ASoC: msm: Add compress audio playback code base

Support bits and pieces to make msm-compress-q6-v2.c usable
will be added in subsequent commits.

Signed-off-by: Matt Wagantall <mattw@codeaurora.org>

sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c

@@ -0,0 +1,950 @@
+/* Copyright (c) 2012-2013, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/math64.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6asm-v2.h>
+#include <sound/pcm_params.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio_ion.h>
+
+#include <sound/timer.h>
+#include <sound/tlv.h>
+
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+#include <sound/compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+
+#include "msm-pcm-routing-v2.h"
+#include "audio_ocmem.h"
+
+/* 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 COMPRESSED_LR_VOL_MAX_STEPS	0x2000
+const DECLARE_TLV_DB_LINEAR(msm_compr_vol_gain, 0,
+				COMPRESSED_LR_VOL_MAX_STEPS);
+
+struct msm_compr_pdata {
+	atomic_t audio_ocmem_req;
+	struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX];
+	uint32_t volume[MSM_FRONTEND_DAI_MAX][2]; /* For both L & R */
+};
+
+struct msm_compr_audio {
+	struct snd_compr_stream *cstream;
+	struct snd_compr_caps compr_cap;
+	struct snd_compr_codec_caps codec_caps;
+	struct snd_compr_params codec_param;
+	struct audio_client *audio_client;
+
+	uint32_t codec;
+	void    *buffer; /* virtual address */
+	uint32_t buffer_paddr; /* physical address */
+	uint32_t app_pointer;
+	uint32_t buffer_size;
+	uint32_t byte_offset;
+	uint32_t copied_total;
+	uint32_t bytes_received;
+
+	uint16_t session_id;
+
+	uint32_t sample_rate;
+	uint32_t num_channels;
+
+	uint32_t cmd_ack;
+	uint32_t cmd_interrupt;
+	uint32_t drain_ready;
+
+	atomic_t start;
+	atomic_t eos;
+	atomic_t drain;
+
+	wait_queue_head_t eos_wait;
+	wait_queue_head_t drain_wait;
+	wait_queue_head_t flush_wait;
+
+	spinlock_t lock;
+};
+
+static int msm_compr_set_volume(struct snd_compr_stream *cstream,
+				uint32_t volume_l, uint32_t volume_r)
+{
+	struct msm_compr_audio *prtd;
+	int rc = 0;
+
+	pr_debug("%s: volume_l %d volume_r %d\n",
+		__func__, volume_l, volume_r);
+	prtd = cstream->runtime->private_data;
+	if (prtd && prtd->audio_client) {
+		if (volume_l != volume_r) {
+			pr_debug("%s: call q6asm_set_lrgain\n", __func__);
+			rc = q6asm_set_lrgain(prtd->audio_client,
+						volume_l, volume_r);
+		} else {
+			pr_debug("%s: call q6asm_set_volume\n", __func__);
+			rc = q6asm_set_volume(prtd->audio_client, volume_l);
+		}
+		if (rc < 0) {
+			pr_err("%s: Send Volume command failed rc=%d\n",
+				__func__, rc);
+		}
+	}
+
+	return rc;
+}
+
+static int msm_compr_send_buffer(struct msm_compr_audio *prtd)
+{
+	int buffer_length;
+	int bytes_available;
+	struct audio_aio_write_param param;
+
+	if (!atomic_read(&prtd->start)) {
+		pr_err("%s: stream is not in started state\n", __func__);
+		return -EINVAL;
+	}
+	pr_debug("%s: bytes_received = %d copied_total = %d\n",
+		__func__, prtd->bytes_received, prtd->copied_total);
+	buffer_length = prtd->codec_param.buffer.fragment_size;
+	bytes_available = prtd->bytes_received - prtd->copied_total;
+	if (bytes_available < prtd->codec_param.buffer.fragment_size)
