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-rw-r--r--sound/soc/msm/Kconfig1
-rw-r--r--sound/soc/msm/msm-dai-fe.c18
-rw-r--r--sound/soc/msm/msm8996.c21
-rw-r--r--sound/soc/msm/qdsp6v2/Makefile1
-rw-r--r--sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c4
-rw-r--r--sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c32
-rw-r--r--sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c827
-rw-r--r--sound/soc/msm/qdsp6v2/msm-pcm-routing-devdep.c2
-rw-r--r--sound/soc/msm/qdsp6v2/q6adm.c1
-rw-r--r--sound/soc/msm/qdsp6v2/q6voice.c12
10 files changed, 51 insertions, 868 deletions
diff --git a/sound/soc/msm/Kconfig b/sound/soc/msm/Kconfig
index 0d1e73944a2f..9f3097ffed89 100644
--- a/sound/soc/msm/Kconfig
+++ b/sound/soc/msm/Kconfig
@@ -83,6 +83,7 @@ config SND_SOC_CPE
config SND_SOC_MSM8996
tristate "SoC Machine driver for MSM8996 boards"
depends on ARCH_MSM8996
+ select SND_SOC_COMPRESS
select SND_SOC_QDSP6V2
select SND_SOC_MSM_STUB
select SND_SOC_MSM_HOSTLESS_PCM
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c
index f092f34965d4..2586185a627e 100644
--- a/sound/soc/msm/msm-dai-fe.c
+++ b/sound/soc/msm/msm-dai-fe.c
@@ -239,7 +239,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 48000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia4",
.probe = fe_dai_probe,
},
@@ -315,7 +315,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia7",
.probe = fe_dai_probe,
},
@@ -2132,7 +2132,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia10",
.probe = fe_dai_probe,
},
@@ -2150,7 +2150,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia11",
.probe = fe_dai_probe,
},
@@ -2168,7 +2168,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia12",
.probe = fe_dai_probe,
},
@@ -2186,7 +2186,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia13",
.probe = fe_dai_probe,
},
@@ -2204,7 +2204,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia14",
.probe = fe_dai_probe,
},
@@ -2222,7 +2222,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia15",
.probe = fe_dai_probe,
},
@@ -2240,7 +2240,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.rate_max = 192000,
},
.ops = &msm_fe_Multimedia_dai_ops,
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.name = "MultiMedia16",
.probe = fe_dai_probe,
},
diff --git a/sound/soc/msm/msm8996.c b/sound/soc/msm/msm8996.c
index 2525ac83e911..e182b0bba13c 100644
--- a/sound/soc/msm/msm8996.c
+++ b/sound/soc/msm/msm8996.c
@@ -315,9 +315,8 @@ exit:
static void msm8996_ext_control(struct snd_soc_codec *codec)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
- mutex_lock(&codec->mutex);
pr_debug("%s: msm8996_spk_control = %d", __func__,
msm8996_spk_control);
if (msm8996_spk_control == MSM8996_SPK_ON) {
@@ -327,7 +326,6 @@ static void msm8996_ext_control(struct snd_soc_codec *codec)
snd_soc_dapm_disable_pin(dapm, "Lineout_1 amp");
snd_soc_dapm_disable_pin(dapm, "Lineout_2 amp");
}
- mutex_unlock(&codec->mutex);
snd_soc_dapm_sync(dapm);
}
@@ -357,7 +355,7 @@ static int msm8996_set_spk(struct snd_kcontrol *kcontrol,
static int msm8996_hifi_ctrl(struct snd_soc_codec *codec)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
struct snd_soc_card *card = codec->component.card;
struct msm8996_asoc_mach_data *pdata =
snd_soc_card_get_drvdata(card);
@@ -368,7 +366,6 @@ static int msm8996_hifi_ctrl(struct snd_soc_codec *codec)
pr_err("%s: hph_en1_gpio is invalid\n", __func__);
return -EINVAL;
}
- mutex_lock(&codec->mutex);
if (msm_hifi_control == MSM8996_HIFI_ON) {
gpio_direction_output(pdata->hph_en1_gpio, 1);
/* 5msec delay needed as per HW requirement */
@@ -376,7 +373,6 @@ static int msm8996_hifi_ctrl(struct snd_soc_codec *codec)
} else {
gpio_direction_output(pdata->hph_en1_gpio, 0);
}
- mutex_unlock(&codec->mutex);
snd_soc_dapm_sync(dapm);
return 0;
}
@@ -469,13 +465,15 @@ static int msm_snd_enable_codec_ext_clk(struct snd_soc_codec *codec,
static int msm8996_mclk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
pr_debug("%s: event = %d\n", __func__, event);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- return msm_snd_enable_codec_ext_clk(w->codec, 1, true);
+ return msm_snd_enable_codec_ext_clk(codec, 1, true);
case SND_SOC_DAPM_POST_PMD:
