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-rw-r--r--include/sound/Kbuild1
-rw-r--r--include/sound/ac97_codec.h2
-rw-r--r--include/sound/aci.h90
-rw-r--r--include/sound/ak4113.h321
-rw-r--r--include/sound/ak4114.h12
-rw-r--r--include/sound/ak4xxx-adda.h5
-rw-r--r--include/sound/control.h5
-rw-r--r--include/sound/cs4231-regs.h1
-rw-r--r--include/sound/pcm.h3
-rw-r--r--include/sound/rawmidi.h2
-rw-r--r--include/sound/sh_dac_audio.h21
-rw-r--r--include/sound/soc-dai.h14
-rw-r--r--include/sound/soc-dapm.h17
-rw-r--r--include/sound/soc.h15
-rw-r--r--include/sound/sscape_ioctl.h21
-rw-r--r--include/sound/tlv320dac33-plat.h20
-rw-r--r--include/sound/tpa6130a2-plat.h30
-rw-r--r--include/sound/wm8993.h2
-rw-r--r--include/sound/wss.h1
19 files changed, 539 insertions, 44 deletions
diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index fd054a344324..e9dd9369ecb9 100644
--- a/include/sound/Kbuild
+++ b/include/sound/Kbuild
@@ -2,7 +2,6 @@ header-y += asound_fm.h
header-y += hdsp.h
header-y += hdspm.h
header-y += sfnt_info.h
-header-y += sscape_ioctl.h
unifdef-y += asequencer.h
unifdef-y += asound.h
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 3dae3f799b9b..49400459b477 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -593,7 +593,7 @@ enum {
struct ac97_quirk {
unsigned short subvendor; /* PCI subsystem vendor id */
- unsigned short subdevice; /* PCI sybsystem device 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 */
diff --git a/include/sound/aci.h b/include/sound/aci.h
new file mode 100644
index 000000000000..ee639d355ef0
--- /dev/null
+++ b/include/sound/aci.h
@@ -0,0 +1,90 @@
+#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_ */
+
diff --git a/include/sound/ak4113.h b/include/sound/ak4113.h
new file mode 100644
index 000000000000..8988edae1609
--- /dev/null
+++ b/include/sound/ak4113.h
@@ -0,0 +1,321 @@
+#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[AK4113_WRITABLE_REGS],
+ 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 */
+
diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h
index d293d36a66b8..3ce69fd92523 100644
--- a/include/sound/ak4114.h
+++ b/include/sound/ak4114.h
@@ -95,13 +95,13 @@
/* AK4114_REG_IO0 */
#define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */
-#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */
-#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */
-#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */
+#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 Though Data Selector for TX0 pin */
-#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */
-#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */
+#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 */
diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h
index 891cf1aea8b1..030b87c2f6d4 100644
--- a/include/sound/ak4xxx-adda.h
+++ b/include/sound/ak4xxx-adda.h
@@ -68,7 +68,7 @@ struct snd_akm4xxx {
enum {
SND_AK4524, SND_AK4528, SND_AK4529,
SND_AK4355, SND_AK4358, SND_AK4381,
- SND_AK5365
+ SND_AK5365, SND_AK4620,
} type;
/* (array) information of combined codecs */
@@ -76,6 +76,9 @@ struct snd_akm4xxx {
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,
diff --git a/include/sound/control.h b/include/sound/control.h
index ef96f07aa03b..112374dc0c58 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -56,7 +56,6 @@ struct snd_kcontrol_new {
struct snd_kcontrol_volatile {
struct snd_ctl_file *owner; /* locked */
- pid_t owner_pid;
unsigned int access; /* access rights */
};
@@ -87,10 +86,12 @@ struct snd_kctl_event {
#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;
- pid_t pid;
+ struct pid *pid;
int prefer_pcm_subdevice;
int prefer_rawmidi_subdevice;
wait_queue_head_t change_sleep;
diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h
index 92647532c454..66d28c2cb53d 100644
--- a/include/sound/cs4231-regs.h
+++ b/include/sound/cs4231-regs.h
@@ -70,7 +70,6 @@
#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 CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */
#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 */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index de6d981de5d6..c83a4a79f16b 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -348,6 +348,8 @@ struct snd_pcm_group { /* keep linked substreams */
int count;
};
+struct pid;
+
struct snd_pcm_substream {
struct snd_pcm *pcm;
struct snd_pcm_str *pstr;
@@ -379,6 +381,7 @@ struct snd_pcm_substream {
atomic_t mmap_count;
unsigned int f_flags;
void (*pcm_release)(struct snd_pcm_substream *);
+ struct pid *pid;
#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
/* -- OSS things -- */
struct snd_pcm_oss_substream oss;
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index c23c26585700..2480e7d10dcf 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -46,6 +46,7 @@
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);
@@ -97,6 +98,7 @@ struct snd_rawmidi_substream {
struct snd_rawmidi_str *pstr;
char name[32];
struct snd_rawmidi_runtime *runtime;
+ struct pid *pid;
/* hardware layer */
struct snd_rawmidi_ops *ops;