+		buffer_length = bytes_available;
+
+	if (buffer_length == 0) {
+		pr_debug("Recieved a zero length buffer-break out\n");
+		if (atomic_read(&prtd->drain)) {
+			prtd->drain_ready = 1;
+			wake_up(&prtd->drain_wait);
+			atomic_set(&prtd->drain, 0);
+		}
+		return 0;
+	}
+
+	param.paddr	= prtd->buffer_paddr + prtd->byte_offset;
+	param.len	= buffer_length;
+	param.msw_ts	= 0;
+	param.lsw_ts	= 0;
+	param.flags	= NO_TIMESTAMP;
+	param.uid	= buffer_length;
+	param.metadata_len = 0;
+
+	pr_debug("%s: sending %d bytes to DSP byte_offset = %d\n",
+		__func__, buffer_length, prtd->byte_offset);
+	if (q6asm_async_write(prtd->audio_client, &param) < 0)
+		pr_err("%s:q6asm_async_write failed\n", __func__);
+
+	return 0;
+}
+
+static void compr_event_handler(uint32_t opcode,
+		uint32_t token, uint32_t *payload, void *priv)
+{
+	struct msm_compr_audio *prtd = priv;
+	struct snd_compr_stream *cstream = prtd->cstream;
+	uint32_t chan_mode = 0;
+	uint32_t sample_rate = 0;
+
+	pr_debug("%s opcode =%08x\n", __func__, opcode);
+	switch (opcode) {
+	case ASM_DATA_EVENT_WRITE_DONE_V2:
+		pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n");
+		spin_lock_irq(&prtd->lock);
+		prtd->byte_offset += token;
+		prtd->copied_total += token;
+		if (prtd->byte_offset >= prtd->buffer_size)
+			prtd->byte_offset -= prtd->buffer_size;
+
+		snd_compr_fragment_elapsed(cstream);
+		if (atomic_read(&prtd->start))
+			msm_compr_send_buffer(prtd);
+		spin_unlock_irq(&prtd->lock);
+		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		pr_debug("ASM_DATA_CMDRSP_EOS\n");
+		if (atomic_read(&prtd->eos)) {
+			pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
+			prtd->cmd_ack = 1;
+			wake_up(&prtd->eos_wait);
+			atomic_set(&prtd->eos, 0);
+		}
+		break;
+	case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
+	case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY: {
+		pr_debug("ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY\n");
+		chan_mode = payload[1] >> 16;
+		sample_rate = payload[2] >> 16;
+		if (prtd && (chan_mode != prtd->num_channels ||
+				sample_rate != prtd->sample_rate)) {
+			prtd->num_channels = chan_mode;
+			prtd->sample_rate = sample_rate;
+		}
+	}
+	case APR_BASIC_RSP_RESULT: {
+		switch (payload[0]) {
+		case ASM_SESSION_CMD_RUN_V2:
+			/* check if the first buffer need to be sent to DSP */
+			pr_debug("ASM_SESSION_CMD_RUN_V2\n");
+			if (!prtd->copied_total)
+				msm_compr_send_buffer(prtd);
+			break;
+		case ASM_STREAM_CMD_FLUSH:
+			pr_debug("ASM_STREAM_CMD_FLUSH\n");
+			prtd->cmd_ack = 1;
+			wake_up(&prtd->flush_wait);
+			break;
+		default:
+			break;
+		}
+		break;
+	}
+	case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3:
+		pr_debug("ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n");
+		break;
+	default:
+		pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+		break;
+	}
+}
+
+static void populate_codec_list(struct msm_compr_audio *prtd)
+{
+	pr_debug("%s\n", __func__);
+	prtd->compr_cap.direction = SND_COMPRESS_PLAYBACK;
+	prtd->compr_cap.min_fragment_size =
+			COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+	prtd->compr_cap.max_fragment_size =
