[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC, MIC_BIAS
jiri.prchal at aksignal.cz
Wed Jun 27 09:48:12 CEST 2012
Dne 26.6.2012 15:17, Mark Brown napsal(a):
> On Tue, Jun 26, 2012 at 03:04:27PM +0200, Prchal Jiří wrote:
>> SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
>> + SOC_ENUM("Left AGC Target level", aic3x_enum[LAGC_LEV_ENUM]),
>> + SOC_ENUM("Right AGC Target level", aic3x_enum[RAGC_LEV_ENUM]),
>> + SOC_ENUM("Left AGC Attack time", aic3x_enum[LAGC_ATT_ENUM]),
>> + SOC_ENUM("Right AGC Attack time", aic3x_enum[RAGC_ATT_ENUM]),
>> + SOC_ENUM("Left AGC Decay time", aic3x_enum[LAGC_DEC_ENUM]),
>> + SOC_ENUM("Right AGC Decay time", aic3x_enum[RAGC_DEC_ENUM]),
> Please move everything out of the big array or if you insist on having
> the array then use the indexes in the assignments. Otherwise the code
> is just error prone and hard to read.
I just add some elements to big array.
As I see in other codec drivers (e.g. wm9713) there are big arrays too.
I think it's more readable to use symbolic names instead of indexes.
>> + SOC_ENUM("Mic Bias", aic3x_enum[MIC_BIAS_ENUM]),
> No, this should be platform data or done with a callback from the
> machine driver. It's very unlikely that it's sensible to vary at
> runtime without coordination with other driver code.
I don't know how, can you, please, suggest me. There are three levels of voltage.
>> + /* set to avoid artifacts on the audio output during power-on/off */
>> + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_B, 0x80); /*ac-coupled*/
>> + snd_soc_write(codec, HPOUT_POP_REDUCTION, 0x4e); /* 10 + 4 ms, reference*/
>> + /* short circuit protection */
>> + snd_soc_write(codec, HPRCOM_CFG, 0x04);
> This appears to be unrelated to the change. What is it for?
This sets default values and I think it would be nice to have defaultly reduced pops-up and short circuit protection.
>> /* DAC to Mono Line Out default volume and route to Output mixer */
>> - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
>> - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
>> + /* mix both channels with -6dB level */
>> + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON);
>> + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON);
> This is also an unrelated change to the chip defaults (which we were
> overriding anyway when we shouldn't be...). Code like this should just
> be removed completely.
No, correct mixing two channels to one is 1/2 + 1/2. 1/2 = -6dB.
>> + /* AGC to -10dB, 20 / 500ms, no clip stepping, noise gate -90dB, hysteresis 3dB*/
>> + snd_soc_write(codec, LAGC_CTRL_A, 0x2f);
>> + snd_soc_write(codec, RAGC_CTRL_A, 0x2f);
>> + snd_soc_write(codec, LAGC_CTRL_C, 0xbe);
>> + snd_soc_write(codec, RAGC_CTRL_C, 0xbe);
> Again, just use the chip defaults and let the user override.
Originally the driver sets many defaults to hear playback, because after reset defaults won't hear anything and user
need to set many switches and volumes. Maybe is not necessary to set AGC here.
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