Merge branch 'topic/asoc' into for-linus
This commit is contained in:
Takashi Iwai
commit
b114701c0e
71
Documentation/sound/alsa/soc/jack.txt
Normal file
71
Documentation/sound/alsa/soc/jack.txt
Normal file
|
|
@ -0,0 +1,71 @@
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|||
ASoC jack detection
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===================
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ALSA has a standard API for representing physical jacks to user space,
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the kernel side of which can be seen in include/sound/jack.h. ASoC
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provides a version of this API adding two additional features:
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- It allows more than one jack detection method to work together on one
|
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user visible jack. In embedded systems it is common for multiple
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to be present on a single jack but handled by separate bits of
|
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hardware.
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- Integration with DAPM, allowing DAPM endpoints to be updated
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automatically based on the detected jack status (eg, turning off the
|
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headphone outputs if no headphones are present).
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This is done by splitting the jacks up into three things working
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together: the jack itself represented by a struct snd_soc_jack, sets of
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snd_soc_jack_pins representing DAPM endpoints to update and blocks of
|
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code providing jack reporting mechanisms.
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For example, a system may have a stereo headset jack with two reporting
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mechanisms, one for the headphone and one for the microphone. Some
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systems won't be able to use their speaker output while a headphone is
|
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connected and so will want to make sure to update both speaker and
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headphone when the headphone jack status changes.
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The jack - struct snd_soc_jack
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==============================
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This represents a physical jack on the system and is what is visible to
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user space. The jack itself is completely passive, it is set up by the
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machine driver and updated by jack detection methods.
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Jacks are created by the machine driver calling snd_soc_jack_new().
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snd_soc_jack_pin
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================
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These represent a DAPM pin to update depending on some of the status
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bits supported by the jack. Each snd_soc_jack has zero or more of these
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which are updated automatically. They are created by the machine driver
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and associated with the jack using snd_soc_jack_add_pins(). The status
|
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of the endpoint may configured to be the opposite of the jack status if
|
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required (eg, enabling a built in microphone if a microphone is not
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connected via a jack).
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Jack detection methods
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======================
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Actual jack detection is done by code which is able to monitor some
|
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input to the system and update a jack by calling snd_soc_jack_report(),
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specifying a subset of bits to update. The jack detection code should
|
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be set up by the machine driver, taking configuration for the jack to
|
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update and the set of things to report when the jack is connected.
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Often this is done based on the status of a GPIO - a handler for this is
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provided by the snd_soc_jack_add_gpio() function. Other methods are
|
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also available, for example integrated into CODECs. One example of
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CODEC integrated jack detection can be see in the WM8350 driver.
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|
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Each jack may have multiple reporting mechanisms, though it will need at
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least one to be useful.
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Machine drivers
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===============
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These are all hooked together by the machine driver depending on the
|
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system hardware. The machine driver will set up the snd_soc_jack and
|
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the list of pins to update then set up one or more jack detection
|
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mechanisms to update that jack based on their current status.
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@ -238,6 +238,8 @@ static inline void pxa_ac97_cold_pxa3xx(void)
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bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
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{
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unsigned long gsr;
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|
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#ifdef CONFIG_PXA25x
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if (cpu_is_pxa25x())
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pxa_ac97_warm_pxa25x();
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|
|
@ -254,10 +256,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
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else
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#endif
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BUG();
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|
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if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
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gsr = GSR | gsr_bits;
|
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if (!(gsr & (GSR_PCR | GSR_SCR))) {
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printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
|
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__func__, gsr_bits);
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__func__, gsr);
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|
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return false;
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}
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|
|
@ -268,6 +270,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
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|||
|
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bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
|
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{
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||||
unsigned long gsr;
|
||||
|
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#ifdef CONFIG_PXA25x
|
||||
if (cpu_is_pxa25x())
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pxa_ac97_cold_pxa25x();
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|
|
@ -285,9 +289,10 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
|
|||
#endif
|
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BUG();
|
||||
|
||||
if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
|
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gsr = GSR | gsr_bits;
|
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if (!(gsr & (GSR_PCR | GSR_SCR))) {
|
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printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
|
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__func__, gsr_bits);
|
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__func__, gsr);
|
||||
|
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return false;
|
||||
}
|
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|
|
|
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|
|
@ -122,6 +122,9 @@ struct twl4030_priv {
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|||
unsigned int bypass_state;
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unsigned int codec_powered;
|
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unsigned int codec_muted;
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|
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struct snd_pcm_substream *master_substream;
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struct snd_pcm_substream *slave_substream;
|
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};
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|
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/*
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|
|
@ -1217,6 +1220,50 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
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|||
return 0;
|
||||
}
|
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|
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static int twl4030_startup(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_device *socdev = rtd->socdev;
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struct snd_soc_codec *codec = socdev->codec;
|
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struct twl4030_priv *twl4030 = codec->private_data;
|
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|
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/* If we already have a playback or capture going then constrain
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* this substream to match it.