+			COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+	prtd->compr_cap.min_fragments =
+			COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+	prtd->compr_cap.max_fragments =
+			COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+	prtd->compr_cap.num_codecs = 2;
+	prtd->compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
+	prtd->compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
+}
+
+static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data;
+	uint16_t bits_per_sample = 16;
+	int dir = IN, ret = 0;
+	struct asm_softpause_params softpause = {
+		.enable = SOFT_PAUSE_ENABLE,
+		.period = SOFT_PAUSE_PERIOD,
+		.step = SOFT_PAUSE_STEP,
+		.rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+	};
+	struct asm_softvolume_params softvol = {
+		.period = SOFT_VOLUME_PERIOD,
+		.step = SOFT_VOLUME_STEP,
+		.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+	};
+
+	pr_debug("%s\n", __func__);
+	ret = q6asm_open_write_v2(prtd->audio_client,
+				prtd->codec, bits_per_sample);
+	if (ret < 0) {
+		pr_err("%s: Session out open failed\n", __func__);
+		 return -ENOMEM;
+	}
+
+	pr_debug("%s be_id %d\n", __func__, soc_prtd->dai_link->be_id);
+	msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+				prtd->audio_client->perf_mode,
+				prtd->session_id,
+				SNDRV_PCM_STREAM_PLAYBACK);
+
+	ret = msm_compr_set_volume(cstream, 0, 0);
+	if (ret < 0)
+		pr_err("%s : Set Volume failed : %d", __func__, ret);
+
+	ret = q6asm_set_softpause(prtd->audio_client,
+					&softpause);
+	if (ret < 0)
+		pr_err("%s: Send SoftPause Param failed ret=%d\n",
+			__func__, ret);
+
+	ret = q6asm_set_softvolume(prtd->audio_client, &softvol);
+	if (ret < 0)
+		pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+			__func__, ret);
+
+	ret = q6asm_set_io_mode(prtd->audio_client,
+				(COMPRESSED_IO | ASYNC_IO_MODE));
+	if (ret < 0) {
+		pr_err("%s: Set IO mode failed\n", __func__);
+		return -EINVAL;
+	}
+
+	runtime->fragments = prtd->codec_param.buffer.fragments;
+	runtime->fragment_size = prtd->codec_param.buffer.fragment_size;
+	pr_debug("allocate %d buffers each of size %d\n",
+			runtime->fragments,
+			runtime->fragment_size);
+	ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+					prtd->audio_client,
+					runtime->fragment_size,
+					runtime->fragments);
+	if (ret < 0) {
+		pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+		return -ENOMEM;
+	}
+
+	prtd->byte_offset  = 0;
+	prtd->copied_total = 0;
+	prtd->app_pointer  = 0;
+	prtd->bytes_received = 0;
+	prtd->buffer       = prtd->audio_client->port[dir].buf[0].data;
+	prtd->buffer_paddr = prtd->audio_client->port[dir].buf[0].phys;
+	prtd->buffer_size  = runtime->fragments * runtime->fragment_size;
+
+	return 0;
+}
+
+static int msm_compr_open(struct snd_compr_stream *cstream)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+	struct msm_compr_audio *prtd;
+	struct msm_compr_pdata *pdata =
+			snd_soc_platform_get_drvdata(rtd->platform);
+
+	pr_debug("%s\n", __func__);
+	prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL);
+	if (prtd == NULL) {
+		pr_err("Failed to allocate memory for msm_compr_audio\n");
+		return -ENOMEM;
+	}
+
+	prtd->cstream = cstream;
