[RESEND,v2,12/15] ASoC: qcom: qdsp6: Add support to q6asm dai driver

Message ID 20171214173402.19074-13-srinivas.kandagatla@linaro.org
State New
Headers show
Series
  • Untitled series #7201
Related show

Commit Message

Srinivas Kandagatla Dec. 14, 2017, 5:33 p.m.
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>


This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.
Currently the driver only exposes 2 playback streams for hdmi playback
support, it can be easily extended to add all 8 streams.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

---
 sound/soc/qcom/Kconfig           |   6 +
 sound/soc/qcom/qdsp6/Makefile    |   1 +
 sound/soc/qcom/qdsp6/q6asm-dai.c | 534 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 541 insertions(+)
 create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c

-- 
2.15.0

Comments

Bjorn Andersson Jan. 3, 2018, 12:03 a.m. | #1
On Thu 14 Dec 09:33 PST 2017, srinivas.kandagatla@linaro.org wrote:

[..]
> +

> +enum stream_state {

> +	IDLE = 0,

> +	STOPPED,

> +	RUNNING,


These are too generic.

> +};

> +

> +struct q6asm_dai_rtd {

> +	struct snd_pcm_substream *substream;

> +	dma_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;

> +	bool set_channel_map;

> +	char channel_map[8];


There's a define for this 8

> +};

> +

> +struct q6asm_dai_data {

> +	u64 sid;

> +};

> +

> +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,


If you just put the numbers here, instead of using the PLAYBACK_
defines, it's possible to grok the values of this struct without having
to jump to the defines for each one.

> +	.fifo_size =            0,

> +};

> +

> +/* 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 = {


This is unreferenced.

> +	.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:

> +		q6asm_write_nolock(prtd->audio_client,

> +				   prtd->pcm_count, 0, 0, NO_TIMESTAMP);

> +		break;

> +	case ASM_CLIENT_EVENT_CMD_EOS_DONE:

> +		prtd->state = STOPPED;

> +		break;

> +	case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {

> +		prtd->pcm_irq_pos += prtd->pcm_count;

> +		snd_pcm_period_elapsed(substream);

> +		if (prtd->state == RUNNING)

> +			q6asm_write_nolock(prtd->audio_client,

> +					   prtd->pcm_count, 0, 0, NO_TIMESTAMP);

> +

> +		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 q6asm_dai_data *pdata;

> +	int ret;

> +

> +	pdata = dev_get_drvdata(soc_prtd->platform->dev);

> +	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_dereg_phy_stream(soc_prtd->dai_link->id,

> +					 SNDRV_PCM_STREAM_PLAYBACK);

> +	}

> +

> +	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;

> +	}

> +

> +	ret = q6asm_open_write(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;


Do you need to roll back the q6asm_map_memory_regions?

> +		return -ENOMEM;

> +	}

> +

> +	prtd->session_id = q6asm_get_session_id(prtd->audio_client);

> +	ret = q6routing_reg_phy_stream(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;

> +	}

> +

> +	ret = q6asm_media_format_block_multi_ch_pcm(

> +			prtd->audio_client, runtime->rate,

> +			runtime->channels, !prtd->set_channel_map,

> +			prtd->channel_map, prtd->bits_per_sample);


set_channel_map and channel_map aren't referenced elsewhere. If this
isn't used consider removing it for now.

> +	if (ret < 0)

> +		pr_info("%s: CMD Format block failed\n", __func__);

> +

> +	prtd->state = RUNNING;

> +

> +	return 0;

> +}

> +

[..]
> +static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)

> +{

> +	struct snd_pcm *pcm = rtd->pcm;

> +	struct snd_pcm_substream *substream;

> +	struct snd_card *card = rtd->card->snd_card;

> +	struct device *dev = card->dev;

> +	struct device_node *node = dev->of_node;

> +	struct q6asm_dai_data *pdata = dev_get_drvdata(rtd->platform->dev);

> +	struct of_phandle_args args;

> +

> +	int size, ret = 0;

> +

> +	ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);

> +	if (ret < 0)

> +		pdata->sid = -1;

> +	else

> +		pdata->sid = args.args[0];

> +


Is this really how you're supposed to deal with the iommu?

> +

> +

> +	substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;

> +	size = q6asm_dai_hardware_playback.buffer_bytes_max;

> +	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,

> +				  &substream->dma_buffer);

> +	if (ret) {

> +		dev_err(dev, "Cannot allocate buffer(s)\n");

> +		return ret;


Just fall through.