- return msm_snd_enable_codec_ext_clk(w->codec, 0, true);
+ return msm_snd_enable_codec_ext_clk(codec, 0, true);
}
return 0;
}
@@ -483,7 +481,8 @@ static int msm8996_mclk_event(struct snd_soc_dapm_widget *w,
static int msm_hifi_ctrl_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
- struct snd_soc_card *card = w->codec->component.card;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct snd_soc_card *card = codec->component.card;
struct msm8996_asoc_mach_data *pdata =
snd_soc_card_get_drvdata(card);
int ret = 0;
@@ -1566,7 +1565,7 @@ static int msm_audrx_init(struct snd_soc_pcm_runtime *rtd)
int err;
void *config_data;
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_pcm_runtime *rtd_aux = rtd->card->rtd_aux;
@@ -3095,7 +3094,7 @@ static int msm8996_wsa881x_init(struct snd_soc_component *component)
return -EINVAL;
}
- dapm = &codec->dapm;
+ dapm = snd_soc_codec_get_dapm(codec);
if (!strcmp(component->name_prefix, "SpkrLeft")) {
dev_dbg(codec->dev, "%s: setting left ch map to codec %s\n",
diff --git a/sound/soc/msm/qdsp6v2/Makefile b/sound/soc/msm/qdsp6v2/Makefile
index 7f58070f5ce8..b2824d3e5759 100644
--- a/sound/soc/msm/qdsp6v2/Makefile
+++ b/sound/soc/msm/qdsp6v2/Makefile
@@ -1,6 +1,5 @@
snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o \
msm-compress-q6-v2.o msm-compr-q6-v2.o \
- msm-pcm-lpa-v2.o \
msm-pcm-afe-v2.o msm-pcm-voip-v2.o \
msm-pcm-voice-v2.o msm-dai-q6-hdmi-v2.o \
msm-lsm-client.o msm-pcm-host-voice-v2.o \
diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c
index 2ae99eb5573a..9c57adb863d4 100644
--- a/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c
@@ -254,12 +254,12 @@ static int msm_dai_q6_hdmi_dai_probe(struct snd_soc_dai *dai)
kcontrol = &hdmi_config_controls[0];
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(kcontrol, dai_data));
kcontrol = &hdmi_config_controls[1];
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(kcontrol, dai_data));
dapm = snd_soc_component_get_dapm(dai->component);
diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
index f3dffd8a57f2..cd44237b40b5 100644
--- a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
@@ -1246,7 +1246,7 @@ static int msm_dai_q6_spdif_dai_probe(struct snd_soc_dai *dai)
kcontrol = &spdif_config_controls[1];
dapm = snd_soc_component_get_dapm(dai->component);
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(kcontrol, dai_data));
memset(&intercon, 0 , sizeof(intercon));
@@ -1902,22 +1902,22 @@ static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
switch (dai->id) {
case SLIMBUS_4_TX:
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(&sb_config_controls[0],
dai_data));
break;
case SLIMBUS_2_RX:
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(&sb_config_controls[1],
dai_data));
break;
case RT_PROXY_DAI_001_RX:
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(&rt_proxy_config_controls[0],
dai_data));
break;
case RT_PROXY_DAI_001_TX:
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(&rt_proxy_config_controls[1],
dai_data));
break;
@@ -2799,7 +2799,7 @@ static int msm_dai_q6_dai_mi2s_probe(struct snd_soc_dai *dai)
if (ctrl) {
kcontrol = snd_ctl_new1(ctrl,
&mi2s_dai_data->rx_dai.mi2s_dai_data);
- rc = snd_ctl_add(dai->card->snd_card, kcontrol);
+ rc = snd_ctl_add(dai->component->card->snd_card, kcontrol);
if (IS_ERR_VALUE(rc)) {
dev_err(dai->dev, "%s: err add RX fmt ctl DAI = %s\n",
@@ -2825,13 +2825,14 @@ static int msm_dai_q6_dai_mi2s_probe(struct snd_soc_dai *dai)
}
if (ctrl) {
- rc = snd_ctl_add(dai->card->snd_card,
+ rc = snd_ctl_add(dai->component->card->snd_card,
snd_ctl_new1(ctrl,
&mi2s_dai_data->tx_dai.mi2s_dai_data));
if (IS_ERR_VALUE(rc)) {
if (kcontrol)
- snd_ctl_remove(dai->card->snd_card, kcontrol);
+ snd_ctl_remove(dai->component->card->snd_card,
+ kcontrol);
dev_err(dai->dev, "%s: err add TX fmt ctl DAI = %s\n",
__func__, dai->name);
}
@@ -4932,7 +4933,8 @@ static int msm_dai_q6_dai_tdm_probe(struct snd_soc_dai *dai)
if (data_format_ctrl) {
data_format_kcontrol = snd_ctl_new1(data_format_ctrl,
tdm_dai_data);
- rc = snd_ctl_add(dai->card->snd_card, data_format_kcontrol);
+ rc = snd_ctl_add(dai->component->card->snd_card,
+ data_format_kcontrol);
if (IS_ERR_VALUE(rc)) {
dev_err(dai->dev, "%s: err add data format ctrl DAI = %s\n",
