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add idl4k kernel firmware version 1.13.0.105

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This commit is contained in:
Jaroslav Kysela
2015-03-26 17:22:37 +01:00
parent 5194d2792e
commit e9070cdc77
31064 changed files with 12769984 additions and 0 deletions
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58
kernel/sound/soc/fsl/Kconfig Normal file
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config SND_SOC_OF_SIMPLE
tristate
config SND_MPC52xx_DMA
tristate
# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers
# for the SSI and the Elo DMA controller. You will still need to select
# a platform driver and a codec driver.
config SND_SOC_MPC8610
tristate
depends on MPC8610
config SND_SOC_MPC8610_HPCD
tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
# I2C is necessary for the CS4270 driver
depends on MPC8610_HPCD && I2C
select SND_SOC_MPC8610
select SND_SOC_CS4270
select SND_SOC_CS4270_VD33_ERRATA
default y if MPC8610_HPCD
help
Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
config SND_SOC_MPC5200_I2S
tristate "Freescale MPC5200 PSC in I2S mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in I2S mode.
config SND_SOC_MPC5200_AC97
tristate "Freescale MPC5200 PSC in AC97 mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
select SND_SOC_AC97_BUS
select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in AC97 mode.
config SND_MPC52xx_SOC_PCM030
tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
depends on PPC_MPC5200_SIMPLE
select SND_SOC_MPC5200_AC97
select SND_SOC_WM9712
help
Say Y if you want to add support for sound on the Phytec pcm030
baseboard.
config SND_MPC52xx_SOC_EFIKA
tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
depends on PPC_EFIKA
select SND_SOC_MPC5200_AC97
select SND_SOC_STAC9766
help
Say Y if you want to add support for sound on the Efika.

21
kernel/sound/soc/fsl/Makefile Normal file
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# Simple machine driver that extracts configuration from the OF device tree
obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o
# MPC8610 HPCD Machine Support
snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
# MPC8610 Platform Support
snd-soc-fsl-ssi-objs := fsl_ssi.o
snd-soc-fsl-dma-objs := fsl_dma.o
obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
# MPC5200 Platform Support
obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
# MPC5200 Machine Support
obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o

92
kernel/sound/soc/fsl/efika-audio-fabric.c Normal file
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/*
* Efika driver for the PSC of the Freescale MPC52xx
* configured as AC97 interface
*
* Copyright 2008 Jon Smirl, Digispeaker
* Author: Jon Smirl <jonsmirl@gmail.com>
*
* This file is licensed under the terms of the GNU General Public License
* version 2. This program is licensed "as is" without any warranty of any
* kind, whether express or implied.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/device.h>
#include <linux/delay.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/soc-of-simple.h>
#include "mpc5200_dma.h"
#include "mpc5200_psc_ac97.h"
#include "../codecs/stac9766.h"
#define DRV_NAME "efika-audio-fabric"
static struct snd_soc_device device;
static struct snd_soc_card card;
static struct snd_soc_dai_link efika_fabric_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 Analog",
.codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG],
.cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
},
{
.name = "AC97",
.stream_name = "AC97 IEC958",
.codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL],
.cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
},
};
static __init int efika_fabric_init(void)
{
struct platform_device *pdev;
int rc;
if (!machine_is_compatible("bplan,efika"))
return -ENODEV;
card.platform = &mpc5200_audio_dma_platform;
card.name = "Efika";
card.dai_link = efika_fabric_dai;
card.num_links = ARRAY_SIZE(efika_fabric_dai);
device.card = &card;
device.codec_dev = &soc_codec_dev_stac9766;
pdev = platform_device_alloc("soc-audio", 1);
if (!pdev) {
pr_err("efika_fabric_init: platform_device_alloc() failed\n");
return -ENODEV;
}
platform_set_drvdata(pdev, &device);
device.dev = &pdev->dev;
rc = platform_device_add(pdev);
if (rc) {
pr_err("efika_fabric_init: platform_device_add() failed\n");
return -ENODEV;
}
return 0;
}
module_init(efika_fabric_init);
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver");
MODULE_LICENSE("GPL");

