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Commit addaeea9 authored by Mark Brown's avatar Mark Brown
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Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/intel',... 

Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/intel', 'asoc/topic/jack', 'asoc/topic/jz4740' and 'asoc/topic/lm49453' into asoc-next
parents a6a21e6b 69434097 7bb73cbd 00d647b0 69b93607 68d27bc6
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Showing with 8285 additions and 545 deletions
arch/x86/include/asm/platform_sst_audio.h
View file @ addaeea9
......@@ -16,6 +16,9 @@
#include <linux/sfi.h>
#define MAX_NUM_STREAMS_MRFLD 25
#define MAX_NUM_STREAMS MAX_NUM_STREAMS_MRFLD
enum sst_audio_task_id_mrfld {
SST_TASK_ID_NONE = 0,
SST_TASK_ID_SBA = 1,
......@@ -73,6 +76,65 @@ struct sst_platform_data {
unsigned int strm_map_size;
};
struct sst_info {
u32 iram_start;
u32 iram_end;
bool iram_use;
u32 dram_start;
u32 dram_end;
bool dram_use;
u32 imr_start;
u32 imr_end;
bool imr_use;
u32 mailbox_start;
bool use_elf;
bool lpe_viewpt_rqd;
unsigned int max_streams;
u32 dma_max_len;
u8 num_probes;
};
struct sst_lib_dnld_info {
unsigned int mod_base;
unsigned int mod_end;
unsigned int mod_table_offset;
unsigned int mod_table_size;
bool mod_ddr_dnld;
};
struct sst_res_info {
unsigned int shim_offset;
unsigned int shim_size;
unsigned int shim_phy_addr;
unsigned int ssp0_offset;
unsigned int ssp0_size;
unsigned int dma0_offset;
unsigned int dma0_size;
unsigned int dma1_offset;
unsigned int dma1_size;
unsigned int iram_offset;
unsigned int iram_size;
unsigned int dram_offset;
unsigned int dram_size;
unsigned int mbox_offset;
unsigned int mbox_size;
unsigned int acpi_lpe_res_index;
unsigned int acpi_ddr_index;
unsigned int acpi_ipc_irq_index;
};
struct sst_ipc_info {
int ipc_offset;
unsigned int mbox_recv_off;
};
struct sst_platform_info {
const struct sst_info *probe_data;
const struct sst_ipc_info *ipc_info;
const struct sst_res_info *res_info;
const struct sst_lib_dnld_info *lib_info;
const char *platform;
};
int add_sst_platform_device(void);
#endif
sound/soc/codecs/hdmi.c
View file @ addaeea9
......@@ -47,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = {
SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
.sig_bits = 24,
},
.capture = {
.stream_name = "Capture",
......@@ -75,6 +76,7 @@ static struct snd_soc_codec_driver hdmi_codec = {
.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
.dapm_routes = hdmi_routes,
.num_dapm_routes = ARRAY_SIZE(hdmi_routes),
.ignore_pmdown_time = true,
};
static int hdmi_codec_probe(struct platform_device *pdev)
......
sound/soc/codecs/lm49453.c
View file @ addaeea9
......@@ -1395,15 +1395,7 @@ static struct snd_soc_dai_driver lm49453_dai[] = {
},
};
/* power down chip */
static int lm49453_remove(struct snd_soc_codec *codec)
{
lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
.remove = lm49453_remove,
.set_bias_level = lm49453_set_bias_level,
.controls = lm49453_snd_controls,
.num_controls = ARRAY_SIZE(lm49453_snd_controls),
......
sound/soc/intel/Kconfig
View file @ addaeea9
......@@ -3,6 +3,7 @@ config SND_MFLD_MACHINE
depends on INTEL_SCU_IPC
select SND_SOC_SN95031
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_PCI
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
......@@ -12,10 +13,23 @@ config SND_MFLD_MACHINE
config SND_SST_MFLD_PLATFORM
tristate
config SND_SST_IPC
tristate
config SND_SST_IPC_PCI
tristate
select SND_SST_IPC
config SND_SST_IPC_ACPI
tristate
select SND_SST_IPC
depends on ACPI
config SND_SOC_INTEL_SST
tristate "ASoC support for Intel(R) Smart Sound Technology"
select SND_SOC_INTEL_SST_ACPI if ACPI
depends on (X86 || COMPILE_TEST)
depends on DW_DMAC_CORE
help
This adds support for Intel(R) Smart Sound Technology (SST).
Say Y if you have such a device
......@@ -32,7 +46,8 @@ config SND_SOC_INTEL_BAYTRAIL
config SND_SOC_INTEL_HASWELL_MACH
tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C
depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\
I2C_DESIGNWARE_PLATFORM
select SND_SOC_INTEL_HASWELL
select SND_SOC_RT5640
help
......@@ -61,7 +76,8 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
config SND_SOC_INTEL_BROADWELL_MACH
tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\
I2C_DESIGNWARE_PLATFORM
select SND_SOC_INTEL_HASWELL
select SND_COMPRESS_OFFLOAD
select SND_SOC_RT286
......@@ -70,3 +86,27 @@ config SND_SOC_INTEL_BROADWELL_MACH
Ultrabook platforms.