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*/
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if (twl4030->master_substream) {
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struct snd_pcm_runtime *master_runtime;
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master_runtime = twl4030->master_substream->runtime;
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|
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snd_pcm_hw_constraint_minmax(substream->runtime,
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SNDRV_PCM_HW_PARAM_RATE,
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master_runtime->rate,
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master_runtime->rate);
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|
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snd_pcm_hw_constraint_minmax(substream->runtime,
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SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
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master_runtime->sample_bits,
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master_runtime->sample_bits);
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|
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twl4030->slave_substream = substream;
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} else
|
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twl4030->master_substream = substream;
|
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|
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return 0;
|
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}
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||||
|
||||
static void twl4030_shutdown(struct snd_pcm_substream *substream)
|
||||
{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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||||
struct snd_soc_device *socdev = rtd->socdev;
|
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struct snd_soc_codec *codec = socdev->codec;
|
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struct twl4030_priv *twl4030 = codec->private_data;
|
||||
|
||||
if (twl4030->master_substream == substream)
|
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twl4030->master_substream = twl4030->slave_substream;
|
||||
|
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twl4030->slave_substream = NULL;
|
||||
}
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|
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static int twl4030_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params,
|
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struct snd_soc_dai *dai)
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||||
|
|
@ -1224,8 +1271,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
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|||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
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struct snd_soc_device *socdev = rtd->socdev;
|
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struct snd_soc_codec *codec = socdev->card->codec;
|
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struct twl4030_priv *twl4030 = codec->private_data;
|
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u8 mode, old_mode, format, old_format;
|
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|
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if (substream == twl4030->slave_substream)
|
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/* Ignoring hw_params for slave substream */
|
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return 0;
|
||||
|
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/* bit rate */
|
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old_mode = twl4030_read_reg_cache(codec,
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TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
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|
|
@ -1259,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
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case 48000:
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mode |= TWL4030_APLL_RATE_48000;
|
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break;
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case 96000:
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mode |= TWL4030_APLL_RATE_96000;
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break;
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default:
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printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
|
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params_rate(params));
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|
|
@ -1384,6 +1439,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
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#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
|
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|
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static struct snd_soc_dai_ops twl4030_dai_ops = {
|
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.startup = twl4030_startup,
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.shutdown = twl4030_shutdown,
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.hw_params = twl4030_hw_params,
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.set_sysclk = twl4030_set_dai_sysclk,
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.set_fmt = twl4030_set_dai_fmt,
|
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|
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@ -1395,7 +1452,7 @@ struct snd_soc_dai twl4030_dai = {
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.stream_name = "Playback",
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.channels_min = 2,
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.channels_max = 2,
|
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.rates = TWL4030_RATES,
|
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.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
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.formats = TWL4030_FORMATS,},
|
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.capture = {
|
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.stream_name = "Capture",
|
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|
|
|
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|
|
@ -109,6 +109,7 @@
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#define TWL4030_APLL_RATE_32000 0x80
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#define TWL4030_APLL_RATE_44100 0x90
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#define TWL4030_APLL_RATE_48000 0xA0
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#define TWL4030_APLL_RATE_96000 0xE0
|
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#define TWL4030_SEL_16K 0x04
|
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#define TWL4030_CODECPDZ 0x02
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#define TWL4030_OPT_MODE 0x01
|
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|
|
|
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|
|
@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec)
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return -EIO;
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}
|
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|
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#ifdef CONFIG_PM
|
||||
static int wm9705_soc_suspend(struct platform_device *pdev)
|
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{
|
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
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struct snd_soc_codec *codec = socdev->card->codec;
|
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|
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soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
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|
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return 0;
|
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}
|
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|
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static int wm9705_soc_resume(struct platform_device *pdev)
|
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{
|
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
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struct snd_soc_codec *codec = socdev->card->codec;
|
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int i, ret;
|
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u16 *cache = codec->reg_cache;
|
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|
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ret = wm9705_reset(codec);
|
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if (ret < 0) {
|
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printk(KERN_ERR "could not reset AC97 codec\n");
|
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return ret;
|
||||
}
|
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|
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for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
|
||||
soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
#else
|
||||
#define wm9705_soc_suspend NULL
|
||||
#define wm9705_soc_resume NULL
|
||||
#endif
|
||||
|
||||
static int wm9705_soc_probe(struct platform_device *pdev)
|
||||
{
|
||||
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
||||
|
|
@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev)
|
|||
struct snd_soc_codec_device soc_codec_dev_wm9705 = {
|
||||
.probe = wm9705_soc_probe,
|
||||
.remove = wm9705_soc_remove,
|
||||
.suspend = wm9705_soc_suspend,
|
||||
.resume = wm9705_soc_resume,
|
||||
};
|
||||
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
|
||||
|
||||
|
|
|
|||
|
|
@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
|
|||
else
|
||||
position = in_be32(&dma_channel->dar);
|
||||
|
||||
/*
|
||||
* When capture is started, the SSI immediately starts to fill its FIFO.
