linux/sound/soc/sh/fsi.c

2274 lines
49 KiB
C
Raw Normal View History

/*
* Fifo-attached Serial Interface (FSI) support for SH7724
*
* Copyright (C) 2009 Renesas Solutions Corp.
* Kuninori Morimoto <morimoto.kuninori@renesas.com>
*
* Based on ssi.c
* Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
*
* 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/delay.h>
#include <linux/dma-mapping.h>
#include <linux/pm_runtime.h>
#include <linux/io.h>
#include <linux/scatterlist.h>
#include <linux/sh_dma.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 16:04:11 +08:00
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/workqueue.h>
#include <sound/soc.h>
#include <sound/pcm_params.h>
#include <sound/sh_fsi.h>
/* PortA/PortB register */
#define REG_DO_FMT 0x0000
#define REG_DOFF_CTL 0x0004
#define REG_DOFF_ST 0x0008
#define REG_DI_FMT 0x000C
#define REG_DIFF_CTL 0x0010
#define REG_DIFF_ST 0x0014
#define REG_CKG1 0x0018
#define REG_CKG2 0x001C
#define REG_DIDT 0x0020
#define REG_DODT 0x0024
#define REG_MUTE_ST 0x0028
#define REG_OUT_DMAC 0x002C
#define REG_OUT_SEL 0x0030
#define REG_IN_DMAC 0x0038
/* master register */
#define MST_CLK_RST 0x0210
#define MST_SOFT_RST 0x0214
#define MST_FIFO_SZ 0x0218
/* core register (depend on FSI version) */
#define A_MST_CTLR 0x0180
#define B_MST_CTLR 0x01A0
#define CPU_INT_ST 0x01F4
#define CPU_IEMSK 0x01F8
#define CPU_IMSK 0x01FC
#define INT_ST 0x0200
#define IEMSK 0x0204
#define IMSK 0x0208
/* DO_FMT */
/* DI_FMT */
#define CR_BWS_MASK (0x3 << 20) /* FSI2 */
#define CR_BWS_24 (0x0 << 20) /* FSI2 */
#define CR_BWS_16 (0x1 << 20) /* FSI2 */
#define CR_BWS_20 (0x2 << 20) /* FSI2 */
#define CR_DTMD_PCM (0x0 << 8) /* FSI2 */
#define CR_DTMD_SPDIF_PCM (0x1 << 8) /* FSI2 */
#define CR_DTMD_SPDIF_STREAM (0x2 << 8) /* FSI2 */
#define CR_MONO (0x0 << 4)
#define CR_MONO_D (0x1 << 4)
#define CR_PCM (0x2 << 4)
#define CR_I2S (0x3 << 4)
#define CR_TDM (0x4 << 4)
#define CR_TDM_D (0x5 << 4)
/* OUT_DMAC */
/* IN_DMAC */
#define VDMD_MASK (0x3 << 4)
#define VDMD_FRONT (0x0 << 4) /* Package in front */
#define VDMD_BACK (0x1 << 4) /* Package in back */
#define VDMD_STREAM (0x2 << 4) /* Stream mode(16bit * 2) */
#define DMA_ON (0x1 << 0)
/* DOFF_CTL */
/* DIFF_CTL */
#define IRQ_HALF 0x00100000
#define FIFO_CLR 0x00000001
/* DOFF_ST */
#define ERR_OVER 0x00000010
#define ERR_UNDER 0x00000001
#define ST_ERR (ERR_OVER | ERR_UNDER)
/* CKG1 */
#define ACKMD_MASK 0x00007000
#define BPFMD_MASK 0x00000700
#define DIMD (1 << 4)
#define DOMD (1 << 0)
/* A/B MST_CTLR */
#define BP (1 << 4) /* Fix the signal of Biphase output */
#define SE (1 << 0) /* Fix the master clock */
/* CLK_RST */
#define CRB (1 << 4)
#define CRA (1 << 0)
/* IO SHIFT / MACRO */
#define BI_SHIFT 12
#define BO_SHIFT 8
#define AI_SHIFT 4
#define AO_SHIFT 0
#define AB_IO(param, shift) (param << shift)
/* SOFT_RST */
#define PBSR (1 << 12) /* Port B Software Reset */
#define PASR (1 << 8) /* Port A Software Reset */
#define IR (1 << 4) /* Interrupt Reset */
#define FSISR (1 << 0) /* Software Reset */
/* OUT_SEL (FSI2) */
#define DMMD (1 << 4) /* SPDIF output timing 0: Biphase only */
/* 1: Biphase and serial */
/* FIFO_SZ */
#define FIFO_SZ_MASK 0x7
#define FSI_RATES SNDRV_PCM_RATE_8000_96000
#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
typedef int (*set_rate_func)(struct device *dev, int rate, int enable);
/*
* bus options
*
* 0x000000BA
*
* A : sample widtht 16bit setting
* B : sample widtht 24bit setting
*/
#define SHIFT_16DATA 0
#define SHIFT_24DATA 4
#define PACKAGE_24BITBUS_BACK 0
#define PACKAGE_24BITBUS_FRONT 1
#define PACKAGE_16BITBUS_STREAM 2
#define BUSOP_SET(s, a) ((a) << SHIFT_ ## s ## DATA)
#define BUSOP_GET(s, a) (((a) >> SHIFT_ ## s ## DATA) & 0xF)
/*
* FSI driver use below type name for variable
*
* xxx_num : number of data
* xxx_pos : position of data
* xxx_capa : capacity of data
*/
/*
* period/frame/sample image
*
* ex) PCM (2ch)
*
* period pos period pos
* [n] [n + 1]
* |<-------------------- period--------------------->|
* ==|============================================ ... =|==
* | |
* ||<----- frame ----->|<------ frame ----->| ... |
* |+--------------------+--------------------+- ... |
* ||[ sample ][ sample ]|[ sample ][ sample ]| ... |
* |+--------------------+--------------------+- ... |
* ==|============================================ ... =|==
*/
/*
* FSI FIFO image
*
* | |
* | |
* | [ sample ] |
* | [ sample ] |
* | [ sample ] |
* | [ sample ] |
* --> go to codecs
*/
/*
* FSI clock
*
* FSIxCLK [CPG] (ick) -------> |
* |-> FSI_DIV (div)-> FSI2
* FSIxCK [external] (xck) ---> |
*/
/*
* struct
*/
struct fsi_stream_handler;
struct fsi_stream {
/*
* these are initialized by fsi_stream_init()
*/
struct snd_pcm_substream *substream;
int fifo_sample_capa; /* sample capacity of FSI FIFO */
int buff_sample_capa; /* sample capacity of ALSA buffer */
int buff_sample_pos; /* sample position of ALSA buffer */
int period_samples; /* sample number / 1 period */
int period_pos; /* current period position */
int sample_width; /* sample width */
int uerr_num;
int oerr_num;
/*
* bus options
*/
u32 bus_option;
/*
* thse are initialized by fsi_handler_init()
