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ASoC: qdsp6: q6asm: Add q6asm dai driver

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This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Srinivas Kandagatla 2018-05-18 13:56:08 +01:00 committed by Mark Brown
parent 24c4cbcfac
commit 2a9e92d371
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0
3 changed files with 629 additions and 0 deletions
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4
sound/soc/qcom/Kconfig
View File

@ -62,6 +62,9 @@ config SND_SOC_QDSP6_ROUTING
config SND_SOC_QDSP6_ASM
tristate
config SND_SOC_QDSP6_ASM_DAI
tristate
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
@ -72,6 +75,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_ADM
select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
select SND_SOC_QDSP6_ASM_DAI
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific

1
sound/soc/qcom/qdsp6/Makefile
View File

@ -5,3 +5,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM_DAI) += q6asm-dai.o

624
sound/soc/qcom/qdsp6/q6asm-dai.c Normal file
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@ -0,0 +1,624 @@
// SPDX-License-Identifier: GPL-2.0
// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
// Copyright (c) 2018, Linaro Limited
#include <linux/init.h>
#include <linux/err.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/component.h>
#include <sound/soc.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/of_device.h>
#include <sound/pcm_params.h>
#include "q6asm.h"
#include "q6routing.h"
#include "q6dsp-errno.h"
#define DRV_NAME "q6asm-fe-dai"
#define PLAYBACK_MIN_NUM_PERIODS 2
#define PLAYBACK_MAX_NUM_PERIODS 8
#define PLAYBACK_MAX_PERIOD_SIZE 65536
#define PLAYBACK_MIN_PERIOD_SIZE 128
#define CAPTURE_MIN_NUM_PERIODS 2
#define CAPTURE_MAX_NUM_PERIODS 8
#define CAPTURE_MAX_PERIOD_SIZE 4096
#define CAPTURE_MIN_PERIOD_SIZE 320
#define SID_MASK_DEFAULT 0xF
enum stream_state {
Q6ASM_STREAM_IDLE = 0,
Q6ASM_STREAM_STOPPED,
Q6ASM_STREAM_RUNNING,
};
struct q6asm_dai_rtd {
struct snd_pcm_substream *substream;
phys_addr_t phys;
unsigned int pcm_size;
unsigned int pcm_count;
unsigned int pcm_irq_pos; /* IRQ position */
unsigned int periods;
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
uint16_t session_id;
enum stream_state state;
};
struct q6asm_dai_data {
long long int sid;
};
static struct snd_pcm_hardware q6asm_dai_hardware_capture = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE),
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 4,
.buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS *
CAPTURE_MAX_PERIOD_SIZE,
.period_bytes_min = CAPTURE_MIN_PERIOD_SIZE,
.period_bytes_max = CAPTURE_MAX_PERIOD_SIZE,
.periods_min = CAPTURE_MIN_NUM_PERIODS,
.periods_max = CAPTURE_MAX_NUM_PERIODS,
.fifo_size = 0,
};
static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE),
.rates = SNDRV_PCM_RATE_8000_192000,
.rate_min = 8000,
.rate_max = 192000,
.channels_min = 1,
.channels_max = 8,
.buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
PLAYBACK_MAX_PERIOD_SIZE),
.period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
.period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
.periods_min = PLAYBACK_MIN_NUM_PERIODS,
.periods_max = PLAYBACK_MAX_NUM_PERIODS,
.fifo_size = 0,
};
#define Q6ASM_FEDAI_DRIVER(num) { \
.playback = { \
.stream_name = "MultiMedia"#num" Playback", \
.rates = (SNDRV_PCM_RATE_8000_192000| \
SNDRV_PCM_RATE_KNOT), \
.formats = (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE), \
.channels_min = 1, \
.channels_max = 8, \
.rate_min = 8000, \
.rate_max = 192000, \
}, \
.capture = { \
.stream_name = "MultiMedia"#num" Capture", \
.rates = (SNDRV_PCM_RATE_8000_48000| \
SNDRV_PCM_RATE_KNOT), \
.formats = (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE), \
.channels_min = 1, \
.channels_max = 4, \
.rate_min = 8000, \
.rate_max = 48000, \
}, \
.name = "MultiMedia"#num, \
.probe = fe_dai_probe, \
.id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
}
/* Conventional and unconventional sample rate supported */
static unsigned int supported_sample_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
88200, 96000, 176400, 192000
};
static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
.count = ARRAY_SIZE(supported_sample_rates),
.list = supported_sample_rates,
.mask = 0,
};
static void event_handler(uint32_t opcode, uint32_t token,
uint32_t *payload, void *priv)
{
struct q6asm_dai_rtd *prtd = priv;
struct snd_pcm_substream *substream = prtd->substream;
