SEQ_DEV_ID_MIDISYNTH "seq-midi"
#define SNDRV_SEQ_DEV_ID_OPL3 "opl3-synth"
#endif /* __SOUND_SEQ_DEVICE_H */

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#ifndef __SOUND_SEQ_KERNEL_H
#define __SOUND_SEQ_KERNEL_H
/*
* Main kernel header file for the ALSA sequencer
* Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
*
*
* 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 <linux/time.h>
#include <sound/asequencer.h>
typedef struct snd_seq_real_time snd_seq_real_time_t;
typedef union snd_seq_timestamp snd_seq_timestamp_t;
/* maximum number of events dequeued per schedule interval */
#define SNDRV_SEQ_MAX_DEQUEUE 50
/* maximum number of queues */
#define SNDRV_SEQ_MAX_QUEUES 8
/* max number of concurrent clients */
#define SNDRV_SEQ_MAX_CLIENTS 192
/* max number of concurrent ports */
#define SNDRV_SEQ_MAX_PORTS 254
/* max number of events in memory pool */
#define SNDRV_SEQ_MAX_EVENTS 2000
/* default number of events in memory chunk */
#define SNDRV_SEQ_DEFAULT_CHUNK_EVENTS 64
/* default number of events in memory pool */
#define SNDRV_SEQ_DEFAULT_EVENTS 500
/* max number of events in memory pool for one client (outqueue) */
#define SNDRV_SEQ_MAX_CLIENT_EVENTS 2000
/* default number of events in memory pool for one client (outqueue) */
#define SNDRV_SEQ_DEFAULT_CLIENT_EVENTS 200
/* max delivery path length */
#define SNDRV_SEQ_MAX_HOPS 10
/* max size of event size */
#define SNDRV_SEQ_MAX_EVENT_LEN 0x3fffffff
/* call-backs for kernel port */
struct snd_seq_port_callback {
struct module *owner;
void *private_data;
int (*subscribe)(void *private_data, struct snd_seq_port_subscribe *info);
int (*unsubscribe)(void *private_data, struct snd_seq_port_subscribe *info);
int (*use)(void *private_data, struct snd_seq_port_subscribe *info);
int (*unuse)(void *private_data, struct snd_seq_port_subscribe *info);
int (*event_input)(struct snd_seq_event *ev, int direct, void *private_data, int atomic, int hop);
void (*private_free)(void *private_data);
unsigned int callback_all; /* call subscribe callbacks at each connection/disconnection */
/*...*/
};
/* interface for kernel client */
__printf(3, 4)
int snd_seq_create_kernel_client(struct snd_card *card, int client_index,
const char *name_fmt, ...);
int snd_seq_delete_kernel_client(int client);
int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, int atomic, int hop);
int snd_seq_kernel_client_dispatch(int client, struct snd_seq_event *ev, int atomic, int hop);
int snd_seq_kernel_client_ctl(int client, unsigned int cmd, void *arg);
#define SNDRV_SEQ_EXT_MASK 0xc0000000
#define SNDRV_SEQ_EXT_USRPTR 0x80000000
#define SNDRV_SEQ_EXT_CHAINED 0x40000000
typedef int (*snd_seq_dump_func_t)(void *ptr, void *buf, int count);
int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char *buf,
int in_kernel, int size_aligned);
int snd_seq_dump_var_event(const struct snd_seq_event *event,
snd_seq_dump_func_t func, void *private_data);
/* interface for OSS emulation */
int snd_seq_set_queue_tempo(int client, struct snd_seq_queue_tempo *tempo);
/* port callback routines */
void snd_port_init_callback(struct snd_seq_port_callback *p);
struct snd_seq_port_callback *snd_port_alloc_callback(void);
/* port attach/detach */
int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp,
int cap, int type, int midi_channels, int midi_voices, char *portname);
int snd_seq_event_port_detach(int client, int port);
#ifdef CONFIG_MODULES
void snd_seq_autoload_lock(void);
void snd_seq_autoload_unlock(void);
#else
#define snd_seq_autoload_lock()
#define snd_seq_autoload_unlock()
#endif
#endif /* __SOUND_SEQ_KERNEL_H */

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#ifndef __SOUND_SEQ_MIDI_EMUL_H
#define __SOUND_SEQ_MIDI_EMUL_H
/*
* Midi channel definition for optional channel management.
*
* Copyright (C) 1999 Steve Ratcliffe
*
* 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/seq_kernel.h>
/*
* This structure is used to keep track of the current state on each
* channel. All drivers for hardware that does not understand midi
* directly will probably need to use this structure.
*/
struct snd_midi_channel {
void *private; /* A back pointer to driver data */
int number; /* The channel number */
int client; /* The client associated with this channel */
int port; /* The port associated with this channel */
unsigned char midi_mode; /* GM, GS, XG etc */
unsigned int
drum_channel:1, /* Drum channel */
param_type:1 /* RPN/NRPN */
;
unsigned char midi_aftertouch; /* Aftertouch (key pressure) */
unsigned char midi_pressure; /* Channel pressure */
unsigned char midi_program; /* Instrument number */
short midi_pitchbend; /* Pitch bend amount */
unsigned char control[128]; /* Current value of all controls */
unsigned char note[128]; /* Current status for all notes */
short gm_rpn_pitch_bend_range; /* Pitch bend range */
short gm_rpn_fine_tuning; /* Master fine tuning */
short gm_rpn_coarse_tuning; /* Master coarse tuning */
};
/*
* A structure that represets a set of channels bound to a port. There
* would usually be 16 channels per port. But fewer could be used for
* particular cases.
* The channel set consists of information describing the client and
* port for this midi synth and an array of snd_midi_channel structures.
* A driver that had no need for snd_midi_channel could still use the
* channel set type if it wished with the channel array null.
*/
struct snd_midi_channel_set {
void *private_data; /* Driver data */
int client; /* Client for this port */
int port; /* The port number */
int max_channels; /* Size of the channels array */
struct snd_midi_channel *channels;
unsigned char midi_mode; /* MIDI operating mode */
unsigned char gs_master_volume; /* SYSEX master volume: 0-127 */
unsigned char gs_chorus_mode;
unsigned char gs_reverb_mode;
};
struct snd_midi_op {
void (*note_on)(void *private_data, int note, int vel, struct snd_midi_channel *chan);
void (*note_off)(void *private_data,int note, int vel, struct snd_midi_channel *chan); /* release note */
void (*key_press)(void *private_data, int note, int vel, struct snd_midi_channel *chan);
void (*note_terminate)(void *private_data, int note, struct snd_midi_channel *chan); /* terminate note immediately */
void (*control)(void *private_data, int type, struct snd_midi_channel *chan);
void (*nrpn)(void *private_data, struct snd_midi_channel *chan,
struct snd_midi_channel_set *chset);
void (*sysex)(void *private_data, unsigned char *buf, int len, int parsed,
struct snd_midi_channel_set *chset);
};
/*
* These defines are used so that pitchbend, aftertouch etc, can be
* distinguished from controller values.
*/
/* 0-127 controller values */
#define MIDI_CTL_PITCHBEND 0x80
#define MIDI_CTL_AFTERTOUCH 0x81
#define MIDI_CTL_CHAN_PRESSURE 0x82
/*
* These names exist to allow symbolic access to the controls array.
* The usage is eg: chan->gm_bank_select. Another implementation would
* be really have these members in the struct, and not the array.
*/
#define gm_bank_select control[0]
#define gm_modulation control[1]
#define gm_breath control[2]
#define gm_foot_pedal control[4]
#define gm_portamento_time control[5]
#define gm_data_entry control[6]
#define gm_volume control[7]
#define gm_balance control[8]
#define gm_pan control[10]
#define gm_expression control[11]
#define gm_effect_control1 control[12]
#define gm_effect_control2 control[13]
#define gm_slider1 control[16]
#define gm_slider2 control[17]
#define gm_slider3 control[18]
#define gm_slider4 control[19]
#define gm_bank_select_lsb control[32]
#define gm_modulation_wheel_lsb control[33]
#define gm_breath_lsb control[34]
#define gm_foot_pedal_lsb control[36]
#define gm_portamento_time_lsb control[37]
#define gm_data_entry_lsb control[38]
#define gm_volume_lsb control[39]
#define gm_balance_lsb control[40]
#define gm_pan_lsb control[42]
#define gm_expression_lsb control[43]
#define gm_effect_control1_lsb control[44]
#define gm_effect_control2_lsb control[45]
#define gm_sustain control[MIDI_CTL_SUSTAIN]
#define gm_hold gm_sustain
#define gm_portamento control[MIDI_CTL_PORTAMENTO]
#define gm_sostenuto control[MIDI_CTL_SOSTENUTO]
/*
* These macros give the complete value of the controls that consist
* of coarse and fine pairs. Of course the fine controls are seldom used
* but there is no harm in being complete.
*/
#define SNDRV_GM_BANK_SELECT(cp) (((cp)->control[0]<<7)|((cp)->control[32]))
#define SNDRV_GM_MODULATION_WHEEL(cp) (((cp)->control[1]<<7)|((cp)->control[33]))
#define SNDRV_GM_BREATH(cp) (((cp)->control[2]<<7)|((cp)->control[34]))
#define SNDRV_GM_FOOT_PEDAL(cp) (((cp)->control[4]<<7)|((cp)->control[36]))
#define SNDRV_GM_PORTAMENTO_TIME(cp) (((cp)->control[5]<<7)|((cp)->control[37]))
#define SNDRV_GM_DATA_ENTRY(cp) (((cp)->control[6]<<7)|((cp)->control[38]))
#define SNDRV_GM_VOLUME(cp) (((cp)->control[7]<<7)|((cp)->control[39]))
#define SNDRV_GM_BALANCE(cp) (((cp)->control[8]<<7)|((cp)->control[40]))
#define SNDRV_GM_PAN(cp) (((cp)->control[10]<<7)|((cp)->control[42]))
#define SNDRV_GM_EXPRESSION(cp) (((cp)->control[11]<<7)|((cp)->control[43]))
/* MIDI mode */
#define SNDRV_MIDI_MODE_NONE 0 /* Generic midi */
#define SNDRV_MIDI_MODE_GM 1
#define SNDRV_MIDI_MODE_GS 2
#define SNDRV_MIDI_MODE_XG 3
#define SNDRV_MIDI_MODE_MT32 4
/* MIDI note state */
#define SNDRV_MIDI_NOTE_OFF 0x00
#define SNDRV_MIDI_NOTE_ON 0x01
#define SNDRV_MIDI_NOTE_RELEASED 0x02
#define SNDRV_MIDI_NOTE_SOSTENUTO 0x04
#define SNDRV_MIDI_PARAM_TYPE_REGISTERED 0
#define SNDRV_MIDI_PARAM_TYPE_NONREGISTERED 1
/* SYSEX parse flag */
enum {
SNDRV_MIDI_SYSEX_NOT_PARSED = 0,
SNDRV_MIDI_SYSEX_GM_ON,
SNDRV_MIDI_SYSEX_GS_ON,
SNDRV_MIDI_SYSEX_GS_RESET,
SNDRV_MIDI_SYSEX_GS_CHORUS_MODE,
SNDRV_MIDI_SYSEX_GS_REVERB_MODE,
SNDRV_MIDI_SYSEX_GS_MASTER_VOLUME,
SNDRV_MIDI_SYSEX_GS_PROGRAM,
SNDRV_MIDI_SYSEX_GS_DRUM_CHANNEL,
SNDRV_MIDI_SYSEX_XG_ON,
};
/* Prototypes for midi_process.c */
void snd_midi_process_event(struct snd_midi_op *ops, struct snd_seq_event *ev,
struct snd_midi_channel_set *chanset);
void snd_midi_channel_set_clear(struct snd_midi_channel_set *chset);
struct snd_midi_channel_set *snd_midi_channel_alloc_set(int n);
void snd_midi_channel_free_set(struct snd_midi_channel_set *chset);
#endif /* __SOUND_SEQ_MIDI_EMUL_H */

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#ifndef __SOUND_SEQ_MIDI_EVENT_H
#define __SOUND_SEQ_MIDI_EVENT_H
/*
* MIDI byte <-> sequencer event coder
*
* Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>,
* Jaroslav Kysela <perex@perex.cz>
*
* 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/asequencer.h>
#define MAX_MIDI_EVENT_BUF 256
/* midi status */
struct snd_midi_event {
int qlen; /* queue length */
int read; /* chars read */
int type; /* current event type */
unsigned char lastcmd; /* last command (for MIDI state handling) */
unsigned char nostat; /* no state flag */
int bufsize; /* allocated buffer size */
unsigned char *buf; /* input buffer */
spinlock_t lock;
};
int snd_midi_event_new(int bufsize, struct snd_midi_event **rdev);
void snd_midi_event_free(struct snd_midi_event *dev);
void snd_midi_event_reset_encode(struct snd_midi_event *dev);
void snd_midi_event_reset_decode(struct snd_midi_event *dev);
void snd_midi_event_no_status(struct snd_midi_event *dev, int on);
/* encode from byte stream - return number of written bytes if success */
long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count,
struct snd_seq_event *ev);
int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, struct snd_seq_event *ev);
/* decode from event to bytes - return number of written bytes if success */
long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count,
struct snd_seq_event *ev);
#endif /* __SOUND_SEQ_MIDI_EVENT_H */

