On 03/03/2014 08:34 AM, Mark Brown wrote:
On Wed, Feb 26, 2014 at 11:14:25AM +0200, Jyri Sarha wrote:
This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs.
Always CC maintainers on patches please.
Will do.
+static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{
- if (aic31xx_precious(dev, reg) || aic31xx_real_volatile(dev, reg))
return true;- switch (reg) {
- case AIC31XX_PAGECTL: /* regmap implementation requires this */
- case AIC31XX_RESET: /* always clears after write */
return true;- }
- return false;
+}
This is more complex than you need, just write a standalone volatile function with some comments in it.
Replaced with straight forward aic31xx_volatile() and aic31xx_writable() functions.
+static const struct regmap_range_cfg aic31xx_ranges[] = {
- {
.name = "codec-regmap",Make this meaningful or omit it.
Removed.
+#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[] = {
Use the define in the array size to help check everything lines up.
Done.
+static const char * const ldac_in_text[] = {
- "off", "Left Data", "Right Data", "Mono"
+};
Off not off - be consistent both internally and with the general style.
Done.
+static const struct snd_kcontrol_new aic311x_snd_controls[] = {
- SOC_DOUBLE_R("SP Driver Playback Switch", AIC31XX_SPLGAIN,
AIC31XX_SPRGAIN, 2, 1, 0),SP?
"SP" replaced with "Speaker", also in DAPM switches.
- if (!strcmp(w->name, "DAC Left")) {
mask = AIC31XX_LDACPWRSTATUS_MASK;There's no overlap with the enable bits? In general it would seem nicer to have a switch based on the register bits used for the enable rather than the tree of string comparisions but it's probably not that big a deal overall.
Not with bits, but with regs. I decided this structure would be easier to implement and understand than having a switch(w->reg) with nested switch(w->shift) cases. I replaced the string comparisons with a single switch case using a binary combination of w->reg and w->shift.
dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",w->name, __func__);return -1; switch (w->shift) {
case 7: mask = AIC31XX_LDACPWRSTATUS_MASK;
Return real error codes.
Changed to -EINVAL.
- if (event == SND_SOC_DAPM_POST_PMU)
return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);- else if (event == SND_SOC_DAPM_POST_PMD)
return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);switch.
Done.
- SND_SOC_DAPM_DAC_E("DAC Right", "DAC Right Input",
AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),Don't use stream names, use DAPM to route from the AIF to the widgets.
Renamed DAC stnames to "Left Playback" and "Right Playback".
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
break;params_width().
Done.
AIC31XX_IFACE1_DATALEN_MASK,data);- return aic31xx_setup_pll(codec, params);
This is going to mean that you have a symmetry constraint I think, if you try to set up different rates for playback and capture presumably things will break. Setting symmetric_rates will tell applications about that.
Done.
- case SND_SOC_DAIFMT_CBS_CFM:
- case SND_SOC_DAIFMT_CBM_CFS:
- case SND_SOC_DAIFMT_CBS_CFS:
dev_err(codec->dev, "Unsupported DAI master/slave interface\n");return -EINVAL;- default:
dev_alert(codec->dev, "Invalid DAI master/slave interface\n");return -EINVAL;Just have a default.
Done.
- for (i = 0; aic31xx_divs[i].mclk != freq; i++)
if (i == ARRAY_SIZE(aic31xx_divs)) {dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",__func__, freq);return -EINVAL;}Braces around the for () too please.
Done.
+static int aic31xx_set_power(struct snd_soc_codec *codec, int power) +{
- struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
- int ret;
- dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
aic31xx->power, power);- if (aic31xx->power == power)
return 0;This looks like you need sensible refcounting somewhere? Implementing this as the standard PM operations may be sensible.
Not really. This was just avoid power on sequence when bias level goes from SND_SOC_BIAS_PREPARE back to SND_SOC_BIAS_STANDBY. I'll change this to a "if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)" in set_bias_level like other codecs do.
if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {gpio_set_value(aic31xx->pdata.gpio_reset, 1);udelay(100);}snd_soc_write(codec, AIC31XX_RESET, 0x01);udelay(100);regcache_cache_only(aic31xx->regmap, false);You should be coming out of cache only mode before you issue the reset write. Is it not sensible to skip that if you have a physical reset line?
snd_soc_write(codec, AIC31XX_RESET, 0x01);regcache_cache_only(aic31xx->regmap, true);if (gpio_is_valid(aic31xx->pdata.gpio_reset))gpio_set_value(aic31xx->pdata.gpio_reset, 0);Likewise here. Also if you do reset the CODEC then the dance you did with the regulator notifiers becomes pointless - the goal with that is to avoid the need to resync the cache if the CODEC wasn't actually reset by a power cycle but since the CODEC is going to be explicitly reset before it's powered off there's no benefit.
I rewrote the bias-level/power logic a bit. It should now look more like the other codecs. I hope you like this version better.
- switch (level) {
- /* full On */
- case SND_SOC_BIAS_ON:
/* All power is driven by DAPM system*/break;- /* partial On */
Coding style, please.
Removed the redundant badly indented comments.
Fixed also the Kconfig and Makefile ordering and added dt-include for micbias and changed the default to 2.0V. All are now squashed to a single patch.
In addition to addressing your comments I also added an internal ADC input and DAC output for codec to turn on at playback/capture start even if the alsamixers are not set correctly.
I'll mail the patches shortly.
Best regards, Jyri