ASoc: msm: Refactor some codec functionality into separate files

- Add all codec driver common functionality in common files.
- Add separate files for callback function definitions.
- Add header files according to the platform.

Change-Id: I906811043d4bb33571f719f79988fbdb89f5c385
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>

arch/arm/mach-msm/Kconfig

@@ -242,6 +242,9 @@ config ARCH_MSMCOPPER
 	select SPARSE_IRQ
 	select MSM_RPM_SMD
 	select REGULATOR
+	select MSM_QDSP6_APR
+	select MSM_QDSP6V2_CODECS
+	select MSM_AUDIO_QDSP6V2 if SND_SOC
 
 config ARCH_FSM9XXX
 	bool "FSM9XXX"
@@ -2173,6 +2176,15 @@ config MSM_QDSP6_CODECS
 	  used by audio driver to configure QDSP6's
 	  ASM, ADM and AFE.
 
+config MSM_QDSP6V2_CODECS
+	bool "Audio QDSP6V2 APR support"
+	depends on MSM_SMD
+	help
+	  Enable Audio codecs with APR IPC protocol support between
+	  application processor and QDSP6 for B-family. APR is
+	  used by audio driver to configure QDSP6's
+	  ASM, ADM and AFE.
+
 config MSM_AUDIO_QDSP6
         bool "QDSP6 HW Audio support"
         select SND_SOC_MSM_QDSP6_INTF
@@ -2181,6 +2193,16 @@ config MSM_AUDIO_QDSP6
           Enable HW audio support in QDSP6.
           QDSP6 can support HW encoder & decoder and audio processing
 
+config MSM_AUDIO_QDSP6V2
+        bool "QDSP6V2 HW Audio support"
+        select SND_SOC_MSM_QDSP6V2_INTF
+        help
+          Enable HW audio support in QDSP6V2.
+          QDSP6V2 can support HW encoder & decoder and
+          audio processing. It will enable support for
+          AAC, AMRNB, AMRWB, EVRC, MP3, QCELP among
+          others.
+
 config MSM_ULTRASOUND
 	bool "MSM ultrasound support"
 	depends on MSM_AUDIO_QDSP6

arch/arm/mach-msm/Makefile

@@ -155,6 +155,7 @@ obj-$(CONFIG_MSM_RPCSERVER_HANDSET) += rpc_server_handset.o
 obj-$(CONFIG_MSM_QDSP6) += qdsp6/
 obj-$(CONFIG_MSM8X60_AUDIO) += qdsp6v2/
 obj-$(CONFIG_MSM_AUDIO_QDSP6) += qdsp6v2/
+obj-$(CONFIG_MSM_AUDIO_QDSP6V2) += qdsp6v2/
 obj-$(CONFIG_MSM_HW3D) += hw3d.o
 obj-$(CONFIG_PM) += pm-boot.o
 obj-$(CONFIG_MSM_PM8X60) += pm-8x60.o pm-data.o

arch/arm/mach-msm/include/mach/qdsp6v2/audio_acdb.h

@@ -14,8 +14,11 @@
 #define _AUDIO_ACDB_H
 
 #include <linux/msm_audio_acdb.h>
+#ifdef CONFIG_ARCH_MSMCOPPER
+#include <sound/q6adm-v2.h>
+#else
 #include <sound/q6adm.h>
-
+#endif
 enum {
 	RX_CAL,
 	TX_CAL,