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#ifndef __SOUND_SEQ_OSS_H
#define __SOUND_SEQ_OSS_H
/*
* OSS compatible sequencer driver
*
* Copyright (C) 1998,99 Takashi Iwai
*
* 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/asequencer.h>
#include <sound/seq_kernel.h>
/*
* argument structure for synthesizer operations
*/
struct snd_seq_oss_arg {
/* given by OSS sequencer */
int app_index; /* application unique index */
int file_mode; /* file mode - see below */
int seq_mode; /* sequencer mode - see below */
/* following must be initialized in open callback */
struct snd_seq_addr addr; /* opened port address */
void *private_data; /* private data for lowlevel drivers */
/* note-on event passing mode: initially given by OSS seq,
* but configurable by drivers - see below
*/
int event_passing;
};
/*
* synthesizer operation callbacks
*/
struct snd_seq_oss_callback {
struct module *owner;
int (*open)(struct snd_seq_oss_arg *p, void *closure);
int (*close)(struct snd_seq_oss_arg *p);
int (*ioctl)(struct snd_seq_oss_arg *p, unsigned int cmd, unsigned long arg);
int (*load_patch)(struct snd_seq_oss_arg *p, int format, const char __user *buf, int offs, int count);
int (*reset)(struct snd_seq_oss_arg *p);
int (*raw_event)(struct snd_seq_oss_arg *p, unsigned char *data);
};
/* flag: file_mode */
#define SNDRV_SEQ_OSS_FILE_ACMODE 3
#define SNDRV_SEQ_OSS_FILE_READ 1
#define SNDRV_SEQ_OSS_FILE_WRITE 2
#define SNDRV_SEQ_OSS_FILE_NONBLOCK 4
/* flag: seq_mode */
#define SNDRV_SEQ_OSS_MODE_SYNTH 0
#define SNDRV_SEQ_OSS_MODE_MUSIC 1
/* flag: event_passing */
#define SNDRV_SEQ_OSS_PROCESS_EVENTS 0 /* key == 255 is processed as velocity change */
#define SNDRV_SEQ_OSS_PASS_EVENTS 1 /* pass all events to callback */
#define SNDRV_SEQ_OSS_PROCESS_KEYPRESS 2 /* key >= 128 will be processed as key-pressure */
/* default control rate: fixed */
#define SNDRV_SEQ_OSS_CTRLRATE 100
/* default max queue length: configurable by module option */
#define SNDRV_SEQ_OSS_MAX_QLEN 1024
/*
* data pointer to snd_seq_register_device
*/
struct snd_seq_oss_reg {
int type;
int subtype;
int nvoices;
struct snd_seq_oss_callback oper;
void *private_data;
};
/* device id */
#define SNDRV_SEQ_DEV_ID_OSS "seq-oss"
#endif /* __SOUND_SEQ_OSS_H */

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#ifndef __SOUND_SEQ_OSS_LEGACY_H
#define __SOUND_SEQ_OSS_LEGACY_H
/*
* OSS compatible macro definitions
*
* Copyright (C) 2000 Abramo Bagnara <abramo@alsa-project.org>
*
* 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 <linux/soundcard.h>
#ifndef SAMPLE_TYPE_AWE32
#define SAMPLE_TYPE_AWE32 0x20
#endif
#endif /* __SOUND_SEQ_OSS_LEGACY_H */

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#ifndef __SOUND_SEQ_VIRMIDI_H
#define __SOUND_SEQ_VIRMIDI_H
/*
* Virtual Raw MIDI client on Sequencer
* Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>,
* Jaroslav Kysela <perex@perex.cz>
*
* 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/rawmidi.h>
#include <sound/seq_midi_event.h>
/*
* device file instance:
* This instance is created at each time the midi device file is
* opened. Each instance has its own input buffer and MIDI parser
* (buffer), and is associated with the device instance.
*/
struct snd_virmidi {
struct list_head list;
int seq_mode;
int client;
int port;
unsigned int trigger: 1;
struct snd_midi_event *parser;
struct snd_seq_event event;
struct snd_virmidi_dev *rdev;
struct snd_rawmidi_substream *substream;
};
#define SNDRV_VIRMIDI_SUBSCRIBE (1<<0)
#define SNDRV_VIRMIDI_USE (1<<1)
/*
* device record:
* Each virtual midi device has one device instance. It contains
* common information and the linked-list of opened files,
*/
struct snd_virmidi_dev {
struct snd_card *card; /* associated card */
struct snd_rawmidi *rmidi; /* rawmidi device */
int seq_mode; /* SNDRV_VIRMIDI_XXX */
int device; /* sequencer device */
int client; /* created/attached client */
int port; /* created/attached port */
unsigned int flags; /* SNDRV_VIRMIDI_* */
rwlock_t filelist_lock;
struct list_head filelist;
};
/* sequencer mode:
* ATTACH = input/output events from midi device are routed to the
* attached sequencer port. sequencer port is not created
* by virmidi itself.
* the input to rawmidi must be processed by passing the
* incoming events via snd_virmidi_receive()
* DISPATCH = input/output events are routed to subscribers.
* sequencer port is created in virmidi.
*/
#define SNDRV_VIRMIDI_SEQ_NONE 0
#define SNDRV_VIRMIDI_SEQ_ATTACH 1
#define SNDRV_VIRMIDI_SEQ_DISPATCH 2
int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmidi);
#endif /* __SOUND_SEQ_VIRMIDI */

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/*
* SH_DAC specific configuration, for the dac_audio platform_device
*
* Copyright (C) 2009 Rafael Ignacio Zurita <rizurita@yahoo.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License version 2 as published
* by the Free Software Foundation.
*/
#ifndef __INCLUDE_SH_DAC_AUDIO_H
#define __INCLUDE_SH_DAC_AUDIO_H
struct dac_audio_pdata {
int buffer_size;
int channel;
void (*start)(struct dac_audio_pdata *pd);
void (*stop)(struct dac_audio_pdata *pd);
};
#endif /* __INCLUDE_SH_DAC_AUDIO_H */

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#ifndef __SOUND_FSI_H
#define __SOUND_FSI_H
/*
* Fifo-attached Serial Interface (FSI) support for SH7724
*
* Copyright (C) 2009 Renesas Solutions Corp.
* Kuninori Morimoto <morimoto.kuninori@renesas.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/clk.h>
#include <sound/soc.h>
/*
* flags
*/
#define SH_FSI_FMT_SPDIF (1 << 0) /* spdif for HDMI */
#define SH_FSI_ENABLE_STREAM_MODE (1 << 1) /* for 16bit data */
#define SH_FSI_CLK_CPG (1 << 2) /* FSIxCK + FSI-DIV */
struct sh_fsi_port_info {
unsigned long flags;
int tx_id;
int rx_id;
};
struct sh_fsi_platform_info {
struct sh_fsi_port_info port_a;
struct sh_fsi_port_info port_b;
};
#endif /* __SOUND_FSI_H */

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/*
* ASoC simple sound card support
*
* Copyright (C) 2012 Renesas Solutions Corp.
* Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __SIMPLE_CARD_H
#define __SIMPLE_CARD_H
#include <sound/soc.h>
struct asoc_simple_dai {
const char *name;
unsigned int fmt;
unsigned int sysclk;
};
struct asoc_simple_card_info {
const char *name;
const char *card;
const char *codec;
const char *platform;
unsigned int daifmt;
struct asoc_simple_dai cpu_dai;
struct asoc_simple_dai codec_dai;
/* used in simple-card.c */
struct snd_soc_dai_link snd_link;
struct snd_soc_card snd_card;
};
#endif /* __SIMPLE_CARD_H */

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#ifndef __SOUND_SND_WAVEFRONT_H__
#define __SOUND_SND_WAVEFRONT_H__
#include <sound/mpu401.h>
#include <sound/hwdep.h>
#include <sound/rawmidi.h>
#include <sound/wavefront.h> /* generic OSS/ALSA/user-level wavefront header */
/* MIDI interface */
struct _snd_wavefront_midi;
struct _snd_wavefront_card;
struct _snd_wavefront;
typedef struct _snd_wavefront_midi snd_wavefront_midi_t;
typedef struct _snd_wavefront_card snd_wavefront_card_t;
typedef struct _snd_wavefront snd_wavefront_t;
typedef enum { internal_mpu = 0, external_mpu = 1 } snd_wavefront_mpu_id;
struct _snd_wavefront_midi {
unsigned long base; /* I/O port address */
char isvirtual; /* doing virtual MIDI stuff ? */
char istimer; /* timer is used */
snd_wavefront_mpu_id output_mpu; /* most-recently-used */
snd_wavefront_mpu_id input_mpu; /* most-recently-used */
unsigned int mode[2]; /* MPU401_MODE_XXX */
struct snd_rawmidi_substream *substream_output[2];
struct snd_rawmidi_substream *substream_input[2];
struct timer_list timer;
spinlock_t open;
spinlock_t virtual; /* protects isvirtual */
};
#define OUTPUT_READY 0x40
#define INPUT_AVAIL 0x80
#define MPU_ACK 0xFE
#define UART_MODE_ON 0x3F
extern struct snd_rawmidi_ops snd_wavefront_midi_output;
extern struct snd_rawmidi_ops snd_wavefront_midi_input;
extern void snd_wavefront_midi_enable_virtual (snd_wavefront_card_t *);
extern void snd_wavefront_midi_disable_virtual (snd_wavefront_card_t *);
extern void snd_wavefront_midi_interrupt (snd_wavefront_card_t *);
extern int snd_wavefront_midi_start (snd_wavefront_card_t *);
struct _snd_wavefront {
unsigned long irq; /* "you were one, one of the few ..." */
unsigned long base; /* low i/o port address */
struct resource *res_base; /* i/o port resource allocation */
#define mpu_data_port base
#define mpu_command_port base + 1 /* write semantics */
#define mpu_status_port base + 1 /* read semantics */
#define data_port base + 2
#define status_port base + 3 /* read semantics */
#define control_port base + 3 /* write semantics */
#define block_port base + 4 /* 16 bit, writeonly */
#define last_block_port base + 6 /* 16 bit, writeonly */
/* FX ports. These are mapped through the ICS2115 to the YS225.
The ICS2115 takes care of flipping the relevant pins on the
YS225 so that access to each of these ports does the right
thing. Note: these are NOT documented by Turtle Beach.
*/
#define fx_status base + 8
#define fx_op base + 8
#define fx_lcr base + 9
#define fx_dsp_addr base + 0xa
#define fx_dsp_page base + 0xb
#define fx_dsp_lsb base + 0xc
#define fx_dsp_msb base + 0xd
#define fx_mod_addr base + 0xe
#define fx_mod_data base + 0xf
volatile int irq_ok; /* set by interrupt handler */
volatile int irq_cnt; /* ditto */
char debug; /* debugging flags */
int freemem; /* installed RAM, in bytes */
char fw_version[2]; /* major = [0], minor = [1] */
char hw_version[2]; /* major = [0], minor = [1] */
char israw; /* needs Motorola microcode */
char has_fx; /* has FX processor (Tropez+) */
char fx_initialized; /* FX's register pages initialized */
char prog_status[WF_MAX_PROGRAM]; /* WF_SLOT_* */
char patch_status[WF_MAX_PATCH]; /* WF_SLOT_* */
char sample_status[WF_MAX_SAMPLE]; /* WF_ST_* | WF_SLOT_* */
int samples_used; /* how many */
char interrupts_are_midi; /* h/w MPU interrupts enabled ? */
char rom_samples_rdonly; /* can we write on ROM samples */
spinlock_t irq_lock;
wait_queue_head_t interrupt_sleeper;
snd_wavefront_midi_t midi; /* ICS2115 MIDI interface */
struct snd_card *card;
};
struct _snd_wavefront_card {
snd_wavefront_t wavefront;
#ifdef CONFIG_PNP
struct pnp_dev *wss;
struct pnp_dev *ctrl;
struct pnp_dev *mpu;
struct pnp_dev *synth;
#endif /* CONFIG_PNP */
};
extern void snd_wavefront_internal_interrupt (snd_wavefront_card_t *card);
extern int snd_wavefront_detect_irq (snd_wavefront_t *dev) ;
extern int snd_wavefront_check_irq (snd_wavefront_t *dev, int irq);
extern int snd_wavefront_restart (snd_wavefront_t *dev);
extern int snd_wavefront_start (snd_wavefront_t *dev);
extern int snd_wavefront_detect (snd_wavefront_card_t *card);
extern int snd_wavefront_config_midi (snd_wavefront_t *dev) ;
extern int snd_wavefront_cmd (snd_wavefront_t *, int, unsigned char *,
unsigned char *);
extern int snd_wavefront_synth_ioctl (struct snd_hwdep *,
struct file *,
unsigned int cmd,
unsigned long arg);
extern int snd_wavefront_synth_open (struct snd_hwdep *, struct file *);
extern int snd_wavefront_synth_release (struct snd_hwdep *, struct file *);
/* FX processor - see also yss225.[ch] */
extern int snd_wavefront_fx_start (snd_wavefront_t *);
extern int snd_wavefront_fx_detect (snd_wavefront_t *);
extern int snd_wavefront_fx_ioctl (struct snd_hwdep *,
struct file *,
unsigned int cmd,
unsigned long arg);
extern int snd_wavefront_fx_open (struct snd_hwdep *, struct file *);
extern int snd_wavefront_fx_release (struct snd_hwdep *, struct file *);
/* prefix in all snd_printk() delivered messages */
#define LOGNAME "WaveFront: "
#endif /* __SOUND_SND_WAVEFRONT_H__ */

