[alsa-devel] [PATCH 1/3] ASoC: add 88pm860x codec driver
On Tue, Aug 17, 2010 at 03:47:59PM +0800, Haojian Zhuang wrote:
+enum {
- FILTER_BYPASS = 0,
- FILTER_LOW_PASS_1,
- FILTER_LOW_PASS_2,
- FILTER_HIGH_PASS_3,
- FILTER_MAX,
+};
This looks to be unused.
+/* 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?
- 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.
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
if (dac) {dac |= MODULATOR;/* mute */snd_soc_update_bits(codec, PM860X_DAC_OFFSET,DAC_MUTE, DAC_MUTE);
This mute handling looks very suspicious.
+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.
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.
- /* 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;elsereturn -EINVAL;break;
You're setting the same register configuration for two different DAI clock master configurations here. Presumably one of the settings is inaccurate?
+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?
+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.
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?
On Tue, Aug 17, 2010 at 06:44:44PM +0800, Haojian Zhuang wrote:
On Tue, Aug 17, 2010 at 5:58 PM, Mark Brown
- /* 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.
As requested in the previous review comments you need to provide better documentation of this in the driver. I'd also expect to see a custom mute control somewhere which implements this by keeping the DAC muted unless it has been enabled rather than some sort of automute handling.
- 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.
As I said previously I would expect you to be using a DAPM supply widget for this.
+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?
I say this because this code is manually going in and enabling the microphone bias here. If the microphone bias is being handled using DAPM you shouldn't be going around explicitly writing to the power control bit like this.
- 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.
I don't follow what you're saying at all. This code will unconditonally enable headset and microphone detection while the microphone is enabled, regardless of the user deciding if they want to use that feature or not. The use of enable_pin() was for the microphone bias which is a separate thing.
- /* 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.
I can't associate your comment there with the code at all. The code does nothing different for the two case statements and there's no other code I can see.
+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.
What is "the clock", and if slave mode is not supported surely an error should be returned?
- 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?
There's a number of in tree examples - seach for snd_soc_jack.
On Tue, Aug 17, 2010 at 7:02 PM, Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Tue, Aug 17, 2010 at 06:44:44PM +0800, Haojian Zhuang wrote:
On Tue, Aug 17, 2010 at 5:58 PM, Mark Brown
- 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.
As I said previously I would expect you to be using a DAPM supply widget for this.
Supply widget should be always on while DAPM is working. How about only enable left ADC?
- /* 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.
I can't associate your comment there with the code at all. The code does nothing different for the two case statements and there's no other code I can see.
You can find pm860x_pcm_set_dai_fmt() and pm860x_i2s_set_dai_fmt(). Both functions called set_dai_fmt(). In set_dai_fmt(), all bit operations are based on variable, not registers. Register operation is implemented in pcm_set_dai_fmt() and i2s_set_dai_fmt().
- 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?
There's a number of in tree examples - seach for snd_soc_jack.
OK. How to test it? Still no document?
On Tue, Aug 17, 2010 at 07:37:21PM +0800, Haojian Zhuang wrote:
On Tue, Aug 17, 2010 at 7:02 PM, Mark Brown
As I said previously I would expect you to be using a DAPM supply widget for this.
Supply widget should be always on while DAPM is working. How about only enable left ADC?
What makes you say this? A supply widget is only enabled when the things it supplies are enabled.
I can't associate your comment there with the code at all. The code does nothing different for the two case statements and there's no other code I can see.
You can find pm860x_pcm_set_dai_fmt() and pm860x_i2s_set_dai_fmt(). Both functions called set_dai_fmt(). In set_dai_fmt(), all bit operations are based on variable, not registers. Register operation is implemented in pcm_set_dai_fmt() and i2s_set_dai_fmt().
You need to move this code into the individual functions. What you've got now is clearly buggy - it will accept either configuration when it sets the variable.
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?
There's a number of in tree examples - seach for snd_soc_jack.
OK. How to test it? Still no document?
A standard API input test application such as evtest can be used.
participants (2)
-
Haojian Zhuang -
Mark Brown