*/
struct fsi_stream_handler *handler;
struct fsi_priv *priv;
/*
* these are for DMAEngine
*/
struct dma_chan *chan;
struct sh_dmae_slave slave; /* see fsi_handler_init() */
struct work_struct work;
dma_addr_t dma;
};
struct fsi_clk {
/* see [FSI clock] */
struct clk *own;
struct clk *xck;
struct clk *ick;
struct clk *div;
int (*set_rate)(struct device *dev,
struct fsi_priv *fsi,
unsigned long rate);
unsigned long rate;
unsigned int count;
};
struct fsi_priv {
void __iomem *base;
struct fsi_master *master;
struct sh_fsi_port_info *info;
struct fsi_stream playback;
struct fsi_stream capture;
struct fsi_clk clock;
u32 fmt;
int chan_num:16;
int clk_master:1;
int clk_cpg:1;
int spdif:1;
int enable_stream:1;
int bit_clk_inv:1;
int lr_clk_inv:1;
long rate;
};
struct fsi_stream_handler {
int (*init)(struct fsi_priv *fsi, struct fsi_stream *io);
int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io);
int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev);
int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io);
int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io);
void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io,
int enable);
};
#define fsi_stream_handler_call(io, func, args...) \
(!(io) ? -ENODEV : \
!((io)->handler->func) ? 0 : \
(io)->handler->func(args))
struct fsi_core {
int ver;
u32 int_st;
u32 iemsk;
u32 imsk;
u32 a_mclk;
u32 b_mclk;
};
struct fsi_master {
void __iomem *base;
int irq;
struct fsi_priv fsia;
struct fsi_priv fsib;
struct fsi_core *core;
spinlock_t lock;
};
static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io);
/*
* basic read write function
*/
static void __fsi_reg_write(u32 __iomem *reg, u32 data)
{
/* valid data area is 24bit */
data &= 0x00ffffff;
__raw_writel(data, reg);
}
static u32 __fsi_reg_read(u32 __iomem *reg)
{
return __raw_readl(reg);
}
static void __fsi_reg_mask_set(u32 __iomem *reg, u32 mask, u32 data)
{
u32 val = __fsi_reg_read(reg);
val &= ~mask;
val |= data & mask;
__fsi_reg_write(reg, val);
}
#define fsi_reg_write(p, r, d)\
__fsi_reg_write((p->base + REG_##r), d)
#define fsi_reg_read(p, r)\
__fsi_reg_read((p->base + REG_##r))
#define fsi_reg_mask_set(p, r, m, d)\
__fsi_reg_mask_set((p->base + REG_##r), m, d)
#define fsi_master_read(p, r) _fsi_master_read(p, MST_##r)
#define fsi_core_read(p, r) _fsi_master_read(p, p->core->r)
static u32 _fsi_master_read(struct fsi_master *master, u32 reg)
{
u32 ret;
unsigned long flags;
spin_lock_irqsave(&master->lock, flags);
ret = __fsi_reg_read(master->base + reg);
spin_unlock_irqrestore(&master->lock, flags);
return ret;
}
#define fsi_master_mask_set(p, r, m, d) _fsi_master_mask_set(p, MST_##r, m, d)
#define fsi_core_mask_set(p, r, m, d) _fsi_master_mask_set(p, p->core->r, m, d)
static void _fsi_master_mask_set(struct fsi_master *master,
u32 reg, u32 mask, u32 data)
{
unsigned long flags;
spin_lock_irqsave(&master->lock, flags);
__fsi_reg_mask_set(master->base + reg, mask, data);
spin_unlock_irqrestore(&master->lock, flags);
}
/*
* basic function
*/
static int fsi_version(struct fsi_master *master)
{
return master->core->ver;
}
static struct fsi_master *fsi_get_master(struct fsi_priv *fsi)
{
return fsi->master;
}
static int fsi_is_clk_master(struct fsi_priv *fsi)
{
return fsi->clk_master;
}
static int fsi_is_port_a(struct fsi_priv *fsi)
{
return fsi->master->base == fsi->base;
}
static int fsi_is_spdif(struct fsi_priv *fsi)
{
return fsi->spdif;
}
static int fsi_is_enable_stream(struct fsi_priv *fsi)
{
return fsi->enable_stream;
}
static int fsi_is_play(struct snd_pcm_substream *substream)
{
return substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
}
static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
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>
2010-03-18 04:15:21 +08:00
return rtd->cpu_dai;
}
static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *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>
2010-03-18 04:15:21 +08:00
struct fsi_master *master = snd_soc_dai_get_drvdata(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>
2010-03-18 04:15:21 +08:00
if (dai->id == 0)
return &master->fsia;
else
return &master->fsib;
}
static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
{
return fsi_get_priv_frm_dai(fsi_get_dai(substream));
}
static set_rate_func fsi_get_info_set_rate(struct fsi_priv *fsi)
{
if (!fsi->info)
return NULL;
return fsi->info->set_rate;
}
static u32 fsi_get_info_flags(struct fsi_priv *fsi)
{
if (!fsi->info)
return 0;
return fsi->info->flags;
}
static u32 fsi_get_port_shift(struct fsi_priv *fsi, struct fsi_stream *io)
{
int is_play = fsi_stream_is_play(fsi, io);
int is_porta = fsi_is_port_a(fsi);
u32 shift;
if (is_porta)
shift = is_play ? AO_SHIFT : AI_SHIFT;
else
shift = is_play ? BO_SHIFT : BI_SHIFT;
return shift;
}
static int fsi_frame2sample(struct fsi_priv *fsi, int frames)
{
return frames * fsi->chan_num;
}
static int fsi_sample2frame(struct fsi_priv *fsi, int samples)
{
return samples / fsi->chan_num;
}
static int fsi_get_current_fifo_samples(struct fsi_priv *fsi,
struct fsi_stream *io)
{
int is_play = fsi_stream_is_play(fsi, io);
u32 status;
int frames;
status = is_play ?