880
kernel/sound/soc/fsl/fsl_dma.c Normal file
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/*
* Freescale DMA ALSA SoC PCM driver
*
* Author: Timur Tabi <timur@freescale.com>
*
* Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
* under the terms of the GNU General Public License version 2. This
* program is licensed "as is" without any warranty of any kind, whether
* express or implied.
*
* This driver implements ASoC support for the Elo DMA controller, which is
* the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms,
* the PCM driver is what handles the DMA buffer.
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/platform_device.h>
#include <linux/dma-mapping.h>
#include <linux/interrupt.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <asm/io.h>
#include "fsl_dma.h"
/*
* The formats that the DMA controller supports, which is anything
* that is 8, 16, or 32 bits.
*/
#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
SNDRV_PCM_FMTBIT_U8 | \
SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_U16_LE | \
SNDRV_PCM_FMTBIT_U16_BE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S24_BE | \
SNDRV_PCM_FMTBIT_U24_LE | \
SNDRV_PCM_FMTBIT_U24_BE | \
SNDRV_PCM_FMTBIT_S32_LE | \
SNDRV_PCM_FMTBIT_S32_BE | \
SNDRV_PCM_FMTBIT_U32_LE | \
SNDRV_PCM_FMTBIT_U32_BE)
#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
SNDRV_PCM_RATE_CONTINUOUS)
/* DMA global data. This structure is used by fsl_dma_open() to determine
* which DMA channels to assign to a substream. Unfortunately, ASoC V1 does
* not allow the machine driver to provide this information to the PCM
* driver in advance, and there's no way to differentiate between the two
* DMA controllers. So for now, this driver only supports one SSI device
* using two DMA channels. We cannot support multiple DMA devices.
*
* ssi_stx_phys: bus address of SSI STX register
* ssi_srx_phys: bus address of SSI SRX register
* dma_channel: pointer to the DMA channel's registers
* irq: IRQ for this DMA channel
* assigned: set to 1 if that DMA channel is assigned to a substream
*/
static struct {
dma_addr_t ssi_stx_phys;
dma_addr_t ssi_srx_phys;
struct ccsr_dma_channel __iomem *dma_channel[2];
unsigned int irq[2];
unsigned int assigned[2];
} dma_global_data;
/*
* The number of DMA links to use. Two is the bare minimum, but if you
* have really small links you might need more.
*/
#define NUM_DMA_LINKS 2
/** fsl_dma_private: p-substream DMA data
*
* Each substream has a 1-to-1 association with a DMA channel.
*
* The link[] array is first because it needs to be aligned on a 32-byte
* boundary, so putting it first will ensure alignment without padding the
* structure.
*
* @link[]: array of link descriptors
* @controller_id: which DMA controller (0, 1, ...)
* @channel_id: which DMA channel on the controller (0, 1, 2, ...)
* @dma_channel: pointer to the DMA channel's registers
* @irq: IRQ for this DMA channel
* @substream: pointer to the substream object, needed by the ISR
* @ssi_sxx_phys: bus address of the STX or SRX register to use
* @ld_buf_phys: physical address of the LD buffer
* @current_link: index into link[] of the link currently being processed
* @dma_buf_phys: physical address of the DMA buffer
* @dma_buf_next: physical address of the next period to process
* @dma_buf_end: physical address of the byte after the end of the DMA
* @buffer period_size: the size of a single period
* @num_periods: the number of periods in the DMA buffer
*/
struct fsl_dma_private {
struct fsl_dma_link_descriptor link[NUM_DMA_LINKS];
unsigned int controller_id;
unsigned int channel_id;
struct ccsr_dma_channel __iomem *dma_channel;
unsigned int irq;
struct snd_pcm_substream *substream;
dma_addr_t ssi_sxx_phys;
dma_addr_t ld_buf_phys;
unsigned int current_link;
dma_addr_t dma_buf_phys;
dma_addr_t dma_buf_next;
dma_addr_t dma_buf_end;
size_t period_size;
unsigned int num_periods;
};
/**
* fsl_dma_hardare: define characteristics of the PCM hardware.
*
* The PCM hardware is the Freescale DMA controller. This structure defines
* the capabilities of that hardware.
*
* Since the sampling rate and data format are not controlled by the DMA
* controller, we specify no limits for those values. The only exception is
* period_bytes_min, which is set to a reasonably low value to prevent the
* DMA controller from generating too many interrupts per second.
*
* Since each link descriptor has a 32-bit byte count field, we set
* period_bytes_max to the largest 32-bit number. We also have no maximum
* number of periods.
*
* Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
* limitation in the SSI driver requires the sample rates for playback and
* capture to be the same.
*/
static const struct snd_pcm_hardware fsl_dma_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_JOINT_DUPLEX |
SNDRV_PCM_INFO_PAUSE,
.formats = FSLDMA_PCM_FORMATS,
.rates = FSLDMA_PCM_RATES,
.rate_min = 5512,
.rate_max = 192000,
.period_bytes_min = 512, /* A reasonable limit */
.period_bytes_max = (u32) -1,
.periods_min = NUM_DMA_LINKS,
.periods_max = (unsigned int) -1,
.buffer_bytes_max = 128 * 1024, /* A reasonable limit */
};
/**
* fsl_dma_abort_stream: tell ALSA that the DMA transfer has aborted
*
* This function should be called by the ISR whenever the DMA controller
* halts data transfer.
*/
static void fsl_dma_abort_stream(struct snd_pcm_substream *substream)
{
unsigned long flags;
snd_pcm_stream_lock_irqsave(substream, flags);
if (snd_pcm_running(substream))
snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
snd_pcm_stream_unlock_irqrestore(substream, flags);
}
/**
* fsl_dma_update_pointers - update LD pointers to point to the next period
*
* As each period is completed, this function changes the the link
* descriptor pointers for that period to point to the next period.
*/
static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private)
{
struct fsl_dma_link_descriptor *link =
&dma_private->link[dma_private->current_link];
/* Update our link descriptors to point to the next period */
if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
link->source_addr =
cpu_to_be32(dma_private->dma_buf_next);
else
link->dest_addr =
cpu_to_be32(dma_private->dma_buf_next);
/* Update our variables for next time */
dma_private->dma_buf_next += dma_private->period_size;
if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
dma_private->dma_buf_next = dma_private->dma_buf_phys;
if (++dma_private->current_link >= NUM_DMA_LINKS)
dma_private->current_link = 0;
}
/**
* fsl_dma_isr: interrupt handler for the DMA controller
*
* @irq: IRQ of the DMA channel
* @dev_id: pointer to the dma_private structure for this DMA channel
*/
static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
{
struct fsl_dma_private *dma_private = dev_id;
struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
irqreturn_t ret = IRQ_NONE;
u32 sr, sr2 = 0;
/* We got an interrupt, so read the status register to see what we
were interrupted for.
*/
sr = in_be32(&dma_channel->sr);
if (sr & CCSR_DMA_SR_TE) {
dev_err(dma_private->substream->pcm->card->dev,
"DMA transmit error (controller=%u channel=%u irq=%u\n",
dma_private->controller_id,
dma_private->channel_id, irq);
fsl_dma_abort_stream(dma_private->substream);
sr2 |= CCSR_DMA_SR_TE;
ret = IRQ_HANDLED;
}
if (sr & CCSR_DMA_SR_CH)
ret = IRQ_HANDLED;
if (sr & CCSR_DMA_SR_PE) {
dev_err(dma_private->substream->pcm->card->dev,
"DMA%u programming error (channel=%u irq=%u)\n",
dma_private->controller_id,
dma_private->channel_id, irq);
fsl_dma_abort_stream(dma_private->substream);
sr2 |= CCSR_DMA_SR_PE;
ret = IRQ_HANDLED;
}
if (sr & CCSR_DMA_SR_EOLNI) {
sr2 |= CCSR_DMA_SR_EOLNI;
ret = IRQ_HANDLED;
}
if (sr & CCSR_DMA_SR_CB)
ret = IRQ_HANDLED;
if (sr & CCSR_DMA_SR_EOSI) {
struct snd_pcm_substream *substream = dma_private->substream;
/* Tell ALSA we completed a period. */
snd_pcm_period_elapsed(substream);
/*
* Update our link descriptors to point to the next period. We
* only need to do this if the number of periods is not equal to
* the number of links.
*/
if (dma_private->num_periods != NUM_DMA_LINKS)
fsl_dma_update_pointers(dma_private);
sr2 |= CCSR_DMA_SR_EOSI;
ret = IRQ_HANDLED;
}
if (sr & CCSR_DMA_SR_EOLSI) {
sr2 |= CCSR_DMA_SR_EOLSI;
ret = IRQ_HANDLED;
}
/* Clear the bits that we set */
if (sr2)
out_be32(&dma_channel->sr, sr2);
return ret;
}
/**
* fsl_dma_new: initialize this PCM driver.
*
* This function is called when the codec driver calls snd_soc_new_pcms(),
* once for each .dai_link in the machine driver's snd_soc_card
* structure.
*/
static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
static u64 fsl_dma_dmamask = DMA_BIT_MASK(32);
int ret;
if (!card->dev->dma_mask)
card->dev->dma_mask = &fsl_dma_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = fsl_dma_dmamask;
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
if (ret) {
dev_err(card->dev,
"Can't allocate playback DMA buffer (size=%u)\n",
fsl_dma_hardware.buffer_bytes_max);
return -ENOMEM;
}
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[1].substream->dma_buffer);
if (ret) {
snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
dev_err(card->dev,
"Can't allocate capture DMA buffer (size=%u)\n",
fsl_dma_hardware.buffer_bytes_max);
return -ENOMEM;
}
return 0;
}
/**
* fsl_dma_open: open a new substream.
*
* Each substream has its own DMA buffer.
*
* ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
* descriptors that ping-pong from one period to the next. For example, if
* there are six periods and two link descriptors, this is how they look
* before playback starts:
*
* The last link descriptor
* ____________ points back to the first
* | |
* V |
* ___ ___ |
* | |->| |->|
* |___| |___|
* | |
* | |
* V V
* _________________________________________
* | | | | | | | The DMA buffer is
* | | | | | | | divided into 6 parts
* |______|______|______|______|______|______|
*
* and here's how they look after the first period is finished playing:
*
* ____________
* | |
* V |
* ___ ___ |
* | |->| |->|
* |___| |___|
* | |
* |______________
* | |
* V V
* _________________________________________
* | | | | | | |
* | | | | | | |
* |______|______|______|______|______|______|
*
* The first link descriptor now points to the third period. The DMA
* controller is currently playing the second period. When it finishes, it
* will jump back to the first descriptor and play the third period.
*
* There are four reasons we do this:
*
* 1. The only way to get the DMA controller to automatically restart the
* transfer when it gets to the end of the buffer is to use chaining
* mode. Basic direct mode doesn't offer that feature.
* 2. We need to receive an interrupt at the end of every period. The DMA
* controller can generate an interrupt at the end of every link transfer
* (aka segment). Making each period into a DMA segment will give us the
* interrupts we need.
* 3. By creating only two link descriptors, regardless of the number of
* periods, we do not need to reallocate the link descriptors if the
* number of periods changes.
* 4. All of the audio data is still stored in a single, contiguous DMA
* buffer, which is what ALSA expects. We're just dividing it into
* contiguous parts, and creating a link descriptor for each one.
*/
static int fsl_dma_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private;
struct ccsr_dma_channel __iomem *dma_channel;
dma_addr_t ld_buf_phys;
u64 temp_link; /* Pointer to next link descriptor */
u32 mr;
unsigned int channel;
int ret = 0;
unsigned int i;
/*
* Reject any DMA buffer whose size is not a multiple of the period
* size. We need to make sure that the DMA buffer can be evenly divided
* into periods.
*/
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0) {
dev_err(substream->pcm->card->dev, "invalid buffer size\n");
return ret;
}
channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
if (dma_global_data.assigned[channel]) {
dev_err(substream->pcm->card->dev,
"DMA channel already assigned\n");
return -EBUSY;
}
dma_private = dma_alloc_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL);
if (!dma_private) {
dev_err(substream->pcm->card->dev,
"can't allocate DMA private data\n");
return -ENOMEM;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys;
else
dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys;
dma_private->dma_channel = dma_global_data.dma_channel[channel];
dma_private->irq = dma_global_data.irq[channel];
dma_private->substream = substream;
dma_private->ld_buf_phys = ld_buf_phys;
dma_private->dma_buf_phys = substream->dma_buffer.addr;
/* We only support one DMA controller for now */
dma_private->controller_id = 0;
dma_private->channel_id = channel;
ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
if (ret) {
dev_err(substream->pcm->card->dev,
"can't register ISR for IRQ %u (ret=%i)\n",
dma_private->irq, ret);
dma_free_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private),
dma_private, dma_private->ld_buf_phys);
return ret;
}
dma_global_data.assigned[channel] = 1;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
runtime->private_data = dma_private;
/* Program the fixed DMA controller parameters */
dma_channel = dma_private->dma_channel;
temp_link = dma_private->ld_buf_phys +
sizeof(struct fsl_dma_link_descriptor);
for (i = 0; i < NUM_DMA_LINKS; i++) {
dma_private->link[i].next = cpu_to_be64(temp_link);
temp_link += sizeof(struct fsl_dma_link_descriptor);
}
/* The last link descriptor points to the first */
dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
/* Tell the DMA controller where the first link descriptor is */
out_be32(&dma_channel->clndar,
CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
out_be32(&dma_channel->eclndar,
CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
/* The manual says the BCR must be clear before enabling EMP */
out_be32(&dma_channel->bcr, 0);
/*
* Program the mode register for interrupts, external master control,
* and source/destination hold. Also clear the Channel Abort bit.
*/
mr = in_be32(&dma_channel->mr) &
~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
/*
* We want External Master Start and External Master Pause enabled,
* because the SSI is controlling the DMA controller. We want the DMA
* controller to be set up in advance, and then we signal only the SSI
* to start transferring.
*
* We want End-Of-Segment Interrupts enabled, because this will generate
* an interrupt at the end of each segment (each link descriptor
* represents one segment). Each DMA segment is the same thing as an
* ALSA period, so this is how we get an interrupt at the end of every
* period.
*
* We want Error Interrupt enabled, so that we can get an error if
* the DMA controller is mis-programmed somehow.
*/
mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
CCSR_DMA_MR_EMS_EN;
/* For playback, we want the destination address to be held. For
capture, set the source address to be held. */
mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
out_be32(&dma_channel->mr, mr);
return 0;
}
/**
* fsl_dma_hw_params: continue initializing the DMA links
*
* This function obtains hardware parameters about the opened stream and
* programs the DMA controller accordingly.
*
* One drawback of big-endian is that when copying integers of different
* sizes to a fixed-sized register, the address to which the integer must be
* copied is dependent on the size of the integer.
*
* For example, if P is the address of a 32-bit register, and X is a 32-bit
* integer, then X should be copied to address P. However, if X is a 16-bit
* integer, then it should be copied to P+2. If X is an 8-bit register,
* then it should be copied to P+3.
*
* So for playback of 8-bit samples, the DMA controller must transfer single
* bytes from the DMA buffer to the last byte of the STX0 register, i.e.
* offset by 3 bytes. For 16-bit samples, the offset is two bytes.
*
* For 24-bit samples, the offset is 1 byte. However, the DMA controller
* does not support 3-byte copies (the DAHTS register supports only 1, 2, 4,
* and 8 bytes at a time). So we do not support packed 24-bit samples.
* 24-bit data must be padded to 32 bits.
*/
static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
/* Number of bits per sample */
unsigned int sample_size =
snd_pcm_format_physical_width(params_format(hw_params));
/* Number of bytes per frame */
unsigned int frame_size = 2 * (sample_size / 8);
/* Bus address of SSI STX register */
dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
/* Size of the DMA buffer, in bytes */
size_t buffer_size = params_buffer_bytes(hw_params);
/* Number of bytes per period */
size_t period_size = params_period_bytes(hw_params);
/* Pointer to next period */
dma_addr_t temp_addr = substream->dma_buffer.addr;
/* Pointer to DMA controller */
struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
u32 mr; /* DMA Mode Register */
unsigned int i;
/* Initialize our DMA tracking variables */
dma_private->period_size = period_size;
dma_private->num_periods = params_periods(hw_params);
dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
dma_private->dma_buf_next = dma_private->dma_buf_phys +
(NUM_DMA_LINKS * period_size);
if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
/* This happens if the number of periods == NUM_DMA_LINKS */
dma_private->dma_buf_next = dma_private->dma_buf_phys;
mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK |
CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK);
/* Due to a quirk of the SSI's STX register, the target address
* for the DMA operations depends on the sample size. So we calculate
* that offset here. While we're at it, also tell the DMA controller
* how much data to transfer per sample.
*/
switch (sample_size) {
case 8:
mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
ssi_sxx_phys += 3;
break;
case 16:
mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2;
ssi_sxx_phys += 2;
break;
case 32:
mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4;
break;
default:
/* We should never get here */
dev_err(substream->pcm->card->dev,
"unsupported sample size %u\n", sample_size);
return -EINVAL;
}
/*
* BWC should always be a multiple of the frame size. BWC determines
* how many bytes are sent/received before the DMA controller checks the
* SSI to see if it needs to stop. For playback, the transmit FIFO can
* hold three frames, so we want to send two frames at a time. For
* capture, the receive FIFO is triggered when it contains one frame, so
* we want to receive one frame at a time.
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
mr |= CCSR_DMA_MR_BWC(2 * frame_size);
else
mr |= CCSR_DMA_MR_BWC(frame_size);
out_be32(&dma_channel->mr, mr);
for (i = 0; i < NUM_DMA_LINKS; i++) {
struct fsl_dma_link_descriptor *link = &dma_private->link[i];
link->count = cpu_to_be32(period_size);
/* Even though the DMA controller supports 36-bit addressing,
* for simplicity we allow only 32-bit addresses for the audio
* buffer itself. This was enforced in fsl_dma_new() with the
* DMA mask.
*
* The snoop bit tells the DMA controller whether it should tell
* the ECM to snoop during a read or write to an address. For
* audio, we use DMA to transfer data between memory and an I/O
* device (the SSI's STX0 or SRX0 register). Snooping is only
* needed if there is a cache, so we need to snoop memory
* addresses only. For playback, that means we snoop the source
* but not the destination. For capture, we snoop the
* destination but not the source.
*
* Note that failing to snoop properly is unlikely to cause
* cache incoherency if the period size is larger than the
* size of L1 cache. This is because filling in one period will
* flush out the data for the previous period. So if you
* increased period_bytes_min to a large enough size, you might
* get more performance by not snooping, and you'll still be
* okay.
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
link->source_addr = cpu_to_be32(temp_addr);
link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
link->dest_addr = cpu_to_be32(ssi_sxx_phys);
link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP);
} else {
link->source_addr = cpu_to_be32(ssi_sxx_phys);
link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP);
link->dest_addr = cpu_to_be32(temp_addr);
link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
}
temp_addr += period_size;
}
return 0;
}
/**
* fsl_dma_pointer: determine the current position of the DMA transfer
*
* This function is called by ALSA when ALSA wants to know where in the
* stream buffer the hardware currently is.
*
* For playback, the SAR register contains the physical address of the most
* recent DMA transfer. For capture, the value is in the DAR register.
*
* The base address of the buffer is stored in the source_addr field of the
* first link descriptor.
*/
static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
dma_addr_t position;
snd_pcm_uframes_t frames;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
position = in_be32(&dma_channel->sar);
else
position = in_be32(&dma_channel->dar);
/*
* When capture is started, the SSI immediately starts to fill its FIFO.
* This means that the DMA controller is not started until the FIFO is
* full. However, ALSA calls this function before that happens, when
* MR.DAR is still zero. In this case, just return zero to indicate
* that nothing has been received yet.
*/
if (!position)
return 0;
if ((position < dma_private->dma_buf_phys) ||
(position > dma_private->dma_buf_end)) {
dev_err(substream->pcm->card->dev,
"dma pointer is out of range, halting stream\n");
return SNDRV_PCM_POS_XRUN;
}
frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
/*
* If the current address is just past the end of the buffer, wrap it
* around.
*/
if (frames == runtime->buffer_size)
frames = 0;
return frames;
}
/**
* fsl_dma_hw_free: release resources allocated in fsl_dma_hw_params()
*
* Release the resources allocated in fsl_dma_hw_params() and de-program the
* registers.
*
* This function can be called multiple times.
*/
static int fsl_dma_hw_free(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
if (dma_private) {
struct ccsr_dma_channel __iomem *dma_channel;
dma_channel = dma_private->dma_channel;
/* Stop the DMA */
out_be32(&dma_channel->mr, CCSR_DMA_MR_CA);
out_be32(&dma_channel->mr, 0);
/* Reset all the other registers */
out_be32(&dma_channel->sr, -1);
out_be32(&dma_channel->clndar, 0);
out_be32(&dma_channel->eclndar, 0);
out_be32(&dma_channel->satr, 0);
out_be32(&dma_channel->sar, 0);
out_be32(&dma_channel->datr, 0);
out_be32(&dma_channel->dar, 0);
out_be32(&dma_channel->bcr, 0);
out_be32(&dma_channel->nlndar, 0);
out_be32(&dma_channel->enlndar, 0);
}
return 0;
}
/**
* fsl_dma_close: close the stream.
*/
static int fsl_dma_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
if (dma_private) {
if (dma_private->irq)
free_irq(dma_private->irq, dma_private);
if (dma_private->ld_buf_phys) {
dma_unmap_single(substream->pcm->card->dev,
dma_private->ld_buf_phys,
sizeof(dma_private->link), DMA_TO_DEVICE);
}
/* Deallocate the fsl_dma_private structure */
dma_free_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private),
dma_private, dma_private->ld_buf_phys);
substream->runtime->private_data = NULL;
}
dma_global_data.assigned[dir] = 0;
return 0;
}
/*
* Remove this PCM driver.
*/
static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
unsigned int i;
for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
substream = pcm->streams[i].substream;
if (substream) {
snd_dma_free_pages(&substream->dma_buffer);
substream->dma_buffer.area = NULL;
substream->dma_buffer.addr = 0;
}
}
}
static struct snd_pcm_ops fsl_dma_ops = {
.open = fsl_dma_open,
.close = fsl_dma_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = fsl_dma_hw_params,
.hw_free = fsl_dma_hw_free,
.pointer = fsl_dma_pointer,
};
struct snd_soc_platform fsl_soc_platform = {
.name = "fsl-dma",
.pcm_ops = &fsl_dma_ops,
.pcm_new = fsl_dma_new,
.pcm_free = fsl_dma_free_dma_buffers,
};
EXPORT_SYMBOL_GPL(fsl_soc_platform);
/**
* fsl_dma_configure: store the DMA parameters from the fabric driver.
*
* This function is called by the ASoC fabric driver to give us the DMA and
* SSI channel information.
*
* Unfortunately, ASoC V1 does make it possible to determine the DMA/SSI
* data when a substream is created, so for now we need to store this data
* into a global variable. This means that we can only support one DMA
* controller, and hence only one SSI.
*/
int fsl_dma_configure(struct fsl_dma_info *dma_info)
{
static int initialized;
/* We only support one DMA controller for now */
if (initialized)
return 0;
dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys;
dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys;
dma_global_data.dma_channel[0] = dma_info->dma_channel[0];
dma_global_data.dma_channel[1] = dma_info->dma_channel[1];
dma_global_data.irq[0] = dma_info->dma_irq[0];
dma_global_data.irq[1] = dma_info->dma_irq[1];
dma_global_data.assigned[0] = 0;
dma_global_data.assigned[1] = 0;
initialized = 1;
return 1;
}
EXPORT_SYMBOL_GPL(fsl_dma_configure);
static int __init fsl_soc_platform_init(void)
{
return snd_soc_register_platform(&fsl_soc_platform);
}
module_init(fsl_soc_platform_init);
static void __exit fsl_soc_platform_exit(void)
{
snd_soc_unregister_platform(&fsl_soc_platform);
}
module_exit(fsl_soc_platform_exit);
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
MODULE_LICENSE("GPL");