arch/arm/mach-msm/qdsp6v2/Makefile

@@ -1,4 +1,3 @@
-obj-y += rtac.o
 ifdef CONFIG_ARCH_MSM8X60
 obj-y += audio_dev_ctl.o
 obj-y += board-msm8x60-audio.o
@@ -13,10 +12,15 @@ obj-$(CONFIG_FB_MSM_HDMI_MSM_PANEL) += lpa_if_hdmi.o
 endif
 obj-$(CONFIG_MSM_QDSP6_APR) += apr.o apr_tal.o q6core.o dsp_debug.o
 obj-y += audio_acdb.o
-ifndef CONFIG_ARCH_MSM9615
+ifdef CONFIG_ARCH_MSM9615
-obj-y += aac_in.o qcelp_in.o evrc_in.o amrnb_in.o audio_utils.o
+obj-y += rtac.o
-obj-y += audio_wma.o audio_wmapro.o audio_aac.o audio_multi_aac.o audio_utils_aio.o
-obj-$(CONFIG_MSM_QDSP6_CODECS) += q6audio_v1.o q6audio_v1_aio.o
-obj-$(CONFIG_MSM_ULTRASOUND) += ultrasound/
-obj-y += audio_mp3.o audio_amrnb.o audio_amrwb.o audio_evrc.o audio_qcelp.o amrwb_in.o
 endif
+obj-$(CONFIG_MSM_QDSP6_CODECS) += aac_in.o qcelp_in.o evrc_in.o amrnb_in.o audio_utils.o
+obj-$(CONFIG_MSM_QDSP6_CODECS) += audio_wma.o audio_wmapro.o audio_aac.o audio_multi_aac.o audio_utils_aio.o
+obj-$(CONFIG_MSM_QDSP6_CODECS) += rtac.o q6audio_v1.o q6audio_v1_aio.o
+obj-$(CONFIG_MSM_QDSP6_CODECS) += audio_mp3.o audio_amrnb.o audio_amrwb.o audio_evrc.o audio_qcelp.o amrwb_in.o
+obj-$(CONFIG_MSM_QDSP6V2_CODECS) += aac_in.o qcelp_in.o evrc_in.o amrnb_in.o audio_utils.o
+obj-$(CONFIG_MSM_QDSP6V2_CODECS) += audio_wma.o audio_wmapro.o audio_aac.o audio_multi_aac.o audio_utils_aio.o
+obj-$(CONFIG_MSM_QDSP6V2_CODECS) += rtac_v2.o q6audio_v2.o q6audio_v2_aio.o
+obj-$(CONFIG_MSM_QDSP6V2_CODECS) += audio_mp3.o audio_amrnb.o audio_amrwb.o audio_evrc.o audio_qcelp.o amrwb_in.o
+obj-$(CONFIG_MSM_ULTRASOUND) += ultrasound/

arch/arm/mach-msm/qdsp6v2/q6audio_common.h

@@ -15,8 +15,13 @@
 #ifndef __Q6_AUDIO_COMMON_H__
 #define __Q6_AUDIO_COMMON_H__
 
+#ifdef CONFIG_ARCH_MSMCOPPER
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+#else
 #include <sound/apr_audio.h>
 #include <sound/q6asm.h>
+#endif
 
 void q6_audio_cb(uint32_t opcode, uint32_t token,
 		uint32_t *payload, void *priv);