> +	}

> +

> +	return ret;

> +}

> +

[..]
> +static struct snd_soc_dai_driver q6asm_fe_dais[] = {

> +	{

> +		.playback = {

> +			.stream_name = "MultiMedia1 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,

> +		},

> +		.name = "MM_DL1",

> +		.probe = fe_dai_probe,

> +		.id = MSM_FRONTEND_DAI_MULTIMEDIA1,

> +	},

> +	{

> +		.playback = {

> +			.stream_name = "MultiMedia2 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,


I presume the listed frontend DAIs needs to match the firmware of the
DSP (and features of hardware)? Can we get away with a single list for
all versions of the adsp?

In msm-4.4 the max rate for these where changed to 384000, see:

9c46f74b2724 ("ASoC: msm: add 384KHz playback support")

> +		},

> +		.name = "MM_DL2",

> +		.probe = fe_dai_probe,

> +		.id = MSM_FRONTEND_DAI_MULTIMEDIA2,

> +	},

> +};

> +


Regards,
Bjorn
Srinivas Kandagatla Jan. 3, 2018, 4:27 p.m. | #2
Thanks for the comments.

On 03/01/18 00:03, Bjorn Andersson wrote:
> On Thu 14 Dec 09:33 PST 2017, srinivas.kandagatla@linaro.org wrote:

> 

> [..]

>> +

>> +enum stream_state {

>> +	IDLE = 0,

>> +	STOPPED,

>> +	RUNNING,

> 

> These are too generic.

> 

Yep, I will prefix them with Q6ASM.

>> +};

>> +

>> +struct q6asm_dai_rtd {

>> +	struct snd_pcm_substream *substream;

>> +	dma_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;

>> +	bool set_channel_map;

>> +	char channel_map[8];

> 

> There's a define for this 8


Yes, this is max channels.

> 

>> +};

>> +

>> +struct q6asm_dai_data {

>> +	u64 sid;

>> +};

>> +

>> +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,

> 

> If you just put the numbers here, instead of using the PLAYBACK_

> defines, it's possible to grok the values of this struct without having

> to jump to the defines for each one.


This is usually done this way in may other drivers!,

> 

>> +	.fifo_size =            0,

>> +};

>> +

>> +/* 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 = {

> 


It is used in q6asm_dai_open().
> 


>> +	.count = ARRAY_SIZE(supported_sample_rates),

>> +	.list = supported_sample_rates,

>> +	.mask = 0,

>> +};

>> +

>

>> +

>> +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 q6asm_dai_data *pdata;

>> +	int ret;

>> +

>> +	pdata = dev_get_drvdata(soc_prtd->platform->dev);

>> +	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_dereg_phy_stream(soc_prtd->dai_link->id,

>> +					 SNDRV_PCM_STREAM_PLAYBACK);

>> +	}

>> +

>> +	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;

>> +	}

>> +

>> +	ret = q6asm_open_write(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;

> 

> Do you need to roll back the q6asm_map_memory_regions?

> 

yes you are correct, we should roll back the map.

>> +		return -ENOMEM;

>> +	}

>> +

>> +	prtd->session_id = q6asm_get_session_id(prtd->audio_client);

>> +	ret = q6routing_reg_phy_stream(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;

>> +	}

>> +

>> +	ret = q6asm_media_format_block_multi_ch_pcm(

>> +			prtd->audio_client, runtime->rate,

>> +			runtime->channels, !prtd->set_channel_map,

>> +			prtd->channel_map, prtd->bits_per_sample);

> 

> set_channel_map and channel_map aren't referenced elsewhere. If this

> isn't used consider removing it for now.

>

Will take a closer look before sending next version.

>> +	if (ret < 0)

>> +		pr_info("%s: CMD Format block failed\n", __func__);

>> +

>> +	prtd->state = RUNNING;

>> +

>> +	return 0;

>> +}

>> +

> [..]

>> +static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)

>> +{

>> +	struct snd_pcm *pcm = rtd->pcm;

>> +	struct snd_pcm_substream *substream;

>> +	struct snd_card *card = rtd->card->snd_card;

>> +	struct device *dev = card->dev;

>> +	struct device_node *node = dev->of_node;

>> +	struct q6asm_dai_data *pdata = dev_get_drvdata(rtd->platform->dev);

>> +	struct of_phandle_args args;

>> +

>> +	int size, ret = 0;

>> +

>> +	ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);

>> +	if (ret < 0)

>> +		pdata->sid = -1;

>> +	else

>> +		pdata->sid = args.args[0];

>> +

> 

> Is this really how you're supposed to deal with the iommu?

