From 85488037bb9b533b064be66412dbe1dbcd2734d9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 5 Sep 2009 18:52:16 +0100 Subject: ASoC: Add source argument to PLL configuration More and more devices feature PLLs and FLLs with the ability to select between multiple input clocks. In order to better support these devices a new argument, source, has been added to the set_pll() configuration API. Using set_clkdiv() is often difficult due to the need to stop the PLL/FLL before any reconfiguration can be done. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af414dc..16963d4d5df2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ 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); @@ -136,8 +136,8 @@ 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); /* -- cgit v1.2.2 From 215edda3adf502ccdf3a358ab35b616e7abd25ff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Sep 2009 18:59:05 +0100 Subject: ASoC: Allow per-route connectedness checks for supplies Some chips with complex internal supply (particularly clocking) arragements may have multiple options for some of the supply connections. Since these don't affect user-visible audio routing the expectation would be that they would be managed automatically by one of the drivers. Support these users by allowing routes to have a connected function which is queried before the connectedness of the path is checked as normal. Currently this is only done for supplies, other widgets could be supported but are not currently since the expectation for them is that audio routing will be under the control of userspace. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c1410e3191e3..67224db60348 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -333,6 +333,10 @@ 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 +353,9 @@ 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; -- cgit v1.2.2 From 472df3cbae8da6a949f1392a11958b8d21383735 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Sat, 12 Sep 2009 01:16:29 +0800 Subject: ASoC: Provide API for reordering channels The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 16963d4d5df2..e0c7fa7b1060 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,10 @@ 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 */ @@ -148,6 +152,9 @@ 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); /* -- cgit v1.2.2 From c0a9eedf9acafb083adf3ddbff0a1e4d6d9a6949 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:35 +0200 Subject: ALSA: ak4114 - fix errors in output selector bits * the previous version had a typo - values of AK4114_OPS10-12 were identical with AK4114_OPS00-02 * Since no cards actually use this feature, the bug was not identified earlier Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- include/sound/ak4114.h | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'include') 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 */ -- cgit v1.2.2 From 8f34692f63d66805b51ff408f4067748d3c1c3fd Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:36 +0200 Subject: ALSA: ak4620 support, codec regs listed in proc * complete support for ak4620 * codec regs listed in proc for all codecs/chips * adding total regs for each codec * fixing nb. of steps in input attenuation controls Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- include/sound/ak4xxx-adda.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'include') 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, -- cgit v1.2.2 From 42cfa276aebd28e5cc4350ff6c7d75f1cb84dd98 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Wed, 16 Sep 2009 22:25:37 +0200 Subject: ALSA: ak4113 support * complete support for ak4113 * based on code for ak4114 and ak4117 Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- include/sound/ak4113.h | 321 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 321 insertions(+) create mode 100644 include/sound/ak4113.h (limited to 'include') 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 , + * Copyright (c) by Pavel Hofman , + * + * + * 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 */ + -- cgit v1.2.2 From be2500b8353d41463399e997fe8562f772dcaaba Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Fri, 25 Sep 2009 21:02:49 -0500 Subject: ASoC: Add PDM DAI format definition Add DAI format definition for PDM interfaces. Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index e0c7fa7b1060..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 -- cgit v1.2.2 From acd47100914b2896d0699febefd077f85c4dd272 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 1 Oct 2009 00:10:34 +0200 Subject: ALSA: sscape: convert to firmware loader framework The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/sscape_ioctl.h | 21 --------------------- 1 file changed, 21 deletions(-) delete mode 100644 include/sound/sscape_ioctl.h (limited to 'include') 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 -- cgit v1.2.2 From 88439ac793934a47f47ad285656b63d09f5937c8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 1 Oct 2009 10:32:47 +0300 Subject: ASoC: add support for multiple cards/codecs in debugfs In order to support multiple codecs on the same system in the debugfs the directory hierarchy need to be changed by adding directory per codec under the asoc direcorty: debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg /dapm_pop_time /dapm/{widgets} With the original implementation only the debugfs files are only created for the first codec, other codecs loaded later would fail to create the debugfs files (since they are already exist). Furthermore in this situation any of the codecs has been removed, would cause the debugfs entries to disappear, regardless if the codec, which created them are still loaded (the one which loaded first). Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 475cb7ed6bec..0b1f917a53ba 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -413,6 +413,7 @@ 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; -- cgit v1.2.2 From 7c824f4b69316df55fe243c5a6c7dba2b62285c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 2 Oct 2009 07:22:58 +0200 Subject: ALSA: sscape - Remove sscap_ioctl.h from include/sound/Kbuild Signed-off-by: Takashi Iwai --- include/sound/Kbuild | 1 - 1 file changed, 1 deletion(-) (limited to 'include') 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 -- cgit v1.2.2 From d2b247a8be57647d1745535acd58169fbcbe431a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Oct 2009 15:21:04 +0100 Subject: ASoC: Add virtual enumeration support for DAPM muxes Sometimes it is desirable to have a mux which does not reflect any direct register configuration but which will instead only have an effect implicitly (for example, as a result of changing which parts of the device are powered up). Provide a virtual mux for this purpose. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 67224db60348..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, -- cgit v1.2.2 From 493b67efffc462703d583389aca96f850c18d3b3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 9 Oct 2009 15:55:41 +0300 Subject: ASoC: TPA6130A2 amplifier driver Driver for Texas Instruments TPA6130A2 stereo headphone amplifier. The driver provides playback gain control and also pre-defined DAPM_HP widgets and DAPM routings for power management. The DAPM_HP widget names are: "TPA6130A2 Headphone Left" "TPA6130A2 Headphone Right" From soc machine drivers to use with the tpa6130a2 amplifier, the tpa6130a2_add_controls has to be called, which adds the alsa controls and the DAPM routing needed for the tpa6130a2. After that the machine driver can connect the codec's output with 'TPA6130A2 Left' and 'TPA6130A2 Right': {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"}, {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"}, Internally the left and right channels are powered separately. When none of the channels are needed the amplifier is powered down: hard power: valid GPIO number is passed within platform data soft power: Using the software shutdown of the amplifier Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/sound/tpa6130a2-plat.