149
kernel/sound/soc/fsl/fsl_dma.h Normal file
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/*
* mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef _MPC8610_PCM_H
#define _MPC8610_PCM_H
struct ccsr_dma {
u8 res0[0x100];
struct ccsr_dma_channel {
__be32 mr; /* Mode register */
__be32 sr; /* Status register */
__be32 eclndar; /* Current link descriptor extended addr reg */
__be32 clndar; /* Current link descriptor address register */
__be32 satr; /* Source attributes register */
__be32 sar; /* Source address register */
__be32 datr; /* Destination attributes register */
__be32 dar; /* Destination address register */
__be32 bcr; /* Byte count register */
__be32 enlndar; /* Next link descriptor extended address reg */
__be32 nlndar; /* Next link descriptor address register */
u8 res1[4];
__be32 eclsdar; /* Current list descriptor extended addr reg */
__be32 clsdar; /* Current list descriptor address register */
__be32 enlsdar; /* Next list descriptor extended address reg */
__be32 nlsdar; /* Next list descriptor address register */
__be32 ssr; /* Source stride register */
__be32 dsr; /* Destination stride register */
u8 res2[0x38];
} channel[4];
__be32 dgsr;
};
#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000
#define CCSR_DMA_MR_BWC_SHIFT 24
#define CCSR_DMA_MR_BWC_MASK 0x0F000000
#define CCSR_DMA_MR_BWC(x) \
((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK)
#define CCSR_DMA_MR_EMP_EN 0x00200000
#define CCSR_DMA_MR_EMS_EN 0x00040000
#define CCSR_DMA_MR_DAHTS_MASK 0x00030000
#define CCSR_DMA_MR_DAHTS_1 0x00000000
#define CCSR_DMA_MR_DAHTS_2 0x00010000
#define CCSR_DMA_MR_DAHTS_4 0x00020000
#define CCSR_DMA_MR_DAHTS_8 0x00030000
#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000
#define CCSR_DMA_MR_SAHTS_1 0x00000000
#define CCSR_DMA_MR_SAHTS_2 0x00004000
#define CCSR_DMA_MR_SAHTS_4 0x00008000
#define CCSR_DMA_MR_SAHTS_8 0x0000C000
#define CCSR_DMA_MR_DAHE 0x00002000
#define CCSR_DMA_MR_SAHE 0x00001000
#define CCSR_DMA_MR_SRW 0x00000400
#define CCSR_DMA_MR_EOSIE 0x00000200
#define CCSR_DMA_MR_EOLNIE 0x00000100
#define CCSR_DMA_MR_EOLSIE 0x00000080
#define CCSR_DMA_MR_EIE 0x00000040
#define CCSR_DMA_MR_XFE 0x00000020
#define CCSR_DMA_MR_CDSM_SWSM 0x00000010
#define CCSR_DMA_MR_CA 0x00000008
#define CCSR_DMA_MR_CTM 0x00000004
#define CCSR_DMA_MR_CC 0x00000002
#define CCSR_DMA_MR_CS 0x00000001
#define CCSR_DMA_SR_TE 0x00000080
#define CCSR_DMA_SR_CH 0x00000020
#define CCSR_DMA_SR_PE 0x00000010
#define CCSR_DMA_SR_EOLNI 0x00000008
#define CCSR_DMA_SR_CB 0x00000004
#define CCSR_DMA_SR_EOSI 0x00000002
#define CCSR_DMA_SR_EOLSI 0x00000001
/* ECLNDAR takes bits 32-36 of the CLNDAR register */
static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x)
{
return (x >> 32) & 0xf;
}
#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE)
#define CCSR_DMA_CLNDAR_EOSIE 0x00000008
/* SATR and DATR, combined */
#define CCSR_DMA_ATR_PBATMU 0x20000000
#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000
#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000
#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000
#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000
#define CCSR_DMA_ATR_PCIORDER 0x02000000
#define CCSR_DMA_ATR_SME 0x01000000
#define CCSR_DMA_ATR_NOSNOOP 0x00040000
#define CCSR_DMA_ATR_SNOOP 0x00050000
#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F
/**
* List Descriptor for extended chaining mode DMA operations.
*
* The CLSDAR register points to the first (in a linked-list) List
* Descriptor. Each object must be aligned on a 32-byte boundary. Each
* list descriptor points to a linked-list of link Descriptors.
*/
struct fsl_dma_list_descriptor {
__be64 next; /* Address of next list descriptor */
__be64 first_link; /* Address of first link descriptor */
__be32 source; /* Source stride */
__be32 dest; /* Destination stride */
u8 res[8]; /* Reserved */
} __attribute__ ((aligned(32), packed));
/**
* Link Descriptor for basic and extended chaining mode DMA operations.
*
* A Link Descriptor points to a single DMA buffer. Each link descriptor
* must be aligned on a 32-byte boundary.
*/
struct fsl_dma_link_descriptor {
__be32 source_attr; /* Programmed into SATR register */
__be32 source_addr; /* Programmed into SAR register */
__be32 dest_attr; /* Programmed into DATR register */
__be32 dest_addr; /* Programmed into DAR register */
__be64 next; /* Address of next link descriptor */
__be32 count; /* Byte count */
u8 res[4]; /* Reserved */
} __attribute__ ((aligned(32), packed));
/* DMA information needed to create a snd_soc_dai object
*
* ssi_stx_phys: bus address of SSI STX register to use
* ssi_srx_phys: bus address of SSI SRX register to use
* dma[0]: points to the DMA channel to use for playback
* dma[1]: points to the DMA channel to use for capture
* dma_irq[0]: IRQ of the DMA channel to use for playback
* dma_irq[1]: IRQ of the DMA channel to use for capture
*/
struct fsl_dma_info {
dma_addr_t ssi_stx_phys;
dma_addr_t ssi_srx_phys;
struct ccsr_dma_channel __iomem *dma_channel[2];
unsigned int dma_irq[2];
};
extern struct snd_soc_platform fsl_soc_platform;
int fsl_dma_configure(struct fsl_dma_info *dma_info);
#endif

733
kernel/sound/soc/fsl/fsl_ssi.c Normal file
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/*
* Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver
*
* Author: Timur Tabi <timur@freescale.com>
*
* Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
* under the terms of the GNU General Public License version 2. This
* program is licensed "as is" without any warranty of any kind, whether
* express or implied.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/device.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <asm/immap_86xx.h>
#include "fsl_ssi.h"
/**
* FSLSSI_I2S_RATES: sample rates supported by the I2S
*
* This driver currently only supports the SSI running in I2S slave mode,
* which means the codec determines the sample rate. Therefore, we tell
* ALSA that we support all rates and let the codec driver decide what rates
* are really supported.
*/
#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
SNDRV_PCM_RATE_CONTINUOUS)
/**
* FSLSSI_I2S_FORMATS: audio formats supported by the SSI
*
* This driver currently only supports the SSI running in I2S slave mode.
*
* The SSI has a limitation in that the samples must be in the same byte
* order as the host CPU. This is because when multiple bytes are written
* to the STX register, the bytes and bits must be written in the same
* order. The STX is a shift register, so all the bits need to be aligned
* (bit-endianness must match byte-endianness). Processors typically write
* the bits within a byte in the same order that the bytes of a word are
* written in. So if the host CPU is big-endian, then only big-endian
* samples will be written to STX properly.
*/
#ifdef __BIG_ENDIAN
#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \
SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE)
#else
#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
#endif
/* SIER bitflag of interrupts to enable */
#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
/**
* fsl_ssi_private: per-SSI private data
*
* @name: short name for this device ("SSI0", "SSI1", etc)
* @ssi: pointer to the SSI's registers
* @ssi_phys: physical address of the SSI registers
* @irq: IRQ of this SSI
* @first_stream: pointer to the stream that was opened first
* @second_stream: pointer to second stream
* @dev: struct device pointer
* @playback: the number of playback streams opened
* @capture: the number of capture streams opened
* @asynchronous: 0=synchronous mode, 1=asynchronous mode
* @cpu_dai: the CPU DAI for this device
* @dev_attr: the sysfs device attribute structure
* @stats: SSI statistics
*/
struct fsl_ssi_private {
char name[8];
struct ccsr_ssi __iomem *ssi;
dma_addr_t ssi_phys;
unsigned int irq;
struct snd_pcm_substream *first_stream;
struct snd_pcm_substream *second_stream;
struct device *dev;
unsigned int playback;
unsigned int capture;
int asynchronous;
struct snd_soc_dai cpu_dai;
struct device_attribute dev_attr;
struct {
unsigned int rfrc;
unsigned int tfrc;
unsigned int cmdau;
unsigned int cmddu;
unsigned int rxt;
unsigned int rdr1;
unsigned int rdr0;
unsigned int tde1;
unsigned int tde0;
unsigned int roe1;
unsigned int roe0;
unsigned int tue1;
unsigned int tue0;
unsigned int tfs;
unsigned int rfs;
unsigned int tls;
unsigned int rls;
unsigned int rff1;
unsigned int rff0;
unsigned int tfe1;
unsigned int tfe0;
} stats;
};
/**
* fsl_ssi_isr: SSI interrupt handler
*
* Although it's possible to use the interrupt handler to send and receive
* data to/from the SSI, we use the DMA instead. Programming is more
* complicated, but the performance is much better.
*
* This interrupt handler is used only to gather statistics.
*
* @irq: IRQ of the SSI device
* @dev_id: pointer to the ssi_private structure for this SSI device
*/
static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
{
struct fsl_ssi_private *ssi_private = dev_id;
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
irqreturn_t ret = IRQ_NONE;
__be32 sisr;
__be32 sisr2 = 0;
/* We got an interrupt, so read the status register to see what we
were interrupted for. We mask it with the Interrupt Enable register
so that we only check for events that we're interested in.
*/
sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
if (sisr & CCSR_SSI_SISR_RFRC) {
ssi_private->stats.rfrc++;
sisr2 |= CCSR_SSI_SISR_RFRC;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TFRC) {
ssi_private->stats.tfrc++;
sisr2 |= CCSR_SSI_SISR_TFRC;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_CMDAU) {
ssi_private->stats.cmdau++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_CMDDU) {
ssi_private->stats.cmddu++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RXT) {
ssi_private->stats.rxt++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RDR1) {
ssi_private->stats.rdr1++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RDR0) {
ssi_private->stats.rdr0++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TDE1) {
ssi_private->stats.tde1++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TDE0) {
ssi_private->stats.tde0++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_ROE1) {
ssi_private->stats.roe1++;
sisr2 |= CCSR_SSI_SISR_ROE1;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_ROE0) {
ssi_private->stats.roe0++;
sisr2 |= CCSR_SSI_SISR_ROE0;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TUE1) {
ssi_private->stats.tue1++;
sisr2 |= CCSR_SSI_SISR_TUE1;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TUE0) {
ssi_private->stats.tue0++;
sisr2 |= CCSR_SSI_SISR_TUE0;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TFS) {
ssi_private->stats.tfs++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RFS) {
ssi_private->stats.rfs++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TLS) {
ssi_private->stats.tls++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RLS) {
ssi_private->stats.rls++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RFF1) {
ssi_private->stats.rff1++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_RFF0) {
ssi_private->stats.rff0++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TFE1) {
ssi_private->stats.tfe1++;
ret = IRQ_HANDLED;
}
if (sisr & CCSR_SSI_SISR_TFE0) {
ssi_private->stats.tfe0++;
ret = IRQ_HANDLED;
}
/* Clear the bits that we set */
if (sisr2)
out_be32(&ssi->sisr, sisr2);
return ret;
}
/**
* fsl_ssi_startup: create a new substream
*
* This is the first function called when a stream is opened.
*
* If this is the first stream open, then grab the IRQ and program most of
* the SSI registers.
*/
static int fsl_ssi_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
/*
* If this is the first stream opened, then request the IRQ
* and initialize the SSI registers.
*/
if (!ssi_private->playback && !ssi_private->capture) {
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
int ret;
ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0,
ssi_private->name, ssi_private);
if (ret < 0) {
dev_err(substream->pcm->card->dev,
"could not claim irq %u\n", ssi_private->irq);
return ret;
}
/*
* Section 16.5 of the MPC8610 reference manual says that the
* SSI needs to be disabled before updating the registers we set
* here.
*/
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
/*
* Program the SSI into I2S Slave Non-Network Synchronous mode.
* Also enable the transmit and receive FIFO.
*
* FIXME: Little-endian samples require a different shift dir
*/
clrsetbits_be32(&ssi->scr,
CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
| (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN));
out_be32(&ssi->stcr,
CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
CCSR_SSI_STCR_TSCKP);
out_be32(&ssi->srcr,
CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
CCSR_SSI_SRCR_RSCKP);
/*
* The DC and PM bits are only used if the SSI is the clock
* master.
*/
/* 4. Enable the interrupts and DMA requests */
out_be32(&ssi->sier, SIER_FLAGS);
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
* don't use FIFO 1. Since the SSI only supports stereo, the
* watermark should never be an odd number.
*/
out_be32(&ssi->sfcsr,
CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2));
/*
* We keep the SSI disabled because if we enable it, then the
* DMA controller will start. It's not supposed to start until
* the SCR.TE (or SCR.RE) bit is set, but it does anyway. The
* DMA controller will transfer one "BWC" of data (i.e. the
* amount of data that the MR.BWC bits are set to). The reason
* this is bad is because at this point, the PCM driver has not
* finished initializing the DMA controller.
*/
}
if (!ssi_private->first_stream)
ssi_private->first_stream = substream;
else {
/* This is the second stream open, so we need to impose sample
* rate and maybe sample size constraints. Note that this can
* cause a race condition if the second stream is opened before
* the first stream is fully initialized.
*
* We provide some protection by checking to make sure the first
* stream is initialized, but it's not perfect. ALSA sometimes
* re-initializes the driver with a different sample rate or
* size. If the second stream is opened before the first stream
* has received its final parameters, then the second stream may
* be constrained to the wrong sample rate or size.
*
* FIXME: This code does not handle opening and closing streams
* repeatedly. If you open two streams and then close the first
* one, you may not be able to open another stream until you
* close the second one as well.
*/
struct snd_pcm_runtime *first_runtime =
ssi_private->first_stream->runtime;
if (!first_runtime->sample_bits) {
dev_err(substream->pcm->card->dev,
"set sample size in %s stream first\n",
substream->stream == SNDRV_PCM_STREAM_PLAYBACK
? "capture" : "playback");
return -EAGAIN;
}
/* If we're in synchronous mode, then we need to constrain
* the sample size as well. We don't support independent sample
* rates in asynchronous mode.
*/
if (!ssi_private->asynchronous)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
first_runtime->sample_bits,
first_runtime->sample_bits);
ssi_private->second_stream = substream;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ssi_private->playback++;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
ssi_private->capture++;
return 0;
}
/**
* fsl_ssi_hw_params - program the sample size
*
* Most of the SSI registers have been programmed in the startup function,
* but the word length must be programmed here. Unfortunately, programming
* the SxCCR.WL bits requires the SSI to be temporarily disabled. This can
* cause a problem with supporting simultaneous playback and capture. If
* the SSI is already playing a stream, then that stream may be temporarily
* stopped when you start capture.
*
* Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
* clock master.
*/
static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai)
{
struct fsl_ssi_private *ssi_private = cpu_dai->private_data;
if (substream == ssi_private->first_stream) {
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
unsigned int sample_size =
snd_pcm_format_width(params_format(hw_params));
u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
/* The SSI should always be disabled at this points (SSIEN=0) */
/* In synchronous mode, the SSI uses STCCR for capture */
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
!ssi_private->asynchronous)
clrsetbits_be32(&ssi->stccr,
CCSR_SSI_SxCCR_WL_MASK, wl);
else
clrsetbits_be32(&ssi->srccr,
CCSR_SSI_SxCCR_WL_MASK, wl);
}
return 0;
}
/**
* fsl_ssi_trigger: start and stop the DMA transfer.
*
* This function is called by ALSA to start, stop, pause, and resume the DMA
* transfer of data.
*
* The DMA channel is in external master start and pause mode, which
* means the SSI completely controls the flow of data.
*/
static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
else
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
else
clrbits32(&ssi->scr, CCSR_SSI_SCR_RE);
break;
default:
return -EINVAL;
}
return 0;
}
/**
* fsl_ssi_shutdown: shutdown the SSI
*
* Shutdown the SSI if there are no other substreams open.
*/
static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ssi_private->playback--;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
ssi_private->capture--;
if (ssi_private->first_stream == substream)
ssi_private->first_stream = ssi_private->second_stream;
ssi_private->second_stream = NULL;
/*
* If this is the last active substream, disable the SSI and release
* the IRQ.
*/
if (!ssi_private->playback && !ssi_private->capture) {
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
free_irq(ssi_private->irq, ssi_private);
}
}
/**
* fsl_ssi_set_sysclk: set the clock frequency and direction
*
* This function is called by the machine driver to tell us what the clock
* frequency and direction are.
*
* Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
* and we don't care about the frequency. Return an error if the direction
* is not SND_SOC_CLOCK_IN.
*
* @clk_id: reserved, should be zero
* @freq: the frequency of the given clock ID, currently ignored
* @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
*/
static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
}
/**
* fsl_ssi_set_fmt: set the serial format.
*
* This function is called by the machine driver to tell us what serial
* format to use.
*
* Currently, we only support I2S mode. Return an error if the format is
* not SND_SOC_DAIFMT_I2S.
*
* @format: one of SND_SOC_DAIFMT_xxx
*/
static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
{
return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
}
/**
* fsl_ssi_dai_template: template CPU DAI for the SSI
*/
static struct snd_soc_dai_ops fsl_ssi_dai_ops = {
.startup = fsl_ssi_startup,
.hw_params = fsl_ssi_hw_params,
.shutdown = fsl_ssi_shutdown,
.trigger = fsl_ssi_trigger,
.set_sysclk = fsl_ssi_set_sysclk,
.set_fmt = fsl_ssi_set_fmt,
};
static struct snd_soc_dai fsl_ssi_dai_template = {
.playback = {
/* The SSI does not support monaural audio. */
.channels_min = 2,
.channels_max = 2,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
.ops = &fsl_ssi_dai_ops,
};
/* Show the statistics of a flag only if its interrupt is enabled. The
* compiler will optimze this code to a no-op if the interrupt is not
* enabled.
*/
#define SIER_SHOW(flag, name) \
do { \
if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
length += sprintf(buf + length, #name "=%u\n", \
ssi_private->stats.name); \
} while (0)
/**
* fsl_sysfs_ssi_show: display SSI statistics
*
* Display the statistics for the current SSI device. To avoid confusion,
* we only show those counts that are enabled.
*/
static ssize_t fsl_sysfs_ssi_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct fsl_ssi_private *ssi_private =
container_of(attr, struct fsl_ssi_private, dev_attr);
ssize_t length = 0;
SIER_SHOW(RFRC_EN, rfrc);
SIER_SHOW(TFRC_EN, tfrc);
SIER_SHOW(CMDAU_EN, cmdau);
SIER_SHOW(CMDDU_EN, cmddu);
SIER_SHOW(RXT_EN, rxt);
SIER_SHOW(RDR1_EN, rdr1);
SIER_SHOW(RDR0_EN, rdr0);
SIER_SHOW(TDE1_EN, tde1);
SIER_SHOW(TDE0_EN, tde0);
SIER_SHOW(ROE1_EN, roe1);
SIER_SHOW(ROE0_EN, roe0);
SIER_SHOW(TUE1_EN, tue1);
SIER_SHOW(TUE0_EN, tue0);
SIER_SHOW(TFS_EN, tfs);
SIER_SHOW(RFS_EN, rfs);
SIER_SHOW(TLS_EN, tls);
SIER_SHOW(RLS_EN, rls);
SIER_SHOW(RFF1_EN, rff1);
SIER_SHOW(RFF0_EN, rff0);
SIER_SHOW(TFE1_EN, tfe1);
SIER_SHOW(TFE0_EN, tfe0);
return length;
}
/**
* fsl_ssi_create_dai: create a snd_soc_dai structure
*
* This function is called by the machine driver to create a snd_soc_dai
* structure. The function creates an ssi_private object, which contains
* the snd_soc_dai. It also creates the sysfs statistics device.
*/
struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
{
struct snd_soc_dai *fsl_ssi_dai;
struct fsl_ssi_private *ssi_private;
int ret = 0;
struct device_attribute *dev_attr;
ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL);
if (!ssi_private) {
dev_err(ssi_info->dev, "could not allocate DAI object\n");
return NULL;
}
memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template,
sizeof(struct snd_soc_dai));
fsl_ssi_dai = &ssi_private->cpu_dai;
dev_attr = &ssi_private->dev_attr;
sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id);
ssi_private->ssi = ssi_info->ssi;
ssi_private->ssi_phys = ssi_info->ssi_phys;
ssi_private->irq = ssi_info->irq;
ssi_private->dev = ssi_info->dev;
ssi_private->asynchronous = ssi_info->asynchronous;
dev_set_drvdata(ssi_private->dev, fsl_ssi_dai);
/* Initialize the the device_attribute structure */
dev_attr->attr.name = "ssi-stats";
dev_attr->attr.mode = S_IRUGO;
dev_attr->show = fsl_sysfs_ssi_show;
ret = device_create_file(ssi_private->dev, dev_attr);
if (ret) {
dev_err(ssi_info->dev, "could not create sysfs %s file\n",
ssi_private->dev_attr.attr.name);
kfree(fsl_ssi_dai);
return NULL;
}
fsl_ssi_dai->private_data = ssi_private;
fsl_ssi_dai->name = ssi_private->name;
fsl_ssi_dai->id = ssi_info->id;
fsl_ssi_dai->dev = ssi_info->dev;
fsl_ssi_dai->symmetric_rates = 1;
ret = snd_soc_register_dai(fsl_ssi_dai);
if (ret != 0) {
dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
kfree(fsl_ssi_dai);
return NULL;
}
return fsl_ssi_dai;
}
EXPORT_SYMBOL_GPL(fsl_ssi_create_dai);
/**
* fsl_ssi_destroy_dai: destroy the snd_soc_dai object
*
* This function undoes the operations of fsl_ssi_create_dai()
*/
void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
{
struct fsl_ssi_private *ssi_private =
container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai);
device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
snd_soc_unregister_dai(&ssi_private->cpu_dai);
kfree(ssi_private);
}
EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
static int __init fsl_ssi_init(void)
{
printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n");
return 0;
}
module_init(fsl_ssi_init);
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
MODULE_LICENSE("GPL");