};
diff --git a/include/sound/sh_dac_audio.h b/include/sound/sh_dac_audio.h
new file mode 100644
index 000000000000..f5deaf1ddb9f
--- /dev/null
+++ b/include/sound/sh_dac_audio.h
@@ -0,0 +1,21 @@
+/*
+ * 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 */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 97ca9af414dc..ca24e7f7a3f5 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -30,6 +30,7 @@ struct snd_pcm_substream;
#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
+#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
@@ -106,7 +107,7 @@ 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, unsigned int freq_in, unsigned int freq_out);
+ 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);
@@ -114,6 +115,10 @@ 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 */
@@ -136,8 +141,8 @@ struct snd_soc_dai_ops {
*/
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, unsigned int freq_in, unsigned int freq_out);
+ 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);
/*
@@ -148,6 +153,9 @@ struct snd_soc_dai_ops {
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);
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c1410e3191e3..c5c95e1da65b 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -206,6 +206,12 @@
.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_VALUE_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
@@ -260,6 +266,10 @@ 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,
@@ -333,6 +343,10 @@ 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 */
@@ -349,6 +363,9 @@ struct snd_soc_dapm_path {
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ 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;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 475cb7ed6bec..0d7718f9280d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -223,15 +223,15 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
int addr_bits, int data_bits,
enum snd_soc_control_type control);
-#ifdef CONFIG_PM
-int snd_soc_suspend_device(struct device *dev);
-int snd_soc_resume_device(struct device *dev);
-#endif
-
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_init_card(struct snd_soc_device *socdev);
+
+/* Utility functions to get clock rates from various things */
+int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
+int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
/* set runtime hw params */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@@ -333,6 +333,8 @@ struct snd_soc_jack_gpio {
int debounce_time;
struct snd_soc_jack *jack;
struct work_struct work;
+
+ int (*jack_status_check)(void);
};
#endif
@@ -413,6 +415,7 @@ struct snd_soc_codec {
unsigned int num_dai;
#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_codec_root;
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
struct dentry *debugfs_dapm;
diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h
deleted file mode 100644
index 0d8885969c64..000000000000
--- a/include/sound/sscape_ioctl.h
+++ /dev/null
@@ -1,21 +0,0 @@
-#ifndef SSCAPE_IOCTL_H
-#define SSCAPE_IOCTL_H
-
-
-struct sscape_bootblock
-{
- unsigned char code[256];
- unsigned version;
-};
-
-#define SSCAPE_MICROCODE_SIZE 65536
-
-struct sscape_microcode
-{
- unsigned char __user *code;
-};
-
-#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock)
-#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode)
-
-#endif
diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
new file mode 100644
index 000000000000..5858d06a7ffa
--- /dev/null
+++ b/include/sound/tlv320dac33-plat.h
@@ -0,0 +1,20 @@
+/*
+ * Platform header for Texas Instruments TLV320DAC33 codec driver
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@nokia.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;
+};
+
+#endif /* __TLV320DAC33_PLAT_H */
diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
new file mode 100644
index 000000000000..e8c901e749d8
--- /dev/null
+++ b/include/sound/tpa6130a2-plat.h
@@ -0,0 +1,30 @@
+/*
+ * TPA6130A2 driver platform header
+ *
+ * Copyright (C) Nokia Corporation
+ *
+ * Written by Peter Ujfalusi <peter.ujfalusi@nokia.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
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
index 9c661f2f8cda..eee19f63c0d8 100644
--- a/include/sound/wm8993.h
+++ b/include/sound/wm8993.h
@@ -36,7 +36,7 @@ struct wm8993_platform_data {
unsigned int micbias1_lvl:1;
unsigned int micbias2_lvl:1;
- /* Jack detect threashold levels, see datasheet for values */
+ /* Jack detect threshold levels, see datasheet for values */
unsigned int jd_scthr:2;
unsigned int jd_thr:2;
};
diff --git a/include/sound/wss.h b/include/sound/wss.h
index 6d65f322f1d5..fd01f22825cd 100644
--- a/include/sound/wss.h
+++ b/include/sound/wss.h
@@ -154,7 +154,6 @@ int snd_wss_create(struct snd_card *card,
unsigned short hardware,
unsigned short hwshare,
struct snd_wss **rchip);
-int snd_wss_free(struct snd_wss *chip);
int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm);
int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer);
int snd_wss_mixer(struct snd_wss *chip);