@@ -4944,11 +4946,12 @@ static int msm_dai_q6_dai_tdm_probe(struct snd_soc_dai *dai)
if (header_type_ctrl) {
header_type_kcontrol = snd_ctl_new1(header_type_ctrl,
tdm_dai_data);
- rc = snd_ctl_add(dai->card->snd_card, header_type_kcontrol);
+ rc = snd_ctl_add(dai->component->card->snd_card,
+ header_type_kcontrol);
if (IS_ERR_VALUE(rc)) {
if (data_format_kcontrol)
- snd_ctl_remove(dai->card->snd_card,
+ snd_ctl_remove(dai->component->card->snd_card,
data_format_kcontrol);
dev_err(dai->dev, "%s: err add header type ctrl DAI = %s\n",
__func__, dai->name);
@@ -4959,14 +4962,15 @@ static int msm_dai_q6_dai_tdm_probe(struct snd_soc_dai *dai)
if (header_ctrl) {
header_kcontrol = snd_ctl_new1(header_ctrl,
tdm_dai_data);
- rc = snd_ctl_add(dai->card->snd_card, header_kcontrol);
+ rc = snd_ctl_add(dai->component->card->snd_card,
+ header_kcontrol);
if (IS_ERR_VALUE(rc)) {
if (header_type_kcontrol)
- snd_ctl_remove(dai->card->snd_card,
+ snd_ctl_remove(dai->component->card->snd_card,
header_type_kcontrol);
if (data_format_kcontrol)
- snd_ctl_remove(dai->card->snd_card,
+ snd_ctl_remove(dai->component->card->snd_card,
data_format_kcontrol);
dev_err(dai->dev, "%s: err add header ctrl DAI = %s\n",
__func__, dai->name);
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
deleted file mode 100644
index f023f3a04048..000000000000
--- a/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
+++ /dev/null
@@ -1,827 +0,0 @@
-/* Copyright (c) 2012-2015, 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/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/pcm_params.h>
-#include <asm/dma.h>
-#include <linux/dma-mapping.h>
-
-#include <linux/of_device.h>
-#include <linux/msm_audio_ion.h>
-
-#include <sound/compress_params.h>
-#include <sound/compress_offload.h>
-#include <sound/compress_driver.h>
-#include <sound/timer.h>
-#include <sound/pcm_params.h>
-
-#include "msm-pcm-q6-v2.h"
-#include "msm-pcm-routing-v2.h"
-#include <sound/pcm.h>
-#include <sound/tlv.h>
-
-#define LPA_LR_VOL_MAX_STEPS 0x20002000
-
-const DECLARE_TLV_DB_LINEAR(lpa_rx_vol_gain, 0,
- LPA_LR_VOL_MAX_STEPS);
-static struct audio_locks the_locks;
-
-static struct snd_pcm_hardware msm_pcm_hardware = {
- .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 |
- SNDRV_PCM_RATE_KNOT,
- .rate_min = 8000,
- .rate_max = 192000,
- .channels_min = 1,
- .channels_max = 2,
- .buffer_bytes_max = 1024 * 1024,
- .period_bytes_min = 128 * 1024,
- .period_bytes_max = 256 * 1024,
- .periods_min = 4,
- .periods_max = 8,
- .fifo_size = 0,
-};
-
-/* Conventional and unconventional sample rate supported */
-static unsigned int supported_sample_rates[] = {
- 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
- 96000, 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 msm_audio *prtd = priv;
- struct snd_pcm_substream *substream = prtd->substream;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct audio_aio_write_param param;
- struct audio_buffer *buf = NULL;
- struct output_meta_data_st output_meta_data;
- unsigned long flag = 0;
- int i = 0;
-
- memset(&output_meta_data, 0x0, sizeof(struct output_meta_data_st));
- spin_lock_irqsave(&the_locks.event_lock, flag);
- switch (opcode) {
- case ASM_DATA_EVENT_WRITE_DONE_V2: {
- uint32_t *ptrmem = (uint32_t *)&param;
- dma_addr_t temp;
- pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n");
- pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
- prtd->pcm_irq_pos += prtd->pcm_count;
- if (atomic_read(&prtd->start))
- snd_pcm_period_elapsed(substream);
- else
- if (substream->timer_running)
- snd_timer_interrupt(substream->timer, 1);
-
- atomic_inc(&prtd->out_count);
- wake_up(&the_locks.write_wait);
- if (!atomic_read(&prtd->start)) {
- atomic_set(&prtd->pending_buffer, 1);
- break;
- } else
- atomic_set(&prtd->pending_buffer, 0);
-
- buf = prtd->audio_client->port[IN].buf;
- if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
- runtime->render_flag |= SNDRV_RENDER_STOPPED;
- pr_info("%s:lpa driver underrun\n", __func__);
- break;
- }
- pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
- __func__, prtd->pcm_count, prtd->out_head);
- temp = buf[0].phys + (prtd->out_head * prtd->pcm_count);