+	pdata->cstream[rtd->dai_link->be_id] = cstream;
+	prtd->audio_client = q6asm_audio_client_alloc(
+				(app_cb)compr_event_handler, prtd);
+	if (!prtd->audio_client) {
+		pr_err("%s: Could not allocate memory\n", __func__);
+		kfree(prtd);
+		return -ENOMEM;
+	}
+
+	pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+	prtd->audio_client->perf_mode = false;
+	prtd->session_id = prtd->audio_client->session;
+	prtd->codec = FORMAT_MP3;
+	prtd->bytes_received = 0;
+	prtd->copied_total = 0;
+	prtd->byte_offset = 0;
+	prtd->sample_rate = 44100;
+	prtd->num_channels = 2;
+	prtd->drain_ready = 0;
+
+	spin_lock_init(&prtd->lock);
+
+	atomic_set(&prtd->eos, 0);
+	atomic_set(&prtd->start, 0);
+	atomic_set(&prtd->drain, 0);
+
+	init_waitqueue_head(&prtd->eos_wait);
+	init_waitqueue_head(&prtd->drain_wait);
+	init_waitqueue_head(&prtd->flush_wait);
+
+	runtime->private_data = prtd;
+	populate_codec_list(prtd);
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+		if (!atomic_cmpxchg(&pdata->audio_ocmem_req, 0, 1))
+			audio_ocmem_process_req(AUDIO, true);
+		else
+			atomic_inc(&pdata->audio_ocmem_req);
+		pr_debug("%s: ocmem_req: %d\n", __func__,
+				atomic_read(&pdata->audio_ocmem_req));
+	} else {
+		pr_err("%s: Unsupported stream type", __func__);
+	}
+
+	return 0;
+}
+
+static int msm_compr_free(struct snd_compr_stream *cstream)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data;
+	struct msm_compr_pdata *pdata =
+			snd_soc_platform_get_drvdata(soc_prtd->platform);
+	int dir = IN, ret = 0;
+
+	pr_debug("%s\n", __func__);
+	pdata->cstream[soc_prtd->dai_link->be_id] = NULL;
+	if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+		if (atomic_read(&pdata->audio_ocmem_req) > 1)
+			atomic_dec(&pdata->audio_ocmem_req);
+		else if (atomic_cmpxchg(&pdata->audio_ocmem_req, 1, 0))
+			audio_ocmem_process_req(AUDIO, false);
+
+		msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+						SNDRV_PCM_STREAM_PLAYBACK);
+	}
+
+	pr_debug("%s: ocmem_req: %d\n", __func__,
+		atomic_read(&pdata->audio_ocmem_req));
+
+	ret = wait_event_timeout(prtd->eos_wait,
+				prtd->cmd_ack, 5 * HZ);
+	if (!ret)
+		pr_err("%s: CMD_EOS failed\n", __func__);
+
+	q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
+	q6asm_audio_client_buf_free_contiguous(dir,
+					prtd->audio_client);
+
+	q6asm_audio_client_free(prtd->audio_client);
+
+	kfree(prtd);
+
+	return 0;
+}
+
+/* compress stream operations */
+static int msm_compr_set_params(struct snd_compr_stream *cstream,
+				struct snd_compr_params *params)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	int ret = 0;
+
+	pr_debug("%s\n", __func__);
+
+	memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params));
+
+	/* ToDo: remove duplicates */
+	prtd->num_channels = prtd->codec_param.codec.ch_in;
+
+	switch (prtd->codec_param.codec.sample_rate) {
+	case SNDRV_PCM_RATE_8000:
+		prtd->sample_rate = 8000;
+		break;
+	case SNDRV_PCM_RATE_11025:
+		prtd->sample_rate = 11025;
+		break;
+	/* ToDo: What about 12K and 24K sample rates ? */
+	case SNDRV_PCM_RATE_16000:
+		prtd->sample_rate = 16000;
+		break;
+	case SNDRV_PCM_RATE_22050:
+		prtd->sample_rate = 22050;
+		break;
+	case SNDRV_PCM_RATE_32000:
+		prtd->sample_rate = 32000;
+		break;
+	case SNDRV_PCM_RATE_44100:
+		prtd->sample_rate = 44100;
+		break;
+	case SNDRV_PCM_RATE_48000:
+		prtd->sample_rate = 48000;
+		break;
+	}
+
+	pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate);
+
+	switch (params->codec.id) {
+	case SND_AUDIOCODEC_MP3: {
+		pr_debug("SND_AUDIOCODEC_MP3\n");
+		prtd->codec = FORMAT_MP3;
+		break;
+	}
+
+	case SND_AUDIOCODEC_AAC: {
+		pr_debug("SND_AUDIOCODEC_AAC\n");
+		prtd->codec = FORMAT_MPEG4_AAC;
+		break;
+	}
+
+	default:
+		pr_err("codec not supported, id =%d\n", params->codec.id);
+		return -EINVAL;
+	}
+
+	ret = msm_compr_configure_dsp(cstream);
+
+	return ret;
+}
+
+static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+	struct msm_compr_pdata *pdata =
+			snd_soc_platform_get_drvdata(rtd->platform);
+	uint32_t *volume = pdata->volume[rtd->dai_link->be_id];
+	int rc = 0;
+
+	if (cstream->direction != SND_COMPRESS_PLAYBACK) {
+		pr_err("%s: Unsupported stream type\n", __func__);
+		return -EINVAL;
+	}
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__);
+		atomic_set(&prtd->start, 1);
+		q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+
+		msm_compr_set_volume(cstream, volume[0], volume[1]);
+		if (rc)
+			pr_err("%s : Set Volume failed : %d\n",
+				__func__, rc);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		pr_debug("%s: SNDRV_PCM_TRIGGER_STOP\n", __func__);
+		atomic_set(&prtd->start, 0);
+		if (atomic_read(&prtd->eos)) {
+			prtd->cmd_interrupt = 1;
+			prtd->drain_ready = 1;
+			wake_up(&prtd->drain_wait);
+			wake_up(&prtd->eos_wait);
+			atomic_set(&prtd->eos, 0);
+		}
+		/* Issue flush command only if any buffers are left with DSP */
+		spin_lock_irq(&prtd->lock);
+		if (prtd->bytes_received > prtd->copied_total) {
+			prtd->cmd_ack = 0;
+			spin_unlock_irq(&prtd->lock);
+			rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+			if (rc < 0) {
+				pr_err("%s: flush cmd failed rc=%d\n",
+					__func__, rc);
+				return rc;
+			}
+			rc = wait_event_timeout(prtd->flush_wait,
+				prtd->cmd_ack, 1 * HZ);
+			if (!rc)
+				pr_err("Flush cmd timeout\n");
+		} else
+			spin_unlock_irq(&prtd->lock);
+
+		prtd->byte_offset  = 0;
+		prtd->copied_total = 0;
+		prtd->app_pointer  = 0;
+		prtd->bytes_received = 0;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		pr_debug("SNDRV_PCM_TRIGGER_PAUSE_PUSH\n");
+		q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+		atomic_set(&prtd->start, 0);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		pr_debug("SNDRV_PCM_TRIGGER_PAUSE_RELEASE\n");
+		atomic_set(&prtd->start, 1);
+		q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+		break;
+	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+		pr_debug("%s: SND_COMPR_TRIGGER_PARTIAL_DRAIN\n", __func__);
+	case SND_COMPR_TRIGGER_DRAIN:
+		pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__);
+		if (!atomic_read(&prtd->start)) {
+			pr_err("%s: stream is not in started state\n",
+				__func__);
+			break;
+		}
+
+		/* Make sure all the data is sent to DSP before sending EOS */
+		spin_lock_irq(&prtd->lock);
+		if (prtd->bytes_received > prtd->copied_total) {
+			atomic_set(&prtd->drain, 1);
+			prtd->drain_ready = 0;
+			spin_unlock_irq(&prtd->lock);
+			pr_debug("%s: wait till all the data is sent to dsp\n",
+				__func__);
+			rc = wait_event_interruptible(prtd->drain_wait,
+					prtd->drain_ready);
+		} else
+			spin_unlock_irq(&prtd->lock);
+
+		if (!atomic_read(&prtd->start)) {
+			pr_err("%s: stream is not started\n", __func__);
+			break;
+		}
+
+		if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN)
+			break;
+
+		atomic_set(&prtd->eos, 1);
+		prtd->cmd_ack = 0;
+		pr_debug("%s: CMD_EOS\n", __func__);
+		q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+		/* Wait indefinitely for  DRAIN. Flush can also signal this*/
+		rc = wait_event_interruptible(prtd->eos_wait,
+				(prtd->cmd_ack || prtd->cmd_interrupt));
+
+		if (rc < 0)
+			pr_err("%s: EOS cmd interrupted\n", __func__);
+		pr_debug("%s: SNDRV_COMPRESS_DRAIN  out of wait\n", __func__);
+
+		if (prtd->cmd_interrupt)
+			rc = -EINTR;
+
+		prtd->cmd_interrupt = 0;
+		break;
+	case SND_COMPR_TRIGGER_NEXT_TRACK:
+		pr_debug("%s: SND_COMPR_TRIGGER_NEXT_TRACK\n", __func__);
+		break;
+	}
+
+	return 0;
+}
+
+static int msm_compr_pointer(struct snd_compr_stream *cstream,
+				struct snd_compr_tstamp *arg)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	struct snd_compr_tstamp tstamp;
+	uint64_t timestamp = 0;
+	int rc = 0;
+
+	pr_debug("%s\n", __func__);
+	memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
+
+	spin_lock_irq(&prtd->lock);
+	tstamp.sampling_rate = prtd->sample_rate;
+	tstamp.byte_offset = prtd->byte_offset;
+	tstamp.copied_total = prtd->copied_total;
+	spin_unlock_irq(&prtd->lock);
+
+	if (atomic_read(&prtd->start)) {
+		rc = q6asm_get_session_time(prtd->audio_client, &timestamp);
+		if (rc < 0) {
+			pr_err("%s: Get Session Time return value =%lld\n",
+				__func__, timestamp);
+			return -EAGAIN;
+		}
+	}
+
+	/* DSP returns timestamp in usec */
+	pr_debug("%s: timestamp = %lld usec\n", __func__, timestamp);
+	timestamp *= prtd->sample_rate;
+	tstamp.pcm_io_frames = (snd_pcm_uframes_t)div64_u64(timestamp, 1000000);
+	memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp));
+
+	return 0;
+}
+
+static int msm_compr_ack(struct snd_compr_stream *cstream,
+			size_t count)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	void *src, *dstn;
+	size_t copy;
+
+	pr_debug("%s: count = %d\n", __func__, count);
+	if (!prtd->buffer) {
+		pr_err("%s: Buffer is not allocated yet ??\n", __func__);
+		return -EINVAL;
+	}
+	src = runtime->buffer + prtd->app_pointer;
+	dstn = prtd->buffer + prtd->app_pointer;
+	if (count < prtd->buffer_size - prtd->app_pointer) {
+		memcpy(dstn, src, count);
+		prtd->app_pointer += count;
+	} else {
+		copy = prtd->buffer_size - prtd->app_pointer;
+		memcpy(dstn, src, copy);
+		memcpy(prtd->buffer, runtime->buffer, count - copy);
+		prtd->app_pointer = count - copy;
+	}
+
+	/*
+	 * If the stream is started and all the bytes received were
+	 * copied to DSP, the newly received bytes should be
+	 * sent right away
+	 */
+	spin_lock_irq(&prtd->lock);
+
+	if (atomic_read(&prtd->start) &&
+		prtd->bytes_received == prtd->copied_total) {
+		prtd->bytes_received += count;
+		msm_compr_send_buffer(prtd);
+	} else
+		prtd->bytes_received += count;
+
+	spin_unlock_irq(&prtd->lock);
+
+	return 0;
+}
+
+static int msm_compr_copy(struct snd_compr_stream *cstream,
+				char __user *buf, size_t count)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	void *dstn;
+	size_t copy;
+
+	pr_debug("%s: count = %d\n", __func__, count);
+	if (!prtd->buffer) {
+		pr_err("%s: Buffer is not allocated yet ??", __func__);
+		return 0;
+	}
+
+	dstn = prtd->buffer + prtd->app_pointer;
+	if (count < prtd->buffer_size - prtd->app_pointer) {
+		if (copy_from_user(dstn, buf, count))
+			return -EFAULT;
+		prtd->app_pointer += count;
+	} else {
+		copy = prtd->buffer_size - prtd->app_pointer;
+		if (copy_from_user(dstn, buf, copy))
+			return -EFAULT;
+		if (copy_from_user(prtd->buffer, buf + copy, count - copy))
+			return -EFAULT;
+		prtd->app_pointer = count - copy;
+	}
+
+	/*
+	 * If stream is started and all the bytes received were
+	 * copied to DSP, the newly received bytes should be
+	 * copied right away
+	 */
+	spin_lock_irq(&prtd->lock);
+
+	if (atomic_read(&prtd->start) &&
+		prtd->bytes_received == prtd->copied_total) {
+		prtd->bytes_received += count;
+		msm_compr_send_buffer(prtd);
+	} else
+		prtd->bytes_received += count;
+
+	spin_unlock_irq(&prtd->lock);
+
+	return count;
+}
+
+static int msm_compr_get_caps(struct snd_compr_stream *cstream,
+				struct snd_compr_caps *arg)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+
+	pr_debug("%s\n", __func__);
+	memcpy(arg, &prtd->compr_cap, sizeof(struct snd_compr_caps));
+
+	return 0;
+}
+
+static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream,
+				struct snd_compr_codec_caps *codec)
+{
+	pr_debug("%s\n", __func__);
+
+	switch (codec->codec) {
+	case SND_AUDIOCODEC_MP3:
+		codec->num_descriptors = 2;
+		codec->descriptor[0].max_ch = 2;
+		codec->descriptor[0].sample_rates = SNDRV_PCM_RATE_8000_48000;
+		codec->descriptor[0].bit_rate[0] = 320; /* 320kbps */
+		codec->descriptor[0].bit_rate[1] = 128;
+		codec->descriptor[0].num_bitrates = 2;
+		codec->descriptor[0].profiles = 0;
+		codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO;
+		codec->descriptor[0].formats = 0;
+		break;
+	case SND_AUDIOCODEC_AAC:
+		codec->num_descriptors = 2;
+		codec->descriptor[1].max_ch = 2;
+		codec->descriptor[1].sample_rates = SNDRV_PCM_RATE_8000_48000;
+		codec->descriptor[1].bit_rate[0] = 320; /* 320kbps */
+		codec->descriptor[1].bit_rate[1] = 128;
+		codec->descriptor[1].num_bitrates = 2;
+		codec->descriptor[1].profiles = 0;
+		codec->descriptor[1].modes = 0;
+		codec->descriptor[1].formats =
+			(SND_AUDIOSTREAMFORMAT_MP4ADTS |
+				SND_AUDIOSTREAMFORMAT_RAW);
+		break;
+	default:
+		pr_err("%s: Unsupported audio codec %d\n",
+			__func__, codec->codec);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int msm_compr_set_metadata(struct snd_compr_stream *cstream,
+				struct snd_compr_metadata *metadata)
+{
+	pr_debug("%s\n", __func__);
+	return -ENXIO;
+}
+
+static int msm_compr_volume_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+			snd_soc_platform_get_drvdata(platform);
+	struct snd_compr_stream *cstream = pdata->cstream[mc->reg];
+	uint32_t *volume = pdata->volume[mc->reg];