Say Y if you have such a device
If unsure select "N".
config SND_SOC_INTEL_BYTCR_RT5640_MACH
tristate "ASoC Audio DSP Support for MID BYT Platform"
depends on X86
select SND_SOC_RT5640
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) MID Baytrail platform
used as alsa device in audio substem in Intel(R) MID devices
Say Y if you have such a device
If unsure select "N".
config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
depends on X86_INTEL_LPSS
select SND_SOC_RT5670
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5672 audio codec.
Say Y if you have such a device
If unsure select "N".
sound/soc/intel/Makefile
View file @ addaeea9
......@@ -26,8 +26,15 @@ snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-broadwell-objs := broadwell.o
snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
# DSP driver
obj-$(CONFIG_SND_SST_IPC) += sst/
sound/soc/intel/broadwell.c
View file @ addaeea9
......@@ -19,6 +19,7 @@
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <sound/pcm_params.h>
#include "sst-dsp.h"
......@@ -26,8 +27,26 @@
#include "../codecs/rt286.h"
static struct snd_soc_jack broadwell_headset;
/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin broadwell_headset_pins[] = {
{
.pin = "Mic Jack",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
},
};
static const struct snd_kcontrol_new broadwell_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
};
static const struct snd_soc_dapm_widget broadwell_widgets[] = {
SND_SOC_DAPM_HP("Headphones", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_MIC("DMIC1", NULL),
......@@ -42,7 +61,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
{"Speaker", NULL, "SPOL"},
/* HP jack connectors - unknown if we have jack deteck */
{"Headphones", NULL, "HPO Pin"},
{"Headphone Jack", NULL, "HPO Pin"},
/* other jacks */
{"MIC1", NULL, "Mic Jack"},
......@@ -57,6 +76,27 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
};
static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int ret = 0;
ret = snd_soc_jack_new(codec, "Headset",
SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset);
if (ret)
return ret;
ret = snd_soc_jack_add_pins(&broadwell_headset,
ARRAY_SIZE(broadwell_headset_pins),
broadwell_headset_pins);
if (ret)
return ret;
rt286_mic_detect(codec, &broadwell_headset);
return 0;
}
static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
......@@ -116,7 +156,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
}
/* always connected - check HP for jack detect */
snd_soc_dapm_enable_pin(dapm, "Headphones");
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
snd_soc_dapm_enable_pin(dapm, "Line Jack");
......@@ -131,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
/* Front End DAI links */
{
.name = "System PCM",
.stream_name = "System Playback",
.stream_name = "System Playback/Capture",
.cpu_dai_name = "System Pin",
.platform_name = "haswell-pcm-audio",
.dynamic = 1,
......@@ -140,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.init = broadwell_rtd_init,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_playback = 1,
.dpcm_capture = 1,
},
{
.name = "Offload0",
......@@ -174,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
},
{
.name = "Capture PCM",
.stream_name = "Capture",
.cpu_dai_name = "Capture Pin",
.platform_name = "haswell-pcm-audio",
.dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
},
/* Back End DAI links */
{
/* SSP0 - Codec */
......@@ -196,6 +225,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.no_pcm = 1,
.codec_name = "i2c-INT343A:00",
.codec_dai_name = "rt286-aif1",
.init = broadwell_rt286_codec_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ignore_suspend = 1,
......@@ -213,6 +243,8 @@ static struct snd_soc_card broadwell_rt286 = {
.owner = THIS_MODULE,
.dai_link = broadwell_rt286_dais,
.num_links = ARRAY_SIZE(broadwell_rt286_dais),
.controls = broadwell_controls,
.num_controls = ARRAY_SIZE(broadwell_controls),
.dapm_widgets = broadwell_widgets,
.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
.dapm_routes = broadwell_rt286_map,
......