- pr_debug("%s:writing buffer[%d] from 0x%pa\n",
- __func__, prtd->out_head, &temp);
- if (prtd->meta_data_mode) {
- memcpy(&output_meta_data, (char *)(buf->data +
- prtd->out_head * prtd->pcm_count),
- sizeof(struct output_meta_data_st));
- param.len = output_meta_data.frame_size;
- } else {
- param.len = prtd->pcm_count;
- }
- pr_debug("meta_data_length: %d, frame_length: %d\n",
- output_meta_data.meta_data_length,
- output_meta_data.frame_size);
- param.paddr = temp +
- output_meta_data.meta_data_length;
- param.msw_ts = output_meta_data.timestamp_msw;
- param.lsw_ts = output_meta_data.timestamp_lsw;
- param.flags = NO_TIMESTAMP;
- param.uid = prtd->session_id;
- for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
- i++, ++ptrmem)
- pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
- if (q6asm_async_write(prtd->audio_client,
- &param) < 0)
- pr_err("%s:q6asm_async_write failed\n",
- __func__);
- else
- prtd->out_head =
- (prtd->out_head + 1) & (runtime->periods - 1);
- atomic_set(&prtd->pending_buffer, 0);
- break;
- }
- case ASM_DATA_EVENT_RENDERED_EOS:
- pr_debug("ASM_DATA_CMDRSP_EOS\n");
- prtd->cmd_ack = 1;
- wake_up(&the_locks.eos_wait);
- break;
- case APR_BASIC_RSP_RESULT: {
- switch (payload[0]) {
- case ASM_SESSION_CMD_RUN_V2: {
- if (!atomic_read(&prtd->pending_buffer))
- break;
- pr_debug("%s:writing %d bytes of buffer to dsp\n",
- __func__, prtd->pcm_count);
- buf = prtd->audio_client->port[IN].buf;
- pr_debug("%s:writing buffer[%d] from 0x%08x\n",
- __func__, prtd->out_head,
- ((unsigned int)buf[0].phys +
- (prtd->out_head * prtd->pcm_count)));
- if (prtd->meta_data_mode) {
- memcpy(&output_meta_data, (char *)(buf->data +
- prtd->out_head * prtd->pcm_count),
- sizeof(struct output_meta_data_st));
- param.len = output_meta_data.frame_size;
- } else {
- param.len = prtd->pcm_count;
- }
- param.paddr = buf[prtd->out_head].phys +
- output_meta_data.meta_data_length;
- param.msw_ts = output_meta_data.timestamp_msw;
- param.lsw_ts = output_meta_data.timestamp_lsw;
- param.flags = NO_TIMESTAMP;
- param.uid = prtd->session_id;
- if (q6asm_async_write(prtd->audio_client,
- &param) < 0)
- pr_err("%s:q6asm_async_write failed\n",
- __func__);
- else
- prtd->out_head =
- (prtd->out_head + 1)
- & (runtime->periods - 1);
- atomic_set(&prtd->pending_buffer, 0);
- }
- break;
- case ASM_STREAM_CMD_FLUSH:
- pr_debug("ASM_STREAM_CMD_FLUSH\n");
- prtd->cmd_ack = 1;
- wake_up(&the_locks.eos_wait);
- break;
- default:
- break;
- }
- break;
- }
- case RESET_EVENTS:
- pr_debug("%s RESET_EVENTS\n", __func__);
- prtd->cmd_ack = 1;
- prtd->reset_event = true;
- wake_up(&the_locks.eos_wait);
- break;
- default:
- pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
- break;
- }
- spin_unlock_irqrestore(&the_locks.event_lock, flag);
-}
-
-static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
- int ret;
- uint16_t bits_per_sample = 16;
- u32 io_mode = ASYNC_IO_MODE;
-
- pr_debug("%s\n", __func__);
- prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
- prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
- prtd->pcm_irq_pos = 0;
- /* rate and channels are sent to audio driver */
- prtd->samp_rate = runtime->rate;
- prtd->channel_mode = runtime->channels;
- prtd->out_head = 0;
- if (prtd->enabled)
- return 0;
-
- switch (runtime->format) {
- case SNDRV_PCM_FORMAT_S16_LE:
- bits_per_sample = 16;
- break;
-
- case SNDRV_PCM_FORMAT_S24_LE:
- bits_per_sample = 24;
- break;
- }
-
- if (prtd->meta_data_mode)
- io_mode |= COMPRESSED_IO;
-
- ret = q6asm_set_io_mode(prtd->audio_client, io_mode);
- if (ret < 0) {
- pr_err("%s: Set IO mode failed\n", __func__);
- q6asm_audio_client_free(prtd->audio_client);
- prtd->audio_client = NULL;
- return -ENOMEM;
- }
-
- ret = q6asm_media_format_block_pcm_format_support(
- prtd->audio_client, runtime->rate,
- runtime->channels, bits_per_sample);
- if (ret < 0)
- pr_debug("%s: CMD Format block failed\n", __func__);
-
- atomic_set(&prtd->out_count, runtime->periods);
- prtd->enabled = 1;
- prtd->cmd_ack = 0;
- prtd->cmd_interrupt = 0;
-
- return 0;
-}
-
-static int msm_pcm_restart(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
- struct audio_aio_write_param param;
- struct audio_buffer *buf = NULL;