226
kernel/sound/soc/fsl/fsl_ssi.h Normal file
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/*
* fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC
*
* Author: Timur Tabi <timur@freescale.com>
*
* Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
* under the terms of the GNU General Public License version 2. This
* program is licensed "as is" without any warranty of any kind, whether
* express or implied.
*/
#ifndef _MPC8610_I2S_H
#define _MPC8610_I2S_H
/* SSI Register Map */
struct ccsr_ssi {
__be32 stx0; /* 0x.0000 - SSI Transmit Data Register 0 */
__be32 stx1; /* 0x.0004 - SSI Transmit Data Register 1 */
__be32 srx0; /* 0x.0008 - SSI Receive Data Register 0 */
__be32 srx1; /* 0x.000C - SSI Receive Data Register 1 */
__be32 scr; /* 0x.0010 - SSI Control Register */
__be32 sisr; /* 0x.0014 - SSI Interrupt Status Register Mixed */
__be32 sier; /* 0x.0018 - SSI Interrupt Enable Register */
__be32 stcr; /* 0x.001C - SSI Transmit Configuration Register */
__be32 srcr; /* 0x.0020 - SSI Receive Configuration Register */
__be32 stccr; /* 0x.0024 - SSI Transmit Clock Control Register */
__be32 srccr; /* 0x.0028 - SSI Receive Clock Control Register */
__be32 sfcsr; /* 0x.002C - SSI FIFO Control/Status Register */
__be32 str; /* 0x.0030 - SSI Test Register */
__be32 sor; /* 0x.0034 - SSI Option Register */
__be32 sacnt; /* 0x.0038 - SSI AC97 Control Register */
__be32 sacadd; /* 0x.003C - SSI AC97 Command Address Register */
__be32 sacdat; /* 0x.0040 - SSI AC97 Command Data Register */
__be32 satag; /* 0x.0044 - SSI AC97 Tag Register */
__be32 stmsk; /* 0x.0048 - SSI Transmit Time Slot Mask Register */
__be32 srmsk; /* 0x.004C - SSI Receive Time Slot Mask Register */
__be32 saccst; /* 0x.0050 - SSI AC97 Channel Status Register */
__be32 saccen; /* 0x.0054 - SSI AC97 Channel Enable Register */
__be32 saccdis; /* 0x.0058 - SSI AC97 Channel Disable Register */
};
#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800
#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400
#define CCSR_SSI_SCR_TCH_EN 0x00000100
#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080
#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060
#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000
#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020
#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040
#define CCSR_SSI_SCR_SYN 0x00000010
#define CCSR_SSI_SCR_NET 0x00000008
#define CCSR_SSI_SCR_RE 0x00000004
#define CCSR_SSI_SCR_TE 0x00000002
#define CCSR_SSI_SCR_SSIEN 0x00000001
#define CCSR_SSI_SISR_RFRC 0x01000000
#define CCSR_SSI_SISR_TFRC 0x00800000
#define CCSR_SSI_SISR_CMDAU 0x00040000
#define CCSR_SSI_SISR_CMDDU 0x00020000
#define CCSR_SSI_SISR_RXT 0x00010000
#define CCSR_SSI_SISR_RDR1 0x00008000
#define CCSR_SSI_SISR_RDR0 0x00004000
#define CCSR_SSI_SISR_TDE1 0x00002000
#define CCSR_SSI_SISR_TDE0 0x00001000
#define CCSR_SSI_SISR_ROE1 0x00000800
#define CCSR_SSI_SISR_ROE0 0x00000400
#define CCSR_SSI_SISR_TUE1 0x00000200
#define CCSR_SSI_SISR_TUE0 0x00000100
#define CCSR_SSI_SISR_TFS 0x00000080
#define CCSR_SSI_SISR_RFS 0x00000040
#define CCSR_SSI_SISR_TLS 0x00000020
#define CCSR_SSI_SISR_RLS 0x00000010
#define CCSR_SSI_SISR_RFF1 0x00000008
#define CCSR_SSI_SISR_RFF0 0x00000004
#define CCSR_SSI_SISR_TFE1 0x00000002
#define CCSR_SSI_SISR_TFE0 0x00000001
#define CCSR_SSI_SIER_RFRC_EN 0x01000000
#define CCSR_SSI_SIER_TFRC_EN 0x00800000
#define CCSR_SSI_SIER_RDMAE 0x00400000
#define CCSR_SSI_SIER_RIE 0x00200000
#define CCSR_SSI_SIER_TDMAE 0x00100000
#define CCSR_SSI_SIER_TIE 0x00080000
#define CCSR_SSI_SIER_CMDAU_EN 0x00040000
#define CCSR_SSI_SIER_CMDDU_EN 0x00020000
#define CCSR_SSI_SIER_RXT_EN 0x00010000
#define CCSR_SSI_SIER_RDR1_EN 0x00008000
#define CCSR_SSI_SIER_RDR0_EN 0x00004000
#define CCSR_SSI_SIER_TDE1_EN 0x00002000
#define CCSR_SSI_SIER_TDE0_EN 0x00001000
#define CCSR_SSI_SIER_ROE1_EN 0x00000800
#define CCSR_SSI_SIER_ROE0_EN 0x00000400
#define CCSR_SSI_SIER_TUE1_EN 0x00000200
#define CCSR_SSI_SIER_TUE0_EN 0x00000100
#define CCSR_SSI_SIER_TFS_EN 0x00000080
#define CCSR_SSI_SIER_RFS_EN 0x00000040
#define CCSR_SSI_SIER_TLS_EN 0x00000020
#define CCSR_SSI_SIER_RLS_EN 0x00000010
#define CCSR_SSI_SIER_RFF1_EN 0x00000008
#define CCSR_SSI_SIER_RFF0_EN 0x00000004
#define CCSR_SSI_SIER_TFE1_EN 0x00000002
#define CCSR_SSI_SIER_TFE0_EN 0x00000001
#define CCSR_SSI_STCR_TXBIT0 0x00000200
#define CCSR_SSI_STCR_TFEN1 0x00000100
#define CCSR_SSI_STCR_TFEN0 0x00000080
#define CCSR_SSI_STCR_TFDIR 0x00000040
#define CCSR_SSI_STCR_TXDIR 0x00000020
#define CCSR_SSI_STCR_TSHFD 0x00000010
#define CCSR_SSI_STCR_TSCKP 0x00000008
#define CCSR_SSI_STCR_TFSI 0x00000004
#define CCSR_SSI_STCR_TFSL 0x00000002
#define CCSR_SSI_STCR_TEFS 0x00000001
#define CCSR_SSI_SRCR_RXEXT 0x00000400
#define CCSR_SSI_SRCR_RXBIT0 0x00000200
#define CCSR_SSI_SRCR_RFEN1 0x00000100
#define CCSR_SSI_SRCR_RFEN0 0x00000080
#define CCSR_SSI_SRCR_RFDIR 0x00000040
#define CCSR_SSI_SRCR_RXDIR 0x00000020
#define CCSR_SSI_SRCR_RSHFD 0x00000010
#define CCSR_SSI_SRCR_RSCKP 0x00000008
#define CCSR_SSI_SRCR_RFSI 0x00000004
#define CCSR_SSI_SRCR_RFSL 0x00000002
#define CCSR_SSI_SRCR_REFS 0x00000001
/* STCCR and SRCCR */
#define CCSR_SSI_SxCCR_DIV2 0x00040000
#define CCSR_SSI_SxCCR_PSR 0x00020000
#define CCSR_SSI_SxCCR_WL_SHIFT 13
#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000
#define CCSR_SSI_SxCCR_WL(x) \
(((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK)
#define CCSR_SSI_SxCCR_DC_SHIFT 8
#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00
#define CCSR_SSI_SxCCR_DC(x) \
((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK)
#define CCSR_SSI_SxCCR_PM_SHIFT 0
#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF
#define CCSR_SSI_SxCCR_PM(x) \
((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK)
/*
* The xFCNT bits are read-only, and the xFWM bits are read/write. Use the
* CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the
* CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks.
*/
#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28
#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000
#define CCSR_SSI_SFCSR_RFCNT1(x) \
(((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT)
#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24
#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000
#define CCSR_SSI_SFCSR_TFCNT1(x) \
(((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT)
#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20
#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000
#define CCSR_SSI_SFCSR_RFWM1(x) \
(((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK)
#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16
#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000
#define CCSR_SSI_SFCSR_TFWM1(x) \
(((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK)
#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12
#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000
#define CCSR_SSI_SFCSR_RFCNT0(x) \
(((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT)
#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8
#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00
#define CCSR_SSI_SFCSR_TFCNT0(x) \
(((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT)
#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4
#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0
#define CCSR_SSI_SFCSR_RFWM0(x) \
(((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK)
#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0
#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F
#define CCSR_SSI_SFCSR_TFWM0(x) \
(((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK)
#define CCSR_SSI_STR_TEST 0x00008000
#define CCSR_SSI_STR_RCK2TCK 0x00004000
#define CCSR_SSI_STR_RFS2TFS 0x00002000
#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F)
#define CCSR_SSI_STR_TXD2RXD 0x00000080
#define CCSR_SSI_STR_TCK2RCK 0x00000040
#define CCSR_SSI_STR_TFS2RFS 0x00000020
#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F)
#define CCSR_SSI_SOR_CLKOFF 0x00000040
#define CCSR_SSI_SOR_RX_CLR 0x00000020
#define CCSR_SSI_SOR_TX_CLR 0x00000010
#define CCSR_SSI_SOR_INIT 0x00000008
#define CCSR_SSI_SOR_WAIT_SHIFT 1
#define CCSR_SSI_SOR_WAIT_MASK 0x00000006
#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT)
#define CCSR_SSI_SOR_SYNRST 0x00000001
/* Instantiation data for an SSI interface
*
* This structure contains all the information that the the SSI driver needs
* to instantiate an SSI interface with ALSA. The machine driver should
* create this structure, fill it in, call fsl_ssi_create_dai(), and then
* delete the structure.
*
* id: which SSI this is (0, 1, etc. )
* ssi: pointer to the SSI's registers
* ssi_phys: physical address of the SSI registers
* irq: IRQ of this SSI
* dev: struct device, used to create the sysfs statistics file
* asynchronous: 0=synchronous mode, 1=asynchronous mode
*/
struct fsl_ssi_info {
unsigned int id;
struct ccsr_ssi __iomem *ssi;
dma_addr_t ssi_phys;
unsigned int irq;
struct device *dev;
int asynchronous;
};
struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai);
#endif