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) create mode 100644 include/sound/tpa6130a2-plat.h (limited to 'include') 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 + * + * 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 -- cgit v1.2.2 From d2058b0cd039aad89b111d83b9c347e9d8f57a84 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Oct 2009 17:39:56 +0100 Subject: ASoC: Remove snd_soc_suspend_device() The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: Mark Brown --- include/sound/soc.h | 5 ----- 1 file changed, 5 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b1f917a53ba..b1245e3acdfc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -223,11 +223,6 @@ 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); -- cgit v1.2.2 From c8bf93f0fe8c5a509a29e30f3bac823fa0f6d96e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 15 Oct 2009 09:03:56 +0300 Subject: ASoC: Codec driver for Texas Instruments tlv320dac33 codec Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/sound/tlv320dac33-plat.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 include/sound/tlv320dac33-plat.h (limited to 'include') 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 + * + * 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 */ -- cgit v1.2.2 From 0b83ddebc6e884dc0221358cf68c461520fbdd8e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Oct 2009 13:26:45 +0300 Subject: MFD: twl4030: add twl4030_codec MFD as a new child to the core New MFD child to twl4030 MFD device. Reason for the twl4030_codec MFD: the vibra control is actually in the codec part of the twl4030. If both the vibra and the audio functionality is needed from the twl4030 at the same time, than they need to control the codec power and APLL at the same time without breaking the other driver. Also these two has to be able to work without the need for the other driver. This MFD device will be used by the drivers, which needs resources from the twl4030 codec like audio and vibra. The platform specific configuration data is passed along to the child drivers (audio, vibra). Signed-off-by: Peter Ujfalusi Acked-by: Samuel Ortiz Signed-off-by: Mark Brown --- include/linux/i2c/twl4030.h | 18 +++ include/linux/mfd/twl4030-codec.h | 271 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 289 insertions(+) create mode 100644 include/linux/mfd/twl4030-codec.h (limited to 'include') diff --git a/include/linux/i2c/twl4030.h b/include/linux/i2c/twl4030.h index 2d02dfd7076c..42d6c722bd82 100644 --- a/include/linux/i2c/twl4030.h +++ b/include/linux/i2c/twl4030.h @@ -401,6 +401,23 @@ struct twl4030_power_data { extern void twl4030_power_init(struct twl4030_power_data *triton2_scripts); +struct twl4030_codec_audio_data { + unsigned int audio_mclk; + unsigned int ramp_delay_value; + unsigned int hs_extmute:1; + void (*set_hs_extmute)(int mute); +}; + +struct twl4030_codec_vibra_data { + unsigned int audio_mclk; + unsigned int coexist; +}; + +struct twl4030_codec_data { + struct twl4030_codec_audio_data *audio; + struct twl4030_codec_vibra_data *vibra; +}; + struct twl4030_platform_data { unsigned irq_base, irq_end; struct twl4030_bci_platform_data *bci; @@ -409,6 +426,7 @@ struct twl4030_platform_data { struct twl4030_keypad_data *keypad; struct twl4030_usb_data *usb; struct twl4030_power_data *power; + struct twl4030_codec_data *codec; /* LDO regulators */ struct regulator_init_data *vdac; diff --git a/include/linux/mfd/twl4030-codec.h b/include/linux/mfd/twl4030-codec.h new file mode 100644 index 000000000000..ef0a3041d759 --- /dev/null +++ b/include/linux/mfd/twl4030-codec.h @@ -0,0 +1,271 @@ +/* + * MFD driver for twl4030 codec submodule + * + * Author: Peter Ujfalusi + * + * 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. + * + * 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 __TWL4030_CODEC_H__ +#define __TWL4030_CODEC_H__ + +/* Codec registers */ +#define TWL4030_REG_CODEC_MODE 0x01 +#define TWL4030_REG_OPTION 0x02 +#define TWL4030_REG_UNKNOWN 0x03 +#define TWL4030_REG_MICBIAS_CTL 0x04 +#define TWL4030_REG_ANAMICL 0x05 +#define TWL4030_REG_ANAMICR 0x06 +#define TWL4030_REG_AVADC_CTL 0x07 +#define TWL4030_REG_ADCMICSEL 0x08 +#define TWL4030_REG_DIGMIXING 0x09 +#define TWL4030_REG_ATXL1PGA 0x0A +#define TWL4030_REG_ATXR1PGA 0x0B +#define TWL4030_REG_AVTXL2PGA 0x0C +#define TWL4030_REG_AVTXR2PGA 0x0D +#define TWL4030_REG_AUDIO_IF 0x0E +#define TWL4030_REG_VOICE_IF 0x0F +#define TWL4030_REG_ARXR1PGA 0x10 +#define TWL4030_REG_ARXL1PGA 0x11 +#define TWL4030_REG_ARXR2PGA 0x12 +#define TWL4030_REG_ARXL2PGA 0x13 +#define TWL4030_REG_VRXPGA 0x14 +#define TWL4030_REG_VSTPGA 0x15 +#define TWL4030_REG_VRX2ARXPGA 0x16 +#define TWL4030_REG_AVDAC_CTL 0x17 +#define TWL4030_REG_ARX2VTXPGA 0x18 +#define TWL4030_REG_ARXL1_APGA_CTL 0x19 +#define TWL4030_REG_ARXR1_APGA_CTL 0x1A +#define TWL4030_REG_ARXL2_APGA_CTL 0x1B +#define TWL4030_REG_ARXR2_APGA_CTL 0x1C +#define TWL4030_REG_ATX2ARXPGA 0x1D +#define TWL4030_REG_BT_IF 0x1E +#define TWL4030_REG_BTPGA 0x1F +#define TWL4030_REG_BTSTPGA 0x20 +#define TWL4030_REG_EAR_CTL 0x21 +#define TWL4030_REG_HS_SEL 0x22 +#define TWL4030_REG_HS_GAIN_SET 0x23 +#define TWL4030_REG_HS_POPN_SET 0x24 +#define TWL4030_REG_PREDL_CTL 0x25 +#define TWL4030_REG_PREDR_CTL 0x26 +#define TWL4030_REG_PRECKL_CTL 0x27 +#define TWL4030_REG_PRECKR_CTL 0x28 +#define TWL4030_REG_HFL_CTL 0x29 +#define TWL4030_REG_HFR_CTL 0x2A +#define TWL4030_REG_ALC_CTL 0x2B +#define TWL4030_REG_ALC_SET1 0x2C +#define TWL4030_REG_ALC_SET2 0x2D +#define TWL4030_REG_BOOST_CTL 0x2E +#define TWL4030_REG_SOFTVOL_CTL 0x2F +#define TWL4030_REG_DTMF_FREQSEL 0x30 +#define TWL4030_REG_DTMF_TONEXT1H 0x31 +#define TWL4030_REG_DTMF_TONEXT1L 0x32 +#define TWL4030_REG_DTMF_TONEXT2H 0x33 +#define TWL4030_REG_DTMF_TONEXT2L 0x34 +#define TWL4030_REG_DTMF_TONOFF 0x35 +#define TWL4030_REG_DTMF_WANONOFF 0x36 +#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 +#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 +#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 +#define TWL4030_REG_APLL_CTL 0x3A +#define TWL4030_REG_DTMF_CTL 0x3B +#define TWL4030_REG_DTMF_PGA_CTL2 0x3C +#define TWL4030_REG_DTMF_PGA_CTL1 0x3D +#define TWL4030_REG_MISC_SET_1 0x3E +#define TWL4030_REG_PCMBTMUX 0x3F +#define TWL4030_REG_RX_PATH_SEL 0x43 +#define TWL4030_REG_VDL_APGA_CTL 0x44 +#define TWL4030_REG_VIBRA_CTL 0x45 +#define TWL4030_REG_VIBRA_SET 0x46 +#define TWL4030_REG_VIBRA_PWM_SET 0x47 +#define TWL4030_REG_ANAMIC_GAIN 0x48 +#define TWL4030_REG_MISC_SET_2 0x49 + +/* Bitfield Definitions */ + +/* TWL4030_CODEC_MODE (0x01) Fields */ +#define TWL4030_APLL_RATE 0xF0 +#define TWL4030_APLL_RATE_8000 0x00 +#define TWL4030_APLL_RATE_11025 0x10 +#define TWL4030_APLL_RATE_12000 0x20 +#define TWL4030_APLL_RATE_16000 0x40 +#define TWL4030_APLL_RATE_22050 0x50 +#define TWL4030_APLL_RATE_24000 0x60 +#define TWL4030_APLL_RATE_32000 0x80 +#define TWL4030_APLL_RATE_44100 0x90 +#define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_APLL_RATE_96000 0xE0 +#define TWL4030_SEL_16K 0x08 +#define TWL4030_CODECPDZ 0x02 +#define TWL4030_OPT_MODE 0x01 +#define TWL4030_OPTION_1 (1 << 0) +#define TWL4030_OPTION_2 (0 << 0) + +/* TWL4030_OPTION (0x02) Fields */ +#define