|
||||
* This means that the DMA controller is not started until the FIFO is
|
||||
* full. However, ALSA calls this function before that happens, when
|
||||
* MR.DAR is still zero. In this case, just return zero to indicate
|
||||
* that nothing has been received yet.
|
||||
*/
|
||||
if (!position)
|
||||
return 0;
|
||||
|
||||
if ((position < dma_private->dma_buf_phys) ||
|
||||
(position > dma_private->dma_buf_end)) {
|
||||
dev_err(substream->pcm->card->dev,
|
||||
"dma pointer is out of range, halting stream\n");
|
||||
return SNDRV_PCM_POS_XRUN;
|
||||
}
|
||||
|
||||
frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
|
||||
|
||||
/*
|
||||
|
|
|
|||
|
|
@ -60,6 +60,13 @@
|
|||
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
|
||||
#endif
|
||||
|
||||
/* SIER bitflag of interrupts to enable */
|
||||
#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
|
||||
CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
|
||||
CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
|
||||
CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
|
||||
CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
|
||||
|
||||
/**
|
||||
* fsl_ssi_private: per-SSI private data
|
||||
*
|
||||
|
|
@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
|
|||
were interrupted for. We mask it with the Interrupt Enable register
|
||||
so that we only check for events that we're interested in.
|
||||
*/
|
||||
sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier);
|
||||
sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
|
||||
|
||||
if (sisr & CCSR_SSI_SISR_RFRC) {
|
||||
ssi_private->stats.rfrc++;
|
||||
|
|
@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
|
|||
*/
|
||||
|
||||
/* 4. Enable the interrupts and DMA requests */
|
||||
out_be32(&ssi->sier,
|
||||
CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE |
|
||||
CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN |
|
||||
CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN |
|
||||
CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE |
|
||||
CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN);
|
||||
out_be32(&ssi->sier, SIER_FLAGS);
|
||||
|
||||
/*
|
||||
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
|
||||
|
|
@ -466,28 +468,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
|
|||
case SNDRV_PCM_TRIGGER_START:
|
||||
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
|
||||
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
||||
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
||||
setbits32(&ssi->scr,
|
||||
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
|
||||
} else {
|
||||
long timeout = jiffies + 10;
|
||||
|
||||
else
|
||||
setbits32(&ssi->scr,
|
||||
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
|
||||
|
||||
/* Wait until the SSI has filled its FIFO. Without this
|
||||
* delay, ALSA complains about overruns. When the FIFO
|
||||
* is full, the DMA controller initiates its first
|
||||
* transfer. Until then, however, the DMA's DAR
|
||||
* register is zero, which translates to an
|
||||
* out-of-bounds pointer. This makes ALSA think an
|
||||
* overrun has occurred.
|
||||
*/
|
||||
while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
|
||||
(jiffies < timeout));
|
||||
if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
|
||||
return -EIO;
|
||||
}
|
||||
break;
|
||||
|
||||
case SNDRV_PCM_TRIGGER_STOP:
|
||||
|
|
@ -606,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
|
|||
.ops = &fsl_ssi_dai_ops,
|
||||
};
|
||||
|
||||
/* Show the statistics of a flag only if its interrupt is enabled. The
|
||||
* compiler will optimze this code to a no-op if the interrupt is not
|
||||
* enabled.
|
||||
*/
|
||||
#define SIER_SHOW(flag, name) \
|
||||
do { \
|
||||
if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
|
||||
length += sprintf(buf + length, #name "=%u\n", \
|
||||
ssi_private->stats.name); \
|
||||
} while (0)
|
||||
|
||||
|
||||
/**
|
||||
* fsl_sysfs_ssi_show: display SSI statistics
|
||||
*
|
||||
* Display the statistics for the current SSI device.
|
||||
* Display the statistics for the current SSI device. To avoid confusion,
|
||||
* we only show those counts that are enabled.