> 

Any suggestions are welcome, I did not find a better way to append sid 
to iova address from iommu.

Currently downstream abstracts this in ion apis.

>> +

>> +

>> +	substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;

>> +	size = q6asm_dai_hardware_playback.buffer_bytes_max;

>> +	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,

>> +				  &substream->dma_buffer);

>> +	if (ret) {

>> +		dev_err(dev, "Cannot allocate buffer(s)\n");

>> +		return ret;

> 

> Just fall through.

> 

yep

>> +	}

>> +

>> +	return ret;

>> +}

>> +

> [..]

>> +static struct snd_soc_dai_driver q6asm_fe_dais[] = {

>> +	{

>> +		.playback = {

>> +			.stream_name = "MultiMedia1 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,

>> +		},

>> +		.name = "MM_DL1",

>> +		.probe = fe_dai_probe,

>> +		.id = MSM_FRONTEND_DAI_MULTIMEDIA1,

>> +	},

>> +	{

>> +		.playback = {

>> +			.stream_name = "MultiMedia2 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,

> 

> I presume the listed frontend DAIs needs to match the firmware of the

> DSP (and features of hardware)? Can we get away with a single list for

> all versions of the adsp?

> 

Yes, DSP supports 8 concurrent streams both playback and record streams.

For now I have only added two entires to keep the patch simple but this 
should be ideally 8 entries.

> In msm-4.4 the max rate for these where changed to 384000, see:

> 

> 9c46f74b2724 ("ASoC: msm: add 384KHz playback support")

sure i will include that in next version.
> 

>> +		},

>> +		.name = "MM_DL2",

>> +		.probe = fe_dai_probe,

>> +		.id = MSM_FRONTEND_DAI_MULTIMEDIA2,

>> +	},

>> +};

>> +

> 

> Regards,

> Bjorn

>

Patch

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 003ce182691c..ecd1e4ba834d 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -68,6 +68,11 @@  config SND_SOC_QDSP6_AFE_DAI
 	tristate
 	default n
 