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/*
* linux/sound/soc-dai.h -- ALSA SoC Layer
*
* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Digital Audio Interface (DAI) API.
*/
#ifndef __LINUX_SND_SOC_DAI_H
#define __LINUX_SND_SOC_DAI_H
#include <linux/list.h>
struct snd_pcm_substream;
struct snd_soc_dapm_widget;
struct snd_compr_stream;
/*
* DAI hardware audio formats.
*
* Describes the physical PCM data formating and clocking. Add new formats
* to the end.
*/
#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97 6 /* AC97 */
#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */
/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
/*
* DAI Clock gating.
*
* DAI bit clocks can be be gated (disabled) when the DAI is not
* sending or receiving PCM data in a frame. This can be used to save power.
*/
#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
/*
* DAI hardware signal inversions.
*
* Specifies whether the DAI can also support inverted clocks for the specified
* format.
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
/*
* DAI hardware clock masters.
*
* This is wrt the codec, the inverse is true for the interface
* i.e. if the codec is clk and FRM master then the interface is
* clk and frame slave.
*/
#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
#define SND_SOC_DAIFMT_INV_MASK 0x0f00
#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
/*
* Master Clock Directions
*/
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
SNDRV_PCM_FMTBIT_S16_LE |\
SNDRV_PCM_FMTBIT_S16_BE |\
SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S20_3BE |\
SNDRV_PCM_FMTBIT_S24_3LE |\
SNDRV_PCM_FMTBIT_S24_3BE |\
SNDRV_PCM_FMTBIT_S32_LE |\
SNDRV_PCM_FMTBIT_S32_BE)
struct snd_soc_dai_driver;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
/* Digital Audio Interface clocking API.*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir);
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div);
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
unsigned int tx_num, unsigned int *tx_slot,
unsigned int rx_num, unsigned int *rx_slot);
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
int direction);
struct snd_soc_dai_ops {
/*
* DAI clocking configuration, all optional.
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/*
* DAI format configuration
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width);
int (*set_channel_map)(struct snd_soc_dai *dai,
unsigned int tx_num, unsigned int *tx_slot,
unsigned int rx_num, unsigned int *rx_slot);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
* DAI digital mute - optional.
* Called by soc-core to minimise any pops.
*/
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
/*
* ALSA PCM audio operations - all optional.
* Called by soc-core during audio PCM operations.
*/
int (*startup)(struct snd_pcm_substream *,
struct snd_soc_dai *);
void (*shutdown)(struct snd_pcm_substream *,
struct snd_soc_dai *);
int (*hw_params)(struct snd_pcm_substream *,
struct snd_pcm_hw_params *, struct snd_soc_dai *);
int (*hw_free)(struct snd_pcm_substream *,
struct snd_soc_dai *);
int (*prepare)(struct snd_pcm_substream *,
struct snd_soc_dai *);
int (*trigger)(struct snd_pcm_substream *, int,
struct snd_soc_dai *);
int (*bespoke_trigger)(struct snd_pcm_substream *, int,
struct snd_soc_dai *);
/*
* For hardware based FIFO caused delay reporting.
* Optional.
*/
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
struct snd_soc_dai *);
};
/*
* Digital Audio Interface Driver.
*
* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
* operations and capabilities. Codec and platform drivers will register this
* structure for every DAI they have.
*
* This structure covers the clocking, formating and ALSA operations for each
* interface.
*/
struct snd_soc_dai_driver {
/* DAI description */
const char *name;
unsigned int id;
int ac97_control;
unsigned int base;
/* DAI driver callbacks */
int (*probe)(struct snd_soc_dai *dai);
int (*remove)(struct snd_soc_dai *dai);
int (*suspend)(struct snd_soc_dai *dai);
int (*resume)(struct snd_soc_dai *dai);
/* compress dai */
bool compress_dai;
/* ops */
const struct snd_soc_dai_ops *ops;
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
unsigned int symmetric_rates:1;
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
};
/*
* Digital Audio Interface runtime data.
*
* Holds runtime data for a DAI.
*/
struct snd_soc_dai {
const char *name;
int id;
struct device *dev;
void *ac97_pdata; /* platform_data for the ac97 codec */
/* driver ops */
struct snd_soc_dai_driver *driver;
/* DAI runtime info */
unsigned int capture_active:1; /* stream is in use */
unsigned int playback_active:1; /* stream is in use */
unsigned int symmetric_rates:1;
struct snd_pcm_runtime *runtime;
unsigned int active;
unsigned char probed:1;
struct snd_soc_dapm_widget *playback_widget;
struct snd_soc_dapm_widget *capture_widget;
struct snd_soc_dapm_context dapm;
/* DAI DMA data */
void *playback_dma_data;
void *capture_dma_data;
/* Symmetry data - only valid if symmetry is being enforced */
unsigned int rate;
/* parent platform/codec */
struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
};
static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
const struct snd_pcm_substream *ss)
{
return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
dai->playback_dma_data : dai->capture_dma_data;
}
static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
const struct snd_pcm_substream *ss,
void *data)
{
if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
dai->playback_dma_data = data;
else
dai->capture_dma_data = data;
}
static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
void *data)
{
dev_set_drvdata(dai->dev, data);
}
static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
{
return dev_get_drvdata(dai->dev);
}
#endif

608
include/sound/soc-dapm.h Normal file
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/*
* linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management
*
* Author: Liam Girdwood
* Created: Aug 11th 2005
* Copyright: Wolfson Microelectronics. PLC.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __LINUX_SND_SOC_DAPM_H
#define __LINUX_SND_SOC_DAPM_H
#include <linux/types.h>
#include <sound/control.h>
struct device;
/* widget has no PM register bit */
#define SND_SOC_NOPM -1
/*
* SoC dynamic audio power management
*
* We can have up to 4 power domains
* 1. Codec domain - VREF, VMID
* Usually controlled at codec probe/remove, although can be set
* at stream time if power is not needed for sidetone, etc.
* 2. Platform/Machine domain - physically connected inputs and outputs
* Is platform/machine and user action specific, is set in the machine
* driver and by userspace e.g when HP are inserted
* 3. Path domain - Internal codec path mixers
* Are automatically set when mixer and mux settings are
* changed by the user.
* 4. Stream domain - DAC's and ADC's.
* Enabled when stream playback/capture is started.
*/
/* codec domain */
#define SND_SOC_DAPM_VMID(wname) \
{ .id = snd_soc_dapm_vmid, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0}
/* platform domain */
#define SND_SOC_DAPM_SIGGEN(wname) \
{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
#define SND_SOC_DAPM_INPUT(wname) \
{ .id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
#define SND_SOC_DAPM_OUTPUT(wname) \
{ .id = snd_soc_dapm_output, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
#define SND_SOC_DAPM_MIC(wname, wevent) \
{ .id = snd_soc_dapm_mic, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
#define SND_SOC_DAPM_HP(wname, wevent) \
{ .id = snd_soc_dapm_hp, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
#define SND_SOC_DAPM_SPK(wname, wevent) \
{ .id = snd_soc_dapm_spk, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
#define SND_SOC_DAPM_LINE(wname, wevent) \
{ .id = snd_soc_dapm_line, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
/* path domain */
#define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\
wcontrols, wncontrols) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
#define SND_SOC_DAPM_OUT_DRV(wname, wreg, wshift, winvert,\
wcontrols, wncontrols) \
{ .id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
#define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \
wcontrols, wncontrols)\
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
#define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \
wcontrols, wncontrols)\
{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \
.num_kcontrols = wncontrols}
#define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0}
#define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \
{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1}
#define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \
{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1}
#define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \
{ .id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1}
#define SND_SOC_DAPM_VALUE_MUX(wname, wreg, wshift, winvert, wcontrols) \
{ .id = snd_soc_dapm_value_mux, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \
.num_kcontrols = 1}
/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
wcontrols) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \
wcontrols)\
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \
wcontrols)\
{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \
.num_kcontrols = ARRAY_SIZE(wcontrols)}
/* path domain with event - event handler must return 0 for success */
#define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
wncontrols, wevent, wflags) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_OUT_DRV_E(wname, wreg, wshift, winvert, wcontrols, \
wncontrols, wevent, wflags) \
{ .id = snd_soc_dapm_out_drv, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \
wncontrols, wevent, wflags) \
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \
wcontrols, wncontrols, wevent, wflags) \
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, \
.num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_VIRT_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
{ .id = snd_soc_dapm_virt_mux, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = 1, \
.event = wevent, .event_flags = wflags}
/* additional sequencing control within an event type */
#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .event = wevent, .event_flags = wflags, \
.subseq = wsubseq}
#define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \
wflags) \
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags, .subseq = wsubseq}
/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
.event = wevent, .event_flags = wflags}
#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
.event = wevent, .event_flags = wflags}
#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \
wcontrols, wevent, wflags) \
{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
.invert = winvert, .kcontrol_news = wcontrols, \
.num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags}
/* events that are pre and post DAPM */
#define SND_SOC_DAPM_PRE(wname, wevent) \
{ .id = snd_soc_dapm_pre, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD}
#define SND_SOC_DAPM_POST(wname, wevent) \
{ .id = snd_soc_dapm_post, .name = wname, .kcontrol_news = NULL, \
.num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
/* stream domain */
#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
.reg = wreg, .shift = wshift, .invert = winvert }
#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
.reg = wreg, .shift = wshift, .invert = winvert, \
.event = wevent, .event_flags = wflags }
#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
.reg = wreg, .shift = wshift, .invert = winvert }
#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
.reg = wreg, .shift = wshift, .invert = winvert, \
.event = wevent, .event_flags = wflags }
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
wevent, wflags) \
{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert, \
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \
{ .id = snd_soc_dapm_clock_supply, .name = wname, \
.reg = SND_SOC_NOPM, .event = dapm_clock_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
/* generic widgets */
#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
{ .id = wid, .name = wname, .kcontrol_news = NULL, .num_kcontrols = 0, \
.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags}
#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \
{ .id = snd_soc_dapm_regulator_supply, .name = wname, \
.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
.invert = wflags}
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_DAPM_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
.get = snd_soc_dapm_get_enum_double, \
.put = snd_soc_dapm_put_enum_double, \
.private_value = (unsigned long)&xenum }
#define SOC_DAPM_ENUM_VIRT(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
.get = snd_soc_dapm_get_enum_virt, \
.put = snd_soc_dapm_put_enum_virt, \
.private_value = (unsigned long)&xenum }
#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
.get = xget, \
.put = xput, \
.private_value = (unsigned long)&xenum }
#define SOC_DAPM_VALUE_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
.get = snd_soc_dapm_get_value_enum_double, \
.put = snd_soc_dapm_put_value_enum_double, \
.private_value = (unsigned long)&xenum }
#define SOC_DAPM_PIN_SWITCH(xname) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \
.info = snd_soc_dapm_info_pin_switch, \
.get = snd_soc_dapm_get_pin_switch, \
.put = snd_soc_dapm_put_pin_switch, \
.private_value = (unsigned long)xname }
/* dapm stream operations */
#define SND_SOC_DAPM_STREAM_NOP 0x0
#define SND_SOC_DAPM_STREAM_START 0x1
#define SND_SOC_DAPM_STREAM_STOP 0x2
#define SND_SOC_DAPM_STREAM_SUSPEND 0x4
#define SND_SOC_DAPM_STREAM_RESUME 0x8
#define SND_SOC_DAPM_STREAM_PAUSE_PUSH 0x10
#define SND_SOC_DAPM_STREAM_PAUSE_RELEASE 0x20
/* dapm event types */
#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
#define SND_SOC_DAPM_PRE_POST_PMD \
(SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)
/* convenience event type detection */
#define SND_SOC_DAPM_EVENT_ON(e) \
(e & (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU))
#define SND_SOC_DAPM_EVENT_OFF(e) \
(e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD))
/* regulator widget flags */
#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */
struct snd_soc_dapm_widget;
enum snd_soc_dapm_type;
struct snd_soc_dapm_path;
struct snd_soc_dapm_pin;
struct snd_soc_dapm_route;
struct snd_soc_dapm_context;
struct regulator;
struct snd_soc_dapm_widget_list;
int dapm_reg_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
int dapm_regulator_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
int dapm_clock_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
/* dapm controls */
int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget,
int num);
int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
struct snd_soc_dai *dai);
int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
const struct snd_soc_pcm_stream *params,
struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
/* dapm path setup */
int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
/* dapm events */
void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
int event);
void snd_soc_dapm_shutdown(struct snd_soc_card *card);
/* external DAPM widget events */
int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int connect);
int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
struct dentry *parent);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin);
int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
const char *pin);
void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
/* Mostly internal - should not normally be used */
void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
/* dapm path query */
int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
struct snd_soc_dapm_widget_list **list);
/* dapm widget types */
enum snd_soc_dapm_type {
snd_soc_dapm_input = 0, /* input pin */
snd_soc_dapm_output, /* output pin */
snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */
snd_soc_dapm_virt_mux, /* virtual version of snd_soc_dapm_mux */
snd_soc_dapm_value_mux, /* selects 1 analog signal from many inputs */
snd_soc_dapm_mixer, /* mixes several analog signals together */
snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */
snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */
snd_soc_dapm_out_drv, /* output driver */
snd_soc_dapm_adc, /* analog to digital converter */
snd_soc_dapm_dac, /* digital to analog converter */
snd_soc_dapm_micbias, /* microphone bias (power) */
snd_soc_dapm_mic, /* microphone */
snd_soc_dapm_hp, /* headphones */
snd_soc_dapm_spk, /* speaker */
snd_soc_dapm_line, /* line input/output */
snd_soc_dapm_switch, /* analog switch */
snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
snd_soc_dapm_regulator_supply, /* external regulator */
snd_soc_dapm_clock_supply, /* external clock */
snd_soc_dapm_aif_in, /* audio interface input */
snd_soc_dapm_aif_out, /* audio interface output */
snd_soc_dapm_siggen, /* signal generator */
snd_soc_dapm_dai_in, /* link to DAI structure */
snd_soc_dapm_dai_out,
snd_soc_dapm_dai_link, /* link between two DAI structures */
};
enum snd_soc_dapm_subclass {
SND_SOC_DAPM_CLASS_INIT = 0,
SND_SOC_DAPM_CLASS_RUNTIME = 1,
};
/*
* DAPM audio route definition.
*
* Defines an audio route originating at source via control and finishing
* at sink.
*/
struct snd_soc_dapm_route {
const char *sink;
const char *control;
const char *source;
/* Note: currently only supported for links where source is a supply */
int (*connected)(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
};
/* dapm audio path between two widgets */
struct snd_soc_dapm_path {
const char *name;
const char *long_name;
/* source (input) and sink (output) widgets */
struct snd_soc_dapm_widget *source;
struct snd_soc_dapm_widget *sink;
struct snd_kcontrol *kcontrol;
/* status */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
u32 walking:1; /* path is in the process of being walked */
u32 weak:1; /* path ignored for power management */
int (*connected)(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
struct list_head list_source;
struct list_head list_sink;
struct list_head list;
};
/* dapm widget */
struct snd_soc_dapm_widget {
enum snd_soc_dapm_type id;
const char *name; /* widget name */
const char *sname; /* stream name */
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
void *priv; /* widget specific data */
struct regulator *regulator; /* attached regulator */
const struct snd_soc_pcm_stream *params; /* params for dai links */
/* dapm control */
int reg; /* negative reg = no direct dapm */
unsigned char shift; /* bits to shift */
unsigned int value; /* widget current value */
unsigned int mask; /* non-shifted mask */
unsigned int on_val; /* on state value */
unsigned int off_val; /* off state value */
unsigned char power:1; /* block power status */
unsigned char invert:1; /* invert the power bit */
unsigned char active:1; /* active stream on DAC, ADC's */
unsigned char connected:1; /* connected codec pin */
unsigned char new:1; /* cnew complete */
unsigned char ext:1; /* has external widgets */
unsigned char force:1; /* force state */
unsigned char ignore_suspend:1; /* kept enabled over suspend */
unsigned char new_power:1; /* power from this run */
unsigned char power_checked:1; /* power checked this run */
int subseq; /* sort within widget type */
int (*power_check)(struct snd_soc_dapm_widget *w);
/* external events */
unsigned short event_flags; /* flags to specify event types */
int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
/* kcontrols that relate to this widget */
int num_kcontrols;
const struct snd_kcontrol_new *kcontrol_news;
struct snd_kcontrol **kcontrols;
/* widget input and outputs */
struct list_head sources;
struct list_head sinks;
/* used during DAPM updates */
struct list_head power_list;
struct list_head dirty;
int inputs;
int outputs;
struct clk *clk;
};
struct snd_soc_dapm_update {
struct snd_soc_dapm_widget *widget;
struct snd_kcontrol *kcontrol;
int reg;
int mask;
int val;
};
/* DAPM context */
struct snd_soc_dapm_context {
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
struct snd_soc_dapm_update *update;
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
struct device *dev; /* from parent - for debug */
struct snd_soc_codec *codec; /* parent codec */
struct snd_soc_platform *platform; /* parent platform */
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
enum snd_soc_bias_level target_bias_level;
struct list_head list;
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dapm;
#endif
};
/* A list of widgets associated with an object, typically a snd_kcontrol */
struct snd_soc_dapm_widget_list {
int num_widgets;
struct snd_soc_dapm_widget *widgets[0];
};
struct snd_soc_dapm_stats {
int power_checks;
int path_checks;
int neighbour_checks;
};
#endif