sound/soc/intel/bytcr_dpcm_rt5640.c 0 → 100644
View file @ addaeea9
/*
* byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform
*
* Copyright (C) 2014 Intel Corp
* Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* 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/module.h>
#include <linux/platform_device.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <linux/input.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "../codecs/rt5640.h"
#include "sst-atom-controls.h"
static const struct snd_soc_dapm_widget byt_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
};
static const struct snd_soc_dapm_route byt_audio_map[] = {
{"IN2P", NULL, "Headset Mic"},
{"IN2N", NULL, "Headset Mic"},
{"Headset Mic", NULL, "MICBIAS1"},
{"IN1P", NULL, "MICBIAS1"},
{"LDO2", NULL, "Int Mic"},
{"Headphone", NULL, "HPOL"},
{"Headphone", NULL, "HPOR"},
{"Ext Spk", NULL, "SPOLP"},
{"Ext Spk", NULL, "SPOLN"},
{"Ext Spk", NULL, "SPORP"},
{"Ext Spk", NULL, "SPORN"},
{"AIF1 Playback", NULL, "ssp2 Tx"},
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"ssp2 Rx", NULL, "AIF1 Capture"},
};
static const struct snd_kcontrol_new byt_mc_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Int Mic"),
SOC_DAPM_PIN_SWITCH("Ext Spk"),
};
static int byt_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
snd_soc_dai_set_bclk_ratio(codec_dai, 50);
ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
params_rate(params) * 512,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec clock %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
params_rate(params) * 50,
params_rate(params) * 512);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
return ret;
}
return 0;
}
static const struct snd_soc_pcm_stream byt_dai_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/* The DSP will covert the FE rate to 48k, stereo, 24bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
SNDRV_PCM_HW_PARAM_FIRST_MASK],
SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
static unsigned int rates_48000[] = {
48000,
};
static struct snd_pcm_hw_constraint_list constraints_48000 = {
.count = ARRAY_SIZE(rates_48000),
.list = rates_48000,
};
static int byt_aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_48000);
}
static struct snd_soc_ops byt_aif1_ops = {
.startup = byt_aif1_startup,
};
static struct snd_soc_ops byt_be_ssp2_ops = {
.hw_params = byt_aif1_hw_params,
};
static struct snd_soc_dai_link byt_dailink[] = {
[MERR_DPCM_AUDIO] = {
.name = "Baytrail Audio Port",
.stream_name = "Baytrail Audio",
.cpu_dai_name = "media-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.ignore_suspend = 1,
.dynamic = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &byt_aif1_ops,
},
[MERR_DPCM_COMPR] = {
.name = "Baytrail Compressed Port",
.stream_name = "Baytrail Compress",
.cpu_dai_name = "compress-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
},
/* back ends */
{
.name = "SSP2-Codec",
.be_id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "rt5640-aif1",
.codec_name = "i2c-10EC5640:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.be_hw_params_fixup = byt_codec_fixup,
.ignore_suspend = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &byt_be_ssp2_ops,
},
};
/* SoC card */
static struct snd_soc_card snd_soc_card_byt = {
.name = "baytrailcraudio",
.dai_link = byt_dailink,
.num_links = ARRAY_SIZE(byt_dailink),
.dapm_widgets = byt_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets),
.dapm_routes = byt_audio_map,
.num_dapm_routes = ARRAY_SIZE(byt_audio_map),
.controls = byt_mc_controls,
.num_controls = ARRAY_SIZE(byt_mc_controls),
};
static int snd_byt_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
/* register the soc card */
snd_soc_card_byt.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt);
if (ret_val) {
dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, &snd_soc_card_byt);
return ret_val;
}
static struct platform_driver snd_byt_mc_driver = {
.driver = {
.owner = THIS_MODULE,
.name = "bytt100_rt5640",
.pm = &snd_soc_pm_ops,
},
.probe = snd_byt_mc_probe,
};
module_platform_driver(snd_byt_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bytrt5640-audio");
sound/soc/intel/cht_bsw_rt5672.c 0 → 100644
View file @ addaeea9
/*
* cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
* Cherrytrail and Braswell, with RT5672 codec.
*
* Copyright (C) 2014 Intel Corp
* Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
* Mengdong Lin <mengdong.lin@intel.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* 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/platform_device.h>
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "../codecs/rt5670.h"
#include "sst-atom-controls.h"
/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
#define CHT_PLAT_CLK_3_HZ 19200000
#define CHT_CODEC_DAI "rt5670-aif1"
static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
{
int i;
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_pcm_runtime *rtd;
rtd = card->rtd + i;
if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
strlen(CHT_CODEC_DAI)))
return rtd->codec_dai;
}
return NULL;
}
static int platform_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_card *card = dapm->card;
struct snd_soc_dai *codec_dai;
codec_dai = cht_get_codec_dai(card);
if (!codec_dai) {
dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
return -EIO;
}
if (!SND_SOC_DAPM_EVENT_OFF(event))
return 0;
/* Set codec sysclk source to its internal clock because codec PLL will
* be off when idle and MCLK will also be off by ACPI when codec is
* runtime suspended. Codec needs clock for jack detection and button
* press.
*/
snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
0, SND_SOC_CLOCK_IN);
return 0;
}
static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_POST_PMD),
};
static const struct snd_soc_dapm_route cht_audio_map[] = {
{"IN1P", NULL, "Headset Mic"},
{"IN1N", NULL, "Headset Mic"},
{"DMIC L1", NULL, "Int Mic"},
{"DMIC R1", NULL, "Int Mic"},
{"Headphone", NULL, "HPOL"},
{"Headphone", NULL, "HPOR"},
{"Ext Spk", NULL, "SPOLP"},
{"Ext Spk", NULL, "SPOLN"},
{"Ext Spk", NULL, "SPORP"},
{"Ext Spk", NULL, "SPORN"},
{"AIF1 Playback", NULL, "ssp2 Tx"},
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"ssp2 Rx", NULL, "AIF1 Capture"},
