alistair23-linux/sound/soc/davinci/davinci-mcasp.c

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/*
* ALSA SoC McASP Audio Layer for TI DAVINCI processor
*
* Multi-channel Audio Serial Port Driver
*
* Author: Nirmal Pandey <n-pandey@ti.com>,
* Suresh Rajashekara <suresh.r@ti.com>
* Steve Chen <schen@.mvista.com>
*
* Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com>
* Copyright: (C) 2009 Texas Instruments, India
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/device.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 02:04:11 -06:00
#include <linux/slab.h>
#include <linux/delay.h>
#include <linux/io.h>
#include <linux/pm_runtime.h>
#include <linux/of.h>
#include <linux/of_platform.h>
#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include "davinci-pcm.h"
#include "davinci-mcasp.h"
/*
* McASP register definitions
*/
#define DAVINCI_MCASP_PID_REG 0x00
#define DAVINCI_MCASP_PWREMUMGT_REG 0x04
#define DAVINCI_MCASP_PFUNC_REG 0x10
#define DAVINCI_MCASP_PDIR_REG 0x14
#define DAVINCI_MCASP_PDOUT_REG 0x18
#define DAVINCI_MCASP_PDSET_REG 0x1c
#define DAVINCI_MCASP_PDCLR_REG 0x20
#define DAVINCI_MCASP_TLGC_REG 0x30
#define DAVINCI_MCASP_TLMR_REG 0x34
#define DAVINCI_MCASP_GBLCTL_REG 0x44
#define DAVINCI_MCASP_AMUTE_REG 0x48
#define DAVINCI_MCASP_LBCTL_REG 0x4c
#define DAVINCI_MCASP_TXDITCTL_REG 0x50
#define DAVINCI_MCASP_GBLCTLR_REG 0x60
#define DAVINCI_MCASP_RXMASK_REG 0x64
#define DAVINCI_MCASP_RXFMT_REG 0x68
#define DAVINCI_MCASP_RXFMCTL_REG 0x6c
#define DAVINCI_MCASP_ACLKRCTL_REG 0x70
#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74
#define DAVINCI_MCASP_RXTDM_REG 0x78
#define DAVINCI_MCASP_EVTCTLR_REG 0x7c
#define DAVINCI_MCASP_RXSTAT_REG 0x80
#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84
#define DAVINCI_MCASP_RXCLKCHK_REG 0x88
#define DAVINCI_MCASP_REVTCTL_REG 0x8c
#define DAVINCI_MCASP_GBLCTLX_REG 0xa0
#define DAVINCI_MCASP_TXMASK_REG 0xa4
#define DAVINCI_MCASP_TXFMT_REG 0xa8
#define DAVINCI_MCASP_TXFMCTL_REG 0xac
#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0
#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4
#define DAVINCI_MCASP_TXTDM_REG 0xb8
#define DAVINCI_MCASP_EVTCTLX_REG 0xbc
#define DAVINCI_MCASP_TXSTAT_REG 0xc0
#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4
#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8
#define DAVINCI_MCASP_XEVTCTL_REG 0xcc
/* Left(even TDM Slot) Channel Status Register File */
#define DAVINCI_MCASP_DITCSRA_REG 0x100
/* Right(odd TDM slot) Channel Status Register File */
#define DAVINCI_MCASP_DITCSRB_REG 0x118
/* Left(even TDM slot) User Data Register File */
#define DAVINCI_MCASP_DITUDRA_REG 0x130
/* Right(odd TDM Slot) User Data Register File */
#define DAVINCI_MCASP_DITUDRB_REG 0x148
/* Serializer n Control Register */
#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180
#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \
(n << 2))
/* Transmit Buffer for Serializer n */
#define DAVINCI_MCASP_TXBUF_REG 0x200
/* Receive Buffer for Serializer n */
#define DAVINCI_MCASP_RXBUF_REG 0x280
/* McASP FIFO Registers */
#define DAVINCI_MCASP_WFIFOCTL (0x1010)
#define DAVINCI_MCASP_WFIFOSTS (0x1014)
#define DAVINCI_MCASP_RFIFOCTL (0x1018)
#define DAVINCI_MCASP_RFIFOSTS (0x101C)
#define MCASP_VER3_WFIFOCTL (0x1000)
#define MCASP_VER3_WFIFOSTS (0x1004)
#define MCASP_VER3_RFIFOCTL (0x1008)
#define MCASP_VER3_RFIFOSTS (0x100C)
/*
* DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management
* Register Bits
*/
#define MCASP_FREE BIT(0)
#define MCASP_SOFT BIT(1)
/*
* DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits
*/
#define AXR(n) (1<<n)
#define PFUNC_AMUTE BIT(25)
#define ACLKX BIT(26)
#define AHCLKX BIT(27)
#define AFSX BIT(28)
#define ACLKR BIT(29)
#define AHCLKR BIT(30)
#define AFSR BIT(31)
/*
* DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits
*/
#define AXR(n) (1<<n)
#define PDIR_AMUTE BIT(25)
#define ACLKX BIT(26)
#define AHCLKX BIT(27)
#define AFSX BIT(28)
#define ACLKR BIT(29)
#define AHCLKR BIT(30)
#define AFSR BIT(31)
/*
* DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits
*/
#define DITEN BIT(0) /* Transmit DIT mode enable/disable */
#define VA BIT(2)
#define VB BIT(3)
/*
* DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits
*/
#define TXROT(val) (val)
#define TXSEL BIT(3)
#define TXSSZ(val) (val<<4)
#define TXPBIT(val) (val<<8)
#define TXPAD(val) (val<<13)
#define TXORD BIT(15)
#define FSXDLY(val) (val<<16)
/*
* DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits
*/
#define RXROT(val) (val)
#define RXSEL BIT(3)
#define RXSSZ(val) (val<<4)
#define RXPBIT(val) (val<<8)
#define RXPAD(val) (val<<13)
#define RXORD BIT(15)
#define FSRDLY(val) (val<<16)
/*
* DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits
*/
#define FSXPOL BIT(0)
#define AFSXE BIT(1)
#define FSXDUR BIT(4)
#define FSXMOD(val) (val<<7)
/*
* DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits
*/
#define FSRPOL BIT(0)
#define AFSRE BIT(1)
#define FSRDUR BIT(4)
#define FSRMOD(val) (val<<7)
/*
* DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits
*/
#define ACLKXDIV(val) (val)
#define ACLKXE BIT(5)
#define TX_ASYNC BIT(6)
#define ACLKXPOL BIT(7)
#define ACLKXDIV_MASK 0x1f
/*
* DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits
*/
#define ACLKRDIV(val) (val)
#define ACLKRE BIT(5)
#define RX_ASYNC BIT(6)
#define ACLKRPOL BIT(7)
#define ACLKRDIV_MASK 0x1f
/*
* DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control
* Register Bits
*/
#define AHCLKXDIV(val) (val)
#define AHCLKXPOL BIT(14)
#define AHCLKXE BIT(15)
#define AHCLKXDIV_MASK 0xfff
/*
* DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control
* Register Bits
*/
#define AHCLKRDIV(val) (val)
#define AHCLKRPOL BIT(14)
#define AHCLKRE BIT(15)
#define AHCLKRDIV_MASK 0xfff
/*
* DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits
*/
#define MODE(val) (val)
#define DISMOD (val)(val<<2)
#define TXSTATE BIT(4)
#define RXSTATE BIT(5)
/*
* DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
*/
#define LBEN BIT(0)
#define LBORD BIT(1)
#define LBGENMODE(val) (val<<2)
/*
* DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration
*/
#define TXTDMS(n) (1<<n)
/*
* DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration
*/
#define RXTDMS(n) (1<<n)
/*
* DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits
*/
#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */
#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */
#define RXSERCLR BIT(2) /* Receiver Serializer Clear */
#define RXSMRST BIT(3) /* Receiver State Machine Reset */
#define RXFSRST BIT(4) /* Frame Sync Generator Reset */
#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */
#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/
#define TXSERCLR BIT(10) /* Transmit Serializer Clear */
#define TXSMRST BIT(11) /* Transmitter State Machine Reset */
#define TXFSRST BIT(12) /* Frame Sync Generator Reset */
/*
* DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits
*/
#define MUTENA(val) (val)
#define MUTEINPOL BIT(2)
#define MUTEINENA BIT(3)
#define MUTEIN BIT(4)
#define MUTER BIT(5)
#define MUTEX BIT(6)
#define MUTEFSR BIT(7)
#define MUTEFSX BIT(8)
#define MUTEBADCLKR BIT(9)
#define MUTEBADCLKX BIT(10)
#define MUTERXDMAERR BIT(11)
#define MUTETXDMAERR BIT(12)
/*
* DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits
*/
#define RXDATADMADIS BIT(0)
/*
* DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits
*/
#define TXDATADMADIS BIT(0)
/*
* DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
*/
#define FIFO_ENABLE BIT(16)
#define NUMEVT_MASK (0xFF << 8)
#define NUMDMA_MASK (0xFF)
#define DAVINCI_MCASP_NUM_SERIALIZER 16
static inline void mcasp_set_bits(void __iomem *reg, u32 val)
{
__raw_writel(__raw_readl(reg) | val, reg);
}
static inline void mcasp_clr_bits(void __iomem *reg, u32 val)
{
__raw_writel((__raw_readl(reg) & ~(val)), reg);
}
static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask)
{
__raw_writel((__raw_readl(reg) & ~mask) | val, reg);
}
static inline void mcasp_set_reg(void __iomem *reg, u32 val)
{
__raw_writel(val, reg);
}
static inline u32 mcasp_get_reg(void __iomem *reg)
{
return (unsigned int)__raw_readl(reg);
}
static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
{
int i = 0;
mcasp_set_bits(regs, val);
/* programming GBLCTL needs to read back from GBLCTL and verfiy */
/* loop count is to avoid the lock-up */
for (i = 0; i < 1000; i++) {
if ((mcasp_get_reg(regs) & val) == val)
break;
}
if (i == 1000 && ((mcasp_get_reg(regs) & val) != val))
printk(KERN_ERR "GBLCTL write error\n");
}
static void mcasp_start_rx(struct davinci_audio_dev *dev)
{
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
}
static void mcasp_start_tx(struct davinci_audio_dev *dev)
{
u8 offset = 0, i;
u32 cnt;
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
for (i = 0; i < dev->num_serializer; i++) {
if (dev->serial_dir[i] == TX_MODE) {