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/*
* linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
*
* Author: Liam Girdwood <lrg@ti.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __LINUX_SND_SOC_DPCM_H
#define __LINUX_SND_SOC_DPCM_H
#include <linux/list.h>
#include <sound/pcm.h>
struct snd_soc_pcm_runtime;
/*
* Types of runtime_update to perform. e.g. originated from FE PCM ops
* or audio route changes triggered by muxes/mixers.
*/
enum snd_soc_dpcm_update {
SND_SOC_DPCM_UPDATE_NO = 0,
SND_SOC_DPCM_UPDATE_BE,
SND_SOC_DPCM_UPDATE_FE,
};
/*
* Dynamic PCM Frontend -> Backend link management states.
*/
enum snd_soc_dpcm_link_state {
SND_SOC_DPCM_LINK_STATE_NEW = 0, /* newly created link */
SND_SOC_DPCM_LINK_STATE_FREE, /* link to be dismantled */
};
/*
* Dynamic PCM Frontend -> Backend link PCM states.
*/
enum snd_soc_dpcm_state {
SND_SOC_DPCM_STATE_NEW = 0,
SND_SOC_DPCM_STATE_OPEN,
SND_SOC_DPCM_STATE_HW_PARAMS,
SND_SOC_DPCM_STATE_PREPARE,
SND_SOC_DPCM_STATE_START,
SND_SOC_DPCM_STATE_STOP,
SND_SOC_DPCM_STATE_PAUSED,
SND_SOC_DPCM_STATE_SUSPEND,
SND_SOC_DPCM_STATE_HW_FREE,
SND_SOC_DPCM_STATE_CLOSE,
};
/*
* Dynamic PCM trigger ordering. Triggering flexibility is required as some
* DSPs require triggering before/after their CPU platform and DAIs.
*
* i.e. some clients may want to manually order this call in their PCM
* trigger() whilst others will just use the regular core ordering.
*/
enum snd_soc_dpcm_trigger {
SND_SOC_DPCM_TRIGGER_PRE = 0,
SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_BESPOKE,
};
/*
* Dynamic PCM link
* This links together a FE and BE DAI at runtime and stores the link
* state information and the hw_params configuration.
*/
struct snd_soc_dpcm {
/* FE and BE DAIs*/
struct snd_soc_pcm_runtime *be;
struct snd_soc_pcm_runtime *fe;
/* link state */
enum snd_soc_dpcm_link_state state;
/* list of BE and FE for this DPCM link */
struct list_head list_be;
struct list_head list_fe;
/* hw params for this link - may be different for each link */
struct snd_pcm_hw_params hw_params;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_state;
#endif
};
/*
* Dynamic PCM runtime data.
*/
struct snd_soc_dpcm_runtime {
struct list_head be_clients;
struct list_head fe_clients;
int users;
struct snd_pcm_runtime *runtime;
struct snd_pcm_hw_params hw_params;
/* state and update */
enum snd_soc_dpcm_update runtime_update;
enum snd_soc_dpcm_state state;
};
/* can this BE stop and free */
int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream);
/* can this BE perform a hw_params() */
int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream);
/* is the current PCM operation for this FE ? */
int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);
/* is the current PCM operation for this BE ? */
int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream);
/* get the substream for this BE */
struct snd_pcm_substream *
snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
/* get the BE runtime state */
enum snd_soc_dpcm_state
snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
/* set the BE runtime state */
void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
enum snd_soc_dpcm_state state);
/* internal use only */
int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *);
#endif

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#ifndef __SOUND_SOUNDFONT_H
#define __SOUND_SOUNDFONT_H
/*
* Soundfont defines and definitions.
*
* Copyright (C) 1999 Steve Ratcliffe
* Copyright (c) 1999-2000 Takashi iwai <tiwai@suse.de>
*
* 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/sfnt_info.h>
#include <sound/util_mem.h>
#define SF_MAX_INSTRUMENTS 128 /* maximum instrument number */
#define SF_MAX_PRESETS 256 /* drums are mapped from 128 to 256 */
#define SF_IS_DRUM_BANK(z) ((z) == 128)
struct snd_sf_zone {
struct snd_sf_zone *next; /* Link to next */
unsigned char bank; /* Midi bank for this zone */
unsigned char instr; /* Midi program for this zone */
unsigned char mapped; /* True if mapped to something else */
struct soundfont_voice_info v; /* All the soundfont parameters */
int counter;
struct snd_sf_sample *sample; /* Link to sample */
/* The following deals with preset numbers (programs) */
struct snd_sf_zone *next_instr; /* Next zone of this instrument */
struct snd_sf_zone *next_zone; /* Next zone in play list */
};
struct snd_sf_sample {
struct soundfont_sample_info v;
int counter;
struct snd_util_memblk *block; /* allocated data block */
struct snd_sf_sample *next;
};
/*
* This represents all the information relating to a soundfont.
*/
struct snd_soundfont {
struct snd_soundfont *next; /* Link to next */
/*struct snd_soundfont *prev;*/ /* Link to previous */
short id; /* file id */
short type; /* font type */
unsigned char name[SNDRV_SFNT_PATCH_NAME_LEN]; /* identifier */
struct snd_sf_zone *zones; /* Font information */
struct snd_sf_sample *samples; /* The sample headers */
};
/*
* Type of the sample access callback
*/
struct snd_sf_callback {
void *private_data;
int (*sample_new)(void *private_data, struct snd_sf_sample *sp,
struct snd_util_memhdr *hdr,
const void __user *buf, long count);
int (*sample_free)(void *private_data, struct snd_sf_sample *sp,
struct snd_util_memhdr *hdr);
void (*sample_reset)(void *private);
};
/*
* List of soundfonts.
*/
struct snd_sf_list {
struct snd_soundfont *currsf; /* The currently open soundfont */
int open_client; /* client pointer for lock */
int mem_used; /* used memory size */
struct snd_sf_zone *presets[SF_MAX_PRESETS];
struct snd_soundfont *fonts; /* The list of soundfonts */
int fonts_size; /* number of fonts allocated */
int zone_counter; /* last allocated time for zone */
int sample_counter; /* last allocated time for sample */
int zone_locked; /* locked time for zone */
int sample_locked; /* locked time for sample */
struct snd_sf_callback callback; /* callback functions */
int presets_locked;
struct mutex presets_mutex;
spinlock_t lock;
struct snd_util_memhdr *memhdr;
};
/* Prototypes for soundfont.c */
int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
long count, int client);
int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data,
long count, int client);
int snd_soundfont_close_check(struct snd_sf_list *sflist, int client);
struct snd_sf_list *snd_sf_new(struct snd_sf_callback *callback,
struct snd_util_memhdr *hdr);
void snd_sf_free(struct snd_sf_list *sflist);
int snd_soundfont_remove_samples(struct snd_sf_list *sflist);
int snd_soundfont_remove_unlocked(struct snd_sf_list *sflist);
int snd_soundfont_search_zone(struct snd_sf_list *sflist, int *notep, int vel,
int preset, int bank,
int def_preset, int def_bank,
struct snd_sf_zone **table, int max_layers);
/* Parameter conversions */
int snd_sf_calc_parm_hold(int msec);
int snd_sf_calc_parm_attack(int msec);
int snd_sf_calc_parm_decay(int msec);
#define snd_sf_calc_parm_delay(msec) (0x8000 - (msec) * 1000 / 725)
extern int snd_sf_vol_table[128];
int snd_sf_linear_to_log(unsigned int amount, int offset, int ratio);
#endif /* __SOUND_SOUNDFONT_H */

35
include/sound/spear_dma.h Normal file
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/*
* linux/spear_dma.h
*
* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com)
*
* 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
*
*/
#ifndef SPEAR_DMA_H
#define SPEAR_DMA_H
#include <linux/dmaengine.h>
struct spear_dma_data {
void *data;
dma_addr_t addr;
u32 max_burst;
enum dma_slave_buswidth addr_width;
bool (*filter)(struct dma_chan *chan, void *slave);
};
#endif /* SPEAR_DMA_H */

29
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/*
* Copyright (ST) 2012 Vipin Kumar (vipin.kumar@st.com)
*
* 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
*/
#ifndef __SOUND_SPDIF_H
#define __SOUND_SPDIF_H
struct spear_spdif_platform_data {
/* DMA params */
void *dma_params;
bool (*filter)(struct dma_chan *chan, void *slave);
void (*reset_perip)(void);
};
#endif /* SOUND_SPDIF_H */

35
include/sound/sta32x.h Normal file
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/*
* Platform data for ST STA32x ASoC codec driver.
*
* Copyright: 2011 Raumfeld GmbH
* Author: Johannes Stezenbach <js@sig21.net>
*
* 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.
*/
#ifndef __LINUX_SND__STA32X_H
#define __LINUX_SND__STA32X_H
#define STA32X_OCFG_2CH 0
#define STA32X_OCFG_2_1CH 1
#define STA32X_OCFG_1CH 3
#define STA32X_OM_CH1 0
#define STA32X_OM_CH2 1
#define STA32X_OM_CH3 2
#define STA32X_THERMAL_ADJUSTMENT_ENABLE 1
#define STA32X_THERMAL_RECOVERY_ENABLE 2
struct sta32x_platform_data {
int output_conf;
int ch1_output_mapping;
int ch2_output_mapping;
int ch3_output_mapping;
int thermal_conf;
int needs_esd_watchdog;
};
#endif /* __LINUX_SND__STA32X_H */

7
include/sound/tas5086.h Normal file
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#ifndef _SND_SOC_CODEC_TAS5086_H_
#define _SND_SOC_CODEC_TAS5086_H_
#define TAS5086_CLK_IDX_MCLK 0
#define TAS5086_CLK_IDX_SCLK 1
#endif /* _SND_SOC_CODEC_TAS5086_H_ */

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include/sound/tea575x-tuner.h Normal file
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#ifndef __SOUND_TEA575X_TUNER_H
#define __SOUND_TEA575X_TUNER_H
/*
* ALSA driver for TEA5757/5759 Philips AM/FM tuner chips
*
* Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
*
* 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 <linux/videodev2.h>
#include <media/v4l2-ctrls.h>
#include <media/v4l2-dev.h>
#include <media/v4l2-device.h>
#define TEA575X_FMIF 10700
#define TEA575X_AMIF 450
#define TEA575X_DATA (1 << 0)
#define TEA575X_CLK (1 << 1)
#define TEA575X_WREN (1 << 2)
#define TEA575X_MOST (1 << 3)
struct snd_tea575x;
struct snd_tea575x_ops {
/* Drivers using snd_tea575x must either define read_ and write_val */
void (*write_val)(struct snd_tea575x *tea, u32 val);
u32 (*read_val)(struct snd_tea575x *tea);
/* Or define the 3 pin functions */
void (*set_pins)(struct snd_tea575x *tea, u8 pins);
u8 (*get_pins)(struct snd_tea575x *tea);
void (*set_direction)(struct snd_tea575x *tea, bool output);
};
struct snd_tea575x {
struct v4l2_device *v4l2_dev;
struct v4l2_file_operations fops;
struct video_device vd; /* video device */
int radio_nr; /* radio_nr */
bool tea5759; /* 5759 chip is present */
bool has_am; /* Device can tune to AM freqs */
bool cannot_read_data; /* Device cannot read the data pin */
bool cannot_mute; /* Device cannot mute */
bool mute; /* Device is muted? */
bool stereo; /* receiving stereo */
bool tuned; /* tuned to a station */
unsigned int val; /* hw value */
u32 band; /* 0: FM, 1: FM-Japan, 2: AM */
u32 freq; /* frequency */
struct mutex mutex;
struct snd_tea575x_ops *ops;
void *private_data;
u8 card[32];
u8 bus_info[32];
struct v4l2_ctrl_handler ctrl_handler;
int (*ext_init)(struct snd_tea575x *tea);
};
int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner);
void snd_tea575x_exit(struct snd_tea575x *tea);
void snd_tea575x_set_freq(struct snd_tea575x *tea);
#endif /* __SOUND_TEA575X_TUNER_H */