- struct output_meta_data_st output_meta_data;
-
- pr_debug("%s: restart\n", __func__);
- memset(&output_meta_data, 0x0, sizeof(struct output_meta_data_st));
- if (runtime->render_flag & SNDRV_RENDER_STOPPED) {
- buf = prtd->audio_client->port[IN].buf;
- pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
- __func__, prtd->pcm_count, prtd->out_head);
- pr_debug("%s:writing buffer[%d] from 0x%08x\n",
- __func__, prtd->out_head,
- ((unsigned int)buf[0].phys +
- (prtd->out_head * prtd->pcm_count)));
- if (prtd->meta_data_mode) {
- memcpy(&output_meta_data, (char *)(buf->data +
- prtd->out_head * prtd->pcm_count),
- sizeof(struct output_meta_data_st));
- param.len = output_meta_data.frame_size;
- } else {
- param.len = prtd->pcm_count;
- }
- pr_debug("meta_data_length: %d, frame_length: %d\n",
- output_meta_data.meta_data_length,
- output_meta_data.frame_size);
- pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
- output_meta_data.timestamp_msw,
- output_meta_data.timestamp_lsw);
- param.paddr = (buf[0].phys +
- (prtd->out_head * prtd->pcm_count) +
- output_meta_data.meta_data_length);
- param.msw_ts = output_meta_data.timestamp_msw;
- param.lsw_ts = output_meta_data.timestamp_lsw;
- param.flags = NO_TIMESTAMP;
- param.uid = prtd->session_id;
- if (q6asm_async_write(prtd->audio_client, &param) < 0)
- pr_err("%s:q6asm_async_write failed\n",
- __func__);
- else
- prtd->out_head =
- (prtd->out_head + 1) & (runtime->periods - 1);
- runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
- return 0;
- }
- return 0;
-}
-
-static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- int ret = 0;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
- pr_debug("%s\n", __func__);
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- prtd->pcm_irq_pos = 0;
- atomic_set(&prtd->pending_buffer, 1);
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- pr_debug("SNDRV_PCM_TRIGGER_START\n");
- q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
- atomic_set(&prtd->start, 1);
- atomic_set(&prtd->stop, 0);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
- atomic_set(&prtd->start, 0);
- atomic_set(&prtd->stop, 1);
- runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
- break;
- break;
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
- q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
- atomic_set(&prtd->start, 0);
- runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
- break;
- default:
- ret = -EINVAL;
- break;
- }
-
- return ret;
-}
-
-static int msm_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd;
- int ret = 0;
-
- pr_debug("%s\n", __func__);
- prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
- if (prtd == NULL) {
- pr_err("Failed to allocate memory for msm_audio\n");
- return -ENOMEM;
- }
- runtime->hw = msm_pcm_hardware;
- prtd->substream = substream;
- prtd->reset_event = false;
- runtime->render_flag = SNDRV_DMA_MODE;
- prtd->audio_client = q6asm_audio_client_alloc(
- (app_cb)event_handler, prtd);
- if (!prtd->audio_client) {
- pr_debug("%s: Could not allocate memory\n", __func__);
- kfree(prtd);
- return -ENOMEM;
- }
-
- prtd->meta_data_mode = false;
- /* Capture path */
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
- return -EPERM;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_sample_rates);
- if (ret < 0)
- pr_debug("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_debug("snd_pcm_hw_constraint_integer failed\n");
-
- prtd->dsp_cnt = 0;
- atomic_set(&prtd->pending_buffer, 1);
- atomic_set(&prtd->stop, 1);
- runtime->private_data = prtd;
- return 0;
-}
-
-static int lpa_set_volume(struct msm_audio *prtd, uint32_t volume)
-{
- int rc = 0;
- if (prtd && prtd->audio_client) {
- rc = q6asm_set_lrgain(prtd->audio_client,
- (volume >> 16) & 0xFFFF, volume & 0xFFFF);
- if (rc < 0) {
- pr_err("%s: Send Volume command failed rc=%d\n",
- __func__, rc);
- } else {
- prtd->volume = volume;
- }
- }
- return rc;
-}
-
-static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct msm_audio *prtd = runtime->private_data;
- int dir = 0;
- int rc = 0;
-
- /*
- If routing is still enabled, we need to issue EOS to
- the DSP
- To issue EOS to dsp, we need to be run state otherwise
- EOS is not honored.
