text: // 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 <sound/soc.h>
  #include <sound/soc-dapm.h>
  #include <sound/pcm.h>
  #include <linux/spinlock.h>
  #include <sound/compress_driver.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
  
  /* Default values used if user space does not set */
  #define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
  #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
  #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
  #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
  
  #define ALAC_CH_LAYOUT_MONO   ((101 << 16) | 1)
  #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
  
  enum stream_state {
  	Q6ASM_STREAM_IDLE = 0,
  	Q6ASM_STREAM_STOPPED,
  	Q6ASM_STREAM_RUNNING,
  };
  
  struct q6asm_dai_rtd {
  	struct snd_pcm_substream *substream;
  	struct snd_compr_stream *cstream;
  	struct snd_codec codec;
  	struct snd_dma_buffer dma_buffer;
  	spinlock_t lock;
  	phys_addr_t phys;
  	unsigned int pcm_size;
  	unsigned int pcm_count;
  	unsigned int pcm_irq_pos;       /* IRQ position */
  	unsigned int periods;
  	unsigned int bytes_sent;
  	unsigned int bytes_received;
  	unsigned int copied_total;
  	uint16_t bits_per_sample;
  	uint16_t source; /* Encoding source bit mask */
  	struct audio_client *audio_client;
  	uint32_t next_track_stream_id;
  	bool next_track;
  	uint32_t stream_id;
  	uint16_t session_id;
  	enum stream_state state;
  	uint32_t initial_samples_drop;
  	uint32_t trailing_samples_drop;
  	bool notify_on_drain;
  };
  
  struct q6asm_dai_data {
  	struct snd_soc_dai_driver *dais;
  	int num_dais;
  	long long int sid;
  };
  
  static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
  	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
  				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_BATCH |
  				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,				\
  		.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 const struct snd_compr_codec_caps q6asm_compr_caps = {
  	.num_descriptors = 1,
  	.descriptor[0].max_ch = 2,
  	.descriptor[0].sample_rates = {	8000, 11025, 12000, 16000, 22050,
  					24000, 32000, 44100, 48000, 88200,
  					96000, 176400, 192000 },
  	.descriptor[0].num_sample_rates = 13,
  	.descriptor[0].bit_rate[0] = 320,
  	.descriptor[0].bit_rate[1] = 128,
  	.descriptor[0].num_bitrates = 2,
  	.descriptor[0].profiles = 0,
  	.descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
  	.descriptor[0].formats = 0,
  };
  
  static void event_handler(uint32_t opcode, uint32_t token,
  			  void *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->stream_id,
  				   prtd->pcm_count, 0, 0, 0);
  		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->stream_id,
  					   prtd->pcm_count, 0, 0, 0);
  
  		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, prtd->stream_id);
  
  		break;
  	default:
  		break;
  	}
  }
  
  static int q6asm_dai_prepare(struct snd_soc_component *component,
  			     struct snd_pcm_substream *substream)
  {
  	struct snd_pcm_runtime *runtime = substream->runtime;
  	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	struct q6asm_dai_data *pdata;
  	struct device *dev = component->dev;
  	int ret, i;
  
  	pdata = snd_soc_component_get_drvdata(component);
  	if (!pdata)
  		return -EINVAL;
  
  	if (!prtd || !prtd->audio_client) {
  		dev_err(dev, "%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, prtd->stream_id, 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) {
  		dev_err(dev, "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, prtd->stream_id,
  				       FORMAT_LINEAR_PCM,
  				       0, prtd->bits_per_sample, false);
  	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
  		ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
  				      FORMAT_LINEAR_PCM,
  				      prtd->bits_per_sample);
  	}
  
  	if (ret < 0) {
  		dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
  		goto open_err;
  	}
  
  	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) {
  		dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
  		goto routing_err;
  	}
  
  	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
  		ret = q6asm_media_format_block_multi_ch_pcm(
  				prtd->audio_client, prtd->stream_id,
  				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,
  							   prtd->stream_id,
  							   runtime->rate,
  							   runtime->channels,
  							   prtd->bits_per_sample);
  
  		/* Queue the buffers */
  		for (i = 0; i < runtime->periods; i++)
  			q6asm_read(prtd->audio_client, prtd->stream_id);
  
  	}
  	if (ret < 0)
  		dev_info(dev, "%s: CMD Format block failed\n", __func__);
  	else
  		prtd->state = Q6ASM_STREAM_RUNNING;
  
  	return ret;
  
  routing_err:
  	q6asm_cmd(prtd->audio_client, prtd->stream_id,  CMD_CLOSE);
  open_err:
  	q6asm_unmap_memory_regions(substream->stream, prtd->audio_client);
  	q6asm_audio_client_free(prtd->audio_client);
  	prtd->audio_client = NULL;
  
  	return ret;
  }
  
  static int q6asm_dai_trigger(struct snd_soc_component *component,
  			     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, prtd->stream_id,
  				       0, 0, 0);
  		break;
  	case SNDRV_PCM_TRIGGER_STOP:
  		prtd->state = Q6ASM_STREAM_STOPPED;
  		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
  				       CMD_EOS);
  		break;
  	case SNDRV_PCM_TRIGGER_SUSPEND:
  	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
  		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
  				       CMD_PAUSE);
  		break;
  	default:
  		ret = -EINVAL;
  		break;
  	}
  
  	return ret;
  }
  
  static int q6asm_dai_open(struct snd_soc_component *component,
  			  struct snd_pcm_substream *substream)
  {
  	struct snd_pcm_runtime *runtime = substream->runtime;
  	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
  	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
  	struct q6asm_dai_rtd *prtd;
  	struct q6asm_dai_data *pdata;
  	struct device *dev = component->dev;
  	int ret = 0;
  	int stream_id;
  
  	stream_id = cpu_dai->driver->id;
  
  	pdata = snd_soc_component_get_drvdata(component);
  	if (!pdata) {
  		dev_err(dev, "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 (IS_ERR(prtd->audio_client)) {
  		dev_info(dev, "%s: Could not allocate memory\n", __func__);
  		ret = PTR_ERR(prtd->audio_client);
  		kfree(prtd);
  		return ret;
  	}
  
  	/* DSP expects stream id from 1 */
  	prtd->stream_id = 1;
  
  	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)
  		dev_info(dev, "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)
  		dev_info(dev, "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) {
  			dev_err(dev, "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) {
  		dev_err(dev, "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) {
  		dev_err(dev, "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);
  
  	return 0;
  }
  
  static int q6asm_dai_close(struct snd_soc_component *component,
  			   struct snd_pcm_substream *substream)
  {
  	struct snd_pcm_runtime *runtime = substream->runtime;
  	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  
  	if (prtd->audio_client) {
  		if (prtd->state)
  			q6asm_cmd(prtd->audio_client, prtd->stream_id,
  				  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_soc_component *component,
  					   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_hw_params(struct snd_soc_component *component,
  			       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 void compress_event_handler(uint32_t opcode, uint32_t token,
  				   void *payload, void *priv)
  {
  	struct q6asm_dai_rtd *prtd = priv;
  	struct snd_compr_stream *substream = prtd->cstream;
  	unsigned long flags;
  	u32 wflags = 0;
  	uint64_t avail;
  	uint32_t bytes_written, bytes_to_write;
  	bool is_last_buffer = false;
  
  	switch (opcode) {
  	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
  		spin_lock_irqsave(&prtd->lock, flags);
  		if (!prtd->bytes_sent) {
  			q6asm_stream_remove_initial_silence(prtd->audio_client,
  						    prtd->stream_id,
  						    prtd->initial_samples_drop);
  
  			q6asm_write_async(prtd->audio_client, prtd->stream_id,
  					  prtd->pcm_count, 0, 0, 0);
  			prtd->bytes_sent += prtd->pcm_count;
  		}
  
  		spin_unlock_irqrestore(&prtd->lock, flags);
  		break;
  
  	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
  		spin_lock_irqsave(&prtd->lock, flags);
  		if (prtd->notify_on_drain) {
  			if (substream->partial_drain) {
  				/*
  				 * Close old stream and make it stale, switch
  				 * the active stream now!
  				 */
  				q6asm_cmd_nowait(prtd->audio_client,
  						 prtd->stream_id,
  						 CMD_CLOSE);
  				/*
  				 * vaild stream ids start from 1, So we are
  				 * toggling this between 1 and 2.
  				 */
  				prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
  			}
  
  			snd_compr_drain_notify(prtd->cstream);
  			prtd->notify_on_drain = false;
  
  		} else {
  			prtd->state = Q6ASM_STREAM_STOPPED;
  		}
  		spin_unlock_irqrestore(&prtd->lock, flags);
  		break;
  
  	case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
  		spin_lock_irqsave(&prtd->lock, flags);
  
  		bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
  		prtd->copied_total += bytes_written;
  		snd_compr_fragment_elapsed(substream);
  
  		if (prtd->state != Q6ASM_STREAM_RUNNING) {
  			spin_unlock_irqrestore(&prtd->lock, flags);
  			break;
  		}
  
  		avail = prtd->bytes_received - prtd->bytes_sent;
  		if (avail > prtd->pcm_count) {
  			bytes_to_write = prtd->pcm_count;
  		} else {
  			if (substream->partial_drain || prtd->notify_on_drain)
  				is_last_buffer = true;
  			bytes_to_write = avail;
  		}
  
  		if (bytes_to_write) {
  			if (substream->partial_drain && is_last_buffer) {
  				wflags |= ASM_LAST_BUFFER_FLAG;
  				q6asm_stream_remove_trailing_silence(prtd->audio_client,
  						     prtd->stream_id,
  						     prtd->trailing_samples_drop);
  			}
  
  			q6asm_write_async(prtd->audio_client, prtd->stream_id,
  					  bytes_to_write, 0, 0, wflags);
  
  			prtd->bytes_sent += bytes_to_write;
  		}
  
  		if (prtd->notify_on_drain && is_last_buffer)
  			q6asm_cmd_nowait(prtd->audio_client,
  					 prtd->stream_id, CMD_EOS);
  
  		spin_unlock_irqrestore(&prtd->lock, flags);
  		break;
  
  	default:
  		break;
  	}
  }
  
  static int q6asm_dai_compr_open(struct snd_soc_component *component,
  				struct snd_compr_stream *stream)
  {
  	struct snd_soc_pcm_runtime *rtd = stream->private_data;
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
  	struct q6asm_dai_data *pdata;
  	struct device *dev = component->dev;
  	struct q6asm_dai_rtd *prtd;
  	int stream_id, size, ret;
  
  	stream_id = cpu_dai->driver->id;
  	pdata = snd_soc_component_get_drvdata(component);
  	if (!pdata) {
  		dev_err(dev, "Drv data not found ..\n");
  		return -EINVAL;
  	}
  
  	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
  	if (!prtd)
  		return -ENOMEM;
  
  	/* DSP expects stream id from 1 */
  	prtd->stream_id = 1;
  
  	prtd->cstream = stream;
  	prtd->audio_client = q6asm_audio_client_alloc(dev,
  					(q6asm_cb)compress_event_handler,
  					prtd, stream_id, LEGACY_PCM_MODE);
  	if (IS_ERR(prtd->audio_client)) {
  		dev_err(dev, "Could not allocate memory\n");
  		ret = PTR_ERR(prtd->audio_client);
  		goto free_prtd;
  	}
  
  	size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
  			COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
  	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
  				  &prtd->dma_buffer);
  	if (ret) {
  		dev_err(dev, "Cannot allocate buffer(s)\n");
  		goto free_client;
  	}
  
  	if (pdata->sid < 0)
  		prtd->phys = prtd->dma_buffer.addr;
  	else
  		prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
  
  	snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
  	spin_lock_init(&prtd->lock);
  	runtime->private_data = prtd;
  
  	return 0;
  
  free_client:
  	q6asm_audio_client_free(prtd->audio_client);
  free_prtd:
  	kfree(prtd);
  
  	return ret;
  }
  
  static int q6asm_dai_compr_free(struct snd_soc_component *component,
  				struct snd_compr_stream *stream)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	struct snd_soc_pcm_runtime *rtd = stream->private_data;
  
  	if (prtd->audio_client) {
  		if (prtd->state) {
  			q6asm_cmd(prtd->audio_client, prtd->stream_id,
  				  CMD_CLOSE);
  			if (prtd->next_track_stream_id) {
  				q6asm_cmd(prtd->audio_client,
  					  prtd->next_track_stream_id,
  					  CMD_CLOSE);
  			}
  		}
  
  		snd_dma_free_pages(&prtd->dma_buffer);
  		q6asm_unmap_memory_regions(stream->direction,
  					   prtd->audio_client);
  		q6asm_audio_client_free(prtd->audio_client);
  		prtd->audio_client = NULL;
  	}
  	q6routing_stream_close(rtd->dai_link->id, stream->direction);
  	kfree(prtd);
  
  	return 0;
  }
  
  static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
  					      struct snd_compr_stream *stream,
  					      struct snd_codec *codec,
  					      int stream_id)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	struct q6asm_flac_cfg flac_cfg;
  	struct q6asm_wma_cfg wma_cfg;
  	struct q6asm_alac_cfg alac_cfg;
  	struct q6asm_ape_cfg ape_cfg;
  	unsigned int wma_v9 = 0;
  	struct device *dev = component->dev;
  	int ret;
  	union snd_codec_options *codec_options;
  	struct snd_dec_flac *flac;
  	struct snd_dec_wma *wma;
  	struct snd_dec_alac *alac;
  	struct snd_dec_ape *ape;
  
  	codec_options = &(prtd->codec.options);
  
  	memcpy(&prtd->codec, codec, sizeof(*codec));
  
  	switch (codec->id) {
  	case SND_AUDIOCODEC_FLAC:
  
  		memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
  		flac = &codec_options->flac_d;
  
  		flac_cfg.ch_cfg = codec->ch_in;
  		flac_cfg.sample_rate = codec->sample_rate;
  		flac_cfg.stream_info_present = 1;
  		flac_cfg.sample_size = flac->sample_size;
  		flac_cfg.min_blk_size = flac->min_blk_size;
  		flac_cfg.max_blk_size = flac->max_blk_size;
  		flac_cfg.max_frame_size = flac->max_frame_size;
  		flac_cfg.min_frame_size = flac->min_frame_size;
  
  		ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
  							   stream_id,
  							   &flac_cfg);
  		if (ret < 0) {
  			dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
  			return -EIO;
  		}
  		break;
  
  	case SND_AUDIOCODEC_WMA:
  		wma = &codec_options->wma_d;
  
  		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
  
  		wma_cfg.sample_rate =  codec->sample_rate;
  		wma_cfg.num_channels = codec->ch_in;
  		wma_cfg.bytes_per_sec = codec->bit_rate / 8;
  		wma_cfg.block_align = codec->align;
  		wma_cfg.bits_per_sample = prtd->bits_per_sample;
  		wma_cfg.enc_options = wma->encoder_option;
  		wma_cfg.adv_enc_options = wma->adv_encoder_option;
  		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
  
  		if (wma_cfg.num_channels == 1)
  			wma_cfg.channel_mask = 4; /* Mono Center */
  		else if (wma_cfg.num_channels == 2)
  			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
  		else
  			return -EINVAL;
  
  		/* check the codec profile */
  		switch (codec->profile) {
  		case SND_AUDIOPROFILE_WMA9:
  			wma_cfg.fmtag = 0x161;
  			wma_v9 = 1;
  			break;
  
  		case SND_AUDIOPROFILE_WMA10:
  			wma_cfg.fmtag = 0x166;
  			break;
  
  		case SND_AUDIOPROFILE_WMA9_PRO:
  			wma_cfg.fmtag = 0x162;
  			break;
  
  		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
  			wma_cfg.fmtag = 0x163;
  			break;
  
  		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
  			wma_cfg.fmtag = 0x167;
  			break;
  
  		default:
  			dev_err(dev, "Unknown WMA profile:%x\n",
  				codec->profile);
  			return -EIO;
  		}
  
  		if (wma_v9)
  			ret = q6asm_stream_media_format_block_wma_v9(
  					prtd->audio_client, stream_id,
  					&wma_cfg);
  		else
  			ret = q6asm_stream_media_format_block_wma_v10(
  					prtd->audio_client, stream_id,
  					&wma_cfg);
  		if (ret < 0) {
  			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
  			return -EIO;
  		}
  		break;
  
  	case SND_AUDIOCODEC_ALAC:
  		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
  		alac = &codec_options->alac_d;
  
  		alac_cfg.sample_rate = codec->sample_rate;
  		alac_cfg.avg_bit_rate = codec->bit_rate;
  		alac_cfg.bit_depth = prtd->bits_per_sample;
  		alac_cfg.num_channels = codec->ch_in;
  
  		alac_cfg.frame_length = alac->frame_length;
  		alac_cfg.pb = alac->pb;
  		alac_cfg.mb = alac->mb;
  		alac_cfg.kb = alac->kb;
  		alac_cfg.max_run = alac->max_run;
  		alac_cfg.compatible_version = alac->compatible_version;
  		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
  
  		switch (codec->ch_in) {
  		case 1:
  			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
  			break;
  		case 2:
  			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
  			break;
  		}
  		ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
  							   stream_id,
  							   &alac_cfg);
  		if (ret < 0) {
  			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
  			return -EIO;
  		}
  		break;
  
  	case SND_AUDIOCODEC_APE:
  		memset(&ape_cfg, 0x0, sizeof(ape_cfg));
  		ape = &codec_options->ape_d;
  
  		ape_cfg.sample_rate = codec->sample_rate;
  		ape_cfg.num_channels = codec->ch_in;
  		ape_cfg.bits_per_sample = prtd->bits_per_sample;
  
  		ape_cfg.compatible_version = ape->compatible_version;
  		ape_cfg.compression_level = ape->compression_level;
  		ape_cfg.format_flags = ape->format_flags;
  		ape_cfg.blocks_per_frame = ape->blocks_per_frame;
  		ape_cfg.final_frame_blocks = ape->final_frame_blocks;
  		ape_cfg.total_frames = ape->total_frames;
  		ape_cfg.seek_table_present = ape->seek_table_present;
  
  		ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
  							  stream_id,
  							  &ape_cfg);
  		if (ret < 0) {
  			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
  			return -EIO;
  		}
  		break;
  
  	default:
  		break;
  	}
  
  	return 0;
  }
  
  static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
  				      struct snd_compr_stream *stream,
  				      struct snd_compr_params *params)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	struct snd_soc_pcm_runtime *rtd = stream->private_data;
  	int dir = stream->direction;
  	struct q6asm_dai_data *pdata;
  	struct device *dev = component->dev;
  	int ret;
  
  	pdata = snd_soc_component_get_drvdata(component);
  	if (!pdata)
  		return -EINVAL;
  
  	if (!prtd || !prtd->audio_client) {
  		dev_err(dev, "private data null or audio client freed\n");
  		return -EINVAL;
  	}
  
  	prtd->periods = runtime->fragments;
  	prtd->pcm_count = runtime->fragment_size;
  	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
  	prtd->bits_per_sample = 16;
  
  	if (dir == SND_COMPRESS_PLAYBACK) {
  		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
  				params->codec.profile, prtd->bits_per_sample,
  				true);
  
  		if (ret < 0) {
  			dev_err(dev, "q6asm_open_write failed\n");
  			q6asm_audio_client_free(prtd->audio_client);
  			prtd->audio_client = NULL;
  			return ret;
  		}
  	}
  
  	prtd->session_id = q6asm_get_session_id(prtd->audio_client);
  	ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
  			      prtd->session_id, dir);
  	if (ret) {
  		dev_err(dev, "Stream reg failed ret:%d\n", ret);
  		return ret;
  	}
  
  	ret = __q6asm_dai_compr_set_codec_params(component, stream,
  						 &params->codec,
  						 prtd->stream_id);
  	if (ret) {
  		dev_err(dev, "codec param setup failed ret:%d\n", ret);
  		return ret;
  	}
  
  	ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
  				       (prtd->pcm_size / prtd->periods),
  				       prtd->periods);
  
  	if (ret < 0) {
  		dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
  		return -ENOMEM;
  	}
  
  	prtd->state = Q6ASM_STREAM_RUNNING;
  
  	return 0;
  }
  
  static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
  					struct snd_compr_stream *stream,
  					struct snd_compr_metadata *metadata)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	int ret = 0;
  
  	switch (metadata->key) {
  	case SNDRV_COMPRESS_ENCODER_PADDING:
  		prtd->trailing_samples_drop = metadata->value[0];
  		break;
  	case SNDRV_COMPRESS_ENCODER_DELAY:
  		prtd->initial_samples_drop = metadata->value[0];
  		if (prtd->next_track_stream_id) {
  			ret = q6asm_open_write(prtd->audio_client,
  					       prtd->next_track_stream_id,
  					       prtd->codec.id,
  					       prtd->codec.profile,
  					       prtd->bits_per_sample,
  				       true);
  			if (ret < 0) {
  				dev_err(component->dev, "q6asm_open_write failed\n");
  				return ret;
  			}
  			ret = __q6asm_dai_compr_set_codec_params(component, stream,
  								 &prtd->codec,
  								 prtd->next_track_stream_id);
  			if (ret < 0) {
  				dev_err(component->dev, "q6asm_open_write failed\n");
  				return ret;
  			}
  
  			ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
  						    prtd->next_track_stream_id,
  						    prtd->initial_samples_drop);
  			prtd->next_track_stream_id = 0;
  
  		}
  
  		break;
  	default:
  		ret = -EINVAL;
  		break;
  	}
  
  	return ret;
  }
  
  static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
  				   struct snd_compr_stream *stream, int cmd)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	int ret = 0;
  
  	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, prtd->stream_id,
  				       0, 0, 0);
  		break;
  	case SNDRV_PCM_TRIGGER_STOP:
  		prtd->state = Q6ASM_STREAM_STOPPED;
  		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
  				       CMD_EOS);
  		break;
  	case SNDRV_PCM_TRIGGER_SUSPEND:
  	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
  		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
  				       CMD_PAUSE);
  		break;
  	case SND_COMPR_TRIGGER_NEXT_TRACK:
  		prtd->next_track = true;
  		prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
  		break;
  	case SND_COMPR_TRIGGER_DRAIN:
  	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
  		prtd->notify_on_drain = true;
  		break;
  	default:
  		ret = -EINVAL;
  		break;
  	}
  
  	return ret;
  }
  
  static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
  				   struct snd_compr_stream *stream,
  				   struct snd_compr_tstamp *tstamp)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	unsigned long flags;
  
  	spin_lock_irqsave(&prtd->lock, flags);
  
  	tstamp->copied_total = prtd->copied_total;
  	tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
  
  	spin_unlock_irqrestore(&prtd->lock, flags);
  
  	return 0;
  }
  
  static int q6asm_compr_copy(struct snd_soc_component *component,
  			    struct snd_compr_stream *stream, char __user *buf,
  			    size_t count)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	unsigned long flags;
  	u32 wflags = 0;
  	int avail, bytes_in_flight = 0;
  	void *dstn;
  	size_t copy;
  	u32 app_pointer;
  	u32 bytes_received;
  
  	bytes_received = prtd->bytes_received;
  
  	/**
  	 * Make sure that next track data pointer is aligned at 32 bit boundary
  	 * This is a Mandatory requirement from DSP data buffers alignment
  	 */
  	if (prtd->next_track)
  		bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
  
  	app_pointer = bytes_received/prtd->pcm_size;
  	app_pointer = bytes_received -  (app_pointer * prtd->pcm_size);
  	dstn = prtd->dma_buffer.area + app_pointer;
  
  	if (count < prtd->pcm_size - app_pointer) {
  		if (copy_from_user(dstn, buf, count))
  			return -EFAULT;
  	} else {
  		copy = prtd->pcm_size - app_pointer;
  		if (copy_from_user(dstn, buf, copy))
  			return -EFAULT;
  		if (copy_from_user(prtd->dma_buffer.area, buf + copy,
  				   count - copy))
  			return -EFAULT;
  	}
  
  	spin_lock_irqsave(&prtd->lock, flags);
  
  	bytes_in_flight = prtd->bytes_received - prtd->copied_total;
  
  	if (prtd->next_track) {
  		prtd->next_track = false;
  		prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
  		prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
  	}
  
  	prtd->bytes_received = bytes_received + count;
  
  	/* Kick off the data to dsp if its starving!! */
  	if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
  		uint32_t bytes_to_write = prtd->pcm_count;
  
  		avail = prtd->bytes_received - prtd->bytes_sent;
  
  		if (avail < prtd->pcm_count)
  			bytes_to_write = avail;
  
  		q6asm_write_async(prtd->audio_client, prtd->stream_id,
  				  bytes_to_write, 0, 0, wflags);
  		prtd->bytes_sent += bytes_to_write;
  	}
  
  	spin_unlock_irqrestore(&prtd->lock, flags);
  
  	return count;
  }
  
  static int q6asm_dai_compr_mmap(struct snd_soc_component *component,
  				struct snd_compr_stream *stream,
  				struct vm_area_struct *vma)
  {
  	struct snd_compr_runtime *runtime = stream->runtime;
  	struct q6asm_dai_rtd *prtd = runtime->private_data;
  	struct device *dev = component->dev;
  
  	return dma_mmap_coherent(dev, vma,
  			prtd->dma_buffer.area, prtd->dma_buffer.addr,
  			prtd->dma_buffer.bytes);
  }
  
  static int q6asm_dai_compr_get_caps(struct snd_soc_component *component,
  				    struct snd_compr_stream *stream,
  				    struct snd_compr_caps *caps)
  {
  	caps->direction = SND_COMPRESS_PLAYBACK;
  	caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
  	caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
  	caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
  	caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
  	caps->num_codecs = 5;
  	caps->codecs[0] = SND_AUDIOCODEC_MP3;
  	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
  	caps->codecs[2] = SND_AUDIOCODEC_WMA;
  	caps->codecs[3] = SND_AUDIOCODEC_ALAC;
  	caps->codecs[4] = SND_AUDIOCODEC_APE;
  
  	return 0;
  }
  
  static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component,
  					  struct snd_compr_stream *stream,
  					  struct snd_compr_codec_caps *codec)
  {
  	switch (codec->codec) {
  	case SND_AUDIOCODEC_MP3:
  		*codec = q6asm_compr_caps;
  		break;
  	default:
  		break;
  	}
  
  	return 0;
  }
  
  static const struct snd_compress_ops q6asm_dai_compress_ops = {
  	.open		= q6asm_dai_compr_open,
  	.free		= q6asm_dai_compr_free,
  	.set_params	= q6asm_dai_compr_set_params,
  	.set_metadata	= q6asm_dai_compr_set_metadata,
  	.pointer	= q6asm_dai_compr_pointer,
  	.trigger	= q6asm_dai_compr_trigger,
  	.get_caps	= q6asm_dai_compr_get_caps,
  	.get_codec_caps	= q6asm_dai_compr_get_codec_caps,
  	.mmap		= q6asm_dai_compr_mmap,
  	.copy		= q6asm_compr_copy,
  };
  
  static int q6asm_dai_pcm_new(struct snd_soc_component *component,
  			     struct snd_soc_pcm_runtime *rtd)
  {
  	struct snd_pcm *pcm = rtd->pcm;
  	size_t size = q6asm_dai_hardware_playback.buffer_bytes_max;
  
  	return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
  					    component->dev, size);
  }
  
  static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = {
  	SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0),
  	SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0),
  };
  
  static const struct snd_soc_component_driver q6asm_fe_dai_component = {
  	.name		= DRV_NAME,
  	.open		= q6asm_dai_open,
  	.hw_params	= q6asm_dai_hw_params,
  	.close		= q6asm_dai_close,
  	.prepare	= q6asm_dai_prepare,
  	.trigger	= q6asm_dai_trigger,
  	.pointer	= q6asm_dai_pointer,
  	.pcm_construct	= q6asm_dai_pcm_new,
  	.compress_ops	= &q6asm_dai_compress_ops,
  	.dapm_widgets	= q6asm_dapm_widgets,
  	.num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets),
  };
  
  static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
  	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 of_q6asm_parse_dai_data(struct device *dev,
  				    struct q6asm_dai_data *pdata)
  {
  	struct snd_soc_dai_driver *dai_drv;
  	struct snd_soc_pcm_stream empty_stream;
  	struct device_node *node;
  	int ret, id, dir, idx = 0;
  
  
  	pdata->num_dais = of_get_child_count(dev->of_node);
  	if (!pdata->num_dais) {
  		dev_err(dev, "No dais found in DT\n");
  		return -EINVAL;
  	}
  
  	pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
  				   GFP_KERNEL);
  	if (!pdata->dais)
  		return -ENOMEM;
  
  	memset(&empty_stream, 0, sizeof(empty_stream));
  
  	for_each_child_of_node(dev->of_node, node) {
  		ret = of_property_read_u32(node, "reg", &id);
  		if (ret || id >= MAX_SESSIONS || id < 0) {
  			dev_err(dev, "valid dai id not found:%d\n", ret);
  			continue;
  		}
  
  		dai_drv = &pdata->dais[idx++];
  		*dai_drv = q6asm_fe_dais_template[id];
  
  		ret = of_property_read_u32(node, "direction", &dir);
  		if (ret)
  			continue;
  
  		if (dir == Q6ASM_DAI_RX)
  			dai_drv->capture = empty_stream;
  		else if (dir == Q6ASM_DAI_TX)
  			dai_drv->playback = empty_stream;
  
  		if (of_property_read_bool(node, "is-compress-dai"))
  			dai_drv->compress_new = snd_soc_new_compress;
  	}
  
  	return 0;
  }
  
  static int q6asm_dai_probe(struct platform_device *pdev)
  {
  	struct device *dev = &pdev->dev;
  	struct device_node *node = dev->of_node;
  	struct of_phandle_args args;
  	struct q6asm_dai_data *pdata;
  	int rc;
  
  	pdata = devm_kzalloc(dev, sizeof(*pdata), 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);
  
  	rc = of_q6asm_parse_dai_data(dev, pdata);
  	if (rc)
  		return rc;
  
  	return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
  					       pdata->dais, pdata->num_dais);
  }
  
  #ifdef CONFIG_OF
  static const struct of_device_id q6asm_dai_device_id[] = {
  	{ .compatible = "qcom,q6asm-dais" },
  	{},
  };
  MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
  #endif
  
  static struct platform_driver q6asm_dai_platform_driver = {
  	.driver = {
  		.name = "q6asm-dai",
  		.of_match_table = of_match_ptr(q6asm_dai_device_id),
  	},
  	.probe = q6asm_dai_probe,
  };
  module_platform_driver(q6asm_dai_platform_driver);
  
  MODULE_DESCRIPTION("Q6ASM dai driver");
  MODULE_LICENSE("GPL v2");
  