arch/arm/mach-msm/qdsp6v2/q6audio_v2.c

@@ -0,0 +1,90 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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/module.h>
+#include <linux/fs.h>
+#include <linux/miscdevice.h>
+#include <linux/uaccess.h>
+#include <linux/sched.h>
+#include <linux/wait.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <asm/atomic.h>
+#include <asm/ioctls.h>
+#include "audio_utils.h"
+
+void q6asm_in_cb(uint32_t opcode, uint32_t token,
+		uint32_t *payload, void *priv)
+{
+	struct q6audio_in *audio = (struct q6audio_in *)priv;
+	unsigned long flags;
+
+	pr_debug("%s:session id %d: opcode[0x%x]\n", __func__,
+			audio->ac->session, opcode);
+
+	spin_lock_irqsave(&audio->dsp_lock, flags);
+	switch (opcode) {
+	case ASM_DATA_EVENT_READ_DONE_V2:
+		audio_in_get_dsp_frames(audio, token, payload);
+		break;
+	case ASM_DATA_EVENT_WRITE_DONE_V2:
+		atomic_inc(&audio->in_count);
+		wake_up(&audio->write_wait);
+		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		audio->eos_rsp = 1;
+		wake_up(&audio->read_wait);
+		break;
+	case ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2:
+		break;
+	case ASM_SESSION_EVENTX_OVERFLOW:
+		pr_err("%s:session id %d: ASM_SESSION_EVENT_TX_OVERFLOW\n",
+			__func__, audio->ac->session);
+		break;
+	default:
+		pr_debug("%s:session id %d: Ignore opcode[0x%x]\n", __func__,
+			audio->ac->session, opcode);
+		break;
+	}
+	spin_unlock_irqrestore(&audio->dsp_lock, flags);
+}
+
+void  audio_in_get_dsp_frames(void *priv,
+	uint32_t token,	uint32_t *payload)
+{
+	struct q6audio_in *audio = (struct q6audio_in *)priv;
+	uint32_t index;
+
+	index = token;
+	pr_debug("%s:session id %d: index=%d nr frames=%d offset[%d]\n",
+			__func__, audio->ac->session, token, payload[9],
+			payload[5]);
+	pr_debug("%s:session id %d: timemsw=%d lsw=%d\n", __func__,
+			audio->ac->session, payload[7], payload[6]);
+	pr_debug("%s:session id %d: uflags=0x%8x uid=0x%8x\n", __func__,
+			audio->ac->session, payload[8], payload[10]);
+	pr_debug("%s:session id %d: enc_framesotal_size=0x%8x\n", __func__,
+			audio->ac->session, payload[4]);
+
+	audio->out_frame_info[index][0] = payload[9];
+	audio->out_frame_info[index][1] = payload[5];
+
+	/* statistics of read */
+	atomic_add(payload[4], &audio->in_bytes);
+	atomic_add(payload[9], &audio->in_samples);
+
+	if (atomic_read(&audio->out_count) <= audio->str_cfg.buffer_count) {
+		atomic_inc(&audio->out_count);
+		wake_up(&audio->read_wait);
+	}
+}

arch/arm/mach-msm/qdsp6v2/q6audio_v2_aio.c

@@ -0,0 +1,112 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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/module.h>
+#include <linux/fs.h>
+#include <linux/miscdevice.h>
+#include <linux/uaccess.h>
+#include <linux/sched.h>
+#include <linux/wait.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <asm/atomic.h>
+#include <asm/ioctls.h>
+#include "audio_utils_aio.h"
+
+void q6_audio_cb(uint32_t opcode, uint32_t token,
+		uint32_t *payload, void *priv)
+{
+	struct q6audio_aio *audio = (struct q6audio_aio *)priv;
+
+	pr_debug("%s:opcode = %x token = 0x%x\n", __func__, opcode, token);
+	switch (opcode) {
+	case ASM_DATA_EVENT_WRITE_DONE_V2:
+	case ASM_DATA_EVENT_READ_DONE_V2:
+	case ASM_DATA_EVENT_RENDERED_EOS:
+	case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+	case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+	case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
+	case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY:
+		audio_aio_cb(opcode, token, payload, audio);
+		break;
+	default:
+		pr_debug("%s:Unhandled event = 0x%8x\n", __func__, opcode);
+		break;
+	}
+}
+
+void audio_aio_cb(uint32_t opcode, uint32_t token,
+		uint32_t *payload,  void *priv/*struct q6audio_aio *audio*/)
+{
+	struct q6audio_aio *audio = (struct q6audio_aio *)priv;
+	union msm_audio_event_payload e_payload;
+
+	switch (opcode) {
+	case ASM_DATA_EVENT_WRITE_DONE_V2:
+		pr_debug("%s[%p]:ASM_DATA_EVENT_WRITE_DONE token = 0x%x\n",
+			__func__, audio, token);
+		audio_aio_async_write_ack(audio, token, payload);
+		break;
+	case ASM_DATA_EVENT_READ_DONE_V2:
+		pr_debug("%s[%p]:ASM_DATA_EVENT_READ_DONE token = 0x%x\n",
+			__func__, audio, token);
+		audio_aio_async_read_ack(audio, token, payload);
+		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		/* EOS Handle */
+		pr_debug("%s[%p]:ASM_DATA_CMDRSP_EOS\n", __func__, audio);
+		if (audio->feedback) { /* Non-Tunnel mode */
+			audio->eos_rsp = 1;
+			/* propagate input EOS i/p buffer,
+			after receiving DSP acknowledgement */
+			if (audio->eos_flag &&
+				(audio->eos_write_payload.aio_buf.buf_addr)) {
+				audio_aio_post_event(audio,
+						AUDIO_EVENT_WRITE_DONE,
+						audio->eos_write_payload);
+				memset(&audio->eos_write_payload , 0,
+					sizeof(union msm_audio_event_payload));
+				audio->eos_flag = 0;
+			}
+		} else { /* Tunnel mode */
+			audio->eos_rsp = 1;
+			wake_up(&audio->write_wait);
+			wake_up(&audio->cmd_wait);
+		}
+		break;
+	case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+	case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+		pr_debug("%s[%p]:payload0[%x] payloa1d[%x]opcode= 0x%x\n",
+			__func__, audio, payload[0], payload[1], opcode);
+		break;
+	case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
+	case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY:
+
+		pr_debug("%s[%p]: ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY, payload[0]-sr = %d, payload[1]-chl = %d, payload[2] = %d, payload[3] = %d\n",
+					 __func__, audio, payload[0],
+					 payload[1], payload[2], payload[3]);
+
+		pr_debug("%s[%p]: ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY, sr(prev) = %d, chl(prev) = %d,",
+				__func__, audio, audio->pcm_cfg.sample_rate,
+				audio->pcm_cfg.channel_count);
+
+		audio->pcm_cfg.sample_rate = payload[0];
+		audio->pcm_cfg.channel_count = payload[1] & 0xFFFF;
+		e_payload.stream_info.chan_info = audio->pcm_cfg.channel_count;
+		e_payload.stream_info.sample_rate = audio->pcm_cfg.sample_rate;
+		audio_aio_post_event(audio, AUDIO_EVENT_STREAM_INFO, e_payload);
+		break;
+	default:
+		break;
+	}
+}

arch/arm/mach-msm/qdsp6v2/rtac.c

@@ -22,8 +22,8 @@
 #include <asm/atomic.h>
 #include <mach/qdsp6v2/audio_acdb.h>
 #include <mach/qdsp6v2/rtac.h>
-#include <sound/q6asm.h>
+#include "q6audio_common.h"
-#include <sound/q6adm.h>
+#include <sound/q6afe.h>
 
 #ifndef CONFIG_RTAC
 
@@ -400,10 +400,8 @@ void rtac_copy_adm_payload_to_user(void *payload, u32 payload_size)
 	memcpy(rtac_adm_buffer, &payload_size, sizeof(u32));
 	if (payload_size != 0) {
 		if (payload_size > rtac_adm_user_buf_size) {
-			pr_err("%s: Buffer set not big enough for "
+			pr_err("%s: Buffer set not big enough for returned data, buf size = %d,ret data = %d\n",
-				"returned data, buf size = %d, "
+				__func__, rtac_adm_user_buf_size, payload_size);
-				"ret data = %d\n", __func__,
-				rtac_adm_user_buf_size, payload_size);
 			goto done;
 		}
 		memcpy(rtac_adm_buffer + sizeof(u32), payload, payload_size);
@@ -443,7 +441,6 @@ u32 send_adm_apr(void *buf, u32 opcode)
 
 
 	if (payload_size > MAX_PAYLOAD_SIZE) {
-
 		pr_err("%s: Invalid payload size = %d\n",
 			__func__, payload_size);
 		goto done;
@@ -522,8 +519,8 @@ u32 send_adm_apr(void *buf, u32 opcode)
 	if (rtac_adm_payload_size != 0) {
 		if (copy_to_user(buf, rtac_adm_buffer,
 			rtac_adm_payload_size + sizeof(u32))) {
-			pr_err("%s: Could not copy buffer to user,"
+			pr_err("%s: Could not copy buffer to user, size = %d\n",
-				"size = %d\n", __func__, payload_size);
+				 __func__, payload_size);
 			goto done;
 		}
 	}
@@ -573,10 +570,8 @@ void rtac_copy_asm_payload_to_user(void *payload, u32 payload_size)
 	memcpy(rtac_asm_buffer, &payload_size, sizeof(u32));
 	if (payload_size) {
 		if (payload_size > rtac_asm_user_buf_size) {
-			pr_err("%s: Buffer set not big enough for "
+			pr_err("%s: Buffer set not big enough for returned data, buf size = %d, ret data = %d\n",
-				"returned data, buf size = %d, "
+			 __func__, rtac_asm_user_buf_size, payload_size);
-				"ret data = %d\n", __func__,
-				rtac_asm_user_buf_size, payload_size);
 			goto done;
 		}
 		memcpy(rtac_asm_buffer + sizeof(u32), payload, payload_size);
@@ -614,7 +609,6 @@ u32 send_rtac_asm_apr(void *buf, u32 opcode)
 	}
 
 	if (payload_size > MAX_PAYLOAD_SIZE) {
-
 		pr_err("%s: Invalid payload size = %d\n",
 			__func__, payload_size);
 		goto done;
@@ -692,8 +686,8 @@ u32 send_rtac_asm_apr(void *buf, u32 opcode)
 	if (rtac_asm_payload_size != 0) {
 		if (copy_to_user(buf, rtac_asm_buffer,
 			rtac_asm_payload_size + sizeof(u32))) {
-			pr_err("%s: Could not copy buffer to user,"
+			pr_err("%s: Could not copy buffer to user,size = %d\n",
-				"size = %d\n", __func__, payload_size);
+				 __func__, payload_size);
 			goto done;
 		}
 	}
@@ -715,7 +709,6 @@ err:
 void rtac_set_voice_handle(u32 mode, void *handle)
 {
 	pr_debug("%s\n", __func__);
-
 	mutex_lock(&rtac_voice_apr_mutex);
 	rtac_voice_apr_data[mode].apr_handle = handle;
 	mutex_unlock(&rtac_voice_apr_mutex);
@@ -743,10 +736,8 @@ void rtac_copy_voice_payload_to_user(void *payload, u32 payload_size)
 	memcpy(rtac_voice_buffer, &payload_size, sizeof(u32));
 	if (payload_size) {
 		if (payload_size > rtac_voice_user_buf_size) {
-			pr_err("%s: Buffer set not big enough for "
+			pr_err("%s: Buffer set not big enough for returned data, buf size = %d, ret data = %d\n",
-				"returned data, buf size = %d, "
+			 __func__, rtac_voice_user_buf_size, payload_size);
-				"ret data = %d\n", __func__,
-				rtac_voice_user_buf_size, payload_size);
 			goto done;
 		}
 		memcpy(rtac_voice_buffer + sizeof(u32), payload, payload_size);
@@ -860,8 +851,8 @@ u32 send_voice_apr(u32 mode, void *buf, u32 opcode)
 	if (rtac_voice_payload_size != 0) {
 		if (copy_to_user(buf, rtac_voice_buffer,
 				rtac_voice_payload_size + sizeof(u32))) {
-			pr_err("%s: Could not copy buffer to user,"
+			pr_err("%s: Could not copy buffer to user,size = %d\n",
-				"size = %d\n", __func__, payload_size);
+						 __func__, payload_size);
 			goto done;
 		}
 	}
@@ -972,7 +963,7 @@ static int __init rtac_init(void)
 	mutex_init(&rtac_adm_mutex);
 	mutex_init(&rtac_adm_apr_mutex);
 
-	rtac_adm_buffer = kmalloc(RTAC_BUF_SIZE, GFP_KERNEL);
+	rtac_adm_buffer = kzalloc(RTAC_BUF_SIZE, GFP_KERNEL);
 	if (rtac_adm_buffer == NULL) {
 		pr_err("%s: Could not allocate payload of size = %d\n",
 			__func__, RTAC_BUF_SIZE);
@@ -987,10 +978,11 @@ static int __init rtac_init(void)
 	}
 	mutex_init(&rtac_asm_apr_mutex);
 
-	rtac_asm_buffer = kmalloc(RTAC_BUF_SIZE, GFP_KERNEL);
+	rtac_asm_buffer = kzalloc(RTAC_BUF_SIZE, GFP_KERNEL);
 	if (rtac_asm_buffer == NULL) {
 		pr_err("%s: Could not allocate payload of size = %d\n",
 			__func__, RTAC_BUF_SIZE);
+		kzfree(rtac_adm_buffer);
 		goto nomem;
 	}
 
@@ -1004,10 +996,12 @@ static int __init rtac_init(void)
 	mutex_init(&rtac_voice_mutex);
 	mutex_init(&rtac_voice_apr_mutex);
 
-	rtac_voice_buffer = kmalloc(RTAC_BUF_SIZE, GFP_KERNEL);
+	rtac_voice_buffer = kzalloc(RTAC_BUF_SIZE, GFP_KERNEL);
 	if (rtac_voice_buffer == NULL) {
 		pr_err("%s: Could not allocate payload of size = %d\n",
 			__func__, RTAC_BUF_SIZE);
+		kzfree(rtac_adm_buffer);
+		kzfree(rtac_asm_buffer);
 		goto nomem;
 	}
 

sound/soc/msm/Kconfig

@@ -105,6 +105,16 @@ config SND_SOC_MSM_QDSP6_INTF
 	help
 	 To add support for SoC audio on MSM8960.
 
+config SND_SOC_MSM_QDSP6V2_INTF
+	bool "SoC Q6 audio driver for MSMCOPPER"
+	depends on MSM_QDSP6_APR
+	help
+	 To add support for SoC audio on MSMCOPPER.
+	 This will enable all the platform specific
+	 interactions towards DSP. It includes asm,
+	 adm and afe interfaces on the DSP.
+
+
 config SND_SOC_VOICE
 	bool "SoC Q6 voice driver for MSM8960"
 	depends on SND_SOC_MSM_QDSP6_INTF
@@ -119,6 +129,15 @@ config SND_SOC_QDSP6
 	help
 	 To add support for MSM QDSP6 Soc Audio.
 
+config SND_SOC_QDSP6V2
+	tristate "SoC ALSA audio driver for QDSP6V2"
+	select SND_SOC_MSM_QDSP6V2_INTF
+	help
+	 To add support for MSM QDSP6V2 Soc Audio.
+	 This will enable sound soc platform specific
+	 audio drivers. This includes q6asm, q6adm,
+	 q6afe interfaces to DSP using apr.
+
 config SND_SOC_MSM8960
 	tristate "SoC Machine driver for MSM8960 and APQ8064 boards"
 	depends on ARCH_MSM8960 || ARCH_APQ8064
@@ -134,6 +153,20 @@ config SND_SOC_MSM8960
 	help
 	 To add support for SoC audio on MSM8960 and APQ8064 boards
 
+config SND_SOC_MSM8974
+	tristate "SoC Machine driver for MSMCOPPER boards"
+	depends on ARCH_MSMCOPPER
+	select SND_SOC_QDSP6V2
+	select SND_SOC_MSM_STUB
+	select SND_SOC_MSM_HOSTLESS_PCM
+	select SND_DYNAMIC_MINORS
+	help
+	 To add support for SoC audio on MSMCOPPER.
+	 This will enable sound soc drivers which
+	 interfaces with DSP, also it will enable
+	 the machine drivers and the corresponding
+	 DAI-links.
+
 config SND_SOC_MDM9615
 	tristate "SoC Machine driver for MDM9615 boards"
 	depends on ARCH_MSM9615