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#ifndef __SOUND_TEA6330T_H
#define __SOUND_TEA6330T_H
/*
* Routines for control of TEA6330T circuit.
* Sound fader control circuit for car radios.
*
* 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/i2c.h> /* generic i2c support */
int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer);
int snd_tea6330t_update_mixer(struct snd_card *card, struct snd_i2c_bus *bus,
int equalizer, int fader);
#endif /* __SOUND_TEA6330T_H */

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#ifndef __SOUND_TIMER_H
#define __SOUND_TIMER_H
/*
* Timer abstract layer
* Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Abramo Bagnara <abramo@alsa-project.org>
*
*
* 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/asound.h>
#include <linux/interrupt.h>
#define snd_timer_chip(timer) ((timer)->private_data)
#define SNDRV_TIMER_DEVICES 16
#define SNDRV_TIMER_DEV_FLG_PCM 0x10000000
#define SNDRV_TIMER_HW_AUTO 0x00000001 /* auto trigger is supported */
#define SNDRV_TIMER_HW_STOP 0x00000002 /* call stop before start */
#define SNDRV_TIMER_HW_SLAVE 0x00000004 /* only slave timer (variable resolution) */
#define SNDRV_TIMER_HW_FIRST 0x00000008 /* first tick can be incomplete */
#define SNDRV_TIMER_HW_TASKLET 0x00000010 /* timer is called from tasklet */
#define SNDRV_TIMER_IFLG_SLAVE 0x00000001
#define SNDRV_TIMER_IFLG_RUNNING 0x00000002
#define SNDRV_TIMER_IFLG_START 0x00000004
#define SNDRV_TIMER_IFLG_AUTO 0x00000008 /* auto restart */
#define SNDRV_TIMER_IFLG_FAST 0x00000010 /* fast callback (do not use tasklet) */
#define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */
#define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */
#define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */
#define SNDRV_TIMER_FLG_CHANGE 0x00000001
#define SNDRV_TIMER_FLG_RESCHED 0x00000002 /* need reschedule */
struct snd_timer;
struct snd_timer_hardware {
/* -- must be filled with low-level driver */
unsigned int flags; /* various flags */
unsigned long resolution; /* average timer resolution for one tick in nsec */
unsigned long resolution_min; /* minimal resolution */
unsigned long resolution_max; /* maximal resolution */
unsigned long ticks; /* max timer ticks per interrupt */
/* -- low-level functions -- */
int (*open) (struct snd_timer * timer);
int (*close) (struct snd_timer * timer);
unsigned long (*c_resolution) (struct snd_timer * timer);
int (*start) (struct snd_timer * timer);
int (*stop) (struct snd_timer * timer);
int (*set_period) (struct snd_timer * timer, unsigned long period_num, unsigned long period_den);
int (*precise_resolution) (struct snd_timer * timer, unsigned long *num, unsigned long *den);
};
struct snd_timer {
int tmr_class;
struct snd_card *card;
struct module *module;
int tmr_device;
int tmr_subdevice;
char id[64];
char name[80];
unsigned int flags;
int running; /* running instances */
unsigned long sticks; /* schedule ticks */
void *private_data;
void (*private_free) (struct snd_timer *timer);
struct snd_timer_hardware hw;
spinlock_t lock;
struct list_head device_list;
struct list_head open_list_head;
struct list_head active_list_head;
struct list_head ack_list_head;
struct list_head sack_list_head; /* slow ack list head */
struct tasklet_struct task_queue;
};
struct snd_timer_instance {
struct snd_timer *timer;
char *owner;
unsigned int flags;
void *private_data;
void (*private_free) (struct snd_timer_instance *ti);
void (*callback) (struct snd_timer_instance *timeri,
unsigned long ticks, unsigned long resolution);
void (*ccallback) (struct snd_timer_instance * timeri,
int event,
struct timespec * tstamp,
unsigned long resolution);
void *callback_data;
unsigned long ticks; /* auto-load ticks when expired */
unsigned long cticks; /* current ticks */
unsigned long pticks; /* accumulated ticks for callback */
unsigned long resolution; /* current resolution for tasklet */
unsigned long lost; /* lost ticks */
int slave_class;
unsigned int slave_id;
struct list_head open_list;
struct list_head active_list;
struct list_head ack_list;
struct list_head slave_list_head;
struct list_head slave_active_head;
struct snd_timer_instance *master;
};
/*
* Registering
*/
int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, struct snd_timer **rtimer);
void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstamp);
int snd_timer_global_new(char *id, int device, struct snd_timer **rtimer);
int snd_timer_global_free(struct snd_timer *timer);
int snd_timer_global_register(struct snd_timer *timer);
int snd_timer_open(struct snd_timer_instance **ti, char *owner, struct snd_timer_id *tid, unsigned int slave_id);
int snd_timer_close(struct snd_timer_instance *timeri);
unsigned long snd_timer_resolution(struct snd_timer_instance *timeri);
int snd_timer_start(struct snd_timer_instance *timeri, unsigned int ticks);
int snd_timer_stop(struct snd_timer_instance *timeri);
int snd_timer_continue(struct snd_timer_instance *timeri);
int snd_timer_pause(struct snd_timer_instance *timeri);
void snd_timer_interrupt(struct snd_timer *timer, unsigned long ticks_left);
#endif /* __SOUND_TIMER_H */

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#ifndef __SOUND_TLV_H
#define __SOUND_TLV_H
/*
* Advanced Linux Sound Architecture - ALSA - Driver
* Copyright (c) 2006 by Jaroslav Kysela <perex@perex.cz>
*
*
* 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
*
*/
/*
* TLV structure is right behind the struct snd_ctl_tlv:
* unsigned int type - see SNDRV_CTL_TLVT_*
* unsigned int length
* .... data aligned to sizeof(unsigned int), use
* block_length = (length + (sizeof(unsigned int) - 1)) &
* ~(sizeof(unsigned int) - 1)) ....
*/
#define SNDRV_CTL_TLVT_CONTAINER 0 /* one level down - group of TLVs */
#define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */
#define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */
#define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */
#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */
#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */
#define TLV_ITEM(type, ...) \
(type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
#define TLV_LENGTH(...) \
((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
#define TLV_CONTAINER_ITEM(...) \
TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
#define DECLARE_TLV_CONTAINER(name, ...) \
unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) }
#define TLV_DB_SCALE_MASK 0xffff
#define TLV_DB_SCALE_MUTE 0x10000
#define TLV_DB_SCALE_ITEM(min, step, mute) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \
(min), \
((step) & TLV_DB_SCALE_MASK) | \
((mute) ? TLV_DB_SCALE_MUTE : 0))
#define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
/* dB scale specified with min/max values instead of step */
#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) }
/* linear volume between min_dB and max_dB (.01dB unit) */
#define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
#define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB) \
unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) }
/* dB range container:
* Items in dB range container must be ordered by their values and by their
* dB values. This implies that larger values must correspond with larger
* dB values (which is also required for all other mixer controls).
*/
/* Each item is: <min> <max> <TLV> */
#define TLV_DB_RANGE_ITEM(...) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
#define DECLARE_TLV_DB_RANGE(name, ...) \
unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) }
/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */
#define TLV_DB_RANGE_HEAD(num) \
SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int)
#define TLV_DB_GAIN_MUTE -9999999
/*
* channel-mapping TLV items
* TLV length must match with num_channels
*/
#define SNDRV_CTL_TLVT_CHMAP_FIXED 0x101 /* fixed channel position */
#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */
#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */
#endif /* __SOUND_TLV_H */

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/*
* tlv320aic32x4.h -- TLV320AIC32X4 Soc Audio driver platform data
*
* Copyright 2011 Vista Silicon S.L.
*
* Author: Javier Martin <javier.martin@vista-silicon.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef _AIC32X4_PDATA_H
#define _AIC32X4_PDATA_H
#define AIC32X4_PWR_MICBIAS_2075_LDOIN 0x00000001
#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE 0x00000002
#define AIC32X4_PWR_AIC32X4_LDO_ENABLE 0x00000004
#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36 0x00000008
#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED 0x00000010
#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K 0x00000001
#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K 0x00000002
struct aic32x4_pdata {
u32 power_cfg;
u32 micpga_routing;
bool swapdacs;
int rstn_gpio;
};
#endif

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/*
* Platform data for Texas Instruments TLV320AIC3x codec
*
* Author: Jarkko Nikula <jarkko.nikula@bitmer.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __TLV320AIC3x_H__
#define __TLV320AIC3x_H__
/* GPIO API */
enum {
AIC3X_GPIO1_FUNC_DISABLED = 0,
AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1,
AIC3X_GPIO1_FUNC_CLOCK_MUX = 2,
AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3,
AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4,
AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5,
AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6,
AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7,
AIC3X_GPIO1_FUNC_INPUT = 8,
AIC3X_GPIO1_FUNC_OUTPUT = 9,
AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10,
AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11,
AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12,
AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13,
AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14,
AIC3X_GPIO1_FUNC_ALL_IRQ = 16
};
enum {
AIC3X_GPIO2_FUNC_DISABLED = 0,
AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2,
AIC3X_GPIO2_FUNC_INPUT = 3,
AIC3X_GPIO2_FUNC_OUTPUT = 4,
AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5,
AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8,
AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9,
AIC3X_GPIO2_FUNC_ALL_IRQ = 10,
AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11,
AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12,
AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13,
AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14,
AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15
};
enum aic3x_micbias_voltage {
AIC3X_MICBIAS_OFF = 0,
AIC3X_MICBIAS_2_0V = 1,
AIC3X_MICBIAS_2_5V = 2,
AIC3X_MICBIAS_AVDDV = 3,
};
struct aic3x_setup_data {
unsigned int gpio_func[2];
};
struct aic3x_pdata {
int gpio_reset; /* < 0 if not used */
struct aic3x_setup_data *setup;
/* Selects the micbias voltage */
enum aic3x_micbias_voltage micbias_vg;
};
#endif

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/*
* Platform header for Texas Instruments TLV320DAC33 codec driver
*
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* Copyright: (C) 2009 Nokia Corporation
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __TLV320DAC33_PLAT_H
#define __TLV320DAC33_PLAT_H
struct tlv320dac33_platform_data {
int power_gpio;
int mode1_latency; /* latency caused by the i2c writes in us */
int auto_fifo_config; /* FIFO config based on the period size */
int keep_bclk; /* Keep the BCLK running in FIFO modes */
u8 burst_bclkdiv;
};
#endif /* __TLV320DAC33_PLAT_H */

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/*
* TPA6130A2 driver platform header
*
* Copyright (C) Nokia Corporation
*
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* 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., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef TPA6130A2_PLAT_H
#define TPA6130A2_PLAT_H
struct tpa6130a2_platform_data {
int power_gpio;
};
#endif

39
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/*
* uda134x.h -- UDA134x ALSA SoC Codec driver
*
* Copyright 2007 Dension Audio Systems Ltd.
* Author: Zoltan Devai
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef _UDA134X_H
#define _UDA134X_H
#include <sound/l3.h>
struct uda134x_platform_data {
struct l3_pins l3;
void (*power) (int);
int model;
/*
ALSA SOC usually puts the device in standby mode when it's not used
for sometime. If you unset is_powered_on_standby the driver will
turn off the ADC/DAC when this callback is invoked and turn it back
on when needed. Unfortunately this will result in a very light bump
(it can be audible only with good earphones). If this bothers you
set is_powered_on_standby, you will have slightly higher power
consumption. Please note that sending the L3 command for ADC is
enough to make the bump, so it doesn't make difference if you
completely take off power from the codec.
*/
int is_powered_on_standby;
#define UDA134X_UDA1340 1
#define UDA134X_UDA1341 2
#define UDA134X_UDA1344 3
#define UDA134X_UDA1345 4
};
#endif /* _UDA134X_H */

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/*
* UDA1380 ALSA SoC Codec driver
*
* Copyright 2009 Philipp Zabel
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __UDA1380_H
#define __UDA1380_H
struct uda1380_platform_data {
int gpio_power;
int gpio_reset;
int dac_clk;
#define UDA1380_DAC_CLK_SYSCLK 0
#define UDA1380_DAC_CLK_WSPLL 1
};
#endif /* __UDA1380_H */

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#ifndef __SOUND_UTIL_MEM_H
#define __SOUND_UTIL_MEM_H
#include <linux/mutex.h>
/*
* Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
*
* Generic memory management routines for soundcard memory allocation
*
* 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
*/
/*
* memory block
*/
struct snd_util_memblk {
unsigned int size; /* size of this block */
unsigned int offset; /* zero-offset of this block */
struct list_head list; /* link */
};
#define snd_util_memblk_argptr(blk) (void*)((char*)(blk) + sizeof(struct snd_util_memblk))
/*
* memory management information
*/
struct snd_util_memhdr {
unsigned int size; /* size of whole data */
struct list_head block; /* block linked-list header */
int nblocks; /* # of allocated blocks */
unsigned int used; /* used memory size */
int block_extra_size; /* extra data size of chunk */
struct mutex block_mutex; /* lock */
};
/*
* prototypes
*/
struct snd_util_memhdr *snd_util_memhdr_new(int memsize);
void snd_util_memhdr_free(struct snd_util_memhdr *hdr);
struct snd_util_memblk *snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size);
int snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk);
int snd_util_mem_avail(struct snd_util_memhdr *hdr);
/* functions without mutex */
struct snd_util_memblk *__snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size);
void __snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk);
struct snd_util_memblk *__snd_util_memblk_new(struct snd_util_memhdr *hdr,
unsigned int units,
struct list_head *prev);
#endif /* __SOUND_UTIL_MEM_H */

