A-Team_Digital_Solutions/android_kernel_motorola_sm6375
Watch
2
0
Fork
You've already forked android_kernel_motorola_sm6375
 0
mirror of https://github.com/BobTheBlinker/android_kernel_motorola_sm6375.git synced 2026-10-09 13:49:24 -04:00
Code Issues Projects Releases Packages Wiki Activity

ASoC: dsp: find CAL block using buffer number

Browse source 
Use FEdai_id to calculate the buffer number and
use the same to identify the right CAL block for
ASM CAL types.

Change-Id: Iab83892f5c2f2bc54c8fa97ac084d8a04586f712
Signed-off-by: Viraja Kommaraju <virajak@codeaurora.org>
Signed-off-by: Hongtao Peng <hongtaop@codeaurora.org>
This commit is contained in:
Viraja Kommaraju 2020-11-02 15:45:03 +05:30 • committed by Hongtao Peng
commit 83338b5ec0
8 changed files with 167 additions and 21 deletions
Download patch file Download diff file Expand all files Collapse all files

9
asoc/msm-compress-q6-v2.c
View file

@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
/* Copyright (c) 2012-2020, The Linux Foundation. All rights reserved.
/* Copyright (c) 2012-2021, The Linux Foundation. All rights reserved.
*/
@ -1609,6 +1609,9 @@ static int msm_compr_configure_dsp_for_playback
else if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE)
bits_per_sample = 32;
ac->fedai_id = soc_prtd->dai_link->id;
ac->stream_type = SNDRV_PCM_STREAM_PLAYBACK;
if (prtd->compr_passthr != LEGACY_PCM) {
ret = q6asm_open_write_compressed(ac, prtd->codec,
prtd->compr_passthr);
@ -1764,6 +1767,9 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
break;
}
prtd->audio_client->stream_type = SNDRV_PCM_STREAM_CAPTURE;
prtd->audio_client->fedai_id = soc_prtd->dai_link->id;
pr_debug("%s: stream_id %d bits_per_sample %d compr_passthr %d\n",
__func__, ac->stream_id, bits_per_sample,
prtd->compr_passthr);
@ -3029,6 +3035,7 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
pr_debug("%s: open_write stream_id %d bits_per_sample %d",
__func__, stream_id, bits_per_sample);
prtd->audio_client->fedai_id = (int)fe_id;
if (q6core_get_avcs_api_version_per_service(
APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >=
ADSP_ASM_API_VERSION_V2)

2
asoc/msm-pcm-loopback-v2.c
View file

@ -343,6 +343,8 @@ static int msm_pcm_open(struct snd_pcm_substream *substream)
mutex_unlock(&pcm->lock);
return -ENOMEM;
}
pcm->audio_client->fedai_id = rtd->dai_link->id;
pcm->session_id = pcm->audio_client->session;
pcm->audio_client->perf_mode = pdata->perf_mode;
ret = q6asm_open_loopback_v2(pcm->audio_client,

3
asoc/msm-pcm-q6-noirq.c
View file

@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
/* Copyright (c) 2016-2020, The Linux Foundation. All rights reserved.
/* Copyright (c) 2016-2021, The Linux Foundation. All rights reserved.
*/
#include <linux/init.h>
@ -367,6 +367,7 @@ static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
config.bufsz = params_buffer_bytes(params) / params_periods(params);
config.bufcnt = params_periods(params);
prtd->audio_client->fedai_id = soc_prtd->dai_link->id;
ret = q6asm_open_shared_io(prtd->audio_client, &config, dir,
use_default_chmap, chmap);
if (ret) {

6
asoc/msm-pcm-q6-v2.c
View file

@ -369,6 +369,9 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
prtd->audio_client->perf_mode = pdata->perf_mode;
pr_debug("%s: perf: %x\n", __func__, pdata->perf_mode);
prtd->audio_client->stream_type = SNDRV_PCM_STREAM_PLAYBACK;
prtd->audio_client->fedai_id = soc_prtd->dai_link->id;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S32_LE:
bits_per_sample = 32;
@ -532,6 +535,9 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
__func__, params_channels(params),
prtd->audio_client->perf_mode);
prtd->audio_client->stream_type = SNDRV_PCM_STREAM_CAPTURE;
prtd->audio_client->fedai_id = soc_prtd->dai_link->id;
if ((q6core_get_avcs_api_version_per_service(
APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >=
ADSP_ASM_API_VERSION_V2))

1
asoc/msm-transcode-loopback-q6-v2.c
View file

@ -540,6 +540,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream,
loopback_event_handler);
trans->session_id = trans->audio_client->session;
trans->audio_client->perf_mode = trans->sink.perf_mode;
trans->audio_client->fedai_id = rtd->dai_link->id;
ret = q6asm_open_transcode_loopback(trans->audio_client,
bit_width,
trans->source.codec_format,