fsi_reg_read(fsi, DOFF_ST) :
fsi_reg_read(fsi, DIFF_ST);
frames = 0x1ff & (status >> 8);
return fsi_frame2sample(fsi, frames);
}
static void fsi_count_fifo_err(struct fsi_priv *fsi)
{
u32 ostatus = fsi_reg_read(fsi, DOFF_ST);
u32 istatus = fsi_reg_read(fsi, DIFF_ST);
if (ostatus & ERR_OVER)
fsi->playback.oerr_num++;
if (ostatus & ERR_UNDER)
fsi->playback.uerr_num++;
if (istatus & ERR_OVER)
fsi->capture.oerr_num++;
if (istatus & ERR_UNDER)
fsi->capture.uerr_num++;
fsi_reg_write(fsi, DOFF_ST, 0);
fsi_reg_write(fsi, DIFF_ST, 0);
}
/*
* fsi_stream_xx() function
*/
static inline int fsi_stream_is_play(struct fsi_priv *fsi,
struct fsi_stream *io)
{
return &fsi->playback == io;
}
static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi,
struct snd_pcm_substream *substream)
{
return fsi_is_play(substream) ? &fsi->playback : &fsi->capture;
}
static int fsi_stream_is_working(struct fsi_priv *fsi,
struct fsi_stream *io)
{
struct fsi_master *master = fsi_get_master(fsi);
unsigned long flags;
int ret;
spin_lock_irqsave(&master->lock, flags);
ret = !!(io->substream && io->substream->runtime);
spin_unlock_irqrestore(&master->lock, flags);
return ret;
}
static struct fsi_priv *fsi_stream_to_priv(struct fsi_stream *io)
{
return io->priv;
}
static void fsi_stream_init(struct fsi_priv *fsi,
struct fsi_stream *io,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsi_master *master = fsi_get_master(fsi);
unsigned long flags;
spin_lock_irqsave(&master->lock, flags);
io->substream = substream;
io->buff_sample_capa = fsi_frame2sample(fsi, runtime->buffer_size);
io->buff_sample_pos = 0;
io->period_samples = fsi_frame2sample(fsi, runtime->period_size);
io->period_pos = 0;
io->sample_width = samples_to_bytes(runtime, 1);
io->bus_option = 0;
io->oerr_num = -1; /* ignore 1st err */
io->uerr_num = -1; /* ignore 1st err */
fsi_stream_handler_call(io, init, fsi, io);
spin_unlock_irqrestore(&master->lock, flags);
}
static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io)
{
struct snd_soc_dai *dai = fsi_get_dai(io->substream);
struct fsi_master *master = fsi_get_master(fsi);
unsigned long flags;
spin_lock_irqsave(&master->lock, flags);
if (io->oerr_num > 0)
dev_err(dai->dev, "over_run = %d\n", io->oerr_num);
if (io->uerr_num > 0)
dev_err(dai->dev, "under_run = %d\n", io->uerr_num);
fsi_stream_handler_call(io, quit, fsi, io);
io->substream = NULL;
io->buff_sample_capa = 0;
io->buff_sample_pos = 0;
io->period_samples = 0;
io->period_pos = 0;
io->sample_width = 0;
io->bus_option = 0;
io->oerr_num = 0;
io->uerr_num = 0;
spin_unlock_irqrestore(&master->lock, flags);
}
static int fsi_stream_transfer(struct fsi_stream *io)
{
struct fsi_priv *fsi = fsi_stream_to_priv(io);
if (!fsi)
return -EIO;
return fsi_stream_handler_call(io, transfer, fsi, io);
}
#define fsi_stream_start(fsi, io)\
fsi_stream_handler_call(io, start_stop, fsi, io, 1)
#define fsi_stream_stop(fsi, io)\
fsi_stream_handler_call(io, start_stop, fsi, io, 0)
static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev)
{
struct fsi_stream *io;
int ret1, ret2;
io = &fsi->playback;
ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev);
io = &fsi->capture;
ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev);
if (ret1 < 0)
return ret1;
if (ret2 < 0)
return ret2;
return 0;
}
static int fsi_stream_remove(struct fsi_priv *fsi)
{
struct fsi_stream *io;
int ret1, ret2;
io = &fsi->playback;
ret1 = fsi_stream_handler_call(io, remove, fsi, io);
io = &fsi->capture;
ret2 = fsi_stream_handler_call(io, remove, fsi, io);
if (ret1 < 0)
return ret1;
if (ret2 < 0)
return ret2;
return 0;
}
/*
* format/bus/dma setting
*/
static void fsi_format_bus_setup(struct fsi_priv *fsi, struct fsi_stream *io,
u32 bus, struct device *dev)
{
struct fsi_master *master = fsi_get_master(fsi);
int is_play = fsi_stream_is_play(fsi, io);
u32 fmt = fsi->fmt;
if (fsi_version(master) >= 2) {
u32 dma = 0;
/*
* FSI2 needs DMA/Bus setting
*/
switch (bus) {
case PACKAGE_24BITBUS_FRONT:
fmt |= CR_BWS_24;
dma |= VDMD_FRONT;
dev_dbg(dev, "24bit bus / package in front\n");
break;
case PACKAGE_16BITBUS_STREAM:
fmt |= CR_BWS_16;
dma |= VDMD_STREAM;
dev_dbg(dev, "16bit bus / stream mode\n");
break;
case PACKAGE_24BITBUS_BACK:
default:
fmt |= CR_BWS_24;
dma |= VDMD_BACK;
dev_dbg(dev, "24bit bus / package in back\n");
break;
}
if (is_play)
fsi_reg_write(fsi, OUT_DMAC, dma);
else
fsi_reg_write(fsi, IN_DMAC, dma);
}
if (is_play)
fsi_reg_write(fsi, DO_FMT, fmt);
else
fsi_reg_write(fsi, DI_FMT, fmt);
}
/*
* irq function
*/
static void fsi_irq_enable(struct fsi_priv *fsi, struct fsi_stream *io)
{
u32 data = AB_IO(1, fsi_get_port_shift(fsi, io));
struct fsi_master *master = fsi_get_master(fsi);
fsi_core_mask_set(master, imsk, data, data);
fsi_core_mask_set(master, iemsk, data, data);
}
static void fsi_irq_disable(struct fsi_priv *fsi, struct fsi_stream *io)
{
u32 data = AB_IO(1, fsi_get_port_shift(fsi, io));
struct fsi_master *master = fsi_get_master(fsi);
fsi_core_mask_set(master, imsk, data, 0);
fsi_core_mask_set(master, iemsk, data, 0);
}
static u32 fsi_irq_get_status(struct fsi_master *master)
{
return fsi_core_read(master, int_st);
}
static void fsi_irq_clear_status(struct fsi_priv *fsi)
{
u32 data = 0;
struct fsi_master *master = fsi_get_master(fsi);
data |= AB_IO(1, fsi_get_port_shift(fsi, &fsi->playback));
data |= AB_IO(1, fsi_get_port_shift(fsi, &fsi->capture));
/* clear interrupt factor */
fsi_core_mask_set(master, int_st, data, 0);
}
/*
* SPDIF master clock function
*
* These functions are used later FSI2
*/
static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
{
struct fsi_master *master = fsi_get_master(fsi);
u32 mask, val;
mask = BP | SE;
val = enable ? mask : 0;
fsi_is_port_a(fsi) ?