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
q6asm_write_async(prtd->audio_client,
prtd->pcm_count, 0, 0, NO_TIMESTAMP);
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
prtd->state = Q6ASM_STREAM_STOPPED;
break;
case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
q6asm_write_async(prtd->audio_client,
prtd->pcm_count, 0, 0, NO_TIMESTAMP);
break;
}
case ASM_CLIENT_EVENT_DATA_READ_DONE:
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
q6asm_read(prtd->audio_client);
break;
default:
break;
}
}
static int q6asm_dai_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct q6asm_dai_rtd *prtd = runtime->private_data;
struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
struct q6asm_dai_data *pdata;
int ret, i;
pdata = snd_soc_component_get_drvdata(c);
if (!pdata)
return -EINVAL;
if (!prtd || !prtd->audio_client) {
pr_err("%s: private data null or audio client freed\n",
__func__);
return -EINVAL;
}
prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
prtd->pcm_irq_pos = 0;
/* rate and channels are sent to audio driver */
if (prtd->state) {
/* clear the previous setup if any */
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6routing_stream_close(soc_prtd->dai_link->id,
substream->stream);
}
ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
prtd->phys,
(prtd->pcm_size / prtd->periods),
prtd->periods);
if (ret < 0) {
pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
ret);
return -ENOMEM;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
prtd->bits_per_sample);
}
if (ret < 0) {
pr_err("%s: q6asm_open_write failed\n", __func__);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
return -ENOMEM;
}
prtd->session_id = q6asm_get_session_id(prtd->audio_client);
ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
prtd->session_id, substream->stream);
if (ret) {
pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
return ret;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_media_format_block_multi_ch_pcm(
prtd->audio_client, runtime->rate,
runtime->channels, NULL,
prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
runtime->rate, runtime->channels,
prtd->bits_per_sample);
/* Queue the buffers */
for (i = 0; i < runtime->periods; i++)
q6asm_read(prtd->audio_client);
}
if (ret < 0)
pr_info("%s: CMD Format block failed\n", __func__);
prtd->state = Q6ASM_STREAM_RUNNING;
return 0;
}
static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int q6asm_dai_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
struct q6asm_dai_rtd *prtd;
struct q6asm_dai_data *pdata;
struct device *dev = c->dev;
int ret = 0;
int stream_id;
stream_id = cpu_dai->driver->id;
pdata = snd_soc_component_get_drvdata(c);
if (!pdata) {
pr_err("Drv data not found ..\n");
return -EINVAL;
}
prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
if (prtd == NULL)
return -ENOMEM;
prtd->substream = substream;
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)event_handler, prtd, stream_id,
LEGACY_PCM_MODE);
if (!prtd->audio_client) {
pr_info("%s: Could not allocate memory\n", __func__);
kfree(prtd);
return -ENOMEM;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = q6asm_dai_hardware_playback;
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
runtime->hw = q6asm_dai_hardware_capture;
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_sample_rates);
if (ret < 0)
pr_info("snd_pcm_hw_constraint_list failed\n");
/* Ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
pr_info("snd_pcm_hw_constraint_integer failed\n");
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
if (ret < 0) {
pr_err("constraint for buffer bytes min max ret = %d\n",
ret);
}
}
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
if (ret < 0) {
pr_err("constraint for period bytes step ret = %d\n",
ret);
}
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
if (ret < 0) {
pr_err("constraint for buffer bytes step ret = %d\n",
ret);
}
runtime->private_data = prtd;
snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
if (pdata->sid < 0)
prtd->phys = substream->dma_buffer.addr;
else
prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
return 0;
}
static int q6asm_dai_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct q6asm_dai_rtd *prtd = runtime->private_data;
if (prtd->audio_client) {
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
}
q6routing_stream_close(soc_prtd->dai_link->id,
substream->stream);
kfree(prtd);
return 0;
}
static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
if (prtd->pcm_irq_pos >= prtd->pcm_size)
prtd->pcm_irq_pos = 0;
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
}
static int q6asm_dai_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
struct device *dev = c->dev;
return dma_mmap_coherent(dev, vma,
runtime->dma_area, runtime->dma_addr,
runtime->dma_bytes);
}
static int q6asm_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
prtd->pcm_size = params_buffer_bytes(params);
prtd->periods = params_periods(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
prtd->bits_per_sample = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
prtd->bits_per_sample = 24;
break;
}
return 0;
}
static struct snd_pcm_ops q6asm_dai_ops = {
.open = q6asm_dai_open,
.hw_params = q6asm_dai_hw_params,
.close = q6asm_dai_close,
.ioctl = snd_pcm_lib_ioctl,
.prepare = q6asm_dai_prepare,
.trigger = q6asm_dai_trigger,
.pointer = q6asm_dai_pointer,
.mmap = q6asm_dai_mmap,
};
static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm_substream *psubstream, *csubstream;
struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct snd_pcm *pcm = rtd->pcm;
struct device *dev;
int size, ret;
dev = c->dev;
size = q6asm_dai_hardware_playback.buffer_bytes_max;
psubstream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
if (psubstream) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
&psubstream->dma_buffer);
if (ret) {
dev_err(dev, "Cannot allocate buffer(s)\n");
return ret;
}
}
csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
if (csubstream) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
&csubstream->dma_buffer);
if (ret) {
dev_err(dev, "Cannot allocate buffer(s)\n");
if (psubstream)
snd_dma_free_pages(&psubstream->dma_buffer);
return ret;
}
}
return ret;
}
static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
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 const struct snd_soc_dapm_route afe_pcm_routes[] = {
{"MM_DL1", NULL, "MultiMedia1 Playback" },
{"MM_DL2", NULL, "MultiMedia2 Playback" },
{"MM_DL3", NULL, "MultiMedia3 Playback" },
{"MM_DL4", NULL, "MultiMedia4 Playback" },
{"MM_DL5", NULL, "MultiMedia5 Playback" },
{"MM_DL6", NULL, "MultiMedia6 Playback" },
{"MM_DL7", NULL, "MultiMedia7 Playback" },
{"MM_DL7", NULL, "MultiMedia8 Playback" },
{"MultiMedia1 Capture", NULL, "MM_UL1"},
{"MultiMedia2 Capture", NULL, "MM_UL2"},
{"MultiMedia3 Capture", NULL, "MM_UL3"},
{"MultiMedia4 Capture", NULL, "MM_UL4"},
{"MultiMedia5 Capture", NULL, "MM_UL5"},
{"MultiMedia6 Capture", NULL, "MM_UL6"},
{"MultiMedia7 Capture", NULL, "MM_UL7"},
{"MultiMedia8 Capture", NULL, "MM_UL8"},
};
static int fe_dai_probe(struct snd_soc_dai *dai)
{
struct snd_soc_dapm_context *dapm;
dapm = snd_soc_component_get_dapm(dai->component);
snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
ARRAY_SIZE(afe_pcm_routes));
return 0;
}
static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.name = DRV_NAME,
.ops = &q6asm_dai_ops,
.pcm_new = q6asm_dai_pcm_new,
.pcm_free = q6asm_dai_pcm_free,
};
static struct snd_soc_dai_driver q6asm_fe_dais[] = {
Q6ASM_FEDAI_DRIVER(1),
Q6ASM_FEDAI_DRIVER(2),
Q6ASM_FEDAI_DRIVER(3),
Q6ASM_FEDAI_DRIVER(4),
Q6ASM_FEDAI_DRIVER(5),
Q6ASM_FEDAI_DRIVER(6),
Q6ASM_FEDAI_DRIVER(7),
Q6ASM_FEDAI_DRIVER(8),
};
static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
{
struct device_node *node = dev->of_node;
struct of_phandle_args args;
struct q6asm_dai_data *pdata;
int rc;
pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL);
if (!pdata)
return -ENOMEM;
rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
if (rc < 0)
pdata->sid = -1;
else
pdata->sid = args.args[0] & SID_MASK_DEFAULT;
dev_set_drvdata(dev, pdata);
return snd_soc_register_component(dev, &q6asm_fe_dai_component,
q6asm_fe_dais,
ARRAY_SIZE(q6asm_fe_dais));
}
static void q6asm_dai_unbind(struct device *dev, struct device *master,
void *data)
{
struct q6asm_dai_data *pdata = dev_get_drvdata(dev);
snd_soc_unregister_component(dev);
kfree(pdata);
}
static const struct component_ops q6asm_dai_comp_ops = {
.bind = q6asm_dai_bind,
.unbind = q6asm_dai_unbind,
};
static int q6asm_dai_probe(struct platform_device *pdev)
{
return component_add(&pdev->dev, &q6asm_dai_comp_ops);
}
static int q6asm_dai_dev_remove(struct platform_device *pdev)
{
component_del(&pdev->dev, &q6asm_dai_comp_ops);
return 0;
}
static struct platform_driver q6asm_dai_platform_driver = {
.driver = {
.name = "q6asm-dai",
},
.probe = q6asm_dai_probe,
.remove = q6asm_dai_dev_remove,
};
module_platform_driver(q6asm_dai_platform_driver);
MODULE_DESCRIPTION("Q6ASM dai driver");
MODULE_LICENSE("GPL v2");