|
||||
*/
|
||||
static ssize_t fsl_sysfs_ssi_show(struct device *dev,
|
||||
struct device_attribute *attr, char *buf)
|
||||
{
|
||||
struct fsl_ssi_private *ssi_private =
|
||||
container_of(attr, struct fsl_ssi_private, dev_attr);
|
||||
ssize_t length;
|
||||
container_of(attr, struct fsl_ssi_private, dev_attr);
|
||||
ssize_t length = 0;
|
||||
|
||||
length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc);
|
||||
length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc);
|
||||
length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau);
|
||||
length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu);
|
||||
length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt);
|
||||
length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1);
|
||||
length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0);
|
||||
length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1);
|
||||
length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0);
|
||||
length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1);
|
||||
length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0);
|
||||
length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1);
|
||||
length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0);
|
||||
length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs);
|
||||
length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs);
|
||||
length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls);
|
||||
length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls);
|
||||
length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1);
|
||||
length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0);
|
||||
length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1);
|
||||
length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0);
|
||||
SIER_SHOW(RFRC_EN, rfrc);
|
||||
SIER_SHOW(TFRC_EN, tfrc);
|
||||
SIER_SHOW(CMDAU_EN, cmdau);
|
||||
SIER_SHOW(CMDDU_EN, cmddu);
|
||||
SIER_SHOW(RXT_EN, rxt);
|
||||
SIER_SHOW(RDR1_EN, rdr1);
|
||||
SIER_SHOW(RDR0_EN, rdr0);
|
||||
SIER_SHOW(TDE1_EN, tde1);
|
||||
SIER_SHOW(TDE0_EN, tde0);
|
||||
SIER_SHOW(ROE1_EN, roe1);
|
||||
SIER_SHOW(ROE0_EN, roe0);
|
||||
SIER_SHOW(TUE1_EN, tue1);
|
||||
SIER_SHOW(TUE0_EN, tue0);
|
||||
SIER_SHOW(TFS_EN, tfs);
|
||||
SIER_SHOW(RFS_EN, rfs);
|
||||
SIER_SHOW(TLS_EN, tls);
|
||||
SIER_SHOW(RLS_EN, rls);
|
||||
SIER_SHOW(RFF1_EN, rff1);
|
||||
SIER_SHOW(RFF0_EN, rff0);
|
||||
SIER_SHOW(TFE1_EN, tfe1);
|
||||
SIER_SHOW(TFE0_EN, tfe0);
|
||||
|
||||
return length;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -146,6 +146,17 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
|
|||
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
|
||||
int err = 0;
|
||||
|
||||
if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
|
||||
/*
|
||||
* McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
|
||||
* Set constraint for minimum buffer size to the same than FIFO
|
||||
* size in order to avoid underruns in playback startup because
|
||||
* HW is keeping the DMA request active until FIFO is filled.
|
||||
*/
|
||||
snd_pcm_hw_constraint_minmax(substream->runtime,
|
||||
SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
|
||||
}
|
||||
|
||||
if (!cpu_dai->active)
|
||||
err = omap_mcbsp_request(mcbsp_data->bus_id);
|
||||
|
||||
|
|
|
|||
|
|
@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE
|
|||
Say Y if you want to add support for SoC audio on the
|
||||
Marvell Zylonite reference platform.
|
||||
|
||||
config SND_PXA2XX_SOC_MAGICIAN
|
||||
tristate "SoC Audio support for HTC Magician"
|
||||
depends on SND_PXA2XX_SOC && MACH_MAGICIAN
|
||||
select SND_PXA2XX_SOC_I2S
|
||||
select SND_PXA_SOC_SSP
|
||||
select SND_SOC_UDA1380
|
||||
help
|
||||
Say Y if you want to add support for SoC audio on the
|
||||
HTC Magician.
|
||||
|
||||
config SND_PXA2XX_SOC_MIOA701
|
||||
tristate "SoC Audio support for MIO A701"
|
||||
depends on SND_PXA2XX_SOC && MACH_MIOA701
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o
|
|||
snd-soc-em-x270-objs := em-x270.o
|
||||
snd-soc-palm27x-objs := palm27x.o
|
||||
snd-soc-zylonite-objs := zylonite.o
|
||||
snd-soc-magician-objs := magician.o
|
||||
snd-soc-mioa701-objs := mioa701_wm9713.o
|
||||
|
||||
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
|
||||
|
|
@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
|
|||
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
|
||||
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
|
||||
obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
|
||||
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
|
||||
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
|
||||
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
|
||||
|
|
|
|||
560
sound/soc/pxa/magician.c
Normal file
560
sound/soc/pxa/magician.c
Normal file
|
|
@ -0,0 +1,560 @@
|
|||
/*
|
||||
* SoC audio for HTC Magician
|
||||
*
|
||||
* Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
|
||||
*
|
||||
* based on spitz.c,
|
||||
* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
|
||||
* Richard Purdie <richard@openedhand.com>
|
||||
*
|
||||
* This program is free software; you can redistribute it and/or modify it
|
||||
* under the terms of the GNU General Public License as published by the
|
||||
* Free Software Foundation; either version 2 of the License, or (at your
|
||||
* option) any later version.