fsi_core_mask_set(master, a_mclk, mask, val) :
fsi_core_mask_set(master, b_mclk, mask, val);
}
/*
* clock function
*/
static int fsi_clk_init(struct device *dev,
struct fsi_priv *fsi,
int xck,
int ick,
int div,
int (*set_rate)(struct device *dev,
struct fsi_priv *fsi,
unsigned long rate))
{
struct fsi_clk *clock = &fsi->clock;
int is_porta = fsi_is_port_a(fsi);
clock->xck = NULL;
clock->ick = NULL;
clock->div = NULL;
clock->rate = 0;
clock->count = 0;
clock->set_rate = set_rate;
clock->own = devm_clk_get(dev, NULL);
if (IS_ERR(clock->own))
return -EINVAL;
/* external clock */
if (xck) {
clock->xck = devm_clk_get(dev, is_porta ? "xcka" : "xckb");
if (IS_ERR(clock->xck)) {
dev_err(dev, "can't get xck clock\n");
return -EINVAL;
}
if (clock->xck == clock->own) {
dev_err(dev, "cpu doesn't support xck clock\n");
return -EINVAL;
}
}
/* FSIACLK/FSIBCLK */
if (ick) {
clock->ick = devm_clk_get(dev, is_porta ? "icka" : "ickb");
if (IS_ERR(clock->ick)) {
dev_err(dev, "can't get ick clock\n");
return -EINVAL;
}
if (clock->ick == clock->own) {
dev_err(dev, "cpu doesn't support ick clock\n");
return -EINVAL;
}
}
/* FSI-DIV */
if (div) {
clock->div = devm_clk_get(dev, is_porta ? "diva" : "divb");
if (IS_ERR(clock->div)) {
dev_err(dev, "can't get div clock\n");
return -EINVAL;
}
if (clock->div == clock->own) {
dev_err(dev, "cpu doens't support div clock\n");
return -EINVAL;
}
}
return 0;
}
#define fsi_clk_invalid(fsi) fsi_clk_valid(fsi, 0)
static void fsi_clk_valid(struct fsi_priv *fsi, unsigned long rate)
{
fsi->clock.rate = rate;
}
static int fsi_clk_is_valid(struct fsi_priv *fsi)
{
return fsi->clock.set_rate &&
fsi->clock.rate;
}
static int fsi_clk_enable(struct device *dev,
struct fsi_priv *fsi,
unsigned long rate)
{
struct fsi_clk *clock = &fsi->clock;
int ret = -EINVAL;
if (!fsi_clk_is_valid(fsi))
return ret;
if (0 == clock->count) {
ret = clock->set_rate(dev, fsi, rate);
if (ret < 0) {
fsi_clk_invalid(fsi);
return ret;
}
if (clock->xck)
clk_enable(clock->xck);
if (clock->ick)
clk_enable(clock->ick);
if (clock->div)
clk_enable(clock->div);
clock->count++;
}
return ret;
}
static int fsi_clk_disable(struct device *dev,
struct fsi_priv *fsi)
{
struct fsi_clk *clock = &fsi->clock;
if (!fsi_clk_is_valid(fsi))
return -EINVAL;
if (1 == clock->count--) {
if (clock->xck)
clk_disable(clock->xck);
if (clock->ick)
clk_disable(clock->ick);
if (clock->div)
clk_disable(clock->div);
}
return 0;
}
static int fsi_clk_set_ackbpf(struct device *dev,
struct fsi_priv *fsi,
int ackmd, int bpfmd)
{
u32 data = 0;
/* check ackmd/bpfmd relationship */
if (bpfmd > ackmd) {
dev_err(dev, "unsupported rate (%d/%d)\n", ackmd, bpfmd);
return -EINVAL;
}
/* ACKMD */
switch (ackmd) {
case 512:
data |= (0x0 << 12);
break;
case 256:
data |= (0x1 << 12);
break;
case 128:
data |= (0x2 << 12);
break;
case 64:
data |= (0x3 << 12);
break;
case 32:
data |= (0x4 << 12);
break;
default:
dev_err(dev, "unsupported ackmd (%d)\n", ackmd);
return -EINVAL;
}
/* BPFMD */
switch (bpfmd) {
case 32:
data |= (0x0 << 8);
break;
case 64:
data |= (0x1 << 8);
break;
case 128:
data |= (0x2 << 8);
break;
case 256:
data |= (0x3 << 8);
break;
case 512:
data |= (0x4 << 8);
break;
case 16:
data |= (0x7 << 8);
break;
default:
dev_err(dev, "unsupported bpfmd (%d)\n", bpfmd);
return -EINVAL;
}
dev_dbg(dev, "ACKMD/BPFMD = %d/%d\n", ackmd, bpfmd);
fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
udelay(10);
return 0;
}
static int fsi_clk_set_rate_external(struct device *dev,
struct fsi_priv *fsi,
unsigned long rate)
{
struct clk *xck = fsi->clock.xck;
struct clk *ick = fsi->clock.ick;
unsigned long xrate;
int ackmd, bpfmd;
int ret = 0;
/* check clock rate */
xrate = clk_get_rate(xck);
if (xrate % rate) {
dev_err(dev, "unsupported clock rate\n");
return -EINVAL;
}
clk_set_parent(ick, xck);
clk_set_rate(ick, xrate);
bpfmd = fsi->chan_num * 32;
ackmd = xrate / rate;
dev_dbg(dev, "external/rate = %ld/%ld\n", xrate, rate);
ret = fsi_clk_set_ackbpf(dev, fsi, ackmd, bpfmd);
if (ret < 0)
dev_err(dev, "%s failed", __func__);
return ret;
}
static int fsi_clk_set_rate_cpg(struct device *dev,
struct fsi_priv *fsi,
unsigned long rate)
{
struct clk *ick = fsi->clock.ick;
struct clk *div = fsi->clock.div;
unsigned long target = 0; /* 12288000 or 11289600 */
unsigned long actual, cout;
unsigned long diff, min;
unsigned long best_cout, best_act;
int adj;
int ackmd, bpfmd;
int ret = -EINVAL;
if (!(12288000 % rate))
target = 12288000;
if (!(11289600 % rate))
target = 11289600;
if (!target) {
dev_err(dev, "unsupported rate\n");
return ret;
}
bpfmd = fsi->chan_num * 32;
ackmd = target / rate;
ret = fsi_clk_set_ackbpf(dev, fsi, ackmd, bpfmd);
if (ret < 0) {
dev_err(dev, "%s failed", __func__);
return ret;
}
/*
* The clock flow is
*
* [CPG] = cout => [FSI_DIV] = audio => [FSI] => [codec]
*
* But, it needs to find best match of CPG and FSI_DIV
* combination, since it is difficult to generate correct
* frequency of audio clock from ick clock only.
* Because ick is created from its parent clock.