- */
- if (msm_routing_check_backend_enabled(soc_prtd->dai_link->be_id) &&
- (!atomic_read(&prtd->stop))) {
- rc = q6asm_run(prtd->audio_client, 0, 0, 0);
- atomic_set(&prtd->pending_buffer, 0);
- prtd->cmd_ack = 0;
- q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
- pr_debug("%s\n", __func__);
- rc = wait_event_timeout(the_locks.eos_wait,
- prtd->cmd_ack, 5 * HZ);
- if (!rc)
- pr_err("EOS cmd timeout\n");
- prtd->pcm_irq_pos = 0;
- }
-
- if (prtd->audio_client) {
- dir = IN;
- atomic_set(&prtd->pending_buffer, 0);
-
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
- q6asm_audio_client_buf_free_contiguous(dir,
- prtd->audio_client);
-
- atomic_set(&prtd->stop, 1);
- q6asm_audio_client_free(prtd->audio_client);
- pr_debug("%s\n", __func__);
- }
- msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
- SNDRV_PCM_STREAM_PLAYBACK);
-
- prtd->meta_data_mode = false;
-
- pr_debug("%s\n", __func__);
- kfree(prtd);
-
- return 0;
-}
-
-static int msm_pcm_close(struct snd_pcm_substream *substream)
-{
- int ret = 0;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- ret = msm_pcm_playback_close(substream);
- return ret;
-}
-
-static int msm_pcm_prepare(struct snd_pcm_substream *substream)
-{
- int ret = 0;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- ret = msm_pcm_playback_prepare(substream);
- return ret;
-}
-
-static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
-{
-
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
-
- if (prtd->pcm_irq_pos >= prtd->pcm_size)
- prtd->pcm_irq_pos = 0;
- pr_debug("%s: pcm_irq_pos = %d\n", __func__, prtd->pcm_irq_pos);
- return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
-}
-
-static int msm_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
- struct audio_client *ac = prtd->audio_client;
- struct audio_port_data *apd = ac->port;
- struct audio_buffer *ab;
- int dir = -1;
-
- prtd->mmap_flag = 1;
- runtime->render_flag = SNDRV_NON_DMA_MODE;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dir = IN;
- else
- dir = OUT;
- ab = &(apd[dir].buf[0]);
-
- return msm_audio_ion_mmap(ab, vma);
-}
-
-static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
- struct audio_buffer *buf;
- uint16_t bits_per_sample = 16;
- int dir, ret;
-
- 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,
- };
-
- prtd->audio_client->perf_mode = false;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE)
- bits_per_sample = 24;
- ret = q6asm_open_write_v2(prtd->audio_client,
- FORMAT_LINEAR_PCM, bits_per_sample);
- if (ret < 0) {
- pr_err("%s: pcm out open failed\n", __func__);
- q6asm_audio_client_free(prtd->audio_client);
- prtd->audio_client = NULL;
- return -ENOMEM;
- }
- }
-
- pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
- prtd->session_id = prtd->audio_client->session;
- msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
- prtd->audio_client->perf_mode,
- prtd->session_id, substream->stream);
-
- 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);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dir = IN;
- else
- return -EPERM;
- ret = q6asm_audio_client_buf_alloc_contiguous(dir,
- prtd->audio_client,
- params_period_bytes(params),
- params_periods(params));
- if (ret < 0) {
- pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
- ret);
- return -ENOMEM;
- }
- buf = prtd->audio_client->port[dir].buf;
-
- if (buf == NULL || buf[0].data == NULL)
- return -ENOMEM;
-
- pr_debug("%s:buf = %p\n", __func__, buf);
- dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
- dma_buf->dev.dev = substream->pcm->card->dev;
- dma_buf->private_data = NULL;
- dma_buf->area = buf[0].data;
- dma_buf->addr = buf[0].phys;