TWL4030_ATXL1_EN (1 << 0) +#define TWL4030_ATXR1_EN (1 << 1) +#define TWL4030_ATXL2_VTXL_EN (1 << 2) +#define TWL4030_ATXR2_VTXR_EN (1 << 3) +#define TWL4030_ARXL1_VRX_EN (1 << 4) +#define TWL4030_ARXR1_EN (1 << 5) +#define TWL4030_ARXL2_EN (1 << 6) +#define TWL4030_ARXR2_EN (1 << 7) + +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + +/* ANAMICL (0x05) Fields */ +#define TWL4030_CNCL_OFFSET_START 0x80 +#define TWL4030_OFFSET_CNCL_SEL 0x60 +#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 +#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 +#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 +#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 +#define TWL4030_MICAMPL_EN 0x10 +#define TWL4030_CKMIC_EN 0x08 +#define TWL4030_AUXL_EN 0x04 +#define TWL4030_HSMIC_EN 0x02 +#define TWL4030_MAINMIC_EN 0x01 + +/* ANAMICR (0x06) Fields */ +#define TWL4030_MICAMPR_EN 0x10 +#define TWL4030_AUXR_EN 0x04 +#define TWL4030_SUBMIC_EN 0x01 + +/* AVADC_CTL (0x07) Fields */ +#define TWL4030_ADCL_EN 0x08 +#define TWL4030_AVADC_CLK_PRIORITY 0x04 +#define TWL4030_ADCR_EN 0x02 + +/* TWL4030_REG_ADCMICSEL (0x08) Fields */ +#define TWL4030_DIGMIC1_EN 0x08 +#define TWL4030_TX2IN_SEL 0x04 +#define TWL4030_DIGMIC0_EN 0x02 +#define TWL4030_TX1IN_SEL 0x01 + +/* AUDIO_IF (0x0E) Fields */ +#define TWL4030_AIF_SLAVE_EN 0x80 +#define TWL4030_DATA_WIDTH 0x60 +#define TWL4030_DATA_WIDTH_16S_16W 0x00 +#define TWL4030_DATA_WIDTH_32S_16W 0x40 +#define TWL4030_DATA_WIDTH_32S_24W 0x60 +#define TWL4030_AIF_FORMAT 0x18 +#define TWL4030_AIF_FORMAT_CODEC 0x00 +#define TWL4030_AIF_FORMAT_LEFT 0x08 +#define TWL4030_AIF_FORMAT_RIGHT 0x10 +#define TWL4030_AIF_FORMAT_TDM 0x18 +#define TWL4030_AIF_TRI_EN 0x04 +#define TWL4030_CLK256FS_EN 0x02 +#define TWL4030_AIF_EN 0x01 + +/* VOICE_IF (0x0F) Fields */ +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + +/* EAR_CTL (0x21) */ +#define TWL4030_EAR_GAIN 0x30 + +/* HS_GAIN_SET (0x23) Fields */ +#define TWL4030_HSR_GAIN 0x0C +#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 +#define TWL4030_HSR_GAIN_0DB 0x08 +#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C +#define TWL4030_HSL_GAIN 0x03 +#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 +#define TWL4030_HSL_GAIN_0DB 0x02 +#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 + +/* HS_POPN_SET (0x24) Fields */ +#define TWL4030_VMID_EN 0x40 +#define TWL4030_EXTMUTE 0x20 +#define TWL4030_RAMP_DELAY 0x1C +#define TWL4030_RAMP_DELAY_20MS 0x00 +#define TWL4030_RAMP_DELAY_40MS 0x04 +#define TWL4030_RAMP_DELAY_81MS 0x08 +#define TWL4030_RAMP_DELAY_161MS 0x0C +#define TWL4030_RAMP_DELAY_323MS 0x10 +#define TWL4030_RAMP_DELAY_645MS 0x14 +#define TWL4030_RAMP_DELAY_1291MS 0x18 +#define TWL4030_RAMP_DELAY_2581MS 0x1C +#define TWL4030_RAMP_EN 0x02 + +/* PREDL_CTL (0x25) */ +#define TWL4030_PREDL_GAIN 0x30 + +/* PREDR_CTL (0x26) */ +#define TWL4030_PREDR_GAIN 0x30 + +/* PRECKL_CTL (0x27) */ +#define TWL4030_PRECKL_GAIN 0x30 + +/* PRECKR_CTL (0x28) */ +#define TWL4030_PRECKR_GAIN 0x30 + +/* HFL_CTL (0x29, 0x2A) Fields */ +#define TWL4030_HF_CTL_HB_EN 0x04 +#define TWL4030_HF_CTL_LOOP_EN 0x08 +#define TWL4030_HF_CTL_RAMP_EN 0x10 +#define TWL4030_HF_CTL_REF_EN 0x20 + +/* APLL_CTL (0x3A) Fields */ +#define TWL4030_APLL_EN 0x10 +#define TWL4030_APLL_INFREQ 0x0F +#define TWL4030_APLL_INFREQ_19200KHZ 0x05 +#define TWL4030_APLL_INFREQ_26000KHZ 0x06 +#define TWL4030_APLL_INFREQ_38400KHZ 0x0F + +/* REG_MISC_SET_1 (0x3E) Fields */ +#define TWL4030_CLK64_EN 0x80 +#define TWL4030_SCRAMBLE_EN 0x40 +#define TWL4030_FMLOOP_EN 0x20 +#define TWL4030_SMOOTH_ANAVOL_EN 0x02 +#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 + +/* VIBRA_CTL (0x45) */ +#define TWL4030_VIBRA_EN 0x01 +#define TWL4030_VIBRA_DIR 0x02 +#define TWL4030_VIBRA_AUDIO_SEL_L1 (0x00 << 2) +#define TWL4030_VIBRA_AUDIO_SEL_R1 (0x01 << 2) +#define TWL4030_VIBRA_AUDIO_SEL_L2 (0x02 << 2) +#define TWL4030_VIBRA_AUDIO_SEL_R2 (0x03 << 2) +#define TWL4030_VIBRA_SEL 0x10 +#define TWL4030_VIBRA_DIR_SEL 0x20 + +/* TWL4030 codec resource IDs */ +enum twl4030_codec_res { + TWL4030_CODEC_RES_POWER = 0, + TWL4030_CODEC_RES_APLL, + TWL4030_CODEC_RES_MAX, +}; + +int twl4030_codec_disable_resource(enum twl4030_codec_res id); +int twl4030_codec_enable_resource(enum twl4030_codec_res id); + +#endif /* End of __TWL4030_CODEC_H__ */ -- cgit v1.2.2 From fe3e78e073d25308756f38019956061153267769 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 Nov 2009 22:13:13 +0000 Subject: ASoC: Factor out snd_soc_init_card() snd_soc_init_card() is always called as the last part of the CODEC probe function so we can factor it out into the core card setup rather than have each CODEC replicate the code to do the initialiastation. This will be required to support multiple CODECs per card. Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b1245e3acdfc..7f3a4c5028da 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -226,7 +226,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, /* 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); /* set runtime hw params */ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, -- cgit v1.2.2 From 9dcaa7b25f2c8f6a0485854cd3641f585a154072 Mon Sep 17 00:00:00 2001 From: Rafael Ignacio Zurita Date: Tue, 3 Nov 2009 17:16:27 -0300 Subject: ALSA: sh: add SuperH DAC audio driver for ALSA V4 This is a port of the sound/oss/sh_dac_audio.c driver. The driver uses an on-chip 8-bit D/A converter, which has a speaker connected to one of its channels, found in several ancient HP machines. For interrupts it uses a high-resolution timer (hrtimer). Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx). Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver would be obsolete soon, and it could be removed. Signed-off-by: Rafael Ignacio Zurita Acked-by: Paul Mundt Signed-off-by: Takashi Iwai --- include/sound/sh_dac_audio.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) create mode 100644 include/sound/sh_dac_audio.h (limited to 'include') 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 + * + * 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 */ -- cgit v1.2.2 From d114cd84a1c5ce42bb10cd3a2da57b2bbcef909b Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 5 Nov 2009 18:32:41 +0100 Subject: ALSA: cs4236: detect chip in one pass The cs4236 was two step detection with call to the snd_wss_free() between two steps. The snd_wss_free() did not free a sound device created in the snd_wss_create(). This caused an OOPS during module removal as the same sound device was released twice. The same OOPS happened if the cs4236 module loading failed. Fix this by adapting the snd_cs4236_create() to correctly work with chips less capable then cs4236. The snd_cs4236_create() behaves the same as the snd_wss_create() if the chip is less capable than the cs4236. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/wss.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') 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); -- cgit v1.2.2 From 31cef7076ed9409a33f19ea372d6dc5fdefe27ae Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Nov 2009 09:34:16 +0100 Subject: control: remove snd_konctrol_volatile::owner_pid field We do not need to save the ID of the process that locked a control because that information is already available in the owner's file data. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/control.