*
* target = rate x [512/256/128/64]fs
* cout = round(target x adjustment)
* actual = cout / adjustment (by FSI-DIV) ~= target
* audio = actual
*/
min = ~0;
best_cout = 0;
best_act = 0;
for (adj = 1; adj < 0xffff; adj++) {
cout = target * adj;
if (cout > 100000000) /* max clock = 100MHz */
break;
/* cout/actual audio clock */
cout = clk_round_rate(ick, cout);
actual = cout / adj;
/* find best frequency */
diff = abs(actual - target);
if (diff < min) {
min = diff;
best_cout = cout;
best_act = actual;
}
}
ret = clk_set_rate(ick, best_cout);
if (ret < 0) {
dev_err(dev, "ick clock failed\n");
return -EIO;
}
ret = clk_set_rate(div, clk_round_rate(div, best_act));
if (ret < 0) {
dev_err(dev, "div clock failed\n");
return -EIO;
}
dev_dbg(dev, "ick/div = %ld/%ld\n",
clk_get_rate(ick), clk_get_rate(div));
return ret;
}
static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
long rate, int enable)
{
set_rate_func set_rate = fsi_get_info_set_rate(fsi);
int ret;
/*
* CAUTION
*
* set_rate will be deleted
*/
if (!set_rate) {
if (enable)
return fsi_clk_enable(dev, fsi, rate);
else
return fsi_clk_disable(dev, fsi);
}
ret = set_rate(dev, rate, enable);
if (ret < 0) /* error */
return ret;
if (!enable)
return 0;
if (ret > 0) {
u32 data = 0;
switch (ret & SH_FSI_ACKMD_MASK) {
default:
/* FALL THROUGH */
case SH_FSI_ACKMD_512:
data |= (0x0 << 12);
break;
case SH_FSI_ACKMD_256:
data |= (0x1 << 12);
break;
case SH_FSI_ACKMD_128:
data |= (0x2 << 12);
break;
case SH_FSI_ACKMD_64:
data |= (0x3 << 12);
break;
case SH_FSI_ACKMD_32:
data |= (0x4 << 12);
break;
}
switch (ret & SH_FSI_BPFMD_MASK) {
default:
/* FALL THROUGH */
case SH_FSI_BPFMD_32:
data |= (0x0 << 8);
break;
case SH_FSI_BPFMD_64:
data |= (0x1 << 8);
break;
case SH_FSI_BPFMD_128:
data |= (0x2 << 8);
break;
case SH_FSI_BPFMD_256:
data |= (0x3 << 8);
break;
case SH_FSI_BPFMD_512:
data |= (0x4 << 8);
break;
case SH_FSI_BPFMD_16:
data |= (0x7 << 8);
break;
}
fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
udelay(10);
ret = 0;
}
return ret;
}
/*
* pio data transfer handler
*/
static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int samples)
{
int i;
if (fsi_is_enable_stream(fsi)) {
/*
* stream mode
* see
* fsi_pio_push_init()
*/
u32 *buf = (u32 *)_buf;
for (i = 0; i < samples / 2; i++)
fsi_reg_write(fsi, DODT, buf[i]);
} else {
/* normal mode */
u16 *buf = (u16 *)_buf;
for (i = 0; i < samples; i++)
fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8));
}
}
static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int samples)
{
u16 *buf = (u16 *)_buf;
int i;
for (i = 0; i < samples; i++)
*(buf + i) = (u16)(fsi_reg_read(fsi, DIDT) >> 8);
}
static void fsi_pio_push32(struct fsi_priv *fsi, u8 *_buf, int samples)
{
u32 *buf = (u32 *)_buf;
int i;
for (i = 0; i < samples; i++)
fsi_reg_write(fsi, DODT, *(buf + i));
}
static void fsi_pio_pop32(struct fsi_priv *fsi, u8 *_buf, int samples)
{
u32 *buf = (u32 *)_buf;
int i;
for (i = 0; i < samples; i++)
*(buf + i) = fsi_reg_read(fsi, DIDT);
}
static u8 *fsi_pio_get_area(struct fsi_priv *fsi, struct fsi_stream *io)
{
struct snd_pcm_runtime *runtime = io->substream->runtime;
return runtime->dma_area +
samples_to_bytes(runtime, io->buff_sample_pos);
}
static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io,
void (*run16)(struct fsi_priv *fsi, u8 *buf, int samples),
void (*run32)(struct fsi_priv *fsi, u8 *buf, int samples),
int samples)
{
struct snd_pcm_runtime *runtime;
struct snd_pcm_substream *substream;
u8 *buf;
int over_period;
if (!fsi_stream_is_working(fsi, io))
return -EINVAL;
over_period = 0;
substream = io->substream;
runtime = substream->runtime;
/* FSI FIFO has limit.
* So, this driver can not send periods data at a time
*/
if (io->buff_sample_pos >=
io->period_samples * (io->period_pos + 1)) {
over_period = 1;
io->period_pos = (io->period_pos + 1) % runtime->periods;
if (0 == io->period_pos)
io->buff_sample_pos = 0;
}
buf = fsi_pio_get_area(fsi, io);
switch (io->sample_width) {
case 2:
run16(fsi, buf, samples);
break;
case 4:
run32(fsi, buf, samples);
break;
default:
return -EINVAL;
}
/* update buff_sample_pos */
io->buff_sample_pos += samples;
if (over_period)
snd_pcm_period_elapsed(substream);
return 0;
}
static int fsi_pio_pop(struct fsi_priv *fsi, struct fsi_stream *io)
{
int sample_residues; /* samples in FSI fifo */
int sample_space; /* ALSA free samples space */
int samples;
sample_residues = fsi_get_current_fifo_samples(fsi, io);
sample_space = io->buff_sample_capa - io->buff_sample_pos;
samples = min(sample_residues, sample_space);
return fsi_pio_transfer(fsi, io,
fsi_pio_pop16,
fsi_pio_pop32,
samples);
}
static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io)
{
int sample_residues; /* ALSA residue samples */
int sample_space; /* FSI fifo free samples space */
int samples;
sample_residues = io->buff_sample_capa - io->buff_sample_pos;
sample_space = io->fifo_sample_capa -
fsi_get_current_fifo_samples(fsi, io);
samples = min(sample_residues, sample_space);
return fsi_pio_transfer(fsi, io,
fsi_pio_push16,
fsi_pio_push32,
samples);
}
static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
int enable)
{
struct fsi_master *master = fsi_get_master(fsi);
u32 clk = fsi_is_port_a(fsi) ? CRA : CRB;
if (enable)
fsi_irq_enable(fsi, io);
else
fsi_irq_disable(fsi, io);
if (fsi_is_clk_master(fsi))
fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
}
static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io)
{
/*
* we can use 16bit stream mode
* when "playback" and "16bit data"
* and platform allows "stream mode"
* see
* fsi_pio_push16()
*/
if (fsi_is_enable_stream(fsi))
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
else
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_24BITBUS_BACK);
return 0;
}
static int fsi_pio_pop_init(struct fsi_priv *fsi, struct fsi_stream *io)
{
/*
* always 24bit bus, package back when "capture"
*/
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_24BITBUS_BACK);
return 0;
}
static struct fsi_stream_handler fsi_pio_push_handler = {
.init = fsi_pio_push_init,
.transfer = fsi_pio_push,
.start_stop = fsi_pio_start_stop,
};
static struct fsi_stream_handler fsi_pio_pop_handler = {
.init = fsi_pio_pop_init,
.transfer = fsi_pio_pop,
.start_stop = fsi_pio_start_stop,
};
static irqreturn_t fsi_interrupt(int irq, void *data)
{
struct fsi_master *master = data;
u32 int_st = fsi_irq_get_status(master);
/* clear irq status */
fsi_master_mask_set(master, SOFT_RST, IR, 0);
fsi_master_mask_set(master, SOFT_RST, IR, IR);
if (int_st & AB_IO(1, AO_SHIFT))
fsi_stream_transfer(&master->fsia.playback);
if (int_st & AB_IO(1, BO_SHIFT))
fsi_stream_transfer(&master->fsib.playback);
if (int_st & AB_IO(1, AI_SHIFT))
fsi_stream_transfer(&master->fsia.capture);
if (int_st & AB_IO(1, BI_SHIFT))
fsi_stream_transfer(&master->fsib.capture);
fsi_count_fifo_err(&master->fsia);
fsi_count_fifo_err(&master->fsib);
fsi_irq_clear_status(&master->fsia);
fsi_irq_clear_status(&master->fsib);
return IRQ_HANDLED;
}
/*
* dma data transfer handler
*/
static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
{
struct snd_pcm_runtime *runtime = io->substream->runtime;
struct snd_soc_dai *dai = fsi_get_dai(io->substream);
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
/*
* 24bit data : 24bit bus / package in back
* 16bit data : 16bit bus / stream mode
*/
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
io->dma = dma_map_single(dai->dev, runtime->dma_area,
snd_pcm_lib_buffer_bytes(io->substream), dir);
return 0;
}
static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
{
struct snd_soc_dai *dai = fsi_get_dai(io->substream);
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
dma_unmap_single(dai->dev, io->dma,
snd_pcm_lib_buffer_bytes(io->substream), dir);
return 0;
}
static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
{
struct snd_pcm_runtime *runtime = io->substream->runtime;
return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
}
static void fsi_dma_complete(void *data)
{
struct fsi_stream *io = (struct fsi_stream *)data;
struct fsi_priv *fsi = fsi_stream_to_priv(io);
struct snd_pcm_runtime *runtime = io->substream->runtime;
struct snd_soc_dai *dai = fsi_get_dai(io->substream);
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
samples_to_bytes(runtime, io->period_samples), dir);
io->buff_sample_pos += io->period_samples;
io->period_pos++;
if (io->period_pos >= runtime->periods) {
io->period_pos = 0;
io->buff_sample_pos = 0;
}
fsi_count_fifo_err(fsi);
fsi_stream_transfer(io);
snd_pcm_period_elapsed(io->substream);
}
static void fsi_dma_do_work(struct work_struct *work)
{
struct fsi_stream *io = container_of(work, struct fsi_stream, work);
struct fsi_priv *fsi = fsi_stream_to_priv(io);
struct snd_soc_dai *dai;
struct dma_async_tx_descriptor *desc;
struct snd_pcm_runtime *runtime;
enum dma_data_direction dir;
int is_play = fsi_stream_is_play(fsi, io);
int len;
dma_addr_t buf;
if (!fsi_stream_is_working(fsi, io))
return;
dai = fsi_get_dai(io->substream);
runtime = io->substream->runtime;
dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
len = samples_to_bytes(runtime, io->period_samples);
buf = fsi_dma_get_area(io);
dma_sync_single_for_device(dai->dev, buf, len, dir);
desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
if (!desc) {
dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
return;
}
desc->callback = fsi_dma_complete;
desc->callback_param = io;
if (dmaengine_submit(desc) < 0) {
dev_err(dai->dev, "tx_submit() fail\n");
return;
}
dma_async_issue_pending(io->chan);
/*
* FIXME
*
* In DMAEngine case, codec and FSI cannot be started simultaneously
* since FSI is using the scheduler work queue.
