On Tue, Aug 17, 2010 at 5:58 PM, Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Tue, Aug 17, 2010 at 03:47:59PM +0800, Haojian Zhuang wrote:
+/* DAPM Widget Events */ +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+{
- struct snd_soc_codec *codec = w->codec;
- /* unmute DAC */
- snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
Can you explain what's going on with this mute handling please?
Em. Actually there should be automute variable. I shouldn't delete that variable. In order to anti-pop, mute DAC before enabling DAC. Unmute it after enabling DAC. It's required by silicon.
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1);
- snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2);
I still don't follow why you need a custom event for this.
Enabling both bit 0 of ADC_EN_1 and bit 5 of ADC_EN_2 can enable left ADC. Enabling both bit 1 of ADC_EN_1 and bit 4 of ADC_EN_2 can enable right ADC. I can't find any DAPM API can handle this. So I implement the custom event.
+static int pm860x_mic1_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+{
- struct snd_soc_codec *codec = w->codec;
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- /* Enable Mic1 Bias & MICDET, HSDET */
- snd_soc_update_bits(codec, PM860X_ADC_ANA_1, MIC1BIAS_MASK,
- MIC1BIAS_MASK);
As I said last time you should handle this via DAPM.
I registered it as DAPM widget. Why do you say it's not handled via DAPM?
- pm860x_set_bits(codec->control_data, REG_MIC_DET,
- MICDET_MASK, MICDET_MASK);
- pm860x_set_bits(codec->control_data, REG_HS_DET,
- EN_HS_DET, EN_HS_DET);
This should be associated with enabling microphone detection.
Yes, but you said that it could be controlled by enable_pin(). I forced to enable microphone pins in machine driver.
- /* set master/slave audio interface */
- switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBM_CFM:
- case SND_SOC_DAIFMT_CBM_CFS:
- if (pm860x->dir == PM860X_CLK_DIR_OUT)
- *inf |= PCM_INF2_MASTER;
- else
- return -EINVAL;
- break;
You're setting the same register configuration for two different DAI clock master configurations here. Presumably one of the settings is inaccurate?
No, they're different registers. But offsets are same. So I just return pointer.
+static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
- int clk_id, unsigned int freq, int dir)
+{
- struct snd_soc_codec *codec = codec_dai->codec;
- struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
- if (dir == PM860X_CLK_DIR_OUT)
- pm860x->dir = PM860X_CLK_DIR_OUT;
- else
- pm860x->dir = PM860X_CLK_DIR_IN;
- return 0;
+}
What is this actually setting - which clock is being configured here?
While codec is master, the clock is fixed. I needn't set detail clock. While codec is slave, it's not supported in this patch yet.
+static irqreturn_t pm860x_codec_handler(int irq, void *data) +{
- struct pm860x_priv *pm860x = data;
- int status, shrt, report = 0;
- status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
- shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
- if (status & HEADSET_STATUS)
- report |= PM860X_DET_HEADSET;
- if (status & MIC_STATUS)
- report |= PM860X_DET_MIC;
- if (status & HOOK_STATUS)
- report |= PM860X_DET_HOOK;
- if (shrt & (SHORT_LO1 | SHORT_LO2))
- report |= PM860X_SHORT_LINEOUT;
- if (shrt & (SHORT_HS1 | SHORT_HS2))
- report |= PM860X_SHORT_HEADSET;
- dev_dbg(pm860x->codec->dev, "report:0x%x\n", report);
- return IRQ_HANDLED;
It would seem better to just remove the interrupt handling support entirely if you're not going to implement jack detection. Right now all the curernt code will do is waste power by enabling the feature but ignoring the result.
I need a document on illustrating jack on alsa. Could you share one?