|
||||
*
|
||||
*/
|
||||
|
||||
#include <linux/module.h>
|
||||
#include <linux/timer.h>
|
||||
#include <linux/interrupt.h>
|
||||
#include <linux/platform_device.h>
|
||||
#include <linux/delay.h>
|
||||
#include <linux/gpio.h>
|
||||
|
||||
#include <sound/core.h>
|
||||
#include <sound/pcm.h>
|
||||
#include <sound/pcm_params.h>
|
||||
#include <sound/soc.h>
|
||||
#include <sound/soc-dapm.h>
|
||||
|
||||
#include <mach/pxa-regs.h>
|
||||
#include <mach/hardware.h>
|
||||
#include <mach/magician.h>
|
||||
#include <asm/mach-types.h>
|
||||
#include "../codecs/uda1380.h"
|
||||
#include "pxa2xx-pcm.h"
|
||||
#include "pxa2xx-i2s.h"
|
||||
#include "pxa-ssp.h"
|
||||
|
||||
#define MAGICIAN_MIC 0
|
||||
#define MAGICIAN_MIC_EXT 1
|
||||
|
||||
static int magician_hp_switch;
|
||||
static int magician_spk_switch = 1;
|
||||
static int magician_in_sel = MAGICIAN_MIC;
|
||||
|
||||
static void magician_ext_control(struct snd_soc_codec *codec)
|
||||
{
|
||||
if (magician_spk_switch)
|
||||
snd_soc_dapm_enable_pin(codec, "Speaker");
|
||||
else
|
||||
snd_soc_dapm_disable_pin(codec, "Speaker");
|
||||
if (magician_hp_switch)
|
||||
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
|
||||
else
|
||||
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
|
||||
|
||||
switch (magician_in_sel) {
|
||||
case MAGICIAN_MIC:
|
||||
snd_soc_dapm_disable_pin(codec, "Headset Mic");
|
||||
snd_soc_dapm_enable_pin(codec, "Call Mic");
|
||||
break;
|
||||
case MAGICIAN_MIC_EXT:
|
||||
snd_soc_dapm_disable_pin(codec, "Call Mic");
|
||||
snd_soc_dapm_enable_pin(codec, "Headset Mic");
|
||||
break;
|
||||
}
|
||||
|
||||
snd_soc_dapm_sync(codec);
|
||||
}
|
||||
|
||||
static int magician_startup(struct snd_pcm_substream *substream)
|
||||
{
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct snd_soc_codec *codec = rtd->socdev->card->codec;
|
||||
|
||||
/* check the jack status at stream startup */
|
||||
magician_ext_control(codec);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*
|
||||
* Magician uses SSP port for playback.
|
||||
*/
|
||||
static int magician_playback_hw_params(struct snd_pcm_substream *substream,
|
||||
struct snd_pcm_hw_params *params)
|
||||
{
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
|
||||
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
|
||||
unsigned int acps, acds, width, rate;
|
||||
unsigned int div4 = PXA_SSP_CLK_SCDB_4;
|
||||
int ret = 0;
|
||||
|
||||
rate = params_rate(params);
|
||||
width = snd_pcm_format_physical_width(params_format(params));
|
||||
|
||||
/*
|
||||
* rate = SSPSCLK / (2 * width(16 or 32))
|
||||
* SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
|
||||
*/
|
||||
switch (params_rate(params)) {
|
||||
case 8000:
|
||||
/* off by a factor of 2: bug in the PXA27x audio clock? */
|
||||
acps = 32842000;
|
||||
switch (width) {
|
||||
case 16:
|
||||
/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_16;
|
||||
break;
|
||||
case 32:
|
||||
/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_8;
|
||||
}
|
||||
break;
|
||||
case 11025:
|
||||
acps = 5622000;
|
||||
switch (width) {
|
||||
case 16:
|
||||
/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_4;
|
||||
break;
|
||||
case 32:
|
||||
/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_2;
|
||||
}
|
||||
break;
|
||||
case 22050:
|
||||
acps = 5622000;
|
||||
switch (width) {
|
||||
case 16:
|
||||
/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_2;
|
||||
break;
|
||||
case 32:
|
||||
/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_1;
|
||||
}
|
||||
break;
|
||||
case 44100:
|
||||
acps = 5622000;
|
||||
switch (width) {
|
||||
case 16:
|
||||
/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_2;
|
||||
break;
|
||||
case 32:
|
||||
/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_1;
|
||||
}
|
||||
break;
|
||||
case 48000:
|
||||
acps = 12235000;
|
||||
switch (width) {
|
||||
case 16:
|
||||
/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_2;
|
||||
break;
|
||||
case 32:
|
||||
/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_1;
|
||||
}
|
||||
break;
|
||||
case 96000:
|
||||
acps = 12235000;
|
||||
switch (width) {
|
||||
case 16:
|
||||
/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_1;
|
||||
break;
|
||||
case 32:
|
||||
/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
|
||||
acds = PXA_SSP_CLK_AUDIO_DIV_2;
|
||||
div4 = PXA_SSP_CLK_SCDB_1;
|
||||
break;
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
/* set codec DAI configuration */
|
||||
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
|
||||
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set cpu DAI configuration */
|
||||
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
|
||||
SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set audio clock as clock source */
|
||||
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
|
||||
SND_SOC_CLOCK_OUT);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set the SSP audio system clock ACDS divider */
|
||||
ret = snd_soc_dai_set_clkdiv(cpu_dai,
|
||||
PXA_SSP_AUDIO_DIV_ACDS, acds);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set the SSP audio system clock SCDB divider4 */
|
||||
ret = snd_soc_dai_set_clkdiv(cpu_dai,
|
||||
PXA_SSP_AUDIO_DIV_SCDB, div4);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set SSP audio pll clock */
|
||||
ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*
|
||||
* Magician uses I2S for capture.