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/*
* Driver for Digigram VX soundcards
*
* Hardware core part
*
* Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
*
* 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
*/
#ifndef __SOUND_VX_COMMON_H
#define __SOUND_VX_COMMON_H
#include <sound/pcm.h>
#include <sound/hwdep.h>
#include <linux/interrupt.h>
struct firmware;
struct device;
#define VX_DRIVER_VERSION 0x010000 /* 1.0.0 */
/*
*/
#define SIZE_MAX_CMD 0x10
#define SIZE_MAX_STATUS 0x10
struct vx_rmh {
u16 LgCmd; /* length of the command to send (WORDs) */
u16 LgStat; /* length of the status received (WORDs) */
u32 Cmd[SIZE_MAX_CMD];
u32 Stat[SIZE_MAX_STATUS];
u16 DspStat; /* status type, RMP_SSIZE_XXX */
};
typedef u64 pcx_time_t;
#define VX_MAX_PIPES 16
#define VX_MAX_PERIODS 32
#define VX_MAX_CODECS 2
struct vx_ibl_info {
int size; /* the current IBL size (0 = query) in bytes */
int max_size; /* max. IBL size in bytes */
int min_size; /* min. IBL size in bytes */
int granularity; /* granularity */
};
struct vx_pipe {
int number;
unsigned int is_capture: 1;
unsigned int data_mode: 1;
unsigned int running: 1;
unsigned int prepared: 1;
int channels;
unsigned int differed_type;
pcx_time_t pcx_time;
struct snd_pcm_substream *substream;
int hbuf_size; /* H-buffer size in bytes */
int buffer_bytes; /* the ALSA pcm buffer size in bytes */
int period_bytes; /* the ALSA pcm period size in bytes */
int hw_ptr; /* the current hardware pointer in bytes */
int position; /* the current position in frames (playback only) */
int transferred; /* the transferred size (per period) in frames */
int align; /* size of alignment */
u64 cur_count; /* current sample position (for playback) */
unsigned int references; /* an output pipe may be used for monitoring and/or playback */
struct vx_pipe *monitoring_pipe; /* pointer to the monitoring pipe (capture pipe only)*/
struct tasklet_struct start_tq;
};
struct vx_core;
struct snd_vx_ops {
/* low-level i/o */
unsigned char (*in8)(struct vx_core *chip, int reg);
unsigned int (*in32)(struct vx_core *chip, int reg);
void (*out8)(struct vx_core *chip, int reg, unsigned char val);
void (*out32)(struct vx_core *chip, int reg, unsigned int val);
/* irq */
int (*test_and_ack)(struct vx_core *chip);
void (*validate_irq)(struct vx_core *chip, int enable);
/* codec */
void (*write_codec)(struct vx_core *chip, int codec, unsigned int data);
void (*akm_write)(struct vx_core *chip, int reg, unsigned int data);
void (*reset_codec)(struct vx_core *chip);
void (*change_audio_source)(struct vx_core *chip, int src);
void (*set_clock_source)(struct vx_core *chp, int src);
/* chip init */
int (*load_dsp)(struct vx_core *chip, int idx, const struct firmware *fw);
void (*reset_dsp)(struct vx_core *chip);
void (*reset_board)(struct vx_core *chip, int cold_reset);
int (*add_controls)(struct vx_core *chip);
/* pcm */
void (*dma_write)(struct vx_core *chip, struct snd_pcm_runtime *runtime,
struct vx_pipe *pipe, int count);
void (*dma_read)(struct vx_core *chip, struct snd_pcm_runtime *runtime,
struct vx_pipe *pipe, int count);
};
struct snd_vx_hardware {
const char *name;
int type; /* VX_TYPE_XXX */
/* hardware specs */
unsigned int num_codecs;
unsigned int num_ins;
unsigned int num_outs;
unsigned int output_level_max;
const unsigned int *output_level_db_scale;
};
/* hwdep id string */
#define SND_VX_HWDEP_ID "VX Loader"
/* hardware type */
enum {
/* VX222 PCI */
VX_TYPE_BOARD, /* old VX222 PCI */
VX_TYPE_V2, /* VX222 V2 PCI */
VX_TYPE_MIC, /* VX222 Mic PCI */
/* VX-pocket */
VX_TYPE_VXPOCKET, /* VXpocket V2 */
VX_TYPE_VXP440, /* VXpocket 440 */
VX_TYPE_NUMS
};
/* chip status */
enum {
VX_STAT_XILINX_LOADED = (1 << 0), /* devices are registered */
VX_STAT_DEVICE_INIT = (1 << 1), /* devices are registered */
VX_STAT_CHIP_INIT = (1 << 2), /* all operational */
VX_STAT_IN_SUSPEND = (1 << 10), /* in suspend phase */
VX_STAT_IS_STALE = (1 << 15) /* device is stale */
};
/* min/max values for analog output for old codecs */
#define VX_ANALOG_OUT_LEVEL_MAX 0xe3
struct vx_core {
/* ALSA stuff */
struct snd_card *card;
struct snd_pcm *pcm[VX_MAX_CODECS];
int type; /* VX_TYPE_XXX */
int irq;
/* ports are defined externally */
/* low-level functions */
struct snd_vx_hardware *hw;
struct snd_vx_ops *ops;
spinlock_t lock;
spinlock_t irq_lock;
struct tasklet_struct tq;
unsigned int chip_status;
unsigned int pcm_running;
struct device *dev;
struct snd_hwdep *hwdep;
struct vx_rmh irq_rmh; /* RMH used in interrupts */
unsigned int audio_info; /* see VX_AUDIO_INFO */
unsigned int audio_ins;
unsigned int audio_outs;
struct vx_pipe **playback_pipes;
struct vx_pipe **capture_pipes;
/* clock and audio sources */
unsigned int audio_source; /* current audio input source */
unsigned int audio_source_target;
unsigned int clock_mode; /* clock mode (VX_CLOCK_MODE_XXX) */
unsigned int clock_source; /* current clock source (INTERNAL_QUARTZ or UER_SYNC) */
unsigned int freq; /* current frequency */
unsigned int freq_detected; /* detected frequency from digital in */
unsigned int uer_detected; /* VX_UER_MODE_XXX */
unsigned int uer_bits; /* IEC958 status bits */
struct vx_ibl_info ibl; /* IBL information */
/* mixer setting */
int output_level[VX_MAX_CODECS][2]; /* analog output level */
int audio_gain[2][4]; /* digital audio level (playback/capture) */
unsigned char audio_active[4]; /* mute/unmute on digital playback */
int audio_monitor[4]; /* playback hw-monitor level */
unsigned char audio_monitor_active[4]; /* playback hw-monitor mute/unmute */
struct mutex mixer_mutex;
const struct firmware *firmware[4]; /* loaded firmware data */
};
/*
* constructor
*/
struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw,
struct snd_vx_ops *ops, int extra_size);
int snd_vx_setup_firmware(struct vx_core *chip);
int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *dsp);
int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *dsp);
int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp);
void snd_vx_free_firmware(struct vx_core *chip);
/*
* interrupt handler; exported for pcmcia
*/
irqreturn_t snd_vx_irq_handler(int irq, void *dev);
/*
* lowlevel functions
*/
static inline int vx_test_and_ack(struct vx_core *chip)
{
return chip->ops->test_and_ack(chip);
}
static inline void vx_validate_irq(struct vx_core *chip, int enable)
{
chip->ops->validate_irq(chip, enable);
}
static inline unsigned char snd_vx_inb(struct vx_core *chip, int reg)
{
return chip->ops->in8(chip, reg);
}
static inline unsigned int snd_vx_inl(struct vx_core *chip, int reg)
{
return chip->ops->in32(chip, reg);
}
static inline void snd_vx_outb(struct vx_core *chip, int reg, unsigned char val)
{
chip->ops->out8(chip, reg, val);
}
static inline void snd_vx_outl(struct vx_core *chip, int reg, unsigned int val)
{
chip->ops->out32(chip, reg, val);
}
#define vx_inb(chip,reg) snd_vx_inb(chip, VX_##reg)
#define vx_outb(chip,reg,val) snd_vx_outb(chip, VX_##reg,val)
#define vx_inl(chip,reg) snd_vx_inl(chip, VX_##reg)
#define vx_outl(chip,reg,val) snd_vx_outl(chip, VX_##reg,val)
static inline void vx_reset_dsp(struct vx_core *chip)
{
chip->ops->reset_dsp(chip);
}
int vx_send_msg(struct vx_core *chip, struct vx_rmh *rmh);
int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh);
int vx_send_rih(struct vx_core *chip, int cmd);
int vx_send_rih_nolock(struct vx_core *chip, int cmd);
void vx_reset_codec(struct vx_core *chip, int cold_reset);
/*
* check the bit on the specified register
* returns zero if a bit matches, or a negative error code.
* exported for vxpocket driver
*/
int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int time);
#define vx_check_isr(chip,mask,bit,time) snd_vx_check_reg_bit(chip, VX_ISR, mask, bit, time)
#define vx_wait_isr_bit(chip,bit) vx_check_isr(chip, bit, bit, 200)
#define vx_wait_for_rx_full(chip) vx_wait_isr_bit(chip, ISR_RX_FULL)
/*
* pseudo-DMA transfer
*/
static inline void vx_pseudo_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
struct vx_pipe *pipe, int count)
{
chip->ops->dma_write(chip, runtime, pipe, count);
}
static inline void vx_pseudo_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
struct vx_pipe *pipe, int count)
{
chip->ops->dma_read(chip, runtime, pipe, count);
}
/* error with hardware code,
* the return value is -(VX_ERR_MASK | actual-hw-error-code)
*/
#define VX_ERR_MASK 0x1000000
#define vx_get_error(err) (-(err) & ~VX_ERR_MASK)
/*
* pcm stuff
*/
int snd_vx_pcm_new(struct vx_core *chip);
void vx_pcm_update_intr(struct vx_core *chip, unsigned int events);
/*
* mixer stuff
*/
int snd_vx_mixer_new(struct vx_core *chip);
void vx_toggle_dac_mute(struct vx_core *chip, int mute);
int vx_sync_audio_source(struct vx_core *chip);
int vx_set_monitor_level(struct vx_core *chip, int audio, int level, int active);
/*
* IEC958 & clock stuff
*/
void vx_set_iec958_status(struct vx_core *chip, unsigned int bits);
int vx_set_clock(struct vx_core *chip, unsigned int freq);
void vx_set_internal_clock(struct vx_core *chip, unsigned int freq);
int vx_change_frequency(struct vx_core *chip);
/*
* PM
*/
int snd_vx_suspend(struct vx_core *card);
int snd_vx_resume(struct vx_core *card);
/*
* hardware constants
*/
#define vx_has_new_dsp(chip) ((chip)->type != VX_TYPE_BOARD)
#define vx_is_pcmcia(chip) ((chip)->type >= VX_TYPE_VXPOCKET)
/* audio input source */
enum {
VX_AUDIO_SRC_DIGITAL,
VX_AUDIO_SRC_LINE,
VX_AUDIO_SRC_MIC
};
/* clock source */
enum {
INTERNAL_QUARTZ,
UER_SYNC
};
/* clock mode */
enum {
VX_CLOCK_MODE_AUTO, /* depending on the current audio source */
VX_CLOCK_MODE_INTERNAL, /* fixed to internal quartz */
VX_CLOCK_MODE_EXTERNAL /* fixed to UER sync */
};
/* SPDIF/UER type */
enum {
VX_UER_MODE_CONSUMER,
VX_UER_MODE_PROFESSIONAL,
VX_UER_MODE_NOT_PRESENT,
};
/* register indices */
enum {
VX_ICR,
VX_CVR,
VX_ISR,
VX_IVR,
VX_RXH,
VX_TXH = VX_RXH,
VX_RXM,
VX_TXM = VX_RXM,
VX_RXL,
VX_TXL = VX_RXL,
VX_DMA,
VX_CDSP,
VX_RFREQ,
VX_RUER_V2,
VX_GAIN,
VX_DATA = VX_GAIN,
VX_MEMIRQ,
VX_ACQ,
VX_BIT0,
VX_BIT1,
VX_MIC0,
VX_MIC1,
VX_MIC2,
VX_MIC3,
VX_PLX0,
VX_PLX1,
VX_PLX2,
VX_LOFREQ, // V2: ACQ, VP: RFREQ
VX_HIFREQ, // V2: BIT0, VP: RUER_V2
VX_CSUER, // V2: BIT1, VP: BIT0
VX_RUER, // V2: RUER_V2, VP: BIT1
VX_REG_MAX,
/* aliases for VX board */
VX_RESET_DMA = VX_ISR,
VX_CFG = VX_RFREQ,
VX_STATUS = VX_MEMIRQ,
VX_SELMIC = VX_MIC0,
VX_COMPOT = VX_MIC1,
VX_SCOMPR = VX_MIC2,
VX_GLIMIT = VX_MIC3,
VX_INTCSR = VX_PLX0,
VX_CNTRL = VX_PLX1,
VX_GPIOC = VX_PLX2,
/* aliases for VXPOCKET board */
VX_MICRO = VX_MEMIRQ,
VX_CODEC2 = VX_MEMIRQ,
VX_DIALOG = VX_ACQ,
};
/* RMH status type */
enum {
RMH_SSIZE_FIXED = 0, /* status size given by the driver (in LgStat) */
RMH_SSIZE_ARG = 1, /* status size given in the LSB byte */
RMH_SSIZE_MASK = 2, /* status size given in bitmask */
};
/* bits for ICR register */
#define ICR_HF1 0x10
#define ICR_HF0 0x08
#define ICR_TREQ 0x02 /* Interrupt mode + HREQ set on for transfer (->DSP) request */
#define ICR_RREQ 0x01 /* Interrupt mode + RREQ set on for transfer (->PC) request */
/* bits for CVR register */
#define CVR_HC 0x80
/* bits for ISR register */
#define ISR_HF3 0x10
#define ISR_HF2 0x08
#define ISR_CHK 0x10
#define ISR_ERR 0x08
#define ISR_TX_READY 0x04
#define ISR_TX_EMPTY 0x02
#define ISR_RX_FULL 0x01
/* Constants used to access the DATA register */
#define VX_DATA_CODEC_MASK 0x80
#define VX_DATA_XICOR_MASK 0x80
/* Constants used to access the CSUER register (both for VX2 and VXP) */
#define VX_SUER_FREQ_MASK 0x0c
#define VX_SUER_FREQ_32KHz_MASK 0x0c
#define VX_SUER_FREQ_44KHz_MASK 0x00
#define VX_SUER_FREQ_48KHz_MASK 0x04
#define VX_SUER_DATA_PRESENT_MASK 0x02
#define VX_SUER_CLOCK_PRESENT_MASK 0x01
#define VX_CUER_HH_BITC_SEL_MASK 0x08
#define VX_CUER_MH_BITC_SEL_MASK 0x04
#define VX_CUER_ML_BITC_SEL_MASK 0x02
#define VX_CUER_LL_BITC_SEL_MASK 0x01
#define XX_UER_CBITS_OFFSET_MASK 0x1f
/* bits for audio_info */
#define VX_AUDIO_INFO_REAL_TIME (1<<0) /* real-time processing available */
#define VX_AUDIO_INFO_OFFLINE (1<<1) /* offline processing available */
#define VX_AUDIO_INFO_MPEG1 (1<<5)
#define VX_AUDIO_INFO_MPEG2 (1<<6)
#define VX_AUDIO_INFO_LINEAR_8 (1<<7)
#define VX_AUDIO_INFO_LINEAR_16 (1<<8)
#define VX_AUDIO_INFO_LINEAR_24 (1<<9)
/* DSP Interrupt Request values */
#define VXP_IRQ_OFFSET 0x40 /* add 0x40 offset for vxpocket and vx222/v2 */
/* call with vx_send_irq_dsp() */
#define IRQ_MESS_WRITE_END 0x30
#define IRQ_MESS_WRITE_NEXT 0x32
#define IRQ_MESS_READ_NEXT 0x34
#define IRQ_MESS_READ_END 0x36
#define IRQ_MESSAGE 0x38
#define IRQ_RESET_CHK 0x3A
#define IRQ_CONNECT_STREAM_NEXT 0x26
#define IRQ_CONNECT_STREAM_END 0x28
#define IRQ_PAUSE_START_CONNECT 0x2A
#define IRQ_END_CONNECTION 0x2C
/* Is there async. events pending ( IT Source Test ) */
#define ASYNC_EVENTS_PENDING 0x008000
#define HBUFFER_EVENTS_PENDING 0x004000 // Not always accurate
#define NOTIF_EVENTS_PENDING 0x002000
#define TIME_CODE_EVENT_PENDING 0x001000
#define FREQUENCY_CHANGE_EVENT_PENDING 0x000800
#define END_OF_BUFFER_EVENTS_PENDING 0x000400
#define FATAL_DSP_ERROR 0xff0000
/* Stream Format Header Defines */
#define HEADER_FMT_BASE 0xFED00000
#define HEADER_FMT_MONO 0x000000C0
#define HEADER_FMT_INTEL 0x00008000
#define HEADER_FMT_16BITS 0x00002000
#define HEADER_FMT_24BITS 0x00004000
#define HEADER_FMT_UPTO11 0x00000200 /* frequency is less or equ. to 11k.*/
#define HEADER_FMT_UPTO32 0x00000100 /* frequency is over 11k and less then 32k.*/
/* Constants used to access the Codec */
#define XX_CODEC_SELECTOR 0x20
/* codec commands */
#define XX_CODEC_ADC_CONTROL_REGISTER 0x01
#define XX_CODEC_DAC_CONTROL_REGISTER 0x02
#define XX_CODEC_LEVEL_LEFT_REGISTER 0x03
#define XX_CODEC_LEVEL_RIGHT_REGISTER 0x04
#define XX_CODEC_PORT_MODE_REGISTER 0x05
#define XX_CODEC_STATUS_REPORT_REGISTER 0x06
#define XX_CODEC_CLOCK_CONTROL_REGISTER 0x07
/*
* Audio-level control values
*/
#define CVAL_M110DB 0x000 /* -110dB */
#define CVAL_M99DB 0x02C
#define CVAL_M21DB 0x163
#define CVAL_M18DB 0x16F
#define CVAL_M10DB 0x18F
#define CVAL_0DB 0x1B7
#define CVAL_18DB 0x1FF /* +18dB */
#define CVAL_MAX 0x1FF
#define AUDIO_IO_HAS_MUTE_LEVEL 0x400000
#define AUDIO_IO_HAS_MUTE_MONITORING_1 0x200000
#define AUDIO_IO_HAS_MUTE_MONITORING_2 0x100000
#define VALID_AUDIO_IO_DIGITAL_LEVEL 0x01
#define VALID_AUDIO_IO_MONITORING_LEVEL 0x02
#define VALID_AUDIO_IO_MUTE_LEVEL 0x04
#define VALID_AUDIO_IO_MUTE_MONITORING_1 0x08
#define VALID_AUDIO_IO_MUTE_MONITORING_2 0x10
#endif /* __SOUND_VX_COMMON_H */