589
kernel/sound/soc/fsl/mpc5200_dma.c Normal file
View File

@@ -0,0 +1,589 @@
/*
* Freescale MPC5200 PSC DMA
* ALSA SoC Platform driver
*
* Copyright (C) 2008 Secret Lab Technologies Ltd.
* Copyright (C) 2009 Jon Smirl, Digispeaker
*/
#include <linux/module.h>
#include <linux/of_device.h>
#include <sound/soc.h>
#include <sysdev/bestcomm/bestcomm.h>
#include <sysdev/bestcomm/gen_bd.h>
#include <asm/mpc52xx_psc.h>
#include "mpc5200_dma.h"
/*
* Interrupt handlers
*/
static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
{
struct psc_dma *psc_dma = _psc_dma;
struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
u16 isr;
isr = in_be16(&regs->mpc52xx_psc_isr);
/* Playback underrun error */
if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
psc_dma->stats.underrun_count++;
/* Capture overrun error */
if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
psc_dma->stats.overrun_count++;
out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
return IRQ_HANDLED;
}
/**
* psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
* @s: pointer to stream private data structure
*
* Enqueues another audio period buffer into the bestcomm queue.
*
* Note: The routine must only be called when there is space available in
* the queue. Otherwise the enqueue will fail and the audio ring buffer
* will get out of sync
*/
static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
{
struct bcom_bd *bd;
/* Prepare and enqueue the next buffer descriptor */
bd = bcom_prepare_next_buffer(s->bcom_task);
bd->status = s->period_bytes;
bd->data[0] = s->period_next_pt;
bcom_submit_next_buffer(s->bcom_task, NULL);
/* Update for next period */
s->period_next_pt += s->period_bytes;
if (s->period_next_pt >= s->period_end)
s->period_next_pt = s->period_start;
}
static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s)
{
if (s->appl_ptr > s->runtime->control->appl_ptr) {
/*
* In this case s->runtime->control->appl_ptr has wrapped around.
* Play the data to the end of the boundary, then wrap our own
* appl_ptr back around.
*/
while (s->appl_ptr < s->runtime->boundary) {
if (bcom_queue_full(s->bcom_task))
return;
s->appl_ptr += s->period_size;
psc_dma_bcom_enqueue_next_buffer(s);
}
s->appl_ptr -= s->runtime->boundary;
}
while (s->appl_ptr < s->runtime->control->appl_ptr) {
if (bcom_queue_full(s->bcom_task))
return;
s->appl_ptr += s->period_size;
psc_dma_bcom_enqueue_next_buffer(s);
}
}
/* Bestcomm DMA irq handler */
static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
{
struct psc_dma_stream *s = _psc_dma_stream;
spin_lock(&s->psc_dma->lock);
/* For each finished period, dequeue the completed period buffer
* and enqueue a new one in it's place. */
while (bcom_buffer_done(s->bcom_task)) {
bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
s->period_current_pt += s->period_bytes;
if (s->period_current_pt >= s->period_end)
s->period_current_pt = s->period_start;
}
psc_dma_bcom_enqueue_tx(s);
spin_unlock(&s->psc_dma->lock);
/* If the stream is active, then also inform the PCM middle layer
* of the period finished event. */
if (s->active)
snd_pcm_period_elapsed(s->stream);
return IRQ_HANDLED;
}
static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream)
{
struct psc_dma_stream *s = _psc_dma_stream;
spin_lock(&s->psc_dma->lock);
/* For each finished period, dequeue the completed period buffer
* and enqueue a new one in it's place. */
while (bcom_buffer_done(s->bcom_task)) {
bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
s->period_current_pt += s->period_bytes;
if (s->period_current_pt >= s->period_end)
s->period_current_pt = s->period_start;
psc_dma_bcom_enqueue_next_buffer(s);
}
spin_unlock(&s->psc_dma->lock);
/* If the stream is active, then also inform the PCM middle layer
* of the period finished event. */
if (s->active)
snd_pcm_period_elapsed(s->stream);
return IRQ_HANDLED;
}
static int psc_dma_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
}
/**
* psc_dma_trigger: start and stop the DMA transfer.
*
* This function is called by ALSA to start, stop, pause, and resume the DMA
* transfer of data.
*/
static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct psc_dma_stream *s;
struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
u16 imr;
unsigned long flags;
int i;
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
s = &psc_dma->capture;
else
s = &psc_dma->playback;
dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)"
" stream_id=%i\n",
substream, cmd, substream->pstr->stream);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
s->period_bytes = frames_to_bytes(runtime,
runtime->period_size);
s->period_start = virt_to_phys(runtime->dma_area);
s->period_end = s->period_start +
(s->period_bytes * runtime->periods);
s->period_next_pt = s->period_start;
s->period_current_pt = s->period_start;
s->period_size = runtime->period_size;
s->active = 1;
/* track appl_ptr so that we have a better chance of detecting
* end of stream and not over running it.
*/
s->runtime = runtime;
s->appl_ptr = s->runtime->control->appl_ptr -
(runtime->period_size * runtime->periods);
/* Fill up the bestcomm bd queue and enable DMA.
* This will begin filling the PSC's fifo.
*/
spin_lock_irqsave(&psc_dma->lock, flags);
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
bcom_gen_bd_rx_reset(s->bcom_task);
for (i = 0; i < runtime->periods; i++)
if (!bcom_queue_full(s->bcom_task))
psc_dma_bcom_enqueue_next_buffer(s);
} else {
bcom_gen_bd_tx_reset(s->bcom_task);
psc_dma_bcom_enqueue_tx(s);
}
bcom_enable(s->bcom_task);
spin_unlock_irqrestore(&psc_dma->lock, flags);
out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
break;
case SNDRV_PCM_TRIGGER_STOP:
s->active = 0;
spin_lock_irqsave(&psc_dma->lock, flags);
bcom_disable(s->bcom_task);
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
bcom_gen_bd_rx_reset(s->bcom_task);
else
bcom_gen_bd_tx_reset(s->bcom_task);
spin_unlock_irqrestore(&psc_dma->lock, flags);
break;
default:
dev_dbg(psc_dma->dev, "invalid command\n");
return -EINVAL;
}
/* Update interrupt enable settings */
imr = 0;
if (psc_dma->playback.active)
imr |= MPC52xx_PSC_IMR_TXEMP;
if (psc_dma->capture.active)
imr |= MPC52xx_PSC_IMR_ORERR;
out_be16(&regs->isr_imr.imr, psc_dma->imr | imr);
return 0;
}
/* ---------------------------------------------------------------------
* The PSC DMA 'ASoC platform' driver
*
* Can be referenced by an 'ASoC machine' driver
* This driver only deals with the audio bus; it doesn't have any
* interaction with the attached codec
*/
static const struct snd_pcm_hardware psc_dma_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 2,
.period_bytes_max = 1024 * 1024,
.period_bytes_min = 32,
.periods_min = 2,
.periods_max = 256,
.buffer_bytes_max = 2 * 1024 * 1024,
.fifo_size = 512,
};
static int psc_dma_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
struct psc_dma_stream *s;
int rc;
dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
s = &psc_dma->capture;
else
s = &psc_dma->playback;
snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware);
rc = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (rc < 0) {
dev_err(substream->pcm->card->dev, "invalid buffer size\n");
return rc;
}
s->stream = substream;
return 0;
}
static int psc_dma_close(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
struct psc_dma_stream *s;
dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
s = &psc_dma->capture;
else
s = &psc_dma->playback;
if (!psc_dma->playback.active &&
!psc_dma->capture.active) {
/* Disable all interrupts and reset the PSC */
out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
}
s->stream = NULL;
return 0;
}
static snd_pcm_uframes_t
psc_dma_pointer(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
struct psc_dma_stream *s;
dma_addr_t count;
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
s = &psc_dma->capture;
else
s = &psc_dma->playback;
count = s->period_current_pt - s->period_start;
return bytes_to_frames(substream->runtime, count);
}
static int
psc_dma_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
return 0;
}
static struct snd_pcm_ops psc_dma_ops = {
.open = psc_dma_open,
.close = psc_dma_close,
.hw_free = psc_dma_hw_free,
.ioctl = snd_pcm_lib_ioctl,
.pointer = psc_dma_pointer,
.trigger = psc_dma_trigger,
.hw_params = psc_dma_hw_params,
};
static u64 psc_dma_dmamask = 0xffffffff;
static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
size_t size = psc_dma_hardware.buffer_bytes_max;
int rc = 0;
dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n",
card, dai, pcm);
if (!card->dev->dma_mask)
card->dev->dma_mask = &psc_dma_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = 0xffffffff;
if (pcm->streams[0].substream) {
rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
size, &pcm->streams[0].substream->dma_buffer);
if (rc)
goto playback_alloc_err;
}
if (pcm->streams[1].substream) {
rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
size, &pcm->streams[1].substream->dma_buffer);
if (rc)
goto capture_alloc_err;
}
if (rtd->socdev->card->codec->ac97)
rtd->socdev->card->codec->ac97->private_data = psc_dma;
return 0;
capture_alloc_err:
if (pcm->streams[0].substream)
snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
playback_alloc_err:
dev_err(card->dev, "Cannot allocate buffer(s)\n");
return -ENOMEM;
}
static void psc_dma_free(struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;
struct snd_pcm_substream *substream;
int stream;
dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm);
for (stream = 0; stream < 2; stream++) {
substream = pcm->streams[stream].substream;
if (substream) {
snd_dma_free_pages(&substream->dma_buffer);
substream->dma_buffer.area = NULL;
substream->dma_buffer.addr = 0;
}
}
}
struct snd_soc_platform mpc5200_audio_dma_platform = {
.name = "mpc5200-psc-audio",
.pcm_ops = &psc_dma_ops,
.pcm_new = &psc_dma_new,
.pcm_free = &psc_dma_free,
};
EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform);
int mpc5200_audio_dma_create(struct of_device *op)
{
phys_addr_t fifo;
struct psc_dma *psc_dma;
struct resource res;
int size, irq, rc;
const __be32 *prop;
void __iomem *regs;
int ret;
/* Fetch the registers and IRQ of the PSC */
irq = irq_of_parse_and_map(op->node, 0);
if (of_address_to_resource(op->node, 0, &res)) {
dev_err(&op->dev, "Missing reg property\n");
return -ENODEV;
}
regs = ioremap(res.start, 1 + res.end - res.start);
if (!regs) {
dev_err(&op->dev, "Could not map registers\n");
return -ENODEV;
}
/* Allocate and initialize the driver private data */
psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
if (!psc_dma) {
ret = -ENOMEM;
goto out_unmap;
}
/* Get the PSC ID */
prop = of_get_property(op->node, "cell-index", &size);
if (!prop || size < sizeof *prop) {
ret = -ENODEV;
goto out_free;
}
spin_lock_init(&psc_dma->lock);
mutex_init(&psc_dma->mutex);
psc_dma->id = be32_to_cpu(*prop);
psc_dma->irq = irq;
psc_dma->psc_regs = regs;
psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
psc_dma->dev = &op->dev;
psc_dma->playback.psc_dma = psc_dma;
psc_dma->capture.psc_dma = psc_dma;
snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
/* Find the address of the fifo data registers and setup the
* DMA tasks */
fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
psc_dma->capture.bcom_task =
bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512);
psc_dma->playback.bcom_task =
bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo);
if (!psc_dma->capture.bcom_task ||
!psc_dma->playback.bcom_task) {
dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
ret = -ENODEV;
goto out_free;
}
/* Disable all interrupts and reset the PSC */
out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
/* reset receiver */
out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX);
/* reset transmitter */
out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX);
/* reset error */
out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT);
/* reset mode */
out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1);
/* Set up mode register;
* First write: RxRdy (FIFO Alarm) generates rx FIFO irq
* Second write: register Normal mode for non loopback
*/
out_8(&psc_dma->psc_regs->mode, 0);
out_8(&psc_dma->psc_regs->mode, 0);
/* Set the TX and RX fifo alarm thresholds */
out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
/* Lookup the IRQ numbers */
psc_dma->playback.irq =
bcom_get_task_irq(psc_dma->playback.bcom_task);
psc_dma->capture.irq =
bcom_get_task_irq(psc_dma->capture.bcom_task);
rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
"psc-dma-status", psc_dma);
rc |= request_irq(psc_dma->capture.irq,
&psc_dma_bcom_irq_rx, IRQF_SHARED,
"psc-dma-capture", &psc_dma->capture);
rc |= request_irq(psc_dma->playback.irq,
&psc_dma_bcom_irq_tx, IRQF_SHARED,
"psc-dma-playback", &psc_dma->playback);
if (rc) {
ret = -ENODEV;
goto out_irq;
}
/* Save what we've done so it can be found again later */
dev_set_drvdata(&op->dev, psc_dma);
/* Tell the ASoC OF helpers about it */
return snd_soc_register_platform(&mpc5200_audio_dma_platform);
out_irq:
free_irq(psc_dma->irq, psc_dma);
free_irq(psc_dma->capture.irq, &psc_dma->capture);
free_irq(psc_dma->playback.irq, &psc_dma->playback);
out_free:
kfree(psc_dma);
out_unmap:
iounmap(regs);
return ret;
}
EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
int mpc5200_audio_dma_destroy(struct of_device *op)
{
struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n");
snd_soc_unregister_platform(&mpc5200_audio_dma_platform);
bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
/* Release irqs */
free_irq(psc_dma->irq, psc_dma);
free_irq(psc_dma->capture.irq, &psc_dma->capture);
free_irq(psc_dma->playback.irq, &psc_dma->playback);
iounmap(psc_dma->psc_regs);
kfree(psc_dma);
dev_set_drvdata(&op->dev, NULL);
return 0;
}
EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy);
MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
MODULE_LICENSE("GPL");