* Therefore, in capture case, probably FSI FIFO will have got
* overflow error in this point.
* in that case, DMA cannot start transfer until error was cleared.
*/
if (!is_play) {
if (ERR_OVER & fsi_reg_read(fsi, DIFF_ST)) {
fsi_reg_mask_set(fsi, DIFF_CTL, FIFO_CLR, FIFO_CLR);
fsi_reg_write(fsi, DIFF_ST, 0);
}
}
}
static bool fsi_dma_filter(struct dma_chan *chan, void *param)
{
struct sh_dmae_slave *slave = param;
chan->private = slave;
return true;
}
static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
{
schedule_work(&io->work);
return 0;
}
static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
int start)
{
struct fsi_master *master = fsi_get_master(fsi);
u32 clk = fsi_is_port_a(fsi) ? CRA : CRB;
u32 enable = start ? DMA_ON : 0;
fsi_reg_mask_set(fsi, OUT_DMAC, DMA_ON, enable);
dmaengine_terminate_all(io->chan);
if (fsi_is_clk_master(fsi))
fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
}
static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev)
{
dma_cap_mask_t mask;
dma_cap_zero(mask);
dma_cap_set(DMA_SLAVE, mask);
io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave);
if (!io->chan) {
/* switch to PIO handler */
if (fsi_stream_is_play(fsi, io))
fsi->playback.handler = &fsi_pio_push_handler;
else
fsi->capture.handler = &fsi_pio_pop_handler;
dev_info(dev, "switch handler (dma => pio)\n");
/* probe again */
return fsi_stream_probe(fsi, dev);
}
INIT_WORK(&io->work, fsi_dma_do_work);
return 0;
}
static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
{
cancel_work_sync(&io->work);
fsi_stream_stop(fsi, io);
if (io->chan)
dma_release_channel(io->chan);
io->chan = NULL;
return 0;
}
static struct fsi_stream_handler fsi_dma_push_handler = {
.init = fsi_dma_init,
.quit = fsi_dma_quit,
.probe = fsi_dma_probe,
.transfer = fsi_dma_transfer,
.remove = fsi_dma_remove,
.start_stop = fsi_dma_push_start_stop,
};
/*
* dai ops
*/
static void fsi_fifo_init(struct fsi_priv *fsi,
struct fsi_stream *io,
struct device *dev)
{
struct fsi_master *master = fsi_get_master(fsi);
int is_play = fsi_stream_is_play(fsi, io);
u32 shift, i;
int frame_capa;
/* get on-chip RAM capacity */
shift = fsi_master_read(master, FIFO_SZ);
shift >>= fsi_get_port_shift(fsi, io);
shift &= FIFO_SZ_MASK;
frame_capa = 256 << shift;
dev_dbg(dev, "fifo = %d words\n", frame_capa);
/*
* The maximum number of sample data varies depending
* on the number of channels selected for the format.
*
* FIFOs are used in 4-channel units in 3-channel mode
* and in 8-channel units in 5- to 7-channel mode
* meaning that more FIFOs than the required size of DPRAM
* are used.