{"Headphone", NULL, "Platform Clock"},
{"Headset Mic", NULL, "Platform Clock"},
{"Int Mic", NULL, "Platform Clock"},
{"Ext Spk", NULL, "Platform Clock"},
};
static const struct snd_kcontrol_new cht_mc_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Int Mic"),
SOC_DAPM_PIN_SWITCH("Ext Spk"),
};
static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
return ret;
}
/* set codec sysclk source to PLL */
ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
params_rate(params) * 512,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
return ret;
}
return 0;
}
static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
struct snd_soc_dai *codec_dai = runtime->codec_dai;
/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
if (ret < 0) {
dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
return ret;
}
return 0;
}
static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/* The DSP will covert the FE rate to 48k, stereo, 24bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
SNDRV_PCM_HW_PARAM_FIRST_MASK],
SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
static unsigned int rates_48000[] = {
48000,
};
static struct snd_pcm_hw_constraint_list constraints_48000 = {
.count = ARRAY_SIZE(rates_48000),
.list = rates_48000,
};
static int cht_aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_48000);
}
static struct snd_soc_ops cht_aif1_ops = {
.startup = cht_aif1_startup,
};
static struct snd_soc_ops cht_be_ssp2_ops = {
.hw_params = cht_aif1_hw_params,
};
static struct snd_soc_dai_link cht_dailink[] = {
/* Front End DAI links */
[MERR_DPCM_AUDIO] = {
.name = "Audio Port",
.stream_name = "Audio",
.cpu_dai_name = "media-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.ignore_suspend = 1,
.dynamic = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &cht_aif1_ops,
},
[MERR_DPCM_COMPR] = {
.name = "Compressed Port",
.stream_name = "Compress",
.cpu_dai_name = "compress-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
},
/* Back End DAI links */
{
/* SSP2 - Codec */
.name = "SSP2-Codec",
.be_id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "rt5670-aif1",
.codec_name = "i2c-10EC5670:00",
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.init = cht_codec_init,
.be_hw_params_fixup = cht_codec_fixup,
.ignore_suspend = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &cht_be_ssp2_ops,
},
};
/* SoC card */
static struct snd_soc_card snd_soc_card_cht = {
.name = "cherrytrailcraudio",
.dai_link = cht_dailink,
.num_links = ARRAY_SIZE(cht_dailink),
.dapm_widgets = cht_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
.dapm_routes = cht_audio_map,
.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
.controls = cht_mc_controls,
.num_controls = ARRAY_SIZE(cht_mc_controls),
};
static int snd_cht_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
/* register the soc card */
snd_soc_card_cht.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
if (ret_val) {
dev_err(&pdev->dev,
"snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, &snd_soc_card_cht);
return ret_val;
}
static struct platform_driver snd_cht_mc_driver = {
.driver = {
.owner = THIS_MODULE,
.name = "cht-bsw-rt5672",
.pm = &snd_soc_pm_ops,
},
.probe = snd_cht_mc_probe,
};
module_platform_driver(snd_cht_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:cht-bsw-rt5672");
sound/soc/intel/haswell.c
View file @ addaeea9
......@@ -109,7 +109,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
/* Front End DAI links */
{
.name = "System",
.stream_name = "System Playback",
.stream_name = "System Playback/Capture",
.cpu_dai_name = "System Pin",
.platform_name = "haswell-pcm-audio",
.dynamic = 1,
......@@ -118,6 +118,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
.init = haswell_rtd_init,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_playback = 1,
.dpcm_capture = 1,
},
{
.name = "Offload0",
......@@ -152,17 +153,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
},
{
.name = "Capture",
.stream_name = "Capture",
.cpu_dai_name = "Capture Pin",
.platform_name = "haswell-pcm-audio",
.dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
},
/* Back End DAI links */
{
......
sound/soc/intel/sst-atom-controls.c
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst-atom-controls.h
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst-baytrail-dsp.c
View file @ addaeea9
......@@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
{
struct dma_block_info *block;
struct sst_module *mod;
struct sst_module_data block_data;
struct sst_module_template template;
int count;
memset(&template, 0, sizeof(template));
template.id = module->type;
template.entry = module->entry_point;
template.p.type = SST_MEM_DRAM;
template.p.data_type = SST_DATA_P;
template.s.type = SST_MEM_DRAM;
template.s.data_type = SST_DATA_S;
mod = sst_module_new(fw, &template, NULL);
if (mod == NULL)
......@@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
switch (block->type) {
case SST_BYT_IRAM:
block_data.offset = block->ram_offset +
mod->offset = block->ram_offset +
dsp->addr.iram_offset;
block_data.type = SST_MEM_IRAM;
mod->type = SST_MEM_IRAM;
break;
case SST_BYT_DRAM:
block_data.offset = block->ram_offset +
mod->offset = block->ram_offset +
dsp->addr.dram_offset;
block_data.type = SST_MEM_DRAM;
mod->type = SST_MEM_DRAM;
break;
case SST_BYT_CACHE:
block_data.offset = block->ram_offset +
mod->offset = block->ram_offset +
(dsp->addr.fw_ext - dsp->addr.lpe);
block_data.type = SST_MEM_CACHE;
mod->type = SST_MEM_CACHE;
break;
default:
dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n",
......@@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
return -EINVAL;
}
block_data.size = block->size;
block_data.data_type = SST_DATA_M;
block_data.data = (void *)block + sizeof(*block);
mod->size = block->size;
mod->data = (void *)block + sizeof(*block);
sst_module_insert_fixed_block(mod, &block_data);
sst_module_alloc_blocks(mod);
block = (void *)block + sizeof(*block) + block->size;
}
......
sound/soc/intel/sst-dsp-priv.h
View file @ addaeea9
......@@ -26,6 +26,9 @@ struct sst_mem_block;
struct sst_module;
struct sst_fw;
/* do we need to remove or keep */
#define DSP_DRAM_ADDR_OFFSET 0x400000
/*
* DSP Operations exported by platform Audio DSP driver.