offset = i;
break;
}
}
/* wait for TX ready */
cnt = 0;
while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) &
TXSTATE) && (cnt < 100000))
cnt++;
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
}
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
{
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (dev->txnumevt) { /* enable FIFO */
switch (dev->version) {
case MCASP_VERSION_3:
mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL,
FIFO_ENABLE);
mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL,
FIFO_ENABLE);
break;
default:
mcasp_clr_bits(dev->base +
DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
mcasp_set_bits(dev->base +
DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
}
}
mcasp_start_tx(dev);
} else {
if (dev->rxnumevt) { /* enable FIFO */
switch (dev->version) {
case MCASP_VERSION_3:
mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL,
FIFO_ENABLE);
mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL,
FIFO_ENABLE);
break;
default:
mcasp_clr_bits(dev->base +
DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
mcasp_set_bits(dev->base +
DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
}
}
mcasp_start_rx(dev);
}
}
static void mcasp_stop_rx(struct davinci_audio_dev *dev)
{
mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
}
static void mcasp_stop_tx(struct davinci_audio_dev *dev)
{
mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
}
static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
{
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (dev->txnumevt) { /* disable FIFO */
switch (dev->version) {
case MCASP_VERSION_3:
mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL,
FIFO_ENABLE);
break;
default:
mcasp_clr_bits(dev->base +
DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
}
}
mcasp_stop_tx(dev);
} else {
if (dev->rxnumevt) { /* disable FIFO */
switch (dev->version) {
case MCASP_VERSION_3:
mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL,
FIFO_ENABLE);
break;
default:
mcasp_clr_bits(dev->base +
DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
}
}
mcasp_stop_rx(dev);
}
}
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
void __iomem *base = dev->base;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
case SND_SOC_DAIFMT_AC97:
mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
break;
default:
/* configure a full-word SYNC pulse (LRCLK) */
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
/* make 1st data bit occur one ACLK cycle after the frame sync */
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
break;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
/* codec is clock and frame slave */
mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX);
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* codec is clock master and frame slave */
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
ACLKX | ACLKR);
mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
AFSX | AFSR);
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* codec is clock and frame master */
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_NF:
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
break;
case SND_SOC_DAIFMT_NB_IF:
mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
break;
case SND_SOC_DAIFMT_IB_IF:
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
break;
case SND_SOC_DAIFMT_NB_NF:
mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
break;
default:
return -EINVAL;
}
return 0;
}
static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
{
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
switch (div_id) {
case 0: /* MCLK divider */
mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG,
AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG,
AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
break;
case 1: /* BCLK divider */
mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG,
ACLKXDIV(div - 1), ACLKXDIV_MASK);
mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKRCTL_REG,
ACLKRDIV(div - 1), ACLKRDIV_MASK);
break;
default:
return -EINVAL;
}
return 0;
}
static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
if (dir == SND_SOC_CLOCK_OUT) {
mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
} else {
mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
}
return 0;
}
static int davinci_config_channel_size(struct davinci_audio_dev *dev,
int channel_size)
{
u32 fmt = 0;
u32 mask, rotate;
switch (channel_size) {