+
+	volume[0] = ucontrol->value.integer.value[0];
+	volume[1] = ucontrol->value.integer.value[1];
+	pr_debug("%s: mc->reg %d left_vol %d right_vol %d\n",
+		__func__, mc->reg, volume[0], volume[1]);
+	if (cstream)
+		msm_compr_set_volume(cstream, volume[0], volume[1]);
+	return 0;
+}
+
+static int msm_compr_volume_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct msm_compr_pdata *pdata =
+		snd_soc_platform_get_drvdata(platform);
+	uint32_t *volume = pdata->volume[mc->reg];
+	pr_debug("%s: mc->reg %d\n", __func__, mc->reg);
+	ucontrol->value.integer.value[0] = volume[0];
+	ucontrol->value.integer.value[1] = volume[1];
+
+	return 0;
+}
+
+/* System Pin has no volume control */
+static const struct snd_kcontrol_new msm_compr_volume_controls[] = {
+	SOC_DOUBLE_EXT_TLV("Compress Playback Volume",
+			MSM_FRONTEND_DAI_MULTIMEDIA4,
+			0, 8, COMPRESSED_LR_VOL_MAX_STEPS, 0,
+			msm_compr_volume_get,
+			msm_compr_volume_put,
+			msm_compr_vol_gain),
+};
+
+static int msm_compr_probe(struct snd_soc_platform *platform)
+{
+	struct msm_compr_pdata *pdata;
+	int i;
+
+	pr_debug("%s\n", __func__);
+	pdata = (struct msm_compr_pdata *)
+			kzalloc(sizeof(*pdata), GFP_KERNEL);
+	if (!pdata)
+		return -ENOMEM;
+
+	snd_soc_platform_set_drvdata(platform, pdata);
+
+	atomic_set(&pdata->audio_ocmem_req, 0);
+
+	for (i = 0; i < MSM_FRONTEND_DAI_MAX; i++) {
+		pdata->volume[i][0] = COMPRESSED_LR_VOL_MAX_STEPS;
+		pdata->volume[i][1] = COMPRESSED_LR_VOL_MAX_STEPS;
+		pdata->cstream[i] = NULL;
+	}
+
+	return 0;
+}
+
+static struct snd_compr_ops msm_compr_ops = {
+	.open		= msm_compr_open,
+	.free		= msm_compr_free,
+	.trigger	= msm_compr_trigger,
+	.pointer	= msm_compr_pointer,
+	.set_params	= msm_compr_set_params,
+	.set_metadata	= msm_compr_set_metadata,
+	.ack		= msm_compr_ack,
+	.copy		= msm_compr_copy,
+	.get_caps	= msm_compr_get_caps,
+	.get_codec_caps = msm_compr_get_codec_caps,
+};
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+	.probe		= msm_compr_probe,
+	.compr_ops	= &msm_compr_ops,
+	.controls	= msm_compr_volume_controls,
+	.num_controls	= ARRAY_SIZE(msm_compr_volume_controls),
+};
+
+static int msm_compr_dev_probe(struct platform_device *pdev)
+{
+	if (pdev->dev.of_node)
+		dev_set_name(&pdev->dev, "%s", "msm-compress-dsp");
+
+	pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+	return snd_soc_register_platform(&pdev->dev,
+					&msm_soc_platform);
+}
+
+static int msm_compr_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static const struct of_device_id msm_compr_dt_match[] = {
+	{.compatible = "qcom,msm-compress-dsp"},
+	{}
+};
+MODULE_DEVICE_TABLE(of, msm_compr_dt_match);
+
+static struct platform_driver msm_compr_driver = {
+	.driver = {
+		.name = "msm-compress-dsp",
+		.owner = THIS_MODULE,
+		.of_match_table = msm_compr_dt_match,
+	},
+	.probe = msm_compr_dev_probe,
+	.remove = msm_compr_remove,
+};
+
+static int __init msm_soc_platform_init(void)
+{
+	return platform_driver_register(&msm_compr_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+	platform_driver_unregister(&msm_compr_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("Compress Offload platform driver");
+MODULE_LICENSE("GPL v2");