*/
......@@ -33,6 +36,9 @@ struct sst_ops {
/* DSP core boot / reset */
void (*boot)(struct sst_dsp *);
void (*reset)(struct sst_dsp *);
int (*wake)(struct sst_dsp *);
void (*sleep)(struct sst_dsp *);
void (*stall)(struct sst_dsp *);
/* Shim IO */
void (*write)(void __iomem *addr, u32 offset, u32 value);
......@@ -67,6 +73,8 @@ struct sst_addr {
u32 shim_offset;
u32 iram_offset;
u32 dram_offset;
u32 dsp_iram_offset;
u32 dsp_dram_offset;
void __iomem *lpe;
void __iomem *shim;
void __iomem *pci_cfg;
......@@ -83,15 +91,6 @@ struct sst_mailbox {
size_t out_size;
};
/*
* Audio DSP Firmware data types.
*/
enum sst_data_type {
SST_DATA_M = 0, /* module block data */
SST_DATA_P = 1, /* peristant data (text, data) */
SST_DATA_S = 2, /* scratch data (usually buffers) */
};
/*
* Audio DSP memory block types.
*/
......@@ -124,23 +123,6 @@ struct sst_fw {
void *private; /* core doesn't touch this */
};
/*
* Audio DSP Generic Module data.
*
* This is used to dsecribe any sections of persistent (text and data) and
* scratch (buffers) of module data in ADSP memory space.
*/
struct sst_module_data {
enum sst_mem_type type; /* destination memory type */
enum sst_data_type data_type; /* type of module data */
u32 size; /* size in bytes */
int32_t offset; /* offset in FW file */
u32 data_offset; /* offset in ADSP memory space */
void *data; /* module data */
};
/*
* Audio DSP Generic Module Template.
*
......@@ -150,15 +132,52 @@ struct sst_module_data {
struct sst_module_template {
u32 id;
u32 entry; /* entry point */
struct sst_module_data s; /* scratch data */
struct sst_module_data p; /* peristant data */
u32 scratch_size;
u32 persistent_size;
};
/*
* Block Allocator - Used to allocate blocks of DSP memory.
*/
struct sst_block_allocator {
u32 id;
u32 offset;
int size;
enum sst_mem_type type;
};
/*
* Runtime Module Instance - A module object can be instanciated multiple
* times within the DSP FW.
*/
struct sst_module_runtime {
struct sst_dsp *dsp;
int id;
struct sst_module *module; /* parent module we belong too */
u32 persistent_offset; /* private memory offset */
void *private;
struct list_head list;
struct list_head block_list; /* list of blocks used */
};
/*
* Runtime Module Context - The runtime context must be manually stored by the
* driver prior to enter S3 and restored after leaving S3. This should really be
* part of the memory context saved by the enter D3 message IPC ???
*/
struct sst_module_runtime_context {
dma_addr_t dma_buffer;
u32 *buffer;
};
/*
* Audio DSP Generic Module.
*
* Each Firmware file can consist of 1..N modules. A module can span multiple
* ADSP memory blocks. The simplest FW will be a file with 1 module.
* ADSP memory blocks. The simplest FW will be a file with 1 module. A module
* can be instanciated multiple times in the DSP.
*/
struct sst_module {
struct sst_dsp *dsp;
......@@ -167,10 +186,13 @@ struct sst_module {
/* module configuration */
u32 id;
u32 entry; /* module entry point */
u32 offset; /* module offset in firmware file */
s32 offset; /* module offset in firmware file */
u32 size; /* module size */
struct sst_module_data s; /* scratch data */
struct sst_module_data p; /* peristant data */
u32 scratch_size; /* global scratch memory required */
u32 persistent_size; /* private memory required */
enum sst_mem_type type; /* destination memory type */
u32 data_offset; /* offset in ADSP memory space */
void *data; /* module data */
/* runtime */
u32 usage_count; /* can be unloaded if count == 0 */
......@@ -180,6 +202,7 @@ struct sst_module {
struct list_head block_list; /* Module list of blocks in use */
struct list_head list; /* DSP list of modules */
struct list_head list_fw; /* FW list of modules */
struct list_head runtime_list; /* list of runtime module objects*/
};
/*
......@@ -208,7 +231,6 @@ struct sst_mem_block {
struct sst_block_ops *ops; /* block operations, if any */
/* block status */
enum sst_data_type data_type; /* data type held in this block */
u32 bytes_used; /* bytes in use by modules */
void *private; /* generic core does not touch this */
int users; /* number of modules using this block */
......@@ -253,6 +275,11 @@ struct sst_dsp {
struct list_head module_list;
struct list_head fw_list;
/* scratch buffer */
struct list_head scratch_block_list;
u32 scratch_offset;
u32 scratch_size;
/* platform data */
struct sst_pdata *pdata;
......@@ -290,18 +317,33 @@ void sst_fw_unload(struct sst_fw *sst_fw);
/* Create/Free firmware modules */
struct sst_module *sst_module_new(struct sst_fw *sst_fw,
struct sst_module_template *template, void *private);
void sst_module_free(struct sst_module *sst_module);
int sst_module_insert(struct sst_module *sst_module);
int sst_module_remove(struct sst_module *sst_module);
int sst_module_insert_fixed_block(struct sst_module *module,
struct sst_module_data *data);
void sst_module_free(struct sst_module *module);
struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id);
/* allocate/free pesistent/scratch memory regions managed by drv */
struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp);
void sst_mem_block_free_scratch(struct sst_dsp *dsp,
struct sst_module *scratch);
int sst_block_module_remove(struct sst_module *module);
int sst_module_alloc_blocks(struct sst_module *module);
int sst_module_free_blocks(struct sst_module *module);
/* Create/Free firmware module runtime instances */
struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module,
int id, void *private);
void sst_module_runtime_free(struct sst_module_runtime *runtime);
struct sst_module_runtime *sst_module_runtime_get_from_id(
struct sst_module *module, u32 id);
int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime,
int offset);
int sst_module_runtime_free_blocks(struct sst_module_runtime *runtime);
int sst_module_runtime_save(struct sst_module_runtime *runtime,
struct sst_module_runtime_context *context);
int sst_module_runtime_restore(struct sst_module_runtime *runtime,
struct sst_module_runtime_context *context);
/* generic block allocation */
int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba,
struct list_head *block_list);
int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list);
/* scratch allocation */
int sst_block_alloc_scratch(struct sst_dsp *dsp);
void sst_block_free_scratch(struct sst_dsp *dsp);
/* Register the DSPs memory blocks - would be nice to read from ACPI */
struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
......@@ -309,4 +351,10 @@ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
void *private);
void sst_mem_block_unregister_all(struct sst_dsp *dsp);
/* Create/Free DMA resources */
int sst_dma_new(struct sst_dsp *sst);
void sst_dma_free(struct sst_dma *dma);
u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset,
enum sst_mem_type type);
#endif
sound/soc/intel/sst-dsp.c
View file @ addaeea9
......@@ -245,6 +245,29 @@ int sst_dsp_boot(struct sst_dsp *sst)
}
EXPORT_SYMBOL_GPL(sst_dsp_boot);
int sst_dsp_wake(struct sst_dsp *sst)
{
if (sst->ops->wake)
return sst->ops->wake(sst);
return 0;
}
EXPORT_SYMBOL_GPL(sst_dsp_wake);
void sst_dsp_sleep(struct sst_dsp *sst)
{
if (sst->ops->sleep)
sst->ops->sleep(sst);
}
EXPORT_SYMBOL_GPL(sst_dsp_sleep);
void sst_dsp_stall(struct sst_dsp *sst)
{
if (sst->ops->stall)
sst->ops->stall(sst);
}
EXPORT_SYMBOL_GPL(sst_dsp_stall);
void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg)
{
sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY);
......@@ -352,6 +375,7 @@ struct sst_dsp *sst_dsp_new(struct device *dev,
INIT_LIST_HEAD(&sst->free_block_list);
INIT_LIST_HEAD(&sst->module_list);
INIT_LIST_HEAD(&sst->fw_list);
INIT_LIST_HEAD(&sst->scratch_block_list);
/* Initialise SST Audio DSP */
if (sst->ops->init) {
......@@ -366,6 +390,10 @@ struct sst_dsp *sst_dsp_new(struct device *dev,
if (err)
goto irq_err;
err = sst_dma_new(sst);
if (err)
dev_warn(dev, "sst_dma_new failed %d\n", err);
return sst;
irq_err:
......@@ -381,6 +409,9 @@ void sst_dsp_free(struct sst_dsp *sst)
free_irq(sst->irq, sst);
if (sst->ops->free)
sst->ops->free(sst);
if (sst->dma)
sst_dma_free(sst->dma);
}
EXPORT_SYMBOL_GPL(sst_dsp_free);
......
sound/soc/intel/sst-dsp.h
View file @ addaeea9
......@@ -30,6 +30,9 @@
#define SST_DMA_TYPE_DW 1
#define SST_DMA_TYPE_MID 2
/* autosuspend delay 5s*/
#define SST_RUNTIME_SUSPEND_DELAY (5 * 1000)
/* SST Shim register map
* The register naming can differ between products. Some products also
* contain extra functionality.
......@@ -156,12 +159,18 @@
#define SST_VDRTCTL3 0xaC
/* VDRTCTL0 */
#define SST_VDRTCL0_APLLSE_MASK 1
#define SST_VDRTCL0_DSRAMPGE_SHIFT 16
#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
#define SST_VDRTCL0_ISRAMPGE_SHIFT 6
#define SST_VDRTCL0_D3PGD (1 << 0)
#define SST_VDRTCL0_D3SRAMPGD (1 << 1)
#define SST_VDRTCL0_DSRAMPGE_SHIFT 12
#define SST_VDRTCL0_DSRAMPGE_MASK (0xfffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
#define SST_VDRTCL0_ISRAMPGE_SHIFT 2
#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
/* VDRTCTL2 */
#define SST_VDRTCL2_DCLCGE (1 << 1)
#define SST_VDRTCL2_DTCGE (1 << 10)
#define SST_VDRTCL2_APLLSE_MASK (1 << 31)
/* PMCS */
#define SST_PMCS 0x84
#define SST_PMCS_PS_MASK 0x3
......@@ -245,6 +254,17 @@ void sst_memcpy_fromio_32(struct sst_dsp *sst,
/* DSP reset & boot */
void sst_dsp_reset(struct sst_dsp *sst);
int sst_dsp_boot(struct sst_dsp *sst);
int sst_dsp_wake(struct sst_dsp *sst);
void sst_dsp_sleep(struct sst_dsp *sst);
void sst_dsp_stall(struct sst_dsp *sst);
/* DMA */
int sst_dsp_dma_get_channel(struct sst_dsp *dsp, int chan_id);
void sst_dsp_dma_put_channel(struct sst_dsp *dsp);
int sst_dsp_dma_copyfrom(struct sst_dsp *sst, dma_addr_t dest_addr,
dma_addr_t src_addr, size_t size);
int sst_dsp_dma_copyto(struct sst_dsp *sst, dma_addr_t dest_addr,
dma_addr_t src_addr, size_t size);
/* Msg IO */
void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg);
......
sound/soc/intel/sst-firmware.c
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst-haswell-dsp.c
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst-haswell-ipc.c
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst-haswell-ipc.h
View file @ addaeea9
......@@ -21,8 +21,10 @@
#include <linux/kernel.h>
#include <linux/platform_device.h>
#define SST_HSW_NO_CHANNELS 2
#define SST_HSW_NO_CHANNELS 4
#define SST_HSW_MAX_DX_REGIONS 14
#define SST_HSW_DX_CONTEXT_SIZE (640 * 1024)
#define SST_HSW_CHANNELS_ALL 0xffffffff
#define SST_HSW_FW_LOG_CONFIG_DWORDS 12
#define SST_HSW_GLOBAL_LOG 15
......@@ -40,6 +42,7 @@ struct sst_hsw_stream;
struct sst_hsw_log_stream;
struct sst_pdata;
struct sst_module;
struct sst_module_runtime;
extern struct sst_ops haswell_ops;
/* Stream Allocate Path ID */
......@@ -84,6 +87,7 @@ enum sst_hsw_device_mclk {
enum sst_hsw_device_mode {
SST_HSW_DEVICE_CLOCK_SLAVE = 0,
SST_HSW_DEVICE_CLOCK_MASTER = 1,
SST_HSW_DEVICE_TDM_CLOCK_MASTER = 2,
};
/* DX Power State */
......@@ -295,7 +299,8 @@ struct sst_hsw_ipc_device_config_req {
u32 clock_frequency;
u32 mode;
u16 clock_divider;
u16 reserved;
u8 channels;
u8 reserved;
} __attribute__((packed));
/* Audio Data formats */
......@@ -430,8 +435,7 @@ int sst_hsw_stream_set_map_config(struct sst_hsw *hsw,
int sst_hsw_stream_set_style(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
enum sst_hsw_interleaving style);
int sst_hsw_stream_set_module_info(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id,
u32 entry_point);
struct sst_hsw_stream *stream, struct sst_module_runtime *runtime);
int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 offset, u32 size);
int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
......@@ -484,7 +488,16 @@ int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata);
void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata);
struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw);
void sst_hsw_set_scratch_module(struct sst_hsw *hsw,
struct sst_module *scratch);
/* runtime module management */
struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw,
int mod_id, int offset);
void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime);
/* PM */
int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw);
int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw);
int sst_hsw_dsp_load(struct sst_hsw *hsw);
int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw);
#endif
sound/soc/intel/sst-haswell-pcm.c
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst-mfld-platform-compress.c
View file @ addaeea9
......@@ -67,8 +67,11 @@ static int sst_platform_compr_open(struct snd_compr_stream *cstream)
goto out_ops;
}
stream->compr_ops = sst->compr_ops;
stream->id = 0;
/* Turn on LPE */
sst->compr_ops->power(sst->dev, true);
sst_set_stream_status(stream, SST_PLATFORM_INIT);
runtime->private_data = stream;
return 0;
......@@ -83,6 +86,9 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream)
int ret_val = 0, str_id;
stream = cstream->runtime->private_data;
/* Turn off LPE */
sst->compr_ops->power(sst->dev, false);
/*need to check*/
str_id = stream->id;
if (str_id)
......
sound/soc/intel/sst-mfld-platform-pcm.c
View file @ addaeea9
......@@ -101,35 +101,11 @@ static struct sst_dev_stream_map dpcm_strm_map[] = {
{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
};
/* MFLD - MSIC */
static struct snd_soc_dai_driver sst_platform_dai[] = {
{
.name = "Headset-cpu-dai",
.id = 0,
.playback = {
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S24_LE,
},
.capture = {
.channels_min = 1,
.channels_max = 5,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S24_LE,
},
},
static int sst_media_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
{
.name = "Compress-cpu-dai",
.compress_dai = 1,
.playback = {
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
};
return sst_send_pipe_gains(dai, stream, mute);
}
/* helper functions */
void sst_set_stream_status(struct sst_runtime_stream *stream,
......@@ -451,12 +427,133 @@ static int sst_media_hw_free(struct snd_pcm_substream *substream,
return snd_pcm_lib_free_pages(substream);
}
static int sst_enable_ssp(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int ret = 0;
if (!dai->active) {
ret = sst_handle_vb_timer(dai, true);
if (ret)
return ret;
ret = send_ssp_cmd(dai, dai->name, 1);
}
return ret;
}
static void sst_disable_ssp(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
if (!dai->active) {
send_ssp_cmd(dai, dai->name, 0);
sst_handle_vb_timer(dai, false);
}
}
static struct snd_soc_dai_ops sst_media_dai_ops = {
.startup = sst_media_open,
.shutdown = sst_media_close,
.prepare = sst_media_prepare,
.hw_params = sst_media_hw_params,
.hw_free = sst_media_hw_free,
.mute_stream = sst_media_digital_mute,
};
static struct snd_soc_dai_ops sst_compr_dai_ops = {
.mute_stream = sst_media_digital_mute,
};
static struct snd_soc_dai_ops sst_be_dai_ops = {
.startup = sst_enable_ssp,
.shutdown = sst_disable_ssp,
};
static struct snd_soc_dai_driver sst_platform_dai[] = {
{
.name = "media-cpu-dai",
.ops = &sst_media_dai_ops,
.playback = {
.stream_name = "Headset Playback",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "Headset Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
.name = "compress-cpu-dai",
.compress_dai = 1,
.ops = &sst_compr_dai_ops,
.playback = {
.stream_name = "Compress Playback",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
/* BE CPU Dais */
{
.name = "ssp0-port",
.ops = &sst_be_dai_ops,
.playback = {
.stream_name = "ssp0 Tx",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "ssp0 Rx",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
.name = "ssp1-port",
.ops = &sst_be_dai_ops,
.playback = {
.stream_name = "ssp1 Tx",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "ssp1 Rx",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
.name = "ssp2-port",
.ops = &sst_be_dai_ops,
.playback = {
.stream_name = "ssp2 Tx",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "ssp2 Rx",
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
};
static int sst_platform_open(struct snd_pcm_substream *substream)
......@@ -609,6 +706,7 @@ static int sst_platform_probe(struct platform_device *pdev)
pdata->pdev_strm_map = dpcm_strm_map;
pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map);
drv->pdata = pdata;
drv->pdev = pdev;
mutex_init(&drv->lock);
dev_set_drvdata(&pdev->dev, drv);
......
sound/soc/intel/sst-mfld-platform.h
View file @ addaeea9
......@@ -117,6 +117,7 @@ struct compress_sst_ops {
int (*get_codec_caps)(struct snd_compr_codec_caps *codec);
int (*set_metadata)(struct device *dev, unsigned int str_id,
struct snd_compr_metadata *mdata);
int (*power)(struct device *dev, bool state);
};
struct sst_ops {
......@@ -153,6 +154,10 @@ struct sst_device {
struct sst_data;
int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform);
int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute);
int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable);
int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable);
void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
struct snd_sst_params *str_params, bool is_compress);
......
sound/soc/intel/sst/Makefile 0 → 100644
View file @ addaeea9
snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o
snd-intel-sst-pci-objs += sst_pci.o
snd-intel-sst-acpi-objs += sst_acpi.o
obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o
obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o
obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o
sound/soc/intel/sst/sst.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst.h 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_acpi.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_drv_interface.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_ipc.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_loader.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_pci.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_pvt.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/intel/sst/sst_stream.c 0 → 100644
View file @ addaeea9
This diff is collapsed.
sound/soc/jz4740/qi_lb60.c
View file @ addaeea9
......@@ -77,25 +77,18 @@ static int qi_lb60_probe(struct platform_device *pdev)
{
struct qi_lb60 *qi_lb60;
struct snd_soc_card *card = &qi_lb60_card;
int ret;
qi_lb60 = devm_kzalloc(&pdev->dev, sizeof(*qi_lb60), GFP_KERNEL);
if (!qi_lb60)
return -ENOMEM;
qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd");
qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd", GPIOD_OUT_LOW);
if (IS_ERR(qi_lb60->snd_gpio))
return PTR_ERR(qi_lb60->snd_gpio);
ret = gpiod_direction_output(qi_lb60->snd_gpio, 0);
if (ret)
return ret;
qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp");
qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp", GPIOD_OUT_LOW);
if (IS_ERR(qi_lb60->amp_gpio))
return PTR_ERR(qi_lb60->amp_gpio);
ret = gpiod_direction_output(qi_lb60->amp_gpio, 0);
if (ret)
return ret;
card->dev = &pdev->dev;
......
sound/soc/soc-jack.c
View file @ addaeea9
......@@ -309,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
/* GPIO descriptor */
gpios[i].desc = gpiod_get_index(gpios[i].gpiod_dev,
gpios[i].name,
gpios[i].idx);
gpios[i].idx, GPIOD_IN);
if (IS_ERR(gpios[i].desc)) {
ret = PTR_ERR(gpios[i].desc);
dev_err(gpios[i].gpiod_dev,
......@@ -327,17 +327,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
goto undo;
}
ret = gpio_request(gpios[i].gpio, gpios[i].name);
ret = gpio_request_one(gpios[i].gpio, GPIOF_IN,
gpios[i].name);
if (ret)
goto undo;
gpios[i].desc = gpio_to_desc(gpios[i].gpio);
}
ret = gpiod_direction_input(gpios[i].desc);
if (ret)
goto err;
INIT_DELAYED_WORK(&gpios[i].work, gpio_work);
gpios[i].jack = jack;
......
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