case DAVINCI_AUDIO_WORD_8:
fmt = 0x03;
rotate = 6;
mask = 0x000000ff;
break;
case DAVINCI_AUDIO_WORD_12:
fmt = 0x05;
rotate = 5;
mask = 0x00000fff;
break;
case DAVINCI_AUDIO_WORD_16:
fmt = 0x07;
rotate = 4;
mask = 0x0000ffff;
break;
case DAVINCI_AUDIO_WORD_20:
fmt = 0x09;
rotate = 3;
mask = 0x000fffff;
break;
case DAVINCI_AUDIO_WORD_24:
fmt = 0x0B;
rotate = 2;
mask = 0x00ffffff;
break;
case DAVINCI_AUDIO_WORD_28:
fmt = 0x0D;
rotate = 1;
mask = 0x0fffffff;
break;
case DAVINCI_AUDIO_WORD_32:
fmt = 0x0F;
rotate = 0;
mask = 0xffffffff;
break;
default:
return -EINVAL;
}
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
RXSSZ(fmt), RXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXSSZ(fmt), TXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate),
TXROT(7));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate),
RXROT(7));
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask);
return 0;
}
static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
{
int i;
u8 tx_ser = 0;
u8 rx_ser = 0;
/* Default configuration */
mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
/* All PINS as McASP */
mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG,
TXDATADMADIS);
} else {
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG,
RXDATADMADIS);
}
for (i = 0; i < dev->num_serializer; i++) {
mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
dev->serial_dir[i]);
if (dev->serial_dir[i] == TX_MODE) {
mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
AXR(i));
tx_ser++;
} else if (dev->serial_dir[i] == RX_MODE) {
mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
AXR(i));
rx_ser++;
}
}
if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (dev->txnumevt * tx_ser > 64)
dev->txnumevt = 1;
switch (dev->version) {
case MCASP_VERSION_3:
mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser,
NUMDMA_MASK);
mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL,
((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
break;
default:
mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
tx_ser, NUMDMA_MASK);
mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
}
}
if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
if (dev->rxnumevt * rx_ser > 64)
dev->rxnumevt = 1;
switch (dev->version) {
case MCASP_VERSION_3:
mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser,
NUMDMA_MASK);
mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL,
((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
break;
default:
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
rx_ser, NUMDMA_MASK);
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
}
}
}
static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
{
int i, active_slots;
u32 mask = 0;
active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots;
for (i = 0; i < active_slots; i++)
mask |= (1 << i);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* bit stream is MSB first with no delay */
/* DSP_B mode */
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
else
printk(KERN_ERR "playback tdm slot %d not supported\n",
dev->tdm_slots);
} else {
/* bit stream is MSB first with no delay */
/* DSP_B mode */
mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
else
printk(KERN_ERR "capture tdm slot %d not supported\n",
dev->tdm_slots);
}
}
/* S/PDIF */
static void davinci_hw_dit_param(struct davinci_audio_dev *dev)
{
/* Set the PDIR for Serialiser as output */
mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX);
/* TXMASK for 24 bits */
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF);
/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
and LSB first */
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXROT(6) | TXSSZ(15));
/* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
AFSXE | FSXMOD(0x180));
/* Set the TX tdm : for all the slots */
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
/* Set the TX clock controls : div = 1 and internal */
mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG,
ACLKXE | TX_ASYNC);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
/* Only 44100 and 48000 are valid, both have the same setting */
mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3));
/* Enable the DIT */
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN);
}
static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
struct davinci_pcm_dma_params *dma_params =
&dev->dma_params[substream->stream];
int word_length;
u8 fifo_level;