- dma_buf->bytes = params_period_bytes(params) * params_periods(params);
- if (!dma_buf->area)
- return -ENOMEM;
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- return 0;
-}
-
-static int msm_pcm_ioctl(struct snd_pcm_substream *substream,
- unsigned int cmd, void *arg)
-{
- int rc = 0;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct msm_audio *prtd = runtime->private_data;
- uint64_t timestamp;
- uint64_t temp;
-
- switch (cmd) {
- case SNDRV_COMPRESS_TSTAMP: {
- struct snd_compr_tstamp tstamp;
- pr_debug("SNDRV_COMPRESS_TSTAMP\n");
-
- memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
- rc = q6asm_get_session_time(prtd->audio_client, &timestamp);
- if (rc < 0) {
- pr_err("%s: Fail to get session time stamp, rc:%d\n",
- __func__, rc);
- return -EAGAIN;
- }
- temp = (timestamp * 2 * runtime->channels);
- temp = temp * (runtime->rate/1000);
- temp = div_u64(temp, 1000);
- tstamp.sampling_rate = runtime->rate;
- tstamp.timestamp = timestamp;
- pr_debug("%s: bytes_consumed:timestamp = %lld,\n",
- __func__,
- tstamp.timestamp);
- if (copy_to_user((void *) arg, &tstamp,
- sizeof(struct snd_compr_tstamp)))
- return -EFAULT;
- return 0;
- }
- case SNDRV_PCM_IOCTL1_RESET:
- prtd->cmd_ack = 0;
- rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
- if (rc < 0)
- pr_err("%s: flush cmd failed rc=%d\n", __func__, rc);
- if (prtd->reset_event == true) {
- prtd->cmd_ack = 1;
- prtd->reset_event = false;
- return -ENETRESET;
- }
- rc = wait_event_timeout(the_locks.eos_wait,
- !prtd->reset_event && prtd->cmd_ack, 5 * HZ);
- if (!rc)
- pr_err("Flush cmd timeout\n");
- prtd->pcm_irq_pos = 0;
- break;
- case SNDRV_COMPRESS_METADATA_MODE:
- if (!atomic_read(&prtd->start)) {
- pr_debug("Metadata mode enabled\n");
- prtd->meta_data_mode = true;
- return 0;
- }
- pr_debug("Metadata mode not enabled\n");
- return -EPERM;
- default:
- break;
- }
- return snd_pcm_lib_ioctl(substream, cmd, arg);
-}
-
-static int msm_lpa_volume_ctl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- int rc = 0;
- struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
- struct snd_pcm_substream *substream =
- vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- struct msm_audio *prtd;
- int volume = ucontrol->value.integer.value[0];
-
- pr_debug("%s: volume : %x\n", __func__, volume);
- if (!substream)
- return -ENODEV;
- if (!substream->runtime)
- return 0;
- prtd = substream->runtime->private_data;
- if (prtd)
- rc = lpa_set_volume(prtd, volume);
-
- return rc;
-}
-
-static int msm_lpa_volume_ctl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
- struct snd_pcm_substream *substream =
- vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- struct msm_audio *prtd;
-
- pr_debug("%s\n", __func__);
- if (!substream)
- return -ENODEV;
- if (!substream->runtime)
- return 0;
- prtd = substream->runtime->private_data;
- if (prtd)
- ucontrol->value.integer.value[0] = prtd->volume;
- return 0;
-}
-
-static int msm_lpa_add_controls(struct snd_soc_pcm_runtime *rtd)
-{
- int ret = 0;
- struct snd_pcm *pcm = rtd->pcm->streams[0].pcm;
- struct snd_pcm_volume *volume_info;
- struct snd_kcontrol *kctl;
-
- dev_dbg(rtd->dev, "%s, Volume cntrl add\n", __func__);
- ret = snd_pcm_add_volume_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
- NULL, 1, rtd->dai_link->be_id,
- &volume_info);
- if (ret < 0)
- return ret;
- kctl = volume_info->kctl;
- kctl->put = msm_lpa_volume_ctl_put;
- kctl->get = msm_lpa_volume_ctl_get;
- kctl->tlv.p = lpa_rx_vol_gain;
- return 0;
-}
-
-static struct snd_pcm_ops msm_pcm_ops = {
- .open = msm_pcm_open,
- .hw_params = msm_pcm_hw_params,
- .close = msm_pcm_close,
- .ioctl = msm_pcm_ioctl,