*
* ex) if 256 words of DP-RAM is connected
* 1 channel: 256 (256 x 1 = 256)
* 2 channels: 128 (128 x 2 = 256)
* 3 channels: 64 ( 64 x 3 = 192)
* 4 channels: 64 ( 64 x 4 = 256)
* 5 channels: 32 ( 32 x 5 = 160)
* 6 channels: 32 ( 32 x 6 = 192)
* 7 channels: 32 ( 32 x 7 = 224)
* 8 channels: 32 ( 32 x 8 = 256)
*/
for (i = 1; i < fsi->chan_num; i <<= 1)
frame_capa >>= 1;
dev_dbg(dev, "%d channel %d store\n",
fsi->chan_num, frame_capa);
io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa);
/*
* set interrupt generation factor
* clear FIFO
*/
if (is_play) {
fsi_reg_write(fsi, DOFF_CTL, IRQ_HALF);
fsi_reg_mask_set(fsi, DOFF_CTL, FIFO_CLR, FIFO_CLR);
} else {
fsi_reg_write(fsi, DIFF_CTL, IRQ_HALF);
fsi_reg_mask_set(fsi, DIFF_CTL, FIFO_CLR, FIFO_CLR);
}
}
static int fsi_hw_startup(struct fsi_priv *fsi,
struct fsi_stream *io,
struct device *dev)
{
u32 flags = fsi_get_info_flags(fsi);
u32 data = 0;
/* clock setting */
if (fsi_is_clk_master(fsi))
data = DIMD | DOMD;
fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
/* clock inversion (CKG2) */
data = 0;
if (fsi->bit_clk_inv)
data |= (1 << 0);
if (fsi->lr_clk_inv)
data |= (1 << 4);
if (fsi_is_clk_master(fsi))
data <<= 8;
/* FIXME
*
* SH_FSI_xxx_INV style will be removed
*/
if (SH_FSI_LRM_INV & flags)
data |= 1 << 12;
if (SH_FSI_BRM_INV & flags)
data |= 1 << 8;
if (SH_FSI_LRS_INV & flags)
data |= 1 << 4;
if (SH_FSI_BRS_INV & flags)
data |= 1 << 0;
fsi_reg_write(fsi, CKG2, data);
/* spdif ? */
if (fsi_is_spdif(fsi)) {
fsi_spdif_clk_ctrl(fsi, 1);
fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
}
/*
* get bus settings
*/
data = 0;
switch (io->sample_width) {
case 2:
data = BUSOP_GET(16, io->bus_option);
break;
case 4:
data = BUSOP_GET(24, io->bus_option);
break;
}
fsi_format_bus_setup(fsi, io, data, dev);
/* irq clear */
fsi_irq_disable(fsi, io);
fsi_irq_clear_status(fsi);
/* fifo init */
fsi_fifo_init(fsi, io, dev);
/* start master clock */
if (fsi_is_clk_master(fsi))
return fsi_set_master_clk(dev, fsi, fsi->rate, 1);
return 0;
}
static int fsi_hw_shutdown(struct fsi_priv *fsi,
struct device *dev)
{
/* stop master clock */
if (fsi_is_clk_master(fsi))
return fsi_set_master_clk(dev, fsi, fsi->rate, 0);
return 0;
}
static int fsi_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
fsi_clk_invalid(fsi);
fsi->rate = 0;
return 0;
}
static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
fsi_clk_invalid(fsi);
fsi->rate = 0;
}
static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_stream_get(fsi, substream);
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
fsi_stream_init(fsi, io, substream);
if (!ret)
ret = fsi_hw_startup(fsi, io, dai->dev);
if (!ret)
ret = fsi_stream_transfer(io);
if (!ret)
fsi_stream_start(fsi, io);
break;
case SNDRV_PCM_TRIGGER_STOP:
if (!ret)
ret = fsi_hw_shutdown(fsi, dai->dev);
fsi_stream_stop(fsi, io);
fsi_stream_quit(fsi, io);
break;
}
return ret;
}
static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt)
{
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
fsi->fmt = CR_I2S;
fsi->chan_num = 2;
break;
case SND_SOC_DAIFMT_LEFT_J:
fsi->fmt = CR_PCM;
fsi->chan_num = 2;
break;
default:
return -EINVAL;
}
return 0;
}
static int fsi_set_fmt_spdif(struct fsi_priv *fsi)
{
struct fsi_master *master = fsi_get_master(fsi);
if (fsi_version(master) < 2)
return -EINVAL;
fsi->fmt = CR_DTMD_SPDIF_PCM | CR_PCM;
fsi->chan_num = 2;
return 0;
}
static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai);
set_rate_func set_rate = fsi_get_info_set_rate(fsi);
int ret;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
fsi->clk_master = 1;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* set clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_IF:
fsi->bit_clk_inv = 0;
fsi->lr_clk_inv = 1;
break;
case SND_SOC_DAIFMT_IB_NF:
fsi->bit_clk_inv = 1;
fsi->lr_clk_inv = 0;
break;
case SND_SOC_DAIFMT_IB_IF:
fsi->bit_clk_inv = 1;
fsi->lr_clk_inv = 1;
break;
case SND_SOC_DAIFMT_NB_NF:
default:
fsi->bit_clk_inv = 0;
fsi->lr_clk_inv = 0;
break;
}
if (fsi_is_clk_master(fsi)) {
/*
* CAUTION
*
* set_rate will be deleted
*/
if (set_rate)
dev_warn(dai->dev, "set_rate will be removed soon\n");
if (fsi->clk_cpg)
fsi_clk_init(dai->dev, fsi, 0, 1, 1,
fsi_clk_set_rate_cpg);
else
fsi_clk_init(dai->dev, fsi, 1, 1, 0,
fsi_clk_set_rate_external);
}
/* set format */
if (fsi_is_spdif(fsi))
ret = fsi_set_fmt_spdif(fsi);
else
ret = fsi_set_fmt_dai(fsi, fmt & SND_SOC_DAIFMT_FORMAT_MASK);
return ret;
}
static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
if (fsi_is_clk_master(fsi)) {
fsi->rate = params_rate(params);
fsi_clk_valid(fsi, fsi->rate);
}
return 0;
}
static const struct snd_soc_dai_ops fsi_dai_ops = {
.startup = fsi_dai_startup,
.shutdown = fsi_dai_shutdown,
.trigger = fsi_dai_trigger,
.set_fmt = fsi_dai_set_fmt,
.hw_params = fsi_dai_hw_params,
};
/*
* pcm ops
*/
static struct snd_pcm_hardware fsi_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE,
.formats = FSI_FMTS,
.rates = FSI_RATES,
.rate_min = 8000,
.rate_max = 192000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 64 * 1024,
.period_bytes_min = 32,
.period_bytes_max = 8192,
.periods_min = 1,
.periods_max = 32,
.fifo_size = 256,
};
static int fsi_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
int ret = 0;
snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware);
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
return ret;
}
static int fsi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
}
static int fsi_hw_free(struct snd_pcm_substream *substream)
{
return snd_pcm_lib_free_pages(substream);
}
static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_stream_get(fsi, substream);
return fsi_sample2frame(fsi, io->buff_sample_pos);
}
static struct snd_pcm_ops fsi_pcm_ops = {
.open = fsi_pcm_open,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = fsi_hw_params,
.hw_free = fsi_hw_free,
.pointer = fsi_pointer,
};
/*
* snd_soc_platform
*/
#define PREALLOC_BUFFER (32 * 1024)
#define PREALLOC_BUFFER_MAX (32 * 1024)
static void fsi_pcm_free(struct snd_pcm *pcm)
{
snd_pcm_lib_preallocate_free_for_all(pcm);
}
static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm *pcm = rtd->pcm;
/*
* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
* in MMAP mode (i.e. aplay -M)
*/
return snd_pcm_lib_preallocate_pages_for_all(
pcm,
SNDRV_DMA_TYPE_CONTINUOUS,
snd_dma_continuous_data(GFP_KERNEL),
PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
}
/*
* alsa struct
*/
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>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_driver fsi_soc_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>