162
dsp/q6asm.c
View file

@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
/*
* Copyright (c) 2012-2020, The Linux Foundation. All rights reserved.
* Copyright (c) 2012-2021, The Linux Foundation. All rights reserved.
* Author: Brian Swetland <swetland@google.com>
*
* This software is licensed under the terms of the GNU General Public
@ -128,7 +128,7 @@ static void q6asm_reset_buf_state(struct audio_client *ac);
void *q6asm_mmap_apr_reg(void);
static int q6asm_is_valid_session(struct apr_client_data *data, void *priv);
static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info);
static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info, struct audio_client *ac);
/* for ASM custom topology */
static struct cal_type_data *cal_data[ASM_MAX_CAL_TYPES];
@ -772,6 +772,93 @@ done:
return result;
}
static struct cal_block_data *q6asm_find_cal_by_app_type(int cal_index,
int app_type)
{
struct list_head *ptr, *next;
struct cal_block_data *cal_block = NULL;
struct audio_cal_info_audstrm *audstrm_info = NULL;
pr_debug("%s\n", __func__);
list_for_each_safe(ptr, next,
&cal_data[cal_index]->cal_blocks) {
cal_block = list_entry(ptr,
struct cal_block_data, list);
if (cal_index == ASM_AUDSTRM_CAL) {
audstrm_info = cal_block->cal_info;
if (audstrm_info->app_type == app_type)
return cal_block;
}
}
pr_err("%s: Can't find ASM Cal for cal_index %d app_type %d\n",
__func__, cal_index, app_type);
return NULL;
}
static bool is_q6asm_topology_cal_send_by_v6(int path, int usecase)
{
struct list_head *ptr, *next;
struct cal_block_data *cal_block = NULL;
int buffer_idx_asm;
buffer_idx_asm = path + MAX_PATH_TYPE * usecase + ASM_CAL_SEND_VERSION_OFFSET;
pr_debug("%s: path(%d), usecase(%d), buffer_idx_asm(%d)\n",
__func__, path, usecase, buffer_idx_asm);
list_for_each_safe(ptr, next,
&cal_data[ASM_TOPOLOGY_CAL]->cal_blocks) {
cal_block = list_entry(ptr,
struct cal_block_data, list);
if (cal_block != NULL) {
if (cal_block->buffer_number == buffer_idx_asm)
return true;
}
}
return false;
}
static struct cal_block_data *q6asm_find_cal_by_buf_number(int cal_index,
int app_type, int path, int usecase)
{
struct list_head *ptr, *next;
struct cal_block_data *cal_block = NULL;
struct audio_cal_info_audstrm *audstrm_info = NULL;
int buffer_idx_w_path = usecase;
pr_debug("%s\n", __func__);
if (cal_index == ASM_AUDSTRM_CAL)
buffer_idx_w_path = path + MAX_PATH_TYPE * usecase;
list_for_each_safe(ptr, next,
&cal_data[cal_index]->cal_blocks) {
cal_block = list_entry(ptr,
struct cal_block_data, list);
if (cal_index == ASM_AUDSTRM_CAL) {
audstrm_info = cal_block->cal_info;
if ((audstrm_info->app_type == app_type) &&
(cal_block->buffer_number == buffer_idx_w_path)) {
return cal_block;
}
} else {
if (cal_block->buffer_number == buffer_idx_w_path) {
return cal_block;
}
}
}
pr_err("%s: Can't find ASM Cal for cal_index %d app_type %d buffer_number %d\n",
__func__, cal_index, app_type, buffer_idx_w_path);
return NULL;
}
int q6asm_unmap_cal_data(int cal_type, struct cal_block_data *cal_block)
{
int ret = 0;
@ -3167,7 +3254,7 @@ static int __q6asm_open_read(struct audio_client *ac,
/* Stream prio : High, provide meta info with encoded frames */
open.src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open.preprocopo_id = cal_info.topology_id;
@ -3514,7 +3601,7 @@ static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
open.bits_per_sample = bits_per_sample;
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open.postprocopo_id = cal_info.topology_id;
if (ac->perf_mode != LEGACY_PCM_MODE)
@ -3785,7 +3872,7 @@ static int __q6asm_open_read_write(struct audio_client *ac, uint32_t rd_format,
open.mode_flags = is_meta_data_mode ? BUFFER_META_ENABLE : 0;
open.bits_per_sample = bits_per_sample;
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open.postprocopo_id = cal_info.topology_id;
open.postprocopo_id = overwrite_topology ?
@ -4010,7 +4097,7 @@ int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample)
open.src_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
open.sink_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open.audproc_topo_id = cal_info.topology_id;
ac->app_type = cal_info.app_type;
@ -4042,7 +4129,7 @@ int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample)
open.src_endpointype = 0;
open.sink_endpointype = 0;