+config SND_SOC_QDSP6_ASM_DAI
+	tristate
+	default n
+
+
 config SND_SOC_QDSP6
 	tristate "SoC ALSA audio driver for QDSP6"
 	select SND_SOC_QDSP6_AFE
@@ -76,6 +81,7 @@  config SND_SOC_QDSP6
 	select SND_SOC_QDSP6_CORE
 	select SND_SOC_QDSP6_ROUTING
 	select SND_SOC_QDSP6_AFE_DAI
+	select SND_SOC_QDSP6_ASM_DAI
 	help
 	 To add support for MSM QDSP6 Soc Audio.
 	 This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index bd8bd02bf09e..03576a442fb5 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -4,3 +4,4 @@  obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
 obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
 obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
 obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
+obj-$(CONFIG_SND_SOC_QDSP6_ASM_DAI) += q6asm-dai.o
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000000..709c5de230fa
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,534 @@ 
+/* SPDX-License-Identifier: GPL-2.0
+* Copyright (c) 2011-2016, The Linux Foundation
+* Copyright (c) 2017, Linaro Limited
+*/
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.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 "common.h"
+
+#define PLAYBACK_MIN_NUM_PERIODS    2
+#define PLAYBACK_MAX_NUM_PERIODS   8
+#define PLAYBACK_MAX_PERIOD_SIZE    65536
+#define PLAYBACK_MIN_PERIOD_SIZE    128
+
+enum stream_state {
+	IDLE = 0,
+	STOPPED,
+	RUNNING,
+};
+
+struct q6asm_dai_rtd {
+	struct snd_pcm_substream *substream;
+	dma_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;
+	bool set_channel_map;
+	char channel_map[8];
+};
+
+struct q6asm_dai_data {
+	u64 sid;
+};
+
+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,
+};
+
+/* 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:
+		q6asm_write_nolock(prtd->audio_client,
+				   prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+		break;
+	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+		prtd->state = STOPPED;
+		break;
+	case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
+		prtd->pcm_irq_pos += prtd->pcm_count;
+		snd_pcm_period_elapsed(substream);
+		if (prtd->state == RUNNING)
+			q6asm_write_nolock(prtd->audio_client,
+					   prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+
+		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 q6asm_dai_data *pdata;
+	int ret;
+
+	pdata = dev_get_drvdata(soc_prtd->platform->dev);
+	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_dereg_phy_stream(soc_prtd->dai_link->id,
+					 SNDRV_PCM_STREAM_PLAYBACK);
+	}
+
+	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;
+	}
+
+	ret = q6asm_open_write(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_reg_phy_stream(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;
+	}
+
+	ret = q6asm_media_format_block_multi_ch_pcm(
+			prtd->audio_client, runtime->rate,
+			runtime->channels, !prtd->set_channel_map,
+			prtd->channel_map, prtd->bits_per_sample);
+	if (ret < 0)
+		pr_info("%s: CMD Format block failed\n", __func__);
+
+	prtd->state = 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:
+		ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+		break;
+	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 = 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 q6asm_dai_rtd *prtd;
+	struct q6asm_dai_data *pdata;
+	struct device *dev = soc_prtd->platform->dev;
+	int ret = 0;
+
+	pdata = dev_get_drvdata(dev);
+	if (!pdata) {
+		pr_err("Platform 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,
+				(app_cb)event_handler, prtd);
+	if (!prtd->audio_client) {
+		pr_info("%s: Could not allocate memory\n", __func__);
+		kfree(prtd);
+		return -ENOMEM;
+	}
+
+//	prtd->audio_client->dev = dev;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		runtime->hw = q6asm_dai_hardware_playback;
+
+	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);
+	}
+
+	prtd->set_channel_map = false;
+	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);
+	}
+	q6routing_dereg_phy_stream(soc_prtd->dai_link->id,
+						SNDRV_PCM_STREAM_PLAYBACK);
+	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_card *card = soc_prtd->card->snd_card;
+
+	return dma_mmap_coherent(card->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 *pcm = rtd->pcm;
+	struct snd_pcm_substream *substream;
+	struct snd_card *card = rtd->card->snd_card;
+	struct device *dev = card->dev;
+	struct device_node *node = dev->of_node;
+	struct q6asm_dai_data *pdata = dev_get_drvdata(rtd->platform->dev);
+	struct of_phandle_args args;
+
+	int size, ret = 0;
+
+	ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+	if (ret < 0)
+		pdata->sid = -1;
+	else
+		pdata->sid = args.args[0];
+
+
+
+	substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	size = q6asm_dai_hardware_playback.buffer_bytes_max;
+	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+				  &substream->dma_buffer);
+	if (ret) {
+		dev_err(dev, "Cannot allocate buffer(s)\n");
+		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 struct snd_soc_platform_driver q6asm_soc_platform = {
+	.ops		= &q6asm_dai_ops,
+	.pcm_new	= q6asm_dai_pcm_new,
+	.pcm_free	= q6asm_dai_pcm_free,
+
+};
+
+static const struct snd_soc_dapm_route afe_pcm_routes[] = {
+	{"MM_DL1",  NULL, "MultiMedia1 Playback" },
+	{"MM_DL2",  NULL, "MultiMedia2 Playback" },
+
+};
+
+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		= "q6asm-fe-dai",
+};
+
+static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+	{
+		.playback = {
+			.stream_name = "MultiMedia1 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,
+		},
+		.name = "MM_DL1",
+		.probe = fe_dai_probe,
+		.id = MSM_FRONTEND_DAI_MULTIMEDIA1,
+	},
+	{
+		.playback = {
+			.stream_name = "MultiMedia2 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,
+		},
+		.name = "MM_DL2",
+		.probe = fe_dai_probe,
+		.id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+	},
+};
+
+static int q6asm_dai_probe(struct platform_device *pdev)
+{
+	struct q6asm_dai_data *pdata;
+	struct device *dev = &pdev->dev;
+	int rc;
+
+	pdata = devm_kzalloc(dev, sizeof(struct q6asm_dai_data), GFP_KERNEL);
+	if (!pdata)
+		return -ENOMEM;
+
+
+	dev_set_drvdata(dev, pdata);
+
+	rc = snd_soc_register_platform(dev,  &q6asm_soc_platform);
+	if (rc) {
+		dev_err(&pdev->dev, "err_dai_platform\n");
+		return rc;
+	}
+
+	rc = snd_soc_register_component(dev, &q6asm_fe_dai_component,
+					q6asm_fe_dais,
+					ARRAY_SIZE(q6asm_fe_dais));
+	if (rc)
+		dev_err(dev, "err_dai_component\n");
+
+	return rc;
+
+}
+
+static int q6asm_dai_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver q6asm_dai_driver = {
+	.driver = {
+		.name = "q6asm_dai",
+		.owner = THIS_MODULE,
+	},
+	.probe = q6asm_dai_probe,
+	.remove = q6asm_dai_remove,
+};
+
+module_platform_driver(q6asm_dai_driver);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");