2010-03-18 04:15:21 +08:00
.name = "fsia-dai",
.playback = {
.rates = FSI_RATES,
.formats = FSI_FMTS,
.channels_min = 2,
.channels_max = 2,
},
.capture = {
.rates = FSI_RATES,
.formats = FSI_FMTS,
.channels_min = 2,
.channels_max = 2,
},
.ops = &fsi_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>
2010-03-18 04:15:21 +08:00
.name = "fsib-dai",
.playback = {
.rates = FSI_RATES,
.formats = FSI_FMTS,
.channels_min = 2,
.channels_max = 2,
},
.capture = {
.rates = FSI_RATES,
.formats = FSI_FMTS,
.channels_min = 2,
.channels_max = 2,
},
.ops = &fsi_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>
2010-03-18 04:15:21 +08:00
static struct snd_soc_platform_driver fsi_soc_platform = {
.ops = &fsi_pcm_ops,
.pcm_new = fsi_pcm_new,
.pcm_free = fsi_pcm_free,
};
/*
* platform function
*/
static void fsi_port_info_init(struct fsi_priv *fsi,
struct sh_fsi_port_info *info)
{
if (info->flags & SH_FSI_FMT_SPDIF)
fsi->spdif = 1;
if (info->flags & SH_FSI_CLK_CPG)
fsi->clk_cpg = 1;
if (info->flags & SH_FSI_ENABLE_STREAM_MODE)
fsi->enable_stream = 1;
}
static void fsi_handler_init(struct fsi_priv *fsi,
struct sh_fsi_port_info *info)
{
fsi->playback.handler = &fsi_pio_push_handler; /* default PIO */
fsi->playback.priv = fsi;
fsi->capture.handler = &fsi_pio_pop_handler; /* default PIO */
fsi->capture.priv = fsi;
if (info->tx_id) {
fsi->playback.slave.shdma_slave.slave_id = info->tx_id;
fsi->playback.handler = &fsi_dma_push_handler;
}
}
static int fsi_probe(struct platform_device *pdev)
{
struct fsi_master *master;
const struct platform_device_id *id_entry;
struct sh_fsi_platform_info *info = pdev->dev.platform_data;
struct sh_fsi_port_info nul_info, *pinfo;
struct fsi_priv *fsi;
struct resource *res;
unsigned int irq;
int ret;
nul_info.flags = 0;
nul_info.tx_id = 0;
nul_info.rx_id = 0;
id_entry = pdev->id_entry;
if (!id_entry) {
dev_err(&pdev->dev, "unknown fsi device\n");
return -ENODEV;
}
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
irq = platform_get_irq(pdev, 0);
if (!res || (int)irq <= 0) {
dev_err(&pdev->dev, "Not enough FSI platform resources.\n");
return -ENODEV;
}
master = devm_kzalloc(&pdev->dev, sizeof(*master), GFP_KERNEL);
if (!master) {
dev_err(&pdev->dev, "Could not allocate master\n");
return -ENOMEM;
}
master->base = devm_ioremap_nocache(&pdev->dev,
res->start, resource_size(res));
if (!master->base) {
dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n");
return -ENXIO;
}
/* master setting */
master->irq = irq;
master->core = (struct fsi_core *)id_entry->driver_data;
spin_lock_init(&master->lock);
/* FSI A setting */
pinfo = (info) ? &info->port_a : &nul_info;
fsi = &master->fsia;
fsi->base = master->base;
fsi->master = master;
fsi->info = pinfo;
fsi_port_info_init(fsi, pinfo);
fsi_handler_init(fsi, pinfo);
ret = fsi_stream_probe(fsi, &pdev->dev);
if (ret < 0) {
dev_err(&pdev->dev, "FSIA stream probe failed\n");
return ret;
}
/* FSI B setting */
pinfo = (info) ? &info->port_b : &nul_info;
fsi = &master->fsib;
fsi->base = master->base + 0x40;
fsi->master = master;
fsi->info = pinfo;
fsi_port_info_init(fsi, pinfo);
fsi_handler_init(fsi, pinfo);
ret = fsi_stream_probe(fsi, &pdev->dev);
if (ret < 0) {
dev_err(&pdev->dev, "FSIB stream probe failed\n");
goto exit_fsia;
}
pm_runtime_enable(&pdev->dev);
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>
2010-03-18 04:15:21 +08:00
dev_set_drvdata(&pdev->dev, master);
ret = devm_request_irq(&pdev->dev, irq, &fsi_interrupt, 0,
id_entry->name, master);
if (ret) {
dev_err(&pdev->dev, "irq request err\n");
goto exit_fsib;
}
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>
2010-03-18 04:15:21 +08:00
ret = snd_soc_register_platform(&pdev->dev, &fsi_soc_platform);
if (ret < 0) {
dev_err(&pdev->dev, "cannot snd soc register\n");
goto exit_fsib;
}
ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
ARRAY_SIZE(fsi_soc_dai));
if (ret < 0) {
dev_err(&pdev->dev, "cannot snd dai register\n");
goto exit_snd_soc;
}
return ret;
exit_snd_soc:
snd_soc_unregister_platform(&pdev->dev);
exit_fsib:
pm_runtime_disable(&pdev->dev);
fsi_stream_remove(&master->fsib);
exit_fsia:
fsi_stream_remove(&master->fsia);
return ret;
}
static int fsi_remove(struct platform_device *pdev)
{
struct fsi_master *master;
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>
2010-03-18 04:15:21 +08:00
master = dev_get_drvdata(&pdev->dev);
pm_runtime_disable(&pdev->dev);
snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
snd_soc_unregister_platform(&pdev->dev);
fsi_stream_remove(&master->fsia);
fsi_stream_remove(&master->fsib);
return 0;
}
static void __fsi_suspend(struct fsi_priv *fsi,
struct fsi_stream *io,
struct device *dev)
{
if (!fsi_stream_is_working(fsi, io))
return;
fsi_stream_stop(fsi, io);
fsi_hw_shutdown(fsi, dev);
}
static void __fsi_resume(struct fsi_priv *fsi,
struct fsi_stream *io,
struct device *dev)
{
if (!fsi_stream_is_working(fsi, io))
return;
fsi_hw_startup(fsi, io, dev);
fsi_stream_start(fsi, io);
}
static int fsi_suspend(struct device *dev)
{
struct fsi_master *master = dev_get_drvdata(dev);
struct fsi_priv *fsia = &master->fsia;
struct fsi_priv *fsib = &master->fsib;
__fsi_suspend(fsia, &fsia->playback, dev);
__fsi_suspend(fsia, &fsia->capture, dev);
__fsi_suspend(fsib, &fsib->playback, dev);
__fsi_suspend(fsib, &fsib->capture, dev);
return 0;
}
static int fsi_resume(struct device *dev)
{
struct fsi_master *master = dev_get_drvdata(dev);
struct fsi_priv *fsia = &master->fsia;
struct fsi_priv *fsib = &master->fsib;
__fsi_resume(fsia, &fsia->playback, dev);
__fsi_resume(fsia, &fsia->capture, dev);
__fsi_resume(fsib, &fsib->playback, dev);
__fsi_resume(fsib, &fsib->capture, dev);
return 0;
}
static struct dev_pm_ops fsi_pm_ops = {
.suspend = fsi_suspend,
.resume = fsi_resume,
};
static struct fsi_core fsi1_core = {
.ver = 1,
/* Interrupt */
.int_st = INT_ST,
.iemsk = IEMSK,
.imsk = IMSK,
};
static struct fsi_core fsi2_core = {
.ver = 2,
/* Interrupt */
.int_st = CPU_INT_ST,
.iemsk = CPU_IEMSK,
.imsk = CPU_IMSK,
.a_mclk = A_MST_CTLR,
.b_mclk = B_MST_CTLR,
};
static struct platform_device_id fsi_id_table[] = {
{ "sh_fsi", (kernel_ulong_t)&fsi1_core },
{ "sh_fsi2", (kernel_ulong_t)&fsi2_core },
{},
};
MODULE_DEVICE_TABLE(platform, fsi_id_table);
static struct platform_driver fsi_driver = {
.driver = {
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>
2010-03-18 04:15:21 +08:00
.name = "fsi-pcm-audio",
.pm = &fsi_pm_ops,
},
.probe = fsi_probe,
.remove = fsi_remove,
.id_table = fsi_id_table,
};
module_platform_driver(fsi_driver);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
MODULE_ALIAS("platform:fsi-pcm-audio");