81
kernel/sound/soc/fsl/mpc5200_dma.h Normal file
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/*
* Freescale MPC5200 Audio DMA driver
*/
#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
#define __SOUND_SOC_FSL_MPC5200_DMA_H__
#define PSC_STREAM_NAME_LEN 32
/**
* psc_ac97_stream - Data specific to a single stream (playback or capture)
* @active: flag indicating if the stream is active
* @psc_dma: pointer back to parent psc_dma data structure
* @bcom_task: bestcomm task structure
* @irq: irq number for bestcomm task
* @period_start: physical address of start of DMA region
* @period_end: physical address of end of DMA region
* @period_next_pt: physical address of next DMA buffer to enqueue
* @period_bytes: size of DMA period in bytes
*/
struct psc_dma_stream {
struct snd_pcm_runtime *runtime;
snd_pcm_uframes_t appl_ptr;
int active;
struct psc_dma *psc_dma;
struct bcom_task *bcom_task;
int irq;
struct snd_pcm_substream *stream;
dma_addr_t period_start;
dma_addr_t period_end;
dma_addr_t period_next_pt;
dma_addr_t period_current_pt;
int period_bytes;
int period_size;
};
/**
* psc_dma - Private driver data
* @name: short name for this device ("PSC0", "PSC1", etc)
* @psc_regs: pointer to the PSC's registers
* @fifo_regs: pointer to the PSC's FIFO registers
* @irq: IRQ of this PSC
* @dev: struct device pointer
* @dai: the CPU DAI for this device
* @sicr: Base value used in serial interface control register; mode is ORed
* with this value.
* @playback: Playback stream context data
* @capture: Capture stream context data
*/
struct psc_dma {
char name[32];
struct mpc52xx_psc __iomem *psc_regs;
struct mpc52xx_psc_fifo __iomem *fifo_regs;
unsigned int irq;
struct device *dev;
spinlock_t lock;
struct mutex mutex;
u32 sicr;
uint sysclk;
int imr;
int id;
unsigned int slots;
/* per-stream data */
struct psc_dma_stream playback;
struct psc_dma_stream capture;
/* Statistics */
struct {
unsigned long overrun_count;
unsigned long underrun_count;
} stats;
};
int mpc5200_audio_dma_create(struct of_device *op);
int mpc5200_audio_dma_destroy(struct of_device *op);
extern struct snd_soc_platform mpc5200_audio_dma_platform;
#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */

345
kernel/sound/soc/fsl/mpc5200_psc_ac97.c Normal file
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/*
* linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
*
* Copyright (C) 2009 Jon Smirl, Digispeaker
* Author: Jon Smirl <jonsmirl@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <linux/delay.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <asm/time.h>
#include <asm/delay.h>
#include <asm/mpc52xx_psc.h>
#include "mpc5200_dma.h"
#include "mpc5200_psc_ac97.h"
#define DRV_NAME "mpc5200-psc-ac97"
/* ALSA only supports a single AC97 device so static is recommend here */
static struct psc_dma *psc_dma;
static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
int status;
unsigned int val;
mutex_lock(&psc_dma->mutex);
/* Wait for command send status zero = ready */
status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
MPC52xx_PSC_SR_CMDSEND), 100, 0);
if (status == 0) {
pr_err("timeout on ac97 bus (rdy)\n");
mutex_unlock(&psc_dma->mutex);
return -ENODEV;
}
/* Force clear the data valid bit */
in_be32(&psc_dma->psc_regs->ac97_data);
/* Send the read */
out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
/* Wait for the answer */
status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) &
MPC52xx_PSC_SR_DATA_VAL), 100, 0);
if (status == 0) {
pr_err("timeout on ac97 read (val) %x\n",
in_be16(&psc_dma->psc_regs->sr_csr.status));
mutex_unlock(&psc_dma->mutex);
return -ENODEV;
}
/* Get the data */
val = in_be32(&psc_dma->psc_regs->ac97_data);
if (((val >> 24) & 0x7f) != reg) {
pr_err("reg echo error on ac97 read\n");
mutex_unlock(&psc_dma->mutex);
return -ENODEV;
}
val = (val >> 8) & 0xffff;
mutex_unlock(&psc_dma->mutex);
return (unsigned short) val;
}
static void psc_ac97_write(struct snd_ac97 *ac97,
unsigned short reg, unsigned short val)
{
int status;
mutex_lock(&psc_dma->mutex);
/* Wait for command status zero = ready */
status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
MPC52xx_PSC_SR_CMDSEND), 100, 0);
if (status == 0) {
pr_err("timeout on ac97 bus (write)\n");
goto out;
}
/* Write data */
out_be32(&psc_dma->psc_regs->ac97_cmd,
((reg & 0x7f) << 24) | (val << 8));
out:
mutex_unlock(&psc_dma->mutex);
}
static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
{
struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
udelay(3);
out_be32(&regs->sicr, psc_dma->sicr);
}
static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
{
struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
/* Do a cold reset */
out_8(&regs->op1, MPC52xx_PSC_OP_RES);
udelay(10);
out_8(&regs->op0, MPC52xx_PSC_OP_RES);
msleep(1);
psc_ac97_warm_reset(ac97);
}
struct snd_ac97_bus_ops soc_ac97_ops = {
.read = psc_ac97_read,
.write = psc_ac97_write,
.reset = psc_ac97_cold_reset,
.warm_reset = psc_ac97_warm_reset,
};
EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
struct psc_dma *psc_dma = cpu_dai->private_data;
dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
" periods=%i buffer_size=%i buffer_bytes=%i channels=%i"
" rate=%i format=%i\n",
__func__, substream, params_period_size(params),
params_period_bytes(params), params_periods(params),
params_buffer_size(params), params_buffer_bytes(params),
params_channels(params), params_rate(params),
params_format(params));
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
if (params_channels(params) == 1)
psc_dma->slots |= 0x00000100;
else
psc_dma->slots |= 0x00000300;
} else {
if (params_channels(params) == 1)
psc_dma->slots |= 0x01000000;
else
psc_dma->slots |= 0x03000000;
}
out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
return 0;
}
static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
struct psc_dma *psc_dma = cpu_dai->private_data;
if (params_channels(params) == 1)
out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
else
out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
return 0;
}
static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_STOP:
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
psc_dma->slots &= 0xFFFF0000;
else
psc_dma->slots &= 0x0000FFFF;
out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
break;
}
return 0;
}
static int psc_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *cpu_dai)
{
struct psc_dma *psc_dma = cpu_dai->private_data;
struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
/* Go */
out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
return 0;
}
/* ---------------------------------------------------------------------
* ALSA SoC Bindings
*
* - Digital Audio Interface (DAI) template
* - create/destroy dai hooks
*/
/**
* psc_ac97_dai_template: template CPU Digital Audio Interface
*/
static struct snd_soc_dai_ops psc_ac97_analog_ops = {
.hw_params = psc_ac97_hw_analog_params,
.trigger = psc_ac97_trigger,
};
static struct snd_soc_dai_ops psc_ac97_digital_ops = {
.hw_params = psc_ac97_hw_digital_params,
};
struct snd_soc_dai psc_ac97_dai[] = {
{
.name = "AC97",
.ac97_control = 1,
.probe = psc_ac97_probe,
.playback = {
.channels_min = 1,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S32_BE,
},
.capture = {
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S32_BE,
},
.ops = &psc_ac97_analog_ops,
},
{
.name = "SPDIF",
.ac97_control = 1,
.playback = {
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
},
.ops = &psc_ac97_digital_ops,
} };
EXPORT_SYMBOL_GPL(psc_ac97_dai);
/* ---------------------------------------------------------------------
* OF platform bus binding code:
* - Probe/remove operations
* - OF device match table
*/
static int __devinit psc_ac97_of_probe(struct of_device *op,
const struct of_device_id *match)
{
int rc, i;
struct snd_ac97 ac97;
struct mpc52xx_psc __iomem *regs;
rc = mpc5200_audio_dma_create(op);
if (rc != 0)
return rc;
for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
psc_ac97_dai[i].dev = &op->dev;
rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
if (rc != 0) {
dev_err(&op->dev, "Failed to register DAI\n");
return rc;
}
psc_dma = dev_get_drvdata(&op->dev);
regs = psc_dma->psc_regs;
ac97.private_data = psc_dma;
for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
psc_ac97_dai[i].private_data = psc_dma;
psc_dma->imr = 0;
out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
/* Configure the serial interface mode to AC97 */
psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
out_be32(&regs->sicr, psc_dma->sicr);
/* No slots active */
out_be32(&regs->ac97_slots, 0x00000000);
return 0;
}
static int __devexit psc_ac97_of_remove(struct of_device *op)
{
return mpc5200_audio_dma_destroy(op);
}
/* Match table for of_platform binding */
static struct of_device_id psc_ac97_match[] __devinitdata = {
{ .compatible = "fsl,mpc5200-psc-ac97", },
{ .compatible = "fsl,mpc5200b-psc-ac97", },
{}
};
MODULE_DEVICE_TABLE(of, psc_ac97_match);
static struct of_platform_driver psc_ac97_driver = {
.match_table = psc_ac97_match,
.probe = psc_ac97_of_probe,
.remove = __devexit_p(psc_ac97_of_remove),
.driver = {
.name = "mpc5200-psc-ac97",
.owner = THIS_MODULE,
},
};
/* ---------------------------------------------------------------------
* Module setup and teardown; simply register the of_platform driver
* for the PSC in AC97 mode.
*/
static int __init psc_ac97_init(void)
{
return of_register_platform_driver(&psc_ac97_driver);
}
module_init(psc_ac97_init);
static void __exit psc_ac97_exit(void)
{
of_unregister_platform_driver(&psc_ac97_driver);
}
module_exit(psc_ac97_exit);
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_DESCRIPTION("mpc5200 AC97 module");
MODULE_LICENSE("GPL");

15
kernel/sound/soc/fsl/mpc5200_psc_ac97.h Normal file
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/*
* Freescale MPC5200 PSC in AC97 mode
* ALSA SoC Digital Audio Interface (DAI) driver
*
*/
#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
extern struct snd_soc_dai psc_ac97_dai[];
#define MPC5200_AC97_NORMAL 0
#define MPC5200_AC97_SPDIF 1
#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */

251
kernel/sound/soc/fsl/mpc5200_psc_i2s.c Normal file
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/*
* Freescale MPC5200 PSC in I2S mode
* ALSA SoC Digital Audio Interface (DAI) driver
*
* Copyright (C) 2008 Secret Lab Technologies Ltd.
* Copyright (C) 2009 Jon Smirl, Digispeaker
*/
#include <linux/module.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <asm/mpc52xx_psc.h>
#include "mpc5200_psc_i2s.h"
#include "mpc5200_dma.h"
/**
* PSC_I2S_RATES: sample rates supported by the I2S
*
* This driver currently only supports the PSC running in I2S slave mode,
* which means the codec determines the sample rate. Therefore, we tell
* ALSA that we support all rates and let the codec driver decide what rates
* are really supported.
*/
#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
SNDRV_PCM_RATE_CONTINUOUS)
/**
* PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
*/
#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
u32 mode;
dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
" periods=%i buffer_size=%i buffer_bytes=%i\n",
__func__, substream, params_period_size(params),
params_period_bytes(params), params_periods(params),
params_buffer_size(params), params_buffer_bytes(params));
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
mode = MPC52xx_PSC_SICR_SIM_CODEC_8;
break;
case SNDRV_PCM_FORMAT_S16_BE:
mode = MPC52xx_PSC_SICR_SIM_CODEC_16;
break;
case SNDRV_PCM_FORMAT_S24_BE:
mode = MPC52xx_PSC_SICR_SIM_CODEC_24;
break;
case SNDRV_PCM_FORMAT_S32_BE:
mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
break;
default:
dev_dbg(psc_dma->dev, "invalid format\n");
return -EINVAL;
}
out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
return 0;
}
/**
* psc_i2s_set_sysclk: set the clock frequency and direction
*
* This function is called by the machine driver to tell us what the clock
* frequency and direction are.
*
* Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
* and we don't care about the frequency. Return an error if the direction
* is not SND_SOC_CLOCK_IN.
*
* @clk_id: reserved, should be zero
* @freq: the frequency of the given clock ID, currently ignored
* @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
*/
static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
struct psc_dma *psc_dma = cpu_dai->private_data;
dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
cpu_dai, dir);
return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
}
/**
* psc_i2s_set_fmt: set the serial format.
*
* This function is called by the machine driver to tell us what serial
* format to use.
*
* This driver only supports I2S mode. Return an error if the format is
* not SND_SOC_DAIFMT_I2S.
*
* @format: one of SND_SOC_DAIFMT_xxx
*/
static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
{
struct psc_dma *psc_dma = cpu_dai->private_data;
dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
cpu_dai, format);
return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
}
/* ---------------------------------------------------------------------
* ALSA SoC Bindings
*
* - Digital Audio Interface (DAI) template
* - create/destroy dai hooks
*/
/**
* psc_i2s_dai_template: template CPU Digital Audio Interface
*/
static struct snd_soc_dai_ops psc_i2s_dai_ops = {
.hw_params = psc_i2s_hw_params,
.set_sysclk = psc_i2s_set_sysclk,
.set_fmt = psc_i2s_set_fmt,
};
struct snd_soc_dai psc_i2s_dai[] = {{
.name = "I2S",
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = PSC_I2S_RATES,
.formats = PSC_I2S_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = PSC_I2S_RATES,
.formats = PSC_I2S_FORMATS,
},
.ops = &psc_i2s_dai_ops,
} };
EXPORT_SYMBOL_GPL(psc_i2s_dai);
/* ---------------------------------------------------------------------
* OF platform bus binding code:
* - Probe/remove operations
* - OF device match table
*/
static int __devinit psc_i2s_of_probe(struct of_device *op,
const struct of_device_id *match)
{
int rc;
struct psc_dma *psc_dma;
struct mpc52xx_psc __iomem *regs;
rc = mpc5200_audio_dma_create(op);
if (rc != 0)
return rc;
rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
if (rc != 0) {
pr_err("Failed to register DAI\n");
return 0;
}
psc_dma = dev_get_drvdata(&op->dev);
regs = psc_dma->psc_regs;
/* Configure the serial interface mode; defaulting to CODEC8 mode */
psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
MPC52xx_PSC_SICR_CLKPOL;
out_be32(&psc_dma->psc_regs->sicr,
psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
/* Check for the codec handle. If it is not present then we
* are done */
if (!of_get_property(op->node, "codec-handle", NULL))
return 0;
/* Due to errata in the dma mode; need to line up enabling
* the transmitter with a transition on the frame sync
* line */
/* first make sure it is low */
while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
;
/* then wait for the transition to high */
while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
;
/* Finally, enable the PSC.
* Receiver must always be enabled; even when we only want
* transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
/* Go */
out_8(&psc_dma->psc_regs->command,
MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
return 0;
}
static int __devexit psc_i2s_of_remove(struct of_device *op)
{
return mpc5200_audio_dma_destroy(op);
}
/* Match table for of_platform binding */
static struct of_device_id psc_i2s_match[] __devinitdata = {
{ .compatible = "fsl,mpc5200-psc-i2s", },
{ .compatible = "fsl,mpc5200b-psc-i2s", },
{}
};
MODULE_DEVICE_TABLE(of, psc_i2s_match);
static struct of_platform_driver psc_i2s_driver = {
.match_table = psc_i2s_match,
.probe = psc_i2s_of_probe,
.remove = __devexit_p(psc_i2s_of_remove),
.driver = {
.name = "mpc5200-psc-i2s",
.owner = THIS_MODULE,
},
};
/* ---------------------------------------------------------------------
* Module setup and teardown; simply register the of_platform driver
* for the PSC in I2S mode.
*/
static int __init psc_i2s_init(void)
{
return of_register_platform_driver(&psc_i2s_driver);
}
module_init(psc_i2s_init);
static void __exit psc_i2s_exit(void)
{
of_unregister_platform_driver(&psc_i2s_driver);
}
module_exit(psc_i2s_exit);
MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
MODULE_LICENSE("GPL");

12
kernel/sound/soc/fsl/mpc5200_psc_i2s.h Normal file
View File

@@ -0,0 +1,12 @@
/*
* Freescale MPC5200 PSC in I2S mode
* ALSA SoC Digital Audio Interface (DAI) driver
*
*/
#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
extern struct snd_soc_dai psc_i2s_dai[];
#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */

624
kernel/sound/soc/fsl/mpc8610_hpcd.c Normal file
View File

@@ -0,0 +1,624 @@
/**
* Freescale MPC8610HPCD ALSA SoC Fabric driver
*
* Author: Timur Tabi <timur@freescale.com>
*
* Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
* under the terms of the GNU General Public License version 2. This
* program is licensed "as is" without any warranty of any kind, whether
* express or implied.
*/
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <sound/soc.h>
#include <asm/immap_86xx.h>
#include "../codecs/cs4270.h"
#include "fsl_dma.h"
#include "fsl_ssi.h"
/**
* mpc8610_hpcd_data: fabric-specific ASoC device data
*
* This structure contains data for a single sound platform device on an
* MPC8610 HPCD. Some of the data is taken from the device tree.
*/
struct mpc8610_hpcd_data {
struct snd_soc_device sound_devdata;
struct snd_soc_dai_link dai;
struct snd_soc_card machine;
unsigned int dai_format;
unsigned int codec_clk_direction;
unsigned int cpu_clk_direction;
unsigned int clk_frequency;
struct ccsr_guts __iomem *guts;
struct ccsr_ssi __iomem *ssi;
unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */
unsigned int ssi_irq;
unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */
unsigned int dma_irq[2];
struct ccsr_dma_channel __iomem *dma[2];
unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
};
/**
* mpc8610_hpcd_machine_probe: initalize the board
*
* This function is called when platform_device_add() is called. It is used
* to initialize the board-specific hardware.
*
* Here we program the DMACR and PMUXCR registers.
*/
static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
{
struct mpc8610_hpcd_data *machine_data =
sound_device->dev.platform_data;
/* Program the signal routing between the SSI and the DMA */
guts_set_dmacr(machine_data->guts, machine_data->dma_id,
machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI);
guts_set_dmacr(machine_data->guts, machine_data->dma_id,
machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI);
guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id,
machine_data->dma_channel_id[0], 0);
guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id,
machine_data->dma_channel_id[1], 0);
switch (machine_data->ssi_id) {
case 0:
clrsetbits_be32(&machine_data->guts->pmuxcr,
CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI);
break;
case 1:
clrsetbits_be32(&machine_data->guts->pmuxcr,
CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI);
break;
}
return 0;
}
/**
* mpc8610_hpcd_startup: program the board with various hardware parameters
*
* This function takes board-specific information, like clock frequencies
* and serial data formats, and passes that information to the codec and
* transport drivers.
*/
static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct mpc8610_hpcd_data *machine_data =
rtd->socdev->dev->platform_data;
int ret = 0;
/* Tell the CPU driver what the serial protocol is. */
ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format);
if (ret < 0) {
dev_err(substream->pcm->card->dev,
"could not set CPU driver audio format\n");
return ret;
}
/* Tell the codec driver what the serial protocol is. */
ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format);
if (ret < 0) {
dev_err(substream->pcm->card->dev,
"could not set codec driver audio format\n");
return ret;
}
/*
* Tell the CPU driver what the clock frequency is, and whether it's a
* slave or master.
*/
ret = snd_soc_dai_set_sysclk(cpu_dai, 0,
machine_data->clk_frequency,
machine_data->cpu_clk_direction);
if (ret < 0) {
dev_err(substream->pcm->card->dev,
"could not set CPU driver clock parameters\n");
return ret;
}
/*
* Tell the codec driver what the MCLK frequency is, and whether it's
* a slave or master.
*/
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
machine_data->clk_frequency,
machine_data->codec_clk_direction);
if (ret < 0) {
dev_err(substream->pcm->card->dev,
"could not set codec driver clock params\n");
return ret;
}
return 0;
}
/**
* mpc8610_hpcd_machine_remove: Remove the sound device
*
* This function is called to remove the sound device for one SSI. We
* de-program the DMACR and PMUXCR register.
*/
int mpc8610_hpcd_machine_remove(struct platform_device *sound_device)
{
struct mpc8610_hpcd_data *machine_data =
sound_device->dev.platform_data;
/* Restore the signal routing */
guts_set_dmacr(machine_data->guts, machine_data->dma_id,
machine_data->dma_channel_id[0], 0);
guts_set_dmacr(machine_data->guts, machine_data->dma_id,
machine_data->dma_channel_id[1], 0);
switch (machine_data->ssi_id) {
case 0:
clrsetbits_be32(&machine_data->guts->pmuxcr,
CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA);
break;
case 1:
clrsetbits_be32(&machine_data->guts->pmuxcr,
CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA);
break;
}
return 0;
}
/**
* mpc8610_hpcd_ops: ASoC fabric driver operations
*/
static struct snd_soc_ops mpc8610_hpcd_ops = {
.startup = mpc8610_hpcd_startup,
};
/**
* mpc8610_hpcd_probe: OF probe function for the fabric driver
*
* This function gets called when an SSI node is found in the device tree.
*
* Although this is a fabric driver, the SSI node is the "master" node with
* respect to audio hardware connections. Therefore, we create a new ASoC
* device for each new SSI node that has a codec attached.
*
* FIXME: Currently, we only support one DMA controller, so if there are
* multiple SSI nodes with codecs, only the first will be supported.
*
* FIXME: Even if we did support multiple DMA controllers, we have no
* mechanism for assigning DMA controllers and channels to the individual
* SSI devices. We also probably aren't compatible with the generic Elo DMA
* device driver.
*/
static int mpc8610_hpcd_probe(struct of_device *ofdev,
const struct of_device_id *match)
{
struct device_node *np = ofdev->node;
struct device_node *codec_np = NULL;
struct device_node *guts_np = NULL;
struct device_node *dma_np = NULL;
struct device_node *dma_channel_np = NULL;
const phandle *codec_ph;
const char *sprop;
const u32 *iprop;
struct resource res;
struct platform_device *sound_device = NULL;
struct mpc8610_hpcd_data *machine_data;
struct fsl_ssi_info ssi_info;
struct fsl_dma_info dma_info;
int ret = -ENODEV;
unsigned int playback_dma_channel;
unsigned int capture_dma_channel;
machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
if (!machine_data)
return -ENOMEM;
memset(&ssi_info, 0, sizeof(ssi_info));
memset(&dma_info, 0, sizeof(dma_info));
ssi_info.dev = &ofdev->dev;
/*
* We are only interested in SSIs with a codec phandle in them, so let's
* make sure this SSI has one.
*/
codec_ph = of_get_property(np, "codec-handle", NULL);
if (!codec_ph)
goto error;
codec_np = of_find_node_by_phandle(*codec_ph);
if (!codec_np)
goto error;
/* The MPC8610 HPCD only knows about the CS4270 codec, so reject
anything else. */
if (!of_device_is_compatible(codec_np, "cirrus,cs4270"))
goto error;
/* Get the device ID */
iprop = of_get_property(np, "cell-index", NULL);
if (!iprop) {
dev_err(&ofdev->dev, "cell-index property not found\n");
ret = -EINVAL;
goto error;
}
machine_data->ssi_id = *iprop;
ssi_info.id = *iprop;
/* Get the serial format and clock direction. */
sprop = of_get_property(np, "fsl,mode", NULL);
if (!sprop) {
dev_err(&ofdev->dev, "fsl,mode property not found\n");
ret = -EINVAL;
goto error;
}
if (strcasecmp(sprop, "i2s-slave") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_I2S;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
/*
* In i2s-slave mode, the codec has its own clock source, so we
* need to get the frequency from the device tree and pass it to
* the codec driver.
*/
iprop = of_get_property(codec_np, "clock-frequency", NULL);
if (!iprop || !*iprop) {
dev_err(&ofdev->dev, "codec bus-frequency property "
"is missing or invalid\n");
ret = -EINVAL;
goto error;
}
machine_data->clk_frequency = *iprop;
} else if (strcasecmp(sprop, "i2s-master") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_I2S;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "lj-slave") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "lj-master") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "rj-slave") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "rj-master") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "ac97-slave") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_AC97;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "ac97-master") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_AC97;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else {
dev_err(&ofdev->dev,
"unrecognized fsl,mode property \"%s\"\n", sprop);
ret = -EINVAL;
goto error;
}
if (!machine_data->clk_frequency) {
dev_err(&ofdev->dev, "unknown clock frequency\n");
ret = -EINVAL;
goto error;
}
/* Read the SSI information from the device tree */
ret = of_address_to_resource(np, 0, &res);
if (ret) {
dev_err(&ofdev->dev, "could not obtain SSI address\n");
goto error;
}
if (!res.start) {
dev_err(&ofdev->dev, "invalid SSI address\n");
goto error;
}
ssi_info.ssi_phys = res.start;
machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi));
if (!machine_data->ssi) {
dev_err(&ofdev->dev, "could not map SSI address %x\n",
ssi_info.ssi_phys);
ret = -EINVAL;
goto error;
}
ssi_info.ssi = machine_data->ssi;
/* Get the IRQ of the SSI */
machine_data->ssi_irq = irq_of_parse_and_map(np, 0);
if (!machine_data->ssi_irq) {
dev_err(&ofdev->dev, "could not get SSI IRQ\n");
ret = -EINVAL;
goto error;
}
ssi_info.irq = machine_data->ssi_irq;
/* Do we want to use asynchronous mode? */
ssi_info.asynchronous =
of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0;
if (ssi_info.asynchronous)
dev_info(&ofdev->dev, "using asynchronous mode\n");
/* Map the global utilities registers. */
guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
if (!guts_np) {
dev_err(&ofdev->dev, "could not obtain address of GUTS\n");
ret = -EINVAL;
goto error;
}
machine_data->guts = of_iomap(guts_np, 0);
of_node_put(guts_np);
if (!machine_data->guts) {
dev_err(&ofdev->dev, "could not map GUTS\n");
ret = -EINVAL;
goto error;
}
/* Find the DMA channels to use. Both SSIs need to use the same DMA
* controller, so let's use DMA#1.
*/
for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") {
iprop = of_get_property(dma_np, "cell-index", NULL);
if (iprop && (*iprop == 0)) {
of_node_put(dma_np);
break;
}
}
if (!dma_np) {
dev_err(&ofdev->dev, "could not find DMA node\n");
ret = -EINVAL;
goto error;
}
machine_data->dma_id = *iprop;
/* SSI1 needs to use DMA Channels 0 and 1, and SSI2 needs to use DMA
* channels 2 and 3. This is just how the MPC8610 is wired
* internally.
*/
playback_dma_channel = (machine_data->ssi_id == 0) ? 0 : 2;
capture_dma_channel = (machine_data->ssi_id == 0) ? 1 : 3;
/*
* Find the DMA channels to use.
*/
while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) {
iprop = of_get_property(dma_channel_np, "cell-index", NULL);
if (iprop && (*iprop == playback_dma_channel)) {
/* dma_channel[0] and dma_irq[0] are for playback */
dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0);
dma_info.dma_irq[0] =
irq_of_parse_and_map(dma_channel_np, 0);
machine_data->dma_channel_id[0] = *iprop;
continue;
}
if (iprop && (*iprop == capture_dma_channel)) {
/* dma_channel[1] and dma_irq[1] are for capture */
dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0);
dma_info.dma_irq[1] =
irq_of_parse_and_map(dma_channel_np, 0);
machine_data->dma_channel_id[1] = *iprop;
continue;
}
}
if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] ||
!dma_info.dma_irq[0] || !dma_info.dma_irq[1]) {
dev_err(&ofdev->dev, "could not find DMA channels\n");
ret = -EINVAL;
goto error;
}
dma_info.ssi_stx_phys = ssi_info.ssi_phys +
offsetof(struct ccsr_ssi, stx0);
dma_info.ssi_srx_phys = ssi_info.ssi_phys +
offsetof(struct ccsr_ssi, srx0);
/* We have the DMA information, so tell the DMA driver what it is */
if (!fsl_dma_configure(&dma_info)) {
dev_err(&ofdev->dev, "could not instantiate DMA device\n");
ret = -EBUSY;
goto error;
}
/*
* Initialize our DAI data structure. We should probably get this
* information from the device tree.
*/
machine_data->dai.name = "CS4270";
machine_data->dai.stream_name = "CS4270";
machine_data->dai.cpu_dai = fsl_ssi_create_dai(&ssi_info);
machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */
machine_data->dai.ops = &mpc8610_hpcd_ops;
machine_data->machine.probe = mpc8610_hpcd_machine_probe;
machine_data->machine.remove = mpc8610_hpcd_machine_remove;
machine_data->machine.name = "MPC8610 HPCD";
machine_data->machine.num_links = 1;
machine_data->machine.dai_link = &machine_data->dai;
/* Allocate a new audio platform device structure */
sound_device = platform_device_alloc("soc-audio", -1);
if (!sound_device) {
dev_err(&ofdev->dev, "platform device allocation failed\n");
ret = -ENOMEM;
goto error;
}
machine_data->sound_devdata.card = &machine_data->machine;
machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
machine_data->machine.platform = &fsl_soc_platform;
sound_device->dev.platform_data = machine_data;
/* Set the platform device and ASoC device to point to each other */
platform_set_drvdata(sound_device, &machine_data->sound_devdata);
machine_data->sound_devdata.dev = &sound_device->dev;
/* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(),
if it exists. */
ret = platform_device_add(sound_device);
if (ret) {
dev_err(&ofdev->dev, "platform device add failed\n");
goto error;
}
dev_set_drvdata(&ofdev->dev, sound_device);
return 0;
error:
of_node_put(codec_np);
of_node_put(guts_np);
of_node_put(dma_np);
of_node_put(dma_channel_np);
if (sound_device)
platform_device_unregister(sound_device);
if (machine_data->dai.cpu_dai)
fsl_ssi_destroy_dai(machine_data->dai.cpu_dai);
if (ssi_info.ssi)
iounmap(ssi_info.ssi);
if (ssi_info.irq)
irq_dispose_mapping(ssi_info.irq);
if (dma_info.dma_channel[0])
iounmap(dma_info.dma_channel[0]);
if (dma_info.dma_channel[1])
iounmap(dma_info.dma_channel[1]);
if (dma_info.dma_irq[0])
irq_dispose_mapping(dma_info.dma_irq[0]);
if (dma_info.dma_irq[1])
irq_dispose_mapping(dma_info.dma_irq[1]);
if (machine_data->guts)
iounmap(machine_data->guts);
kfree(machine_data);
return ret;
}
/**
* mpc8610_hpcd_remove: remove the OF device
*
* This function is called when the OF device is removed.
*/
static int mpc8610_hpcd_remove(struct of_device *ofdev)
{
struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev);
struct mpc8610_hpcd_data *machine_data =
sound_device->dev.platform_data;
platform_device_unregister(sound_device);
if (machine_data->dai.cpu_dai)
fsl_ssi_destroy_dai(machine_data->dai.cpu_dai);
if (machine_data->ssi)
iounmap(machine_data->ssi);
if (machine_data->dma[0])
iounmap(machine_data->dma[0]);
if (machine_data->dma[1])
iounmap(machine_data->dma[1]);
if (machine_data->dma_irq[0])
irq_dispose_mapping(machine_data->dma_irq[0]);
if (machine_data->dma_irq[1])
irq_dispose_mapping(machine_data->dma_irq[1]);
if (machine_data->guts)
iounmap(machine_data->guts);
kfree(machine_data);
sound_device->dev.platform_data = NULL;
dev_set_drvdata(&ofdev->dev, NULL);
return 0;
}
static struct of_device_id mpc8610_hpcd_match[] = {
{
.compatible = "fsl,mpc8610-ssi",
},
{}
};
MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match);
static struct of_platform_driver mpc8610_hpcd_of_driver = {
.owner = THIS_MODULE,
.name = "mpc8610_hpcd",
.match_table = mpc8610_hpcd_match,
.probe = mpc8610_hpcd_probe,
.remove = mpc8610_hpcd_remove,
};
/**
* mpc8610_hpcd_init: fabric driver initialization.
*
* This function is called when this module is loaded.
*/
static int __init mpc8610_hpcd_init(void)
{
int ret;
printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n");
ret = of_register_platform_driver(&mpc8610_hpcd_of_driver);
if (ret)
printk(KERN_ERR
"mpc8610-hpcd: failed to register platform driver\n");
return ret;
}
/**
* mpc8610_hpcd_exit: fabric driver exit
*
* This function is called when this driver is unloaded.
*/
static void __exit mpc8610_hpcd_exit(void)
{
of_unregister_platform_driver(&mpc8610_hpcd_of_driver);
}
module_init(mpc8610_hpcd_init);
module_exit(mpc8610_hpcd_exit);
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver");
MODULE_LICENSE("GPL");