695
include/sound/wavefront.h Normal file
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#ifndef __SOUND_WAVEFRONT_H__
#define __SOUND_WAVEFRONT_H__
/*
* Driver for Turtle Beach Wavefront cards (Maui,Tropez,Tropez+)
*
* Copyright (c) by Paul Barton-Davis <pbd@op.net>
*
* 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
*/
#if (!defined(__GNUC__) && !defined(__GNUG__))
You will not be able to compile this file correctly without gcc, because
it is necessary to pack the "wavefront_alias" structure to a size
of 22 bytes, corresponding to 16-bit alignment (as would have been
the case on the original platform, MS-DOS). If this is not done,
then WavePatch-format files cannot be read/written correctly.
The method used to do this here ("__attribute__((packed)") is
completely compiler dependent.
All other wavefront_* types end up aligned to 32 bit values and
still have the same (correct) size.
#else
/* However, note that as of G++ 2.7.3.2, g++ was unable to
correctly parse *type* __attribute__ tags. It will do the
right thing if we use the "packed" attribute on each struct
member, which has the same semantics anyway.
*/
#endif /* __GNUC__ */
/***************************** WARNING ********************************
PLEASE DO NOT MODIFY THIS FILE IN ANY WAY THAT AFFECTS ITS ABILITY TO
BE USED WITH EITHER C *OR* C++.
**********************************************************************/
#ifndef NUM_MIDIKEYS
#define NUM_MIDIKEYS 128
#endif /* NUM_MIDIKEYS */
#ifndef NUM_MIDICHANNELS
#define NUM_MIDICHANNELS 16
#endif /* NUM_MIDICHANNELS */
/* These are very useful/important. the original wavefront interface
was developed on a 16 bit system, where sizeof(int) = 2
bytes. Defining things like this makes the code much more portable, and
easier to understand without having to toggle back and forth
between a 16-bit view of the world and a 32-bit one.
*/
#ifndef __KERNEL__
/* keep them for compatibility */
typedef short s16;
typedef unsigned short u16;
typedef int s32;
typedef unsigned int u32;
typedef char s8;
typedef unsigned char u8;
typedef s16 INT16;
typedef u16 UINT16;
typedef s32 INT32;
typedef u32 UINT32;
typedef s8 CHAR8;
typedef u8 UCHAR8;
#endif
/* Pseudo-commands not part of the WaveFront command set.
These are used for various driver controls and direct
hardware control.
*/
#define WFC_DEBUG_DRIVER 0
#define WFC_FX_IOCTL 1
#define WFC_PATCH_STATUS 2
#define WFC_PROGRAM_STATUS 3
#define WFC_SAMPLE_STATUS 4
#define WFC_DISABLE_INTERRUPTS 5
#define WFC_ENABLE_INTERRUPTS 6
#define WFC_INTERRUPT_STATUS 7
#define WFC_ROMSAMPLES_RDONLY 8
#define WFC_IDENTIFY_SLOT_TYPE 9
/* Wavefront synth commands
*/
#define WFC_DOWNLOAD_SAMPLE 0x80
#define WFC_DOWNLOAD_BLOCK 0x81
#define WFC_DOWNLOAD_MULTISAMPLE 0x82
#define WFC_DOWNLOAD_SAMPLE_ALIAS 0x83
#define WFC_DELETE_SAMPLE 0x84
#define WFC_REPORT_FREE_MEMORY 0x85
#define WFC_DOWNLOAD_PATCH 0x86
#define WFC_DOWNLOAD_PROGRAM 0x87
#define WFC_SET_SYNTHVOL 0x89
#define WFC_SET_NVOICES 0x8B
#define WFC_DOWNLOAD_DRUM 0x90
#define WFC_GET_SYNTHVOL 0x92
#define WFC_GET_NVOICES 0x94
#define WFC_DISABLE_CHANNEL 0x9A
#define WFC_ENABLE_CHANNEL 0x9B
#define WFC_MISYNTH_OFF 0x9D
#define WFC_MISYNTH_ON 0x9E
#define WFC_FIRMWARE_VERSION 0x9F
#define WFC_GET_NSAMPLES 0xA0
#define WFC_DISABLE_DRUM_PROGRAM 0xA2
#define WFC_UPLOAD_PATCH 0xA3
#define WFC_UPLOAD_PROGRAM 0xA4
#define WFC_SET_TUNING 0xA6
#define WFC_GET_TUNING 0xA7
#define WFC_VMIDI_ON 0xA8
#define WFC_VMIDI_OFF 0xA9
#define WFC_MIDI_STATUS 0xAA
#define WFC_GET_CHANNEL_STATUS 0xAB
#define WFC_DOWNLOAD_SAMPLE_HEADER 0xAC
#define WFC_UPLOAD_SAMPLE_HEADER 0xAD
#define WFC_UPLOAD_MULTISAMPLE 0xAE
#define WFC_UPLOAD_SAMPLE_ALIAS 0xAF
#define WFC_IDENTIFY_SAMPLE_TYPE 0xB0
#define WFC_DOWNLOAD_EDRUM_PROGRAM 0xB1
#define WFC_UPLOAD_EDRUM_PROGRAM 0xB2
#define WFC_SET_EDRUM_CHANNEL 0xB3
#define WFC_INSTOUT_LEVELS 0xB4
#define WFC_PEAKOUT_LEVELS 0xB5
#define WFC_REPORT_CHANNEL_PROGRAMS 0xB6
#define WFC_HARDWARE_VERSION 0xCF
#define WFC_UPLOAD_SAMPLE_PARAMS 0xD7
#define WFC_DOWNLOAD_OS 0xF1
#define WFC_NOOP 0xFF
#define WF_MAX_SAMPLE 512
#define WF_MAX_PATCH 256
#define WF_MAX_PROGRAM 128
#define WF_SECTION_MAX 44 /* longest OS section length */
/* # of bytes we send to the board when sending it various kinds of
substantive data, such as samples, patches and programs.
*/
#define WF_PROGRAM_BYTES 32
#define WF_PATCH_BYTES 132
#define WF_SAMPLE_BYTES 27
#define WF_SAMPLE_HDR_BYTES 25
#define WF_ALIAS_BYTES 25
#define WF_DRUM_BYTES 9
#define WF_MSAMPLE_BYTES 259 /* (MIDI_KEYS * 2) + 3 */
#define WF_ACK 0x80
#define WF_DMA_ACK 0x81
/* OR-values for MIDI status bits */
#define WF_MIDI_VIRTUAL_ENABLED 0x1
#define WF_MIDI_VIRTUAL_IS_EXTERNAL 0x2
#define WF_MIDI_IN_TO_SYNTH_DISABLED 0x4
/* slot indexes for struct address_info: makes code a little more mnemonic */
#define WF_SYNTH_SLOT 0
#define WF_INTERNAL_MIDI_SLOT 1
#define WF_EXTERNAL_MIDI_SLOT 2
/* Magic MIDI bytes used to switch I/O streams on the ICS2115 MPU401
emulation. Note these NEVER show up in output from the device and
should NEVER be used in input unless Virtual MIDI mode has been
disabled. If they do show up as input, the results are unpredictable.
*/
#define WF_EXTERNAL_SWITCH 0xFD
#define WF_INTERNAL_SWITCH 0xF9
/* Debugging flags */
#define WF_DEBUG_CMD 0x1
#define WF_DEBUG_DATA 0x2
#define WF_DEBUG_LOAD_PATCH 0x4
#define WF_DEBUG_IO 0x8
/* WavePatch file format stuff */
#define WF_WAVEPATCH_VERSION 120; /* Current version number (1.2) */
#define WF_MAX_COMMENT 64 /* Comment length */
#define WF_NUM_LAYERS 4
#define WF_NAME_LENGTH 32
#define WF_SOURCE_LENGTH 260
#define BankFileID "Bank"
#define DrumkitFileID "DrumKit"
#define ProgramFileID "Program"
struct wf_envelope
{
u8 attack_time:7;
u8 Unused1:1;
u8 decay1_time:7;
u8 Unused2:1;
u8 decay2_time:7;
u8 Unused3:1;
u8 sustain_time:7;
u8 Unused4:1;
u8 release_time:7;
u8 Unused5:1;
u8 release2_time:7;
u8 Unused6:1;
s8 attack_level;
s8 decay1_level;
s8 decay2_level;
s8 sustain_level;
s8 release_level;
u8 attack_velocity:7;
u8 Unused7:1;
u8 volume_velocity:7;
u8 Unused8:1;
u8 keyboard_scaling:7;
u8 Unused9:1;
};
typedef struct wf_envelope wavefront_envelope;
struct wf_lfo
{
u8 sample_number;
u8 frequency:7;
u8 Unused1:1;
u8 am_src:4;
u8 fm_src:4;
s8 fm_amount;
s8 am_amount;
s8 start_level;
s8 end_level;
u8 ramp_delay:7;
u8 wave_restart:1; /* for LFO2 only */
u8 ramp_time:7;
u8 Unused2:1;
};
typedef struct wf_lfo wavefront_lfo;
struct wf_patch
{
s16 frequency_bias; /* ** THIS IS IN MOTOROLA FORMAT!! ** */
u8 amplitude_bias:7;
u8 Unused1:1;
u8 portamento:7;
u8 Unused2:1;
u8 sample_number;
u8 pitch_bend:4;
u8 sample_msb:1;
u8 Unused3:3;
u8 mono:1;
u8 retrigger:1;
u8 nohold:1;
u8 restart:1;
u8 filterconfig:2; /* SDK says "not used" */
u8 reuse:1;
u8 reset_lfo:1;
u8 fm_src2:4;
u8 fm_src1:4;
s8 fm_amount1;
s8 fm_amount2;
u8 am_src:4;
u8 Unused4:4;
s8 am_amount;
u8 fc1_mode:4;
u8 fc2_mode:4;
s8 fc1_mod_amount;
s8 fc1_keyboard_scaling;
s8 fc1_bias;
s8 fc2_mod_amount;
s8 fc2_keyboard_scaling;
s8 fc2_bias;
u8 randomizer:7;
u8 Unused5:1;
struct wf_envelope envelope1;
struct wf_envelope envelope2;
struct wf_lfo lfo1;
struct wf_lfo lfo2;
};
typedef struct wf_patch wavefront_patch;
struct wf_layer
{
u8 patch_number;
u8 mix_level:7;
u8 mute:1;
u8 split_point:7;
u8 play_below:1;
u8 pan_mod_src:2;
u8 pan_or_mod:1;
u8 pan:4;
u8 split_type:1;
};
typedef struct wf_layer wavefront_layer;
struct wf_program
{
struct wf_layer layer[WF_NUM_LAYERS];
};
typedef struct wf_program wavefront_program;
struct wf_sample_offset
{
s32 Fraction:4;
s32 Integer:20;
s32 Unused:8;
};
typedef struct wf_sample_offset wavefront_sample_offset;
/* Sample slot types */
#define WF_ST_SAMPLE 0
#define WF_ST_MULTISAMPLE 1
#define WF_ST_ALIAS 2
#define WF_ST_EMPTY 3
/* pseudo's */
#define WF_ST_DRUM 4
#define WF_ST_PROGRAM 5
#define WF_ST_PATCH 6
#define WF_ST_SAMPLEHDR 7
#define WF_ST_MASK 0xf
/* Flags for slot status. These occupy the upper bits of the same byte
as a sample type.
*/
#define WF_SLOT_USED 0x80 /* XXX don't rely on this being accurate */
#define WF_SLOT_FILLED 0x40
#define WF_SLOT_ROM 0x20
#define WF_SLOT_MASK 0xf0
/* channel constants */
#define WF_CH_MONO 0
#define WF_CH_LEFT 1
#define WF_CH_RIGHT 2
/* Sample formats */
#define LINEAR_16BIT 0
#define WHITE_NOISE 1
#define LINEAR_8BIT 2
#define MULAW_8BIT 3
#define WF_SAMPLE_IS_8BIT(smpl) ((smpl)->SampleResolution&2)
/*
Because most/all of the sample data we pass in via pointers has
never been copied (just mmap-ed into user space straight from the
disk), it would be nice to allow handling of multi-channel sample
data without forcing user-level extraction of the relevant bytes.