- .prepare = msm_pcm_prepare,
- .trigger = msm_pcm_trigger,
- .pointer = msm_pcm_pointer,
- .mmap = msm_pcm_mmap,
- .restart = msm_pcm_restart,
-};
-
-static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_card *card = rtd->card->snd_card;
- int ret = 0;
-
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
-
- ret = msm_lpa_add_controls(rtd);
- if (ret)
- pr_err("%s, kctl add failed\n", __func__);
- return ret;
-}
-
-static struct snd_soc_platform_driver msm_soc_platform = {
- .ops = &msm_pcm_ops,
- .pcm_new = msm_asoc_pcm_new,
-};
-
-static int msm_pcm_probe(struct platform_device *pdev)
-{
-
- dev_info(&pdev->dev, "%s: dev name %s\n",
- __func__, dev_name(&pdev->dev));
- return snd_soc_register_platform(&pdev->dev,
- &msm_soc_platform);
-}
-
-static int msm_pcm_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
-}
-
-static const struct of_device_id msm_pcm_lpa_dt_match[] = {
- {.compatible = "qcom,msm-pcm-lpa"},
- {}
-};
-MODULE_DEVICE_TABLE(of, msm_pcm_lpa_dt_match);
-
-static struct platform_driver msm_pcm_driver = {
- .driver = {
- .name = "msm-pcm-lpa",
- .owner = THIS_MODULE,
- .of_match_table = msm_pcm_lpa_dt_match,
- },
- .probe = msm_pcm_probe,
- .remove = msm_pcm_remove,
-};
-
-static int __init msm_soc_platform_init(void)
-{
- spin_lock_init(&the_locks.event_lock);
- init_waitqueue_head(&the_locks.enable_wait);
- init_waitqueue_head(&the_locks.eos_wait);
- init_waitqueue_head(&the_locks.write_wait);
- init_waitqueue_head(&the_locks.read_wait);
-
- return platform_driver_register(&msm_pcm_driver);
-}
-module_init(msm_soc_platform_init);
-
-static void __exit msm_soc_platform_exit(void)
-{
- platform_driver_unregister(&msm_pcm_driver);
-}
-module_exit(msm_soc_platform_exit);
-
-MODULE_DESCRIPTION("PCM module platform driver");
-MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-devdep.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-devdep.c
index e57db7d9c494..3263a5786290 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-devdep.c
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-devdep.c
@@ -10,6 +10,8 @@
* GNU General Public License for more details.
*/
+#include <linux/kernel.h>
+#include <linux/platform_device.h>
#include <linux/err.h>
#include <linux/module.h>
#include <sound/hwdep.h>
diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c
index b9b1276ea68b..a157a0ae6041 100644
--- a/sound/soc/msm/qdsp6v2/q6adm.c
+++ b/sound/soc/msm/qdsp6v2/q6adm.c
@@ -9,6 +9,7 @@
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*/
+#include <linux/module.h>
#include <linux/slab.h>
#include <linux/wait.h>
#include <linux/sched.h>
diff --git a/sound/soc/msm/qdsp6v2/q6voice.c b/sound/soc/msm/qdsp6v2/q6voice.c
index dce6f02c02dd..898b2704b8b7 100644
--- a/sound/soc/msm/qdsp6v2/q6voice.c
+++ b/sound/soc/msm/qdsp6v2/q6voice.c
@@ -1835,19 +1835,23 @@ static int voice_send_dtmf_rx_detection_cmd(struct voice_data *v,
static void voice_vote_powerstate_to_bms(struct voice_data *v, bool state)
{
+ union power_supply_propval pval = {0, };
if (!v->psy)
v->psy = power_supply_get_by_name("bms");
if (v->psy && !(is_voip_session(v->session_id) ||
is_vowlan_session(v->session_id))) {
+ pval.intval = VMBMS_VOICE_CALL_BIT;
if (state) {
- power_supply_set_hi_power_state(v->psy,
- VMBMS_VOICE_CALL_BIT);
+ power_supply_set_property(v->psy,
+ POWER_SUPPLY_PROP_HI_POWER,
+ &pval);
pr_debug("%s : Vote High power to BMS\n",
__func__);
} else {
- power_supply_set_low_power_state(v->psy,
- VMBMS_VOICE_CALL_BIT);
+ power_supply_set_property(v->psy,
+ POWER_SUPPLY_PROP_LOW_POWER,
+ &pval);
pr_debug("%s: Vote low power to BMS\n",
__func__);
}
generated by cgit v1.2.3 (git 2.25.1) at 2026-05-08 09:34:16 +0000