92
kernel/sound/soc/fsl/pcm030-audio-fabric.c Normal file
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@@ -0,0 +1,92 @@
/*
* Phytec pcm030 driver for the PSC of the Freescale MPC52xx
* configured as AC97 interface
*
* Copyright 2008 Jon Smirl, Digispeaker
* Author: Jon Smirl <jonsmirl@gmail.com>
*
* This file is licensed under the terms of the GNU General Public License
* version 2. This program is licensed "as is" without any warranty of any
* kind, whether express or implied.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/device.h>
#include <linux/delay.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/soc-of-simple.h>
#include "mpc5200_dma.h"
#include "mpc5200_psc_ac97.h"
#include "../codecs/wm9712.h"
#define DRV_NAME "pcm030-audio-fabric"
static struct snd_soc_device device;
static struct snd_soc_card card;
static struct snd_soc_dai_link pcm030_fabric_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 Analog",
.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
.cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
},
{
.name = "AC97",
.stream_name = "AC97 IEC958",
.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
.cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
},
};
static __init int pcm030_fabric_init(void)
{
struct platform_device *pdev;
int rc;
if (!machine_is_compatible("phytec,pcm030"))
return -ENODEV;
card.platform = &mpc5200_audio_dma_platform;
card.name = "pcm030";
card.dai_link = pcm030_fabric_dai;
card.num_links = ARRAY_SIZE(pcm030_fabric_dai);
device.card = &card;
device.codec_dev = &soc_codec_dev_wm9712;
pdev = platform_device_alloc("soc-audio", 1);
if (!pdev) {
pr_err("pcm030_fabric_init: platform_device_alloc() failed\n");
return -ENODEV;
}
platform_set_drvdata(pdev, &device);
device.dev = &pdev->dev;
rc = platform_device_add(pdev);
if (rc) {
pr_err("pcm030_fabric_init: platform_device_add() failed\n");
return -ENODEV;
}
return 0;
}
module_init(pcm030_fabric_init);
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver");
MODULE_LICENSE("GPL");

171
kernel/sound/soc/fsl/soc-of-simple.c Normal file
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/*
* OF helpers for ALSA SoC Layer
*
* Copyright (C) 2008, Secret Lab Technologies Ltd.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
#include <linux/platform_device.h>
#include <linux/of.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-of-simple.h>
#include <sound/initval.h>
MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
static DEFINE_MUTEX(of_snd_soc_mutex);
static LIST_HEAD(of_snd_soc_device_list);
static int of_snd_soc_next_index;
struct of_snd_soc_device {
int id;
struct list_head list;
struct snd_soc_device device;
struct snd_soc_card card;
struct snd_soc_dai_link dai_link;
struct platform_device *pdev;
struct device_node *platform_node;
struct device_node *codec_node;
};
static struct snd_soc_ops of_snd_soc_ops = {
};
static struct of_snd_soc_device *
of_snd_soc_get_device(struct device_node *codec_node)
{
struct of_snd_soc_device *of_soc;
list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
if (of_soc->codec_node == codec_node)
return of_soc;
}
of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
if (!of_soc)
return NULL;
/* Initialize the structure and add it to the global list */
of_soc->codec_node = codec_node;
of_soc->id = of_snd_soc_next_index++;
of_soc->card.dai_link = &of_soc->dai_link;
of_soc->card.num_links = 1;
of_soc->device.card = &of_soc->card;
of_soc->dai_link.ops = &of_snd_soc_ops;
list_add(&of_soc->list, &of_snd_soc_device_list);
return of_soc;
}
static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
{
struct platform_device *pdev;
int rc;
/* Only register the device if both the codec and platform have
* been registered */
if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
return;
pr_info("platform<-->codec match achieved; registering machine\n");
pdev = platform_device_alloc("soc-audio", of_soc->id);
if (!pdev) {
pr_err("of_soc: platform_device_alloc() failed\n");
return;
}
pdev->dev.platform_data = of_soc;
platform_set_drvdata(pdev, &of_soc->device);
of_soc->device.dev = &pdev->dev;
/* The ASoC device is complete; register it */
rc = platform_device_add(pdev);
if (rc) {
pr_err("of_soc: platform_device_add() failed\n");
return;
}
}
int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
void *codec_data, struct snd_soc_dai *dai,
struct device_node *node)
{
struct of_snd_soc_device *of_soc;
int rc = 0;
pr_info("registering ASoC codec driver: %s\n", node->full_name);
mutex_lock(&of_snd_soc_mutex);
of_soc = of_snd_soc_get_device(node);
if (!of_soc) {
rc = -ENOMEM;
goto out;
}
/* Store the codec data */
of_soc->device.codec_data = codec_data;
of_soc->device.codec_dev = codec_dev;
of_soc->dai_link.name = (char *)node->name;
of_soc->dai_link.stream_name = (char *)node->name;
of_soc->dai_link.codec_dai = dai;
/* Now try to register the SoC device */
of_snd_soc_register_device(of_soc);
out:
mutex_unlock(&of_snd_soc_mutex);
return rc;
}
EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
int of_snd_soc_register_platform(struct snd_soc_platform *platform,
struct device_node *node,
struct snd_soc_dai *cpu_dai)
{
struct of_snd_soc_device *of_soc;
struct device_node *codec_node;
const phandle *handle;
int len, rc = 0;
pr_info("registering ASoC platform driver: %s\n", node->full_name);
handle = of_get_property(node, "codec-handle", &len);
if (!handle || len < sizeof(handle))
return -ENODEV;
codec_node = of_find_node_by_phandle(*handle);
if (!codec_node)
return -ENODEV;
pr_info("looking for codec: %s\n", codec_node->full_name);
mutex_lock(&of_snd_soc_mutex);
of_soc = of_snd_soc_get_device(codec_node);
if (!of_soc) {
rc = -ENOMEM;
goto out;
}
of_soc->platform_node = node;
of_soc->dai_link.cpu_dai = cpu_dai;
of_soc->card.platform = platform;
of_soc->card.name = of_soc->dai_link.cpu_dai->name;
/* Now try to register the SoC device */
of_snd_soc_register_device(of_soc);
out:
mutex_unlock(&of_snd_soc_mutex);
return rc;
}
EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);