linux/sound/soc/omap/omap-abe-twl6040.c
Linus Torvalds f5a246eab9 Sound updates for 3.7-rc1
This contains pretty many small commits covering fairly large range of
 files in sound/ directory.  Partly because of additional API support
 and partly because of constantly developed ASoC and ARM stuff.
 
 Some highlights:
 
 - Introduced the helper function and documentation for exposing the
   channel map via control API, as discussed in Plumbers; most of PCI
   drivers are covered, will follow more drivers later
 
 - Most of drivers have been replaced with the new PM callbacks (if
   the bus is supported)
 
 - HD-audio controller got the support of runtime PM and the support of
   D3 clock-stop.  Also changing the power_save option in sysfs kicks
   off immediately to enable / disable the power-save mode.
 
 - Another significant code change in HD-audio is the rewrite of
   firmware loading code.  Other than that, most of changes in HD-audio
   are continued cleanups and standardization for the generic auto
   parser and bug fixes (HBR, device-specific fixups), in addition to
   the support of channel-map API.
 
 - Addition of ASoC bindings for the compressed API, used by the
   mid-x86 drivers.
 
 - Lots of cleanups and API refreshes for ASoC codec drivers and
   DaVinci.
 
 - Conversion of OMAP to dmaengine.
 
 - New machine driver for Wolfson Microelectronics Bells.
 
 - New CODEC driver for Wolfson Microelectronics WM0010.
 
 - Enhancements to the ux500 and wm2000 drivers
 
 - A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...
2012-10-09 07:07:14 +09:00

429 lines
11 KiB
C

/*
* omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
* twl6040 codec
*
* Author: Misael Lopez Cruz <misael.lopez@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/mfd/twl6040.h>
#include <linux/platform_data/omap-abe-twl6040.h>
#include <linux/module.h>
#include <linux/of.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "omap-dmic.h"
#include "omap-mcpdm.h"
#include "omap-pcm.h"
#include "../codecs/twl6040.h"
struct abe_twl6040 {
int jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
struct platform_device *dmic_codec_dev;
};
static int omap_abe_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->codec_dai;
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(rtd->codec);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
freq = priv->mclk_freq;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
return -EINVAL;
/* set the codec mclk */
ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
SND_SOC_CLOCK_IN);
if (ret) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
return ret;
}
static struct snd_soc_ops omap_abe_ops = {
.hw_params = omap_abe_hw_params,
};
static int omap_abe_dmic_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 *cpu_dai = rtd->cpu_dai;
int ret = 0;
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC cpu system clock\n");
return ret;
}
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
SND_SOC_CLOCK_OUT);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC output clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops omap_abe_dmic_ops = {
.hw_params = omap_abe_dmic_hw_params,
};
/* Headset jack */
static struct snd_soc_jack hs_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin hs_jack_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headset Stereophone",
.mask = SND_JACK_HEADPHONE,
},
};
/* SDP4430 machine DAPM */
static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
/* Outputs */
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_SPK("Earphone Spk", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
SND_SOC_DAPM_SPK("Vibrator", NULL),
/* Inputs */
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
/* Digital microphones */
SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* Routings for outputs */
{"Headset Stereophone", NULL, "HSOL"},
{"Headset Stereophone", NULL, "HSOR"},
{"Earphone Spk", NULL, "EP"},
{"Ext Spk", NULL, "HFL"},
{"Ext Spk", NULL, "HFR"},
{"Line Out", NULL, "AUXL"},
{"Line Out", NULL, "AUXR"},
{"Vibrator", NULL, "VIBRAL"},
{"Vibrator", NULL, "VIBRAR"},
/* Routings for inputs */
{"HSMIC", NULL, "Headset Mic"},
{"Headset Mic", NULL, "Headