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From:   srinivas.kandagatla@linaro.org
To:     andy.gross@linaro.org, broonie@kernel.org,
        linux-arm-msm@vger.kernel.org, alsa-devel@alsa-project.org,
        robh+dt@kernel.org, bgoswami@codeaurora.org
Cc:     gregkh@linuxfoundation.org, david.brown@linaro.org,
        mark.rutland@arm.com, lgirdwood@gmail.com, plai@codeaurora.org,
        tiwai@suse.com, perex@perex.cz, devicetree@vger.kernel.org,
        linux-kernel@vger.kernel.org, linux-arm-kernel@lists.infradead.org,
        rohkumar@qti.qualcomm.com, spatakok@qti.qualcomm.com,
        Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Subject: [PATCH v6 21/24] ASoC: qdsp6: q6asm: Add q6asm dai driver
Date:   Thu, 26 Apr 2018 10:46:03 +0100
Message-Id: <20180426094606.4775-22-srinivas.kandagatla@linaro.org>
In-Reply-To: <20180426094606.4775-1-srinivas.kandagatla@linaro.org>
References: <20180426094606.4775-1-srinivas.kandagatla@linaro.org>
Sender: linux-kernel-owner@vger.kernel.org
Precedence: bulk

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

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>
---
 sound/soc/qcom/Kconfig           |   4 +
 sound/soc/qcom/qdsp6/Makefile    |   1 +
 sound/soc/qcom/qdsp6/q6asm-dai.c | 632 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 637 insertions(+)
 create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 8d36f419e0f8..a424430af391 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -65,6 +65,9 @@ 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
@@ -75,6 +78,7 @@ 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
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index bada1aa303c2..c33b3cacbea1 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -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
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000000..73d0c76933b4
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,632 @@
+// 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 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 q6asm_dai_data *pdata;
+	int ret, i;
+
+	pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+	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 q6asm_dai_rtd *prtd;
+	struct q6asm_dai_data *pdata;
+	struct device *dev = soc_prtd->platform->dev;
+	int ret = 0;
+	int stream_id;
+
+	stream_id = cpu_dai->driver->id;
+
+	pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+	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,
+				(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 device *dev = soc_prtd->platform->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 q6asm_dai_data *pdata;
+	struct snd_pcm *pcm = rtd->pcm;
+	struct device *dev;
+	int size, ret;
+
+	dev = rtd->platform->dev;
+	pdata = snd_soc_platform_get_drvdata(rtd->platform);
+
+	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 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" },
+	{"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		= "q6asm-fe-dai",
+};
+
+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);
+
+	rc = snd_soc_register_platform(dev,  &q6asm_soc_platform);
+	if (rc) {
+		dev_err(dev, "err_dai_platform\n");
+		return rc;
+	}
+
+	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_platform(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");
-- 
2.16.2