davinci_hw_common_param(dev, substream->stream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
fifo_level = dev->txnumevt;
else
fifo_level = dev->rxnumevt;
if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
davinci_hw_dit_param(dev);
else
davinci_hw_param(dev, substream->stream);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_U8:
case SNDRV_PCM_FORMAT_S8:
dma_params->data_type = 1;
word_length = DAVINCI_AUDIO_WORD_8;
break;
case SNDRV_PCM_FORMAT_U16_LE:
case SNDRV_PCM_FORMAT_S16_LE:
dma_params->data_type = 2;
word_length = DAVINCI_AUDIO_WORD_16;
break;
case SNDRV_PCM_FORMAT_U24_3LE:
case SNDRV_PCM_FORMAT_S24_3LE:
case SNDRV_PCM_FORMAT_U24_LE:
case SNDRV_PCM_FORMAT_S24_LE:
dma_params->data_type = 3;
word_length = DAVINCI_AUDIO_WORD_24;
break;
case SNDRV_PCM_FORMAT_U32_LE:
case SNDRV_PCM_FORMAT_S32_LE:
dma_params->data_type = 4;
word_length = DAVINCI_AUDIO_WORD_32;
break;
default:
printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
return -EINVAL;
}
if (dev->version == MCASP_VERSION_2 && !fifo_level)
dma_params->acnt = 4;
else
dma_params->acnt = dma_params->data_type;
dma_params->fifo_level = fifo_level;
davinci_config_channel_size(dev, word_length);
return 0;
}
static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *cpu_dai)
{
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ret = pm_runtime_get_sync(dev->dev);
if (IS_ERR_VALUE(ret))
dev_err(dev->dev, "pm_runtime_get_sync() failed\n");
davinci_mcasp_start(dev, substream->stream);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
davinci_mcasp_stop(dev, substream->stream);
ret = pm_runtime_put_sync(dev->dev);
if (IS_ERR_VALUE(ret))
dev_err(dev->dev, "pm_runtime_put_sync() failed\n");
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
davinci_mcasp_stop(dev, substream->stream);
break;
default:
ret = -EINVAL;
}
return ret;
}
ASoC: davinci: fixes for multi-component Multi-component commit f0fba2ad broke a few things which this patch should fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be good if those with access to other Davinci boards could test. -- The multi-component commit put the initialisation of snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c). The initialisation had to be moved from the probe function in these drivers because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver. Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in davinci-pcm.c) before hw_params is called. I have moved the initialisation to a new snd_soc_dai_ops.startup function in each of these drivers. This fix indicates that all platforms that use davinci-pcm must have been broken and need to test with this fix. -- The multi-component commit also changed the McBSP driver name from "davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board level references to the driver name. This change is understandable, as there is a similarly named "davinci-mcasp" driver in davinci-mcasp.c. There is probably no 'correct' name for this driver. The DM6446 datasheet calls it the "ASP" and describes it as a "specialised McBSP". The DM355 datasheet calls it the "ASP" and describes it as a "specialised ASP". The DM365 datasheet calls it the "McBSP". Rather than fix this problem by reverting to "davinci-asp", I've elected to avoid future confusion with the "davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also consistent with the names of the functions in the driver. There are other fixes required, so it was never going to be as simple as a revert anyway. -- The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has 2 ports), so I've changed the the id of the platform_device from 0 to -1. -- In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link structure with the DM355 EVM as they use different cpu DAI names (the DM355 has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port). This also means that the 2 boards need different snd_soc_card structures. -- The codec_name entries in davinci-evm.c didn't match the i2c ids in the board files. I have only checked and fixed the details of the names used for the McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should check the others. Signed-off-by: Chris Paulson-Ellis <chris@edesix.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-16 05:27:09 -07:00
static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
return 0;
}
static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
