forked from Minki/linux
f0fba2ad1b
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
128 lines
7.0 KiB
C
128 lines
7.0 KiB
C
/*
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* File: sound/soc/codecs/ssm2602.h
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* Author: Cliff Cai <Cliff.Cai@analog.com>
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*
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* Created: Tue June 06 2008
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*
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* Modified:
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* Copyright 2008 Analog Devices Inc.
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*
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* Bugs: Enter bugs at http://blackfin.uclinux.org/
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, see the file COPYING, or write
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* to the Free Software Foundation, Inc.,
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* 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef _SSM2602_H
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#define _SSM2602_H
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/* SSM2602 Codec Register definitions */
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#define SSM2602_LINVOL 0x00
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#define SSM2602_RINVOL 0x01
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#define SSM2602_LOUT1V 0x02
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#define SSM2602_ROUT1V 0x03
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#define SSM2602_APANA 0x04
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#define SSM2602_APDIGI 0x05
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#define SSM2602_PWR 0x06
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#define SSM2602_IFACE 0x07
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#define SSM2602_SRATE 0x08
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#define SSM2602_ACTIVE 0x09
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#define SSM2602_RESET 0x0f
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/*SSM2602 Codec Register Field definitions
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*(Mask value to extract the corresponding Register field)
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*/
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/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/
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#define LINVOL_LIN_VOL 0x01F /* Left Channel PGA Volume control */
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#define LINVOL_LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */
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#define LINVOL_LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */
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/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/
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#define RINVOL_RIN_VOL 0x01F /* Right Channel PGA Volume control */
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#define RINVOL_RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */
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#define RINVOL_RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */
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/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/
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#define LOUT1V_LHP_VOL 0x07F /* Left Channel Headphone volume control */
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#define LOUT1V_ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */
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#define LOUT1V_LRHP_BOTH 0x100 /* Left Channel Headphone volume update */
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/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/
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#define ROUT1V_RHP_VOL 0x07F /* Right Channel Headphone volume control */
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#define ROUT1V_ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */
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#define ROUT1V_RLHP_BOTH 0x100 /* Right Channel Headphone volume update */
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/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/
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#define APANA_ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */
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#define APANA_ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */
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#define APANA_ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */
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#define APANA_ENABLE_BYPASS 0x008 /* Line input bypass to line output */
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#define APANA_SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */
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#define APANA_ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */
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#define APANA_SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */
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#define APANA_ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */
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/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/
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#define APDIGI_ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */
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#define APDIGI_DE_EMPHASIS 0x006 /* De-Emphasis Control */
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#define APDIGI_ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */
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#define APDIGI_STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */
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/*Power Down Control (SSM2602_REG_POWER)
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*(1=Enable PowerDown, 0=Disable PowerDown)
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*/
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#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */
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#define PWR_MIC_PDN 0x002 /* Microphone Input & Bias Power Down */
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#define PWR_ADC_PDN 0x004 /* ADC Power Down */
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#define PWR_DAC_PDN 0x008 /* DAC Power Down */
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#define PWR_OUT_PDN 0x010 /* Outputs Power Down */
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#define PWR_OSC_PDN 0x020 /* Oscillator Power Down */
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#define PWR_CLK_OUT_PDN 0x040 /* CLKOUT Power Down */
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#define PWR_POWER_OFF 0x080 /* POWEROFF Mode */
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/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/
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#define IFACE_IFACE_FORMAT 0x003 /* Digital Audio input format control */
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#define IFACE_AUDIO_DATA_LEN 0x00C /* Audio Data word length control */
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#define IFACE_DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */
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#define IFACE_DAC_LR_SWAP 0x020 /* Swap DAC data control */
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#define IFACE_ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */
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#define IFACE_BCLK_INVERT 0x080 /* Bit Clock Inversion control */
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/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/
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#define SRATE_ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */
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#define SRATE_BOS_RATE 0x002 /* Base Over-Sampling rate */
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#define SRATE_SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */
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#define SRATE_CORECLK_DIV2 0x040 /* Core Clock divider select */
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#define SRATE_CLKOUT_DIV2 0x080 /* Clock Out divider select */
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/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/
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#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */
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/*********************************************************************/
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#define SSM2602_CACHEREGNUM 10
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#define SSM2602_SYSCLK 0
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#define SSM2602_DAI 0
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struct ssm2602_setup_data {
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int i2c_bus;
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unsigned short i2c_address;
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};
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#endif
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