|
||||
*/
|
||||
static int magician_capture_hw_params(struct snd_pcm_substream *substream,
|
||||
struct snd_pcm_hw_params *params)
|
||||
{
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
|
||||
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
|
||||
int ret = 0;
|
||||
|
||||
/* set codec DAI configuration */
|
||||
ret = snd_soc_dai_set_fmt(codec_dai,
|
||||
SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
|
||||
SND_SOC_DAIFMT_CBS_CFS);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set cpu DAI configuration */
|
||||
ret = snd_soc_dai_set_fmt(cpu_dai,
|
||||
SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
|
||||
SND_SOC_DAIFMT_CBS_CFS);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* set the I2S system clock as output */
|
||||
ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
|
||||
SND_SOC_CLOCK_OUT);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct snd_soc_ops magician_capture_ops = {
|
||||
.startup = magician_startup,
|
||||
.hw_params = magician_capture_hw_params,
|
||||
};
|
||||
|
||||
static struct snd_soc_ops magician_playback_ops = {
|
||||
.startup = magician_startup,
|
||||
.hw_params = magician_playback_hw_params,
|
||||
};
|
||||
|
||||
static int magician_get_hp(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
ucontrol->value.integer.value[0] = magician_hp_switch;
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int magician_set_hp(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
||||
|
||||
if (magician_hp_switch == ucontrol->value.integer.value[0])
|
||||
return 0;
|
||||
|
||||
magician_hp_switch = ucontrol->value.integer.value[0];
|
||||
magician_ext_control(codec);
|
||||
return 1;
|
||||
}
|
||||
|
||||
static int magician_get_spk(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
ucontrol->value.integer.value[0] = magician_spk_switch;
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int magician_set_spk(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
||||
|
||||
if (magician_spk_switch == ucontrol->value.integer.value[0])
|
||||
return 0;
|
||||
|
||||
magician_spk_switch = ucontrol->value.integer.value[0];
|
||||
magician_ext_control(codec);
|
||||
return 1;
|
||||
}
|
||||
|
||||
static int magician_get_input(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
ucontrol->value.integer.value[0] = magician_in_sel;
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int magician_set_input(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
if (magician_in_sel == ucontrol->value.integer.value[0])
|
||||
return 0;
|
||||
|
||||
magician_in_sel = ucontrol->value.integer.value[0];
|
||||
|
||||
switch (magician_in_sel) {
|
||||
case MAGICIAN_MIC:
|
||||
gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
|
||||
break;
|
||||
case MAGICIAN_MIC_EXT:
|
||||
gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
|
||||
}
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
||||
static int magician_spk_power(struct snd_soc_dapm_widget *w,
|
||||
struct snd_kcontrol *k, int event)
|
||||
{
|
||||
gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int magician_hp_power(struct snd_soc_dapm_widget *w,
|
||||
struct snd_kcontrol *k, int event)
|
||||
{
|
||||
gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int magician_mic_bias(struct snd_soc_dapm_widget *w,
|
||||
struct snd_kcontrol *k, int event)
|
||||
{
|
||||
gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* magician machine dapm widgets */
|
||||
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
|
||||
SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
|
||||
SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
|
||||
SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
|
||||
SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
|
||||
};
|
||||
|
||||
/* magician machine audio_map */
|
||||
static const struct snd_soc_dapm_route audio_map[] = {
|
||||
|
||||
/* Headphone connected to VOUTL, VOUTR */
|
||||
{"Headphone Jack", NULL, "VOUTL"},
|
||||
{"Headphone Jack", NULL, "VOUTR"},
|
||||
|
||||
/* Speaker connected to VOUTL, VOUTR */
|
||||
{"Speaker", NULL, "VOUTL"},
|
||||
{"Speaker", NULL, "VOUTR"},
|
||||
|
||||
/* Mics are connected to VINM */
|
||||
{"VINM", NULL, "Headset Mic"},
|
||||
{"VINM", NULL, "Call Mic"},
|
||||
};
|
||||
|
||||
static const char *input_select[] = {"Call Mic", "Headset Mic"};
|
||||
static const struct soc_enum magician_in_sel_enum =
|
||||
SOC_ENUM_SINGLE_EXT(2, input_select);