ASoC: davinci: fixes for multi-component Multi-component commit f0fba2ad broke a few things which this patch should fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be good if those with access to other Davinci boards could test. -- The multi-component commit put the initialisation of snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c). The initialisation had to be moved from the probe function in these drivers because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver. Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in davinci-pcm.c) before hw_params is called. I have moved the initialisation to a new snd_soc_dai_ops.startup function in each of these drivers. This fix indicates that all platforms that use davinci-pcm must have been broken and need to test with this fix. -- The multi-component commit also changed the McBSP driver name from "davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board level references to the driver name. This change is understandable, as there is a similarly named "davinci-mcasp" driver in davinci-mcasp.c. There is probably no 'correct' name for this driver. The DM6446 datasheet calls it the "ASP" and describes it as a "specialised McBSP". The DM355 datasheet calls it the "ASP" and describes it as a "specialised ASP". The DM365 datasheet calls it the "McBSP". Rather than fix this problem by reverting to "davinci-asp", I've elected to avoid future confusion with the "davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also consistent with the names of the functions in the driver. There are other fixes required, so it was never going to be as simple as a revert anyway. -- The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has 2 ports), so I've changed the the id of the platform_device from 0 to -1. -- In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link structure with the DM355 EVM as they use different cpu DAI names (the DM355 has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port). This also means that the 2 boards need different snd_soc_card structures. -- The codec_name entries in davinci-evm.c didn't match the i2c ids in the board files. I have only checked and fixed the details of the names used for the McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should check the others. Signed-off-by: Chris Paulson-Ellis <chris@edesix.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-16 05:27:09 -07:00
.startup = davinci_mcasp_startup,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
.set_clkdiv = davinci_mcasp_set_clkdiv,
.set_sysclk = davinci_mcasp_set_sysclk,
};
#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
SNDRV_PCM_FMTBIT_U8 | \
SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_U16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_U24_LE | \
SNDRV_PCM_FMTBIT_S24_3LE | \
SNDRV_PCM_FMTBIT_U24_3LE | \
SNDRV_PCM_FMTBIT_S32_LE | \
SNDRV_PCM_FMTBIT_U32_LE)
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
{
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
.name = "davinci-mcasp.0",
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
.ops = &davinci_mcasp_dai_ops,
},
{
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
"davinci-mcasp.1",
.playback = {
.channels_min = 1,
.channels_max = 384,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
.ops = &davinci_mcasp_dai_ops,
},
};
static const struct of_device_id mcasp_dt_ids[] = {
{
.compatible = "ti,dm646x-mcasp-audio",
.data = (void *)MCASP_VERSION_1,
},
{
.compatible = "ti,da830-mcasp-audio",
.data = (void *)MCASP_VERSION_2,
},
{
.compatible = "ti,omap2-mcasp-audio",
.data = (void *)MCASP_VERSION_3,
},
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, mcasp_dt_ids);
static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct snd_platform_data *pdata = NULL;
const struct of_device_id *match =
of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev);
const u32 *of_serial_dir32;
u8 *of_serial_dir;
u32 val;
int i, ret = 0;
if (pdev->dev.platform_data) {
pdata = pdev->dev.platform_data;
return pdata;
} else if (match) {
pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
if (!pdata) {
ret = -ENOMEM;
goto nodata;
}
} else {
/* control shouldn't reach here. something is wrong */
ret = -EINVAL;
goto nodata;
}
if (match->data)
pdata->version = (u8)((int)match->data);
ret = of_property_read_u32(np, "op-mode", &val);
if (ret >= 0)
pdata->op_mode = val;
ret = of_property_read_u32(np, "tdm-slots", &val);
if (ret >= 0)
pdata->tdm_slots = val;
ret = of_property_read_u32(np, "num-serializer", &val);