So, we need a way of specifying which channel to use (the WaveFront
only handles mono samples in a given slot), and the only way to do
this without using some struct other than wavefront_sample as the
interface is the awful hack of using the unused bits in a
wavefront_sample:
Val Meaning
--- -------
0 no channel selection (use channel 1, sample is MONO)
1 use first channel, and skip one
2 use second channel, and skip one
3 use third channel, and skip two
4 use fourth channel, skip three
5 use fifth channel, skip four
6 use six channel, skip five
This can handle up to 4 channels, and anyone downloading >4 channels
of sample data just to select one of them needs to find some tools
like sox ...
NOTE: values 0, 1 and 2 correspond to WF_CH_* above. This is
important.
*/
#define WF_SET_CHANNEL(samp,chn) \
(samp)->Unused1 = chn & 0x1; \
(samp)->Unused2 = chn & 0x2; \
(samp)->Unused3 = chn & 0x4
#define WF_GET_CHANNEL(samp) \
(((samp)->Unused3 << 2)|((samp)->Unused2<<1)|(samp)->Unused1)
typedef struct wf_sample {
struct wf_sample_offset sampleStartOffset;
struct wf_sample_offset loopStartOffset;
struct wf_sample_offset loopEndOffset;
struct wf_sample_offset sampleEndOffset;
s16 FrequencyBias;
u8 SampleResolution:2; /* sample_format */
u8 Unused1:1;
u8 Loop:1;
u8 Bidirectional:1;
u8 Unused2:1;
u8 Reverse:1;
u8 Unused3:1;
} wavefront_sample;
typedef struct wf_multisample {
s16 NumberOfSamples; /* log2 of the number of samples */
s16 SampleNumber[NUM_MIDIKEYS];
} wavefront_multisample;
typedef struct wf_alias {
s16 OriginalSample;
struct wf_sample_offset sampleStartOffset;
struct wf_sample_offset loopStartOffset;
struct wf_sample_offset sampleEndOffset;
struct wf_sample_offset loopEndOffset;
s16 FrequencyBias;
u8 SampleResolution:2;
u8 Unused1:1;
u8 Loop:1;
u8 Bidirectional:1;
u8 Unused2:1;
u8 Reverse:1;
u8 Unused3:1;
/* This structure is meant to be padded only to 16 bits on their
original. Of course, whoever wrote their documentation didn't
realize that sizeof(struct) can be >=
sum(sizeof(struct-fields)) and so thought that giving a C level
description of the structs used in WavePatch files was
sufficient. I suppose it was, as long as you remember the
standard 16->32 bit issues.
*/
u8 sixteen_bit_padding;
} __attribute__((packed)) wavefront_alias;
typedef struct wf_drum {
u8 PatchNumber;
u8 MixLevel:7;
u8 Unmute:1;
u8 Group:4;
u8 Unused1:4;
u8 PanModSource:2;
u8 PanModulated:1;
u8 PanAmount:4;
u8 Unused2:1;
} wavefront_drum;
typedef struct wf_drumkit {
struct wf_drum drum[NUM_MIDIKEYS];
} wavefront_drumkit;
typedef struct wf_channel_programs {
u8 Program[NUM_MIDICHANNELS];
} wavefront_channel_programs;
/* How to get MIDI channel status from the data returned by
a WFC_GET_CHANNEL_STATUS command (a struct wf_channel_programs)
*/
#define WF_CHANNEL_STATUS(ch,wcp) (wcp)[(ch/7)] & (1<<((ch)%7))
typedef union wf_any {
wavefront_sample s;
wavefront_multisample ms;
wavefront_alias a;
wavefront_program pr;
wavefront_patch p;
wavefront_drum d;
} wavefront_any;
/* Hannu Solvainen hoped that his "patch_info" struct in soundcard.h
might work for other wave-table based patch loading situations.
Alas, his fears were correct. The WaveFront doesn't even come with
just "patches", but several different kind of structures that
control the sound generation process.
*/
typedef struct wf_patch_info {
/* the first two fields are used by the OSS "patch loading" interface
only, and are unused by the current user-level library.
*/
s16 key; /* Use WAVEFRONT_PATCH here */
u16 devno; /* fill in when sending */
u8 subkey; /* WF_ST_{SAMPLE,ALIAS,etc.} */
#define WAVEFRONT_FIND_FREE_SAMPLE_SLOT 999
u16 number; /* patch/sample/prog number */
u32 size; /* size of any data included in
one of the fields in `hdrptr', or
as `dataptr'.
NOTE: for actual samples, this is
the size of the *SELECTED CHANNEL*
even if more data is actually available.
So, a stereo sample (2 channels) of
6000 bytes total has `size' = 3000.
See the macros and comments for
WF_{GET,SET}_CHANNEL above.
*/
wavefront_any __user *hdrptr; /* user-space ptr to hdr bytes */
u16 __user *dataptr; /* actual sample data */
wavefront_any hdr; /* kernel-space copy of hdr bytes */
} wavefront_patch_info;
/* The maximum number of bytes we will ever move to or from user space
in response to a WFC_* command. This obviously doesn't cover
actual sample data.
*/
#define WF_MAX_READ sizeof(wavefront_multisample)
#define WF_MAX_WRITE sizeof(wavefront_multisample)
/*
This allows us to execute any WF command except the download/upload
ones, which are handled differently due to copyin/copyout issues as
well as data-nybbling to/from the card.
*/
typedef struct wavefront_control {
int cmd; /* WFC_* */
char status; /* return status to user-space */
unsigned char rbuf[WF_MAX_READ]; /* bytes read from card */
unsigned char wbuf[WF_MAX_WRITE]; /* bytes written to card */
} wavefront_control;
#define WFCTL_WFCMD 0x1
#define WFCTL_LOAD_SPP 0x2
/* Modulator table */
#define WF_MOD_LFO1 0
#define WF_MOD_LFO2 1
#define WF_MOD_ENV1 2
#define WF_MOD_ENV2 3
#define WF_MOD_KEYBOARD 4
#define WF_MOD_LOGKEY 5
#define WF_MOD_VELOCITY 6
#define WF_MOD_LOGVEL 7
#define WF_MOD_RANDOM 8
#define WF_MOD_PRESSURE 9
#define WF_MOD_MOD_WHEEL 10
#define WF_MOD_1 WF_MOD_MOD_WHEEL
#define WF_MOD_BREATH 11
#define WF_MOD_2 WF_MOD_BREATH
#define WF_MOD_FOOT 12
#define WF_MOD_4 WF_MOD_FOOT
#define WF_MOD_VOLUME 13
#define WF_MOD_7 WF_MOD_VOLUME
#define WF_MOD_PAN 14
#define WF_MOD_10 WF_MOD_PAN
#define WF_MOD_EXPR 15
#define WF_MOD_11 WF_MOD_EXPR
/* FX-related material */
typedef struct wf_fx_info {
int request; /* see list below */
long data[4]; /* we don't need much */
} wavefront_fx_info;
/* support for each of these will be forthcoming once I or someone
else has figured out which of the addresses on page 6 and page 7 of
the YSS225 control each parameter. Incidentally, these come from
the Windows driver interface, but again, Turtle Beach didn't
document the API to use them.
*/
#define WFFX_SETOUTGAIN 0
#define WFFX_SETSTEREOOUTGAIN 1
#define WFFX_SETREVERBIN1GAIN 2
#define WFFX_SETREVERBIN2GAIN 3
#define WFFX_SETREVERBIN3GAIN 4
#define WFFX_SETCHORUSINPORT 5
#define WFFX_SETREVERBIN1PORT 6
#define WFFX_SETREVERBIN2PORT 7
#define WFFX_SETREVERBIN3PORT 8
#define WFFX_SETEFFECTPORT 9
#define WFFX_SETAUXPORT 10
#define WFFX_SETREVERBTYPE 11
#define WFFX_SETREVERBDELAY 12
#define WFFX_SETCHORUSLFO 13
#define WFFX_SETCHORUSPMD 14
#define WFFX_SETCHORUSAMD 15
#define WFFX_SETEFFECT 16
#define WFFX_SETBASEALL 17
#define WFFX_SETREVERBALL 18
#define WFFX_SETCHORUSALL 20
#define WFFX_SETREVERBDEF 22
#define WFFX_SETCHORUSDEF 23
#define WFFX_DELAYSETINGAIN 24
#define WFFX_DELAYSETFBGAIN 25
#define WFFX_DELAYSETFBLPF 26
#define WFFX_DELAYSETGAIN 27
#define WFFX_DELAYSETTIME 28
#define WFFX_DELAYSETFBTIME 29
#define WFFX_DELAYSETALL 30
#define WFFX_DELAYSETDEF 32
#define WFFX_SDELAYSETINGAIN 33
#define WFFX_SDELAYSETFBGAIN 34
#define WFFX_SDELAYSETFBLPF 35
#define WFFX_SDELAYSETGAIN 36
#define WFFX_SDELAYSETTIME 37
#define WFFX_SDELAYSETFBTIME 38
#define WFFX_SDELAYSETALL 39
#define WFFX_SDELAYSETDEF 41
#define WFFX_DEQSETINGAIN 42
#define WFFX_DEQSETFILTER 43
#define WFFX_DEQSETALL 44
#define WFFX_DEQSETDEF 46
#define WFFX_MUTE 47
#define WFFX_FLANGESETBALANCE 48
#define WFFX_FLANGESETDELAY 49
#define WFFX_FLANGESETDWFFX_TH 50
#define WFFX_FLANGESETFBGAIN 51
#define WFFX_FLANGESETINGAIN 52
#define WFFX_FLANGESETLFO 53
#define WFFX_FLANGESETALL 54
#define WFFX_FLANGESETDEF 56
#define WFFX_PITCHSETSHIFT 57
#define WFFX_PITCHSETBALANCE 58
#define WFFX_PITCHSETALL 59
#define WFFX_PITCHSETDEF 61
#define WFFX_SRSSETINGAIN 62
#define WFFX_SRSSETSPACE 63
#define WFFX_SRSSETCENTER 64
#define WFFX_SRSSETGAIN 65
#define WFFX_SRSSETMODE 66
#define WFFX_SRSSETDEF 68
/* Allow direct user-space control over FX memory/coefficient data.
In theory this could be used to download the FX microprogram,
but it would be a little slower, and involve some weird code.
*/
#define WFFX_MEMSET 69
#endif /* __SOUND_WAVEFRONT_H__ */

27
include/sound/wm0010.h Normal file
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@@ -0,0 +1,27 @@
/*
* wm0010.h -- Platform data for WM0010 DSP Driver
*
* Copyright 2012 Wolfson Microelectronics PLC.
*
* Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
*
* 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.
*/
#ifndef WM0010_PDATA_H
#define WM0010_PDATA_H
struct wm0010_pdata {
int gpio_reset;
/* Set if there is an inverter between the GPIO controlling
* the reset signal and the device.
*/
int reset_active_high;
int irq_flags;
};
#endif

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