|
||||
|
||||
static const struct snd_kcontrol_new uda1380_magician_controls[] = {
|
||||
SOC_SINGLE_BOOL_EXT("Headphone Switch",
|
||||
(unsigned long)&magician_hp_switch,
|
||||
magician_get_hp, magician_set_hp),
|
||||
SOC_SINGLE_BOOL_EXT("Speaker Switch",
|
||||
(unsigned long)&magician_spk_switch,
|
||||
magician_get_spk, magician_set_spk),
|
||||
SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
|
||||
magician_get_input, magician_set_input),
|
||||
};
|
||||
|
||||
/*
|
||||
* Logic for a uda1380 as connected on a HTC Magician
|
||||
*/
|
||||
static int magician_uda1380_init(struct snd_soc_codec *codec)
|
||||
{
|
||||
int err;
|
||||
|
||||
/* NC codec pins */
|
||||
snd_soc_dapm_nc_pin(codec, "VOUTLHP");
|
||||
snd_soc_dapm_nc_pin(codec, "VOUTRHP");
|
||||
|
||||
/* FIXME: is anything connected here? */
|
||||
snd_soc_dapm_nc_pin(codec, "VINL");
|
||||
snd_soc_dapm_nc_pin(codec, "VINR");
|
||||
|
||||
/* Add magician specific controls */
|
||||
err = snd_soc_add_controls(codec, uda1380_magician_controls,
|
||||
ARRAY_SIZE(uda1380_magician_controls));
|
||||
if (err < 0)
|
||||
return err;
|
||||
|
||||
/* Add magician specific widgets */
|
||||
snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
|
||||
ARRAY_SIZE(uda1380_dapm_widgets));
|
||||
|
||||
/* Set up magician specific audio path interconnects */
|
||||
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
|
||||
|
||||
snd_soc_dapm_sync(codec);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* magician digital audio interface glue - connects codec <--> CPU */
|
||||
static struct snd_soc_dai_link magician_dai[] = {
|
||||
{
|
||||
.name = "uda1380",
|
||||
.stream_name = "UDA1380 Playback",
|
||||
.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
|
||||
.codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
|
||||
.init = magician_uda1380_init,
|
||||
.ops = &magician_playback_ops,
|
||||
},
|
||||
{
|
||||
.name = "uda1380",
|
||||
.stream_name = "UDA1380 Capture",
|
||||
.cpu_dai = &pxa_i2s_dai,
|
||||
.codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
|
||||
.ops = &magician_capture_ops,
|
||||
}
|
||||
};
|
||||
|
||||
/* magician audio machine driver */
|
||||
static struct snd_soc_card snd_soc_card_magician = {
|
||||
.name = "Magician",
|
||||
.dai_link = magician_dai,
|
||||
.num_links = ARRAY_SIZE(magician_dai),
|
||||
.platform = &pxa2xx_soc_platform,
|
||||
};
|
||||
|
||||
/* magician audio private data */
|
||||
static struct uda1380_setup_data magician_uda1380_setup = {
|
||||
.i2c_address = 0x18,
|
||||
.dac_clk = UDA1380_DAC_CLK_WSPLL,
|
||||
};
|
||||
|
||||
/* magician audio subsystem */
|
||||
static struct snd_soc_device magician_snd_devdata = {
|
||||
.card = &snd_soc_card_magician,
|
||||
.codec_dev = &soc_codec_dev_uda1380,
|
||||
.codec_data = &magician_uda1380_setup,
|
||||
};
|
||||
|
||||
static struct platform_device *magician_snd_device;
|
||||
|
||||
static int __init magician_init(void)
|
||||
{
|
||||
int ret;
|
||||
|
||||
if (!machine_is_magician())
|
||||
return -ENODEV;
|
||||
|
||||
ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
|
||||
if (ret)
|
||||
goto err_request_power;
|
||||
ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
|
||||
if (ret)
|
||||
goto err_request_reset;
|
||||
ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
|
||||
if (ret)
|
||||
goto err_request_spk;
|
||||
ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
|
||||
if (ret)
|
||||
goto err_request_ep;
|
||||
ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
|
||||
if (ret)
|
||||
goto err_request_mic;
|
||||
ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
|
||||
if (ret)
|
||||
goto err_request_in_sel0;
|
||||
ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
|
||||
if (ret)
|
||||
goto err_request_in_sel1;
|
||||
|
||||
gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
|
||||
gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
|
||||
|
||||
/* we may need to have the clock running here - pH5 */
|
||||
gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
|
||||
udelay(5);
|
||||
gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
|
||||
|
||||
magician_snd_device = platform_device_alloc("soc-audio", -1);
|
||||
if (!magician_snd_device) {
|
||||
ret = -ENOMEM;
|
||||
goto err_pdev;
|
||||
}
|
||||
|
||||
platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
|
||||
magician_snd_devdata.dev = &magician_snd_device->dev;
|