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/control.h b/include/sound/control.h index ef96f07aa03b..3517745d0a2d 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 */ }; -- cgit v1.2.2 From 25d27eded1f4fc728e64f443adc339b5229be5d7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Nov 2009 09:35:44 +0100 Subject: control: use reference-counted pid Instead of storing the PID number, take a reference to the task's pid structure. This protects against duplicates due to PID overflows, and using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is correct as seen from the current namespace. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/control.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/control.h b/include/sound/control.h index 3517745d0a2d..112374dc0c58 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -86,10 +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; -- cgit v1.2.2 From cfd5324e699a2e74a44642d43dcf03d581f2a7db Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 4 Nov 2009 09:58:17 +0200 Subject: MFD: TWL4030: Add audio_mclk to the codec platform data Add audio_mclk to the platform data struct for the twl4030-codec MFD driver. Signed-off-by: Peter Ujfalusi Acked-by: Samuel Ortiz Signed-off-by: Mark Brown --- include/linux/i2c/twl4030.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/i2c/twl4030.h b/include/linux/i2c/twl4030.h index 42d6c722bd82..c188961efbce 100644 --- a/include/linux/i2c/twl4030.h +++ b/include/linux/i2c/twl4030.h @@ -414,6 +414,7 @@ struct twl4030_codec_vibra_data { }; struct twl4030_codec_data { + unsigned int audio_mclk; struct twl4030_codec_audio_data *audio; struct twl4030_codec_vibra_data *vibra; }; -- cgit v1.2.2 From f9b4639e045c750e2bad37462476403995508350 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 4 Nov 2009 09:58:19 +0200 Subject: MFD: twl4030-codec: APLL_INFREQ handling in the MFD driver Configure the APLL_INFREQ field in the APLL_CTL register based on the platform data. Provide also a function for childs to query the audio_mclk frequency. Signed-off-by: Peter Ujfalusi Acked-by: Samuel Ortiz Signed-off-by: Mark Brown --- include/linux/mfd/twl4030-codec.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/mfd/twl4030-codec.h b/include/linux/mfd/twl4030-codec.h index ef0a3041d759..2ec317c68e59 100644 --- a/include/linux/mfd/twl4030-codec.h +++ b/include/linux/mfd/twl4030-codec.h @@ -267,5 +267,6 @@ enum twl4030_codec_res { int twl4030_codec_disable_resource(enum twl4030_codec_res id); int twl4030_codec_enable_resource(enum twl4030_codec_res id); +unsigned int twl4030_codec_get_mclk(void); #endif /* End of __TWL4030_CODEC_H__ */ -- cgit v1.2.2 From e7373b702f6eab35f315e016a4159860a7a4d686 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Nov 2009 10:13:30 +0100 Subject: sound: pcm: record a substream's owner process Record the pid of the task that opened a PCM substream. For sound cards with hardware mixing, this allows determining which process is associated with a specific substream's volume control. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') 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; -- cgit v1.2.2 From 7584af10cf46e0f4aa1696f1be79fa0f19a945ba Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Nov 2009 10:14:04 +0100 Subject: sound: rawmidi: record a substream's owner process Record the pid of the task that opened a RawMIDI substream. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/rawmidi.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') 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; }; -- cgit v1.2.2 From 7aae816dae150caad8880357e42303935c0179a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 Nov 2009 16:08:04 +0000 Subject: ASoC: Add bit clock rate calculator utility functions Many devices need to calculate the bit clock rate desired to work out the clock configuration required for the device. Provide utility functions to do this using both hw_params structures and raw numbers. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7f3a4c5028da..310a21949a3e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -227,6 +227,11 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, 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); +/* 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_params_to_bclk(struct snd_pcm_hw_params *parms); + /* set runtime hw params */ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); -- cgit v1.2.2 From c871a05315d1a76034ea06feeda92081e1d608bf Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Thu, 12 Nov 2009 17:14:04 +0900 Subject: ASoC: Add jack_status_check callback function for GPIO jacks The jack_status_check callback function is the interface to check the status of the jack. Some target provides the method to distinguish what is the jack inserted - headphone jack, microphone jack, tvout jack, etc, so we can implement it using the jack_status_check function. Signed-off-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 310a21949a3e..13b117aac5d9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -332,6 +332,8 @@ struct snd_soc_jack_gpio { int debounce_time; struct snd_soc_jack *jack; struct work_struct work; + + int (*jack_status_check)(void); }; #endif -- cgit v1.2.2 From b753e03e5e7c6ee60e81cd6335c80dc26519f9d0 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Tue, 17 Nov 2009 18:34:54 +0100 Subject: ALSA: cs4236: update control names Update control names to be more closer to their meaning. Change the "Mono" name to the "Beep" as this line is usually used to forward the PC beeper signal to sound card's output. Update names for both cs423x and wss. Clean up cs4235 controls according to the cs4235 doc. Rename some of the cs4235 controls to be consistent with the cs4236's ones. Also, delete one misnamed cs4231 register define. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/cs4231-regs.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') 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 */ -- cgit v1.2.2 From 9aeba6297151abcb1b34f3237e4c028aae500ce4 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 22 Nov 2009 17:23:45 +0100 Subject: ALSA: opti-miro: make miro.h header available outside the alsa directory Move the miro.h header to the include/sound directory. It can be used in the Miro PCM20 radio driver (v4l). Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/aci.h | 73 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) create mode 100644 include/sound/aci.h (limited to 'include') diff --git a/include/sound/aci.h b/include/sound/aci.h new file mode 100644 index 000000000000..bb796d06d0f2 --- /dev/null +++ b/include/sound/aci.h @@ -0,0 +1,73 @@ +#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 */ + +#endif /* _ACI_H_ */ -- cgit v1.2.2 From 9dc9120c774e1d7e3d939542200bd44829c0059d Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 22 Nov 2009 17:26:34 +0100 Subject: ALSA: opti-miro: expose ACI mixer to outside drivers The ACI mixer is used to control the radio FM module installed on the Miro PCM20 sound card. Expose ACI mixer outside the sound card driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/aci.h | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'include') diff --git a/include/sound/aci.h b/include/sound/aci.h index bb796d06d0f2..ee639d355ef0 100644 --- a/include/sound/aci.h +++ b/include/sound/aci.h @@ -70,4 +70,21 @@ #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_ */ + -- cgit v1.2.2 From c0fa59df7214e546f8a37bc677867ac7b67b5c93 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Nov 2009 11:36:10 +0000 Subject: ASoC: Add BCLK calculation utility for TDM mode too Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 13b117aac5d9..0d7718f9280d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -230,6 +230,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); /* 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 */ -- cgit v1.2.2