ASoC: codecs: Add DA9055 codec driver
Subsystems (3)
PATCH 1/1 ASoC: codecs: Add DA9055 codec driver
This patch adds support for Dialog semiconductor's DA9055 audio codec.
This has been tested on DA9055 EVB with Samsung SMDK6410 board.
--- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1507 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1513 insertions(+), 0 deletions(-)
sound/soc/codecs/Kconfig +4 -0
@@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC
@@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate
sound/soc/codecs/Makefile +2 -0
@@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o
@@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o
sound/soc/codecs/da9055.c new file +1507 -0
1,507 lines not on this page · raw message
Signature · 1 line
1.7.1
sound/soc/codecs/da9055.c line 393
Quoted 7 lines · Ashish Chavan
sound/soc/codecs/da9055.c new file +6
-0+/* LDO voltage level value */ +static const char *const da9055_ldo_lvl_select_txt[] = { +"1.05V" ,"1.1V" ,"1.2V" ,"1.4V" +}; +static const struct soc_enum da9055_ldo_lvl_select = +SOC_ENUM_SINGLE (DA9055_LDO_CTRL ,4 ,4 , da9055_ldo_lvl_select_txt);
There's a regulator API, if configuration is required we should be using that.
Quoted 10 lines
sound/soc/codecs/da9055.c new file +10
-0+/* Digital MIC clock rate select */ +static const char *const da9055_dmic_clk_rate_txt[] = { +"3MHz" ,"1.5MHz" +}; + +static const struct soc_enum da9055_dmic_clk_rate = +SOC_ENUM_SINGLE (DA9055_MIC_CONFIG ,2 ,2 , da9055_dmic_clk_rate_txt); + +/* Digital MIC sample phase select */ +static const char *const da9055_dmic_phase_txt[] = {
sound/soc/codecs/da9055.c line 437
Quoted 10 lines
sound/soc/codecs/da9055.c new file +10
-0+ "Sample on DMICCLK edges", "Sample between DMICCLK edges" +}; + +static const struct soc_enum da9055_dmic_phase = +SOC_ENUM_SINGLE (DA9055_MIC_CONFIG ,1 ,2 , da9055_dmic_phase_txt); + +/* Digital MIC channel select */ +static const char *const da9055_dmic_channel_select_txt[] = { +"Rising Left Falling Right" ,"Falling Left Rising Right" +};
Why is any of this being exposed to userspace? If this should be configured I'd expect it to be static platform data, not something that gets changed at runtime.
sound/soc/codecs/da9055.c line 455
Quoted 4 lines
sound/soc/codecs/da9055.c new file +4
-0+/* MIC bias voltage level select */ +static const char *const da9055_mic_bias_level_txt[] = { +"1.6V" ,"1.8V" ,"2.1V" ,"2.2V" +};
Again, why is this being exposed to userspace/ I'm fairly sure we went through similar stuff with your last driver...
sound/soc/codecs/da9055.c line 618
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+SOC_DOUBLE_R_TLV ("HeadPhone Volume" , +DA9055_HP_L_GAIN ,DA9055_HP_R_GAIN , + 0, 0x3f, 0, hp_vol_tlv),
Headphone.
sound/soc/codecs/da9055.c line 658
Quoted 13 lines
sound/soc/codecs/da9055.c new file +13
-0+/* Mute controls */ +SOC_DOUBLE_R ("Mic Mute Switch" ,DA9055_MIC_L_CTRL , +DA9055_MIC_R_CTRL ,6 ,1 ,0 ), +SOC_DOUBLE_R ("Aux Mute Switch" ,DA9055_AUX_L_CTRL , +DA9055_AUX_R_CTRL ,6 ,1 ,0 ), +SOC_DOUBLE_R ("Mixin PGA Mute Switch" ,DA9055_MIXIN_L_CTRL , +DA9055_MIXIN_R_CTRL ,6 ,1 ,0 ), +SOC_DOUBLE_R ("ADC Mute Switch" ,DA9055_ADC_L_CTRL , +DA9055_ADC_R_CTRL ,6 ,1 ,0 ), +SOC_DOUBLE_R ("HeadPhone Mute Switch" ,DA9055_HP_L_CTRL , +DA9055_HP_R_CTRL ,6 ,1 ,0 ), +SOC_SINGLE ("Lineout Mute Switch" ,DA9055_LINE_CTRL ,6 ,1 ,0 ), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0),
No "Mute". Again, I'm fairly sure we had the same issue last time.