||||
ret = platform_device_add(magician_snd_device);
|
||||
if (ret) {
|
||||
platform_device_put(magician_snd_device);
|
||||
goto err_pdev;
|
||||
}
|
||||
|
||||
return 0;
|
||||
|
||||
err_pdev:
|
||||
gpio_free(EGPIO_MAGICIAN_IN_SEL1);
|
||||
err_request_in_sel1:
|
||||
gpio_free(EGPIO_MAGICIAN_IN_SEL0);
|
||||
err_request_in_sel0:
|
||||
gpio_free(EGPIO_MAGICIAN_MIC_POWER);
|
||||
err_request_mic:
|
||||
gpio_free(EGPIO_MAGICIAN_EP_POWER);
|
||||
err_request_ep:
|
||||
gpio_free(EGPIO_MAGICIAN_SPK_POWER);
|
||||
err_request_spk:
|
||||
gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
|
||||
err_request_reset:
|
||||
gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
|
||||
err_request_power:
|
||||
return ret;
|
||||
}
|
||||
|
||||
static void __exit magician_exit(void)
|
||||
{
|
||||
platform_device_unregister(magician_snd_device);
|
||||
|
||||
gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
|
||||
gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
|
||||
gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
|
||||
gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
|
||||
|
||||
gpio_free(EGPIO_MAGICIAN_IN_SEL1);
|
||||
gpio_free(EGPIO_MAGICIAN_IN_SEL0);
|
||||
gpio_free(EGPIO_MAGICIAN_MIC_POWER);
|
||||
gpio_free(EGPIO_MAGICIAN_EP_POWER);
|
||||
gpio_free(EGPIO_MAGICIAN_SPK_POWER);
|
||||
gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
|
||||
gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
|
||||
}
|
||||
|
||||
module_init(magician_init);
|
||||
module_exit(magician_exit);
|
||||
|
||||
MODULE_AUTHOR("Philipp Zabel");
|
||||
MODULE_DESCRIPTION("ALSA SoC Magician");
|
||||
MODULE_LICENSE("GPL");
|
||||
|
|
@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
|
|||
u32 sscr0;
|
||||
u32 sspsp;
|
||||
int width = snd_pcm_format_physical_width(params_format(params));
|
||||
int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
|
||||
|
||||
/* select correct DMA params */
|
||||
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
|
||||
dma = 1; /* capture DMA offset is 1,3 */
|
||||
if (chn == 2)
|
||||
dma += 2; /* stereo DMA offset is 2, mono is 0 */
|
||||
/* Network mode with one active slot (ttsa == 1) can be used
|
||||
* to force 16-bit frame width on the wire (for S16_LE), even
|
||||
* with two channels. Use 16-bit DMA transfers for this case.
|
||||
*/
|
||||
if (((chn == 2) && (ttsa != 1)) || (width == 32))
|
||||
dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
|
||||
|
||||
cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
|
||||
|
||||
dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
|
||||
|
|
@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
|
|||
/* When we use a network mode, we always require TDM slots
|
||||
* - complain loudly and fail if they've not been set up yet.
|
||||
*/
|
||||
if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
|
||||
if ((sscr0 & SSCR0_MOD) && !ttsa) {
|
||||
dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
|
||||
return -EINVAL;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
|
|||
int err;
|
||||
|
||||
codec->ac97->dev.bus = &ac97_bus_type;
|
||||
codec->ac97->dev.parent = NULL;
|
||||
codec->ac97->dev.parent = codec->card->dev;
|
||||
codec->ac97->dev.release = soc_ac97_device_release;
|
||||
|
||||
dev_set_name(&codec->ac97->dev, "%d-%d:%s",
|
||||
|
|
@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev)
|
|||
{
|
||||
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
||||
struct snd_soc_card *card = socdev->card;
|
||||
struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
|
||||
|
||||
dev_dbg(socdev->dev, "scheduling resume work\n");
|
||||
|
||||
if (!schedule_work(&card->deferred_resume_work))
|
||||
dev_err(socdev->dev, "resume work item may be lost\n");
|
||||
/* AC97 devices might have other drivers hanging off them so
|
||||
* need to resume immediately. Other drivers don't have that
|
||||
* problem and may take a substantial amount of time to resume
|
||||
* due to I/O costs and anti-pop so handle them out of line.
|
||||
*/
|
||||
if (cpu_dai->ac97_control) {
|
||||
dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
|
||||
soc_resume_deferred(&card->deferred_resume_work);
|
||||
} else {
|
||||
dev_dbg(socdev->dev, "Scheduling resume work\n");
|
||||
if (!schedule_work(&card->deferred_resume_work))
|
||||
dev_err(socdev->dev, "resume work item may be lost\n");
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue