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ASoc: msm: Refactor some codec functionality into separate files

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- 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>
This commit is contained in:
Harmandeep Singh 2012-05-26 09:39:25 -07:00 committed by Stephen Boyd
parent d760b2ccd2
commit dcd23d119c
10 changed files with 1358 additions and 75 deletions
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22
arch/arm/mach-msm/Kconfig
View file

@ -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

1
arch/arm/mach-msm/Makefile
View file

@ -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

5
arch/arm/mach-msm/include/mach/qdsp6v2/audio_acdb.h
View file

@ -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,

18
arch/arm/mach-msm/qdsp6v2/Makefile
View file

@ -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
obj-y += aac_in.o qcelp_in.o evrc_in.o amrnb_in.o audio_utils.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
ifdef CONFIG_ARCH_MSM9615
obj-y += rtac.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/

5
arch/arm/mach-msm/qdsp6v2/q6audio_common.h
View file

@ -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);

90
arch/arm/mach-msm/qdsp6v2/q6audio_v2.c Normal file
View file

@ -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);
}
}

112
arch/arm/mach-msm/qdsp6v2/q6audio_v2_aio.c Normal file
View file

@ -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;
}
}

46
arch/arm/mach-msm/qdsp6v2/rtac.c
View file

@ -22,8 +22,8 @@
#include <asm/atomic.h>
#include <mach/qdsp6v2/audio_acdb.h>
#include <mach/qdsp6v2/rtac.h>
#include <sound/q6asm.h>
#include <sound/q6adm.h>
#include "q6audio_common.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 "
"returned data, buf size = %d, "
"ret data = %d\n", __func__,
rtac_adm_user_buf_size, payload_size);
pr_err("%s: Buffer set not big enough for returned data, buf size = %d,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,"
"size = %d\n", __func__, payload_size);
pr_err("%s: Could not copy buffer to user, size = %d\n",
__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 "
"returned data, buf size = %d, "
"ret data = %d\n", __func__,
rtac_asm_user_buf_size, payload_size);
pr_err("%s: Buffer set not big enough for returned data, buf size = %d, 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,"
"size = %d\n", __func__, payload_size);
pr_err("%s: Could not copy buffer to user,size = %d\n",
__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 "
"returned data, buf size = %d, "
"ret data = %d\n", __func__,
rtac_voice_user_buf_size, payload_size);
pr_err("%s: Buffer set not big enough for returned data, buf size = %d, 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,"
"size = %d\n", __func__, payload_size);
pr_err("%s: Could not copy buffer to user,size = %d\n",
__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;
}

1019
arch/arm/mach-msm/qdsp6v2/rtac_v2.c Normal file
View file

File diff suppressed because it is too large Load diff

33
sound/soc/msm/Kconfig
View file

@ -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