sound/soc/codecs/da9055.c line 696
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* LDO control */ +SOC_SINGLE ("LDO Enable" ,DA9055_LDO_CTRL ,7 ,1 ,0 ), + SOC_ENUM("LDO Level Select", da9055_ldo_lvl_select),
The LDO enable should absolutely not be being exposed to userspace!
sound/soc/codecs/da9055.c line 712
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* DMIC controls */ +SOC_DOUBLE_R ("DMIC Enable" ,DA9055_MIXIN_L_SELECT , + DA9055_MIXIN_R_SELECT, 7, 1, 0),
Switch if this is a mute.
sound/soc/codecs/da9055.c line 719
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* MIC PGA input selection controls */ +SOC_ENUM ("Mic Left Input Select" , da9055_mic_l_select), + SOC_ENUM("Mic Right Input Select", da9055_mic_r_select),
DAPM.
sound/soc/codecs/da9055.c line 729
Quoted 4 lines
sound/soc/codecs/da9055.c new file +4
-0+ +/* ALC Controls */ +SOC_DOUBLE_EXT ("ALC Enable" ,DA9055_ALC_CTRL1 ,3 ,7 ,1 ,0 , + snd_soc_get_volsw, da9055_put_alc_sw),
ALC Switch. All enable controls should be switches.
sound/soc/codecs/da9055.c line 1044
Quoted 12 lines
sound/soc/codecs/da9055.c new file +12
-0+static int da9055_hw_params (struct snd_pcm_substream *substream, +struct snd_pcm_hw_params *params, +struct snd_soc_dai *dai) +{ +struct snd_soc_codec *codec = dai->codec; +struct da9055_priv *da9055 =snd_soc_codec_get_drvdata (codec); +u8 aif_ctrl, fs; +u32 sysclk; + +/* Set AIF source to Left and Right ADC */ +snd_soc_write (codec,DA9055_DIG_ROUTING_AIF , +DA9055_AIF_L_SRC |DA9055_AIF_R_SRC );
This should be in DAPM.
sound/soc/codecs/da9055.c line 1046
Quoted 1 line
sound/soc/codecs/da9055.c new file +1
-0+ aif_ctrl =snd_soc_read (codec,DA9055_AIF_CTRL ) &0xf3 ;
Use snd_soc_update_bits() later on.
sound/soc/codecs/da9055.c line 1065
Quoted 1 line
sound/soc/codecs/da9055.c new file +1
-0+ aif_ctrl |= (DA9055_AIF_OE |DA9055_AIF_EN );
DAPM.
sound/soc/codecs/da9055.c line 1276
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* In slave mode, there is only one set of divisors */ +if (!da9055->master) + fout =2822400 ;
Should check the user supplied this value - also, what happens if the user sets the device to slave mode after setting up the PLL?
sound/soc/codecs/da9055.c line 1379
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* Enable VMID reference & master bias */ +snd_soc_write (codec,DA9055_REFERENCES , +DA9055_VMID_EN |DA9055_BIAS_EN );
set_bias_level()
sound/soc/codecs/da9055.c line 1382
Quoted 2 lines
sound/soc/codecs/da9055.c new file +2
-0+/* Enable Mic Bias */ +snd_soc_write (codec,DA9055_MIC_BIAS_CTRL ,DA9055_MIC_BIAS_EN );
DAPM for this and most of the rest of this funciton.
sound/soc/codecs/da9055.c line 1460
Quoted 1 line
sound/soc/codecs/da9055.c new file +1
-0+ da9055->regmap =regmap_init_i2c (i2c, &da9055_regmap_config);
devm_regmap_init_i2c()
Quoted 16 lines · Mark Brown
Quoted 16 lines
sound/soc/codecs/da9055.c new file +16
-0+/* Digital MIC clock rate select */ +static const char *const da9055_dmic_clk_rate_txt[] = { +"3MHz" ,"1.5MHz" +}; + +static const struct soc_enum da9055_dmic_clk_rate = +SOC_ENUM_SINGLE (DA9055_MIC_CONFIG ,2 ,2 , da9055_dmic_clk_rate_txt); + +/* Digital MIC sample phase select */ +static const char *const da9055_dmic_phase_txt[] = { +"Sample on DMICCLK edges" ,"Sample between DMICCLK edges" +}; + +static const struct soc_enum da9055_dmic_phase = +SOC_ENUM_SINGLE (DA9055_MIC_CONFIG ,1 ,2 , da9055_dmic_phase_txt); +
Quoted 9 lines
sound/soc/codecs/da9055.c line 437
Quoted 4 lines
sound/soc/codecs/da9055.c new file +4
-0+/* Digital MIC channel select */ +static const char *const da9055_dmic_channel_select_txt[] = { +"Rising Left Falling Right" ,"Falling Left Rising Right" +};Why is any of this being exposed to userspace? If this should be configured I'd expect it to be static platform data, not something that gets changed at runtime.
These parameters are exposed considering the fact that DMIC itself is not part of the codec. Codec only provides DMIC interface using which an external DMIC can be attached. These parameters depend on the actual DMIC hardware and hence kept configurable to allow runtime plug in of any DMIC hardware. Doesn't it make sense to keep them runtime configurable?
Quoted 6 lines
sound/soc/codecs/da9055.c line 1276
Quoted 4 lines
sound/soc/codecs/da9055.c new file +3
-0+/* In slave mode, there is only one set of divisors */ +if (!da9055->master) + fout =2822400 ;Should check the user supplied this value
Can you explain which value / user supplied value you are referring to? It is not quite clear to me.
Quoted 2 lines
- also, what happens if the
user sets the device to slave mode after setting up the PLL?
Will add required checks in set_fmt().
Quoted 4 lines
sound/soc/codecs/da9055.c line 1382
Quoted 2 lines
sound/soc/codecs/da9055.c new file +2
-0+/* Enable Mic Bias */ +snd_soc_write (codec,DA9055_MIC_BIAS_CTRL ,DA9055_MIC_BIAS_EN );DAPM for this and most of the rest of this funciton.
I guess here a DAPM_SUPPLY widget should be used and its routing should be part of machine driver instead of this codec driver, right?
For other things like input mixers, Headphone and Lineouts, DAPM is already used to control power specific bits. The confusion is because there are two separate control bits for these blocks. One bit is for "Enabling" that block and other is for "Enabling Amplifier" of that block.
e.g for headphone, one bit is for "output enable" while other is for "output amplifier enable".
Chip designers have confirmed that "Amplifier Enable" bits are ones which should be used for power management. This is the reason why "Enable" bits are being set in probe() and "Amplifier Enable" bits are being taken care by DAPM. May be I need to put enough comments in code to clarify this.
Quoted 6 lines · Ashish Chavan
Quoted 3 lines · Mark Brown
Why is any of this being exposed to userspace? If this should be configured I'd expect it to be static platform data, not something that gets changed at runtime.
Quoted 6 lines
These parameters are exposed considering the fact that DMIC itself is not part of the codec. Codec only provides DMIC interface using which an external DMIC can be attached. These parameters depend on the actual DMIC hardware and hence kept configurable to allow runtime plug in of any DMIC hardware. Doesn't it make sense to keep them runtime configurable?
No. The only realistic way to attach a new DMIC to a board is to solder it down, that's not something people are going to do while the system is actve. It's something that's fixed at PCB design time.
Quoted 8 lines
Quoted 5 lines
sound/soc/codecs/da9055.c line 1276
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* In slave mode, there is only one set of divisors */ +if (!da9055->master) + fout =2822400 ;Should check the user supplied this value
Can you explain which value / user supplied value you are referring to? It is not quite clear to me.
The specified output frequency, you just totally ignore it.
Quoted 7 lines
For other things like input mixers, Headphone and Lineouts, DAPM is already used to control power specific bits. The confusion is because there are two separate control bits for these blocks. One bit is for "Enabling" that block and other is for "Enabling Amplifier" of that block.
e.g for headphone, one bit is for "output enable" while other is for "output amplifier enable".
So document this.
FYI.
-----Original Message-----
From: Mark Brown [mailto:[email protected]]
Sent: Wednesday, September 12, 2012 6:20 PM
To: Ashish P. Chavan
Cc: lrg; alsa-devel; linux-kernel; David Dajun Chen
Subject: Re: [alsa-devel] [PATCH] ASoC: codecs: Add DA9055 codec driver
Quoted 6 lines · Ashish Chavan
Quoted 3 lines · Mark Brown
Why is any of this being exposed to userspace? If this should be configured I'd expect it to be static platform data, not something that gets changed at runtime.
Quoted 6 lines
These parameters are exposed considering the fact that DMIC itself is not part of the codec. Codec only provides DMIC interface using which an external DMIC can be attached. These parameters depend on the actual DMIC hardware and hence kept configurable to allow runtime plug in of any DMIC hardware. Doesn't it make sense to keep them runtime configurable?
No. The only realistic way to attach a new DMIC to a board is to solder it down, that's not something people are going to do while the system is actve. It's something that's fixed at PCB design time.
Quoted 8 lines
Quoted 5 lines
sound/soc/codecs/da9055.c line 1276
Quoted 3 lines
sound/soc/codecs/da9055.c new file +3
-0+/* In slave mode, there is only one set of divisors */ +if (!da9055->master) + fout =2822400 ;Should check the user supplied this value
Can you explain which value / user supplied value you are referring to? It is not quite clear to me.
The specified output frequency, you just totally ignore it.
Quoted 7 lines
For other things like input mixers, Headphone and Lineouts, DAPM is already used to control power specific bits. The confusion is because there are two separate control bits for these blocks. One bit is for "Enabling" that block and other is for "Enabling Amplifier" of that block.
e.g for headphone, one bit is for "output enable" while other is for "output amplifier enable".
So document this.
Quoted 4 lines · Mark Brown
sound/soc/codecs/da9055.c line 1065
Quoted 2 lines
sound/soc/codecs/da9055.c new file +1
-0+ aif_ctrl |= (DA9055_AIF_OE |DA9055_AIF_EN );DAPM.
Here the trouble in making it DAPM based is that there is no separate control for AIF input and output. It is confirmed that AIF_EN is the master control bit, which enables both output as well as input. The AIF _OE bit is redundant. If we use the same control bit(i.e AIF_EN) for both DAPM_AIF_IN and DAPM_AIF_OUT, there will be obvious side effects. Is there any other way to take care of this? Or should we leave it outside DAPM?
Quoted 6 lines · Ashish Chavan
Quoted 3 lines · Mark Brown
sound/soc/codecs/da9055.c line 1065
Quoted 1 line
sound/soc/codecs/da9055.c new file +1
-0+ aif_ctrl |= (DA9055_AIF_OE |DA9055_AIF_EN );DAPM.
Quoted 7 lines
Here the trouble in making it DAPM based is that there is no separate control for AIF input and output. It is confirmed that
AIF_ENis the master control bit, which enables both output as well as input. The AIF_OEbit is redundant. If we use the same controlbit(i.e AIF_EN)for bothDAPM_AIF_INandDAPM_AIF_OUT, there will be obvious side effects. Is there any other way to take care of this? Or should we leave it outside DAPM?
Use a supply widget.