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open.postprocopo_id = cal_info.topology_id;
ac->app_type = cal_info.app_type;
@ -4149,7 +4236,7 @@ int q6asm_open_transcode_loopback(struct audio_client *ac,
}
/* source endpoint : matrix */
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open.audproc_topo_id = cal_info.topology_id;
@ -4405,7 +4492,7 @@ int q6asm_open_shared_io(struct audio_client *ac,
open->endpoint_type = ASM_END_POINT_DEVICE_MATRIX;
open->topo_bits_per_sample = config->bits_per_sample;
rc = q6asm_get_asm_topology_apptype(&cal_info);
rc = q6asm_get_asm_topology_apptype(&cal_info, ac);
open->topo_id = cal_info.topology_id;
if (config->format == FORMAT_LINEAR_PCM)
@ -10998,8 +11085,10 @@ done:
* Hence it cannot be reused or resent unless the flag
* is reset.
*/
static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info)
static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info, struct audio_client *ac)
{
int path = 0;
int cal_buf_index = 0;
struct cal_block_data *cal_block = NULL;
cal_info->topology_id = DEFAULT_POPP_TOPOLOGY;
@ -11007,11 +11096,38 @@ static int q6asm_get_asm_topology_apptype(struct q6asm_cal_info *cal_info)
if (cal_data[ASM_TOPOLOGY_CAL] == NULL)
goto done;
if (!ac) {
pr_err("%s: Audio client is NULL\n", __func__);
goto done;
}
if (ac->stream_type == SNDRV_PCM_STREAM_PLAYBACK)
path = RX_DEVICE;
else
path = TX_DEVICE;
mutex_lock(&cal_data[ASM_TOPOLOGY_CAL]->lock);
cal_block = cal_utils_get_only_cal_block(cal_data[ASM_TOPOLOGY_CAL]);
if (cal_block == NULL || cal_utils_is_cal_stale(cal_block))
goto unlock;
if (is_q6asm_topology_cal_send_by_v6(path, ac->fedai_id)) {
pr_debug("%s: ASM topology cal block is send by acdb_loader_send_audio_cal_v6!\n",
__func__);
cal_buf_index = path + MAX_PATH_TYPE * ac->fedai_id + ASM_CAL_SEND_VERSION_OFFSET;
cal_block = q6asm_find_cal_by_buf_number(ASM_TOPOLOGY_CAL,
ac->app_type,
path,
cal_buf_index);
if (cal_block == NULL) {
pr_err("%s: Couldn't find cal_block with buf_number!\n", __func__);
goto unlock;
}
} else {
cal_block = q6asm_find_cal_by_buf_number(ASM_TOPOLOGY_CAL, 0, 0, ac->fedai_id);
if (cal_block == NULL) {
pr_debug("%s: Couldn't find cal_block with buf_number, re-routing "
"search using CAL type only\n", __func__);
cal_block = cal_utils_get_only_cal_block(cal_data[ASM_TOPOLOGY_CAL]);
}
if (cal_block == NULL || cal_utils_is_cal_stale(cal_block))
goto unlock;
}
cal_info->topology_id = ((struct audio_cal_info_asm_top *)
cal_block->cal_info)->topology;
cal_info->app_type = ((struct audio_cal_info_asm_top *)
@ -11041,6 +11157,7 @@ int q6asm_send_cal(struct audio_client *ac)
struct cal_block_data *cal_block = NULL;
struct mem_mapping_hdr mem_hdr;
u32 payload_size = 0;
int path = 0;
int rc = -EINVAL;
pr_debug("%s:\n", __func__);
@ -11061,15 +11178,24 @@ int q6asm_send_cal(struct audio_client *ac)
goto done;
}
if (ac->stream_type == SNDRV_PCM_STREAM_PLAYBACK)
path = RX_DEVICE;
else
path = TX_DEVICE;
memset(&mem_hdr, 0, sizeof(mem_hdr));
mutex_lock(&cal_data[ASM_AUDSTRM_CAL]->lock);
cal_block = cal_utils_get_only_cal_block(cal_data[ASM_AUDSTRM_CAL]);
cal_block = q6asm_find_cal_by_buf_number(ASM_AUDSTRM_CAL, ac->app_type, path, ac->fedai_id);
if (cal_block == NULL) {
pr_err("%s: cal_block is NULL\n",
__func__);
goto unlock;
pr_debug("%s: Couldn't find cal_block with buf_number, re-routing "
"search using app_type only\n", __func__);
cal_block = q6asm_find_cal_by_app_type(ASM_AUDSTRM_CAL, ac->app_type);
if (cal_block == NULL) {
pr_err("%s: cal_block is NULL\n",
__func__);
goto unlock;
}
}
if (cal_utils_is_cal_stale(cal_block)) {
rc = 0; /* not error case */
pr_debug("%s: cal_block is stale\n",

4
include/dsp/q6asm-v2.h
View file

@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
* Copyright (c) 2012-2019, The Linux Foundation. All rights reserved.
* Copyright (c) 2012-2019, 2021 The Linux Foundation. All rights reserved.
*/
#ifndef __Q6_ASM_V2_H__
#define __Q6_ASM_V2_H__
@ -248,6 +248,8 @@ struct audio_client {
/* shared io */
struct audio_buffer shared_pos_buf;
struct shared_io_config config;
int stream_type;
int fedai_id;
};
struct q6asm_cal_info {

1
include/uapi/audio/linux/msm_audio_calibration.h
View file

@ -137,6 +137,7 @@ enum {
};
#define MAX_IOCTL_CMD_SIZE 512
#define ASM_CAL_SEND_VERSION_OFFSET 256
/* common structures */