if (ret >= 0)
pdata->num_serializer = val;
of_serial_dir32 = of_get_property(np, "serial-dir", &val);
val /= sizeof(u32);
if (val != pdata->num_serializer) {
dev_err(&pdev->dev,
"num-serializer(%d) != serial-dir size(%d)\n",
pdata->num_serializer, val);
ret = -EINVAL;
goto nodata;
}
if (of_serial_dir32) {
of_serial_dir = devm_kzalloc(&pdev->dev,
(sizeof(*of_serial_dir) * val),
GFP_KERNEL);
if (!of_serial_dir) {
ret = -ENOMEM;
goto nodata;
}
for (i = 0; i < pdata->num_serializer; i++)
of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]);
pdata->serial_dir = of_serial_dir;
}
ret = of_property_read_u32(np, "tx-num-evt", &val);
if (ret >= 0)
pdata->txnumevt = val;
ret = of_property_read_u32(np, "rx-num-evt", &val);
if (ret >= 0)
pdata->rxnumevt = val;
ret = of_property_read_u32(np, "sram-size-playback", &val);
if (ret >= 0)
pdata->sram_size_playback = val;
ret = of_property_read_u32(np, "sram-size-capture", &val);
if (ret >= 0)
pdata->sram_size_capture = val;
return pdata;
nodata:
if (ret < 0) {
dev_err(&pdev->dev, "Error populating platform data, err %d\n",
ret);
pdata = NULL;
}
return pdata;
}
static int davinci_mcasp_probe(struct platform_device *pdev)
{
struct davinci_pcm_dma_params *dma_data;
struct resource *mem, *ioarea, *res;
struct snd_platform_data *pdata;
struct davinci_audio_dev *dev;
int ret;
if (!pdev->dev.platform_data && !pdev->dev.of_node) {
dev_err(&pdev->dev, "No platform data supplied\n");
return -EINVAL;
}
dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev),
GFP_KERNEL);
if (!dev)
return -ENOMEM;
pdata = davinci_mcasp_set_pdata_from_of(pdev);
if (!pdata) {
dev_err(&pdev->dev, "no platform data\n");
return -EINVAL;
}
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "no mem resource?\n");
return -ENODEV;
}
ioarea = devm_request_mem_region(&pdev->dev, mem->start,
resource_size(mem), pdev->name);
if (!ioarea) {
dev_err(&pdev->dev, "Audio region already claimed\n");
return -EBUSY;
}
pm_runtime_enable(&pdev->dev);
ret = pm_runtime_get_sync(&pdev->dev);
if (IS_ERR_VALUE(ret)) {
dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n");
return ret;
}
dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
if (!dev->base) {
dev_err(&pdev->dev, "ioremap failed\n");
ret = -ENOMEM;
goto err_release_clk;
}
dev->op_mode = pdata->op_mode;
dev->tdm_slots = pdata->tdm_slots;
dev->num_serializer = pdata->num_serializer;
dev->serial_dir = pdata->serial_dir;
dev->codec_fmt = pdata->codec_fmt;
dev->version = pdata->version;
dev->txnumevt = pdata->txnumevt;
dev->rxnumevt = pdata->rxnumevt;
dev->dev = &pdev->dev;
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->sram_size = pdata->sram_size_playback;
dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
mem->start);
/* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!res) {
dev_err(&pdev->dev, "no DMA resource\n");
ret = -ENODEV;
goto err_release_clk;
}
dma_data->channel = res->start;
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->sram_size = pdata->sram_size_capture;
dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
mem->start);
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
dev_err(&pdev->dev, "no DMA resource\n");
ret = -ENODEV;
goto err_release_clk;
}
dma_data->channel = res->start;
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
dev_set_drvdata(&pdev->dev, dev);
ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]);
if (ret != 0)
goto err_release_clk;
ret = davinci_soc_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
goto err_unregister_dai;
}
return 0;
err_unregister_dai:
snd_soc_unregister_dai(&pdev->dev);
err_release_clk:
pm_runtime_put_sync(&pdev->dev);
pm_runtime_disable(&pdev->dev);
return ret;
}
static int davinci_mcasp_remove(struct platform_device *pdev)
{
ASoC: multi-component - ASoC Multi-Component Support 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>
2010-03-17 14:15:21 -06:00
snd_soc_unregister_dai(&pdev->dev);
davinci_soc_platform_unregister(&pdev->dev);
pm_runtime_put_sync(&pdev->dev);
pm_runtime_disable(&pdev->dev);
return 0;
}
static struct platform_driver davinci_mcasp_driver = {
.probe = davinci_mcasp_probe,
.remove = davinci_mcasp_remove,
.driver = {
.name = "davinci-mcasp",
.owner = THIS_MODULE,
.of_match_table = of_match_ptr(mcasp_dt_ids),
},
};
module_platform_driver(davinci_mcasp_driver);
MODULE_AUTHOR("Steve Chen");
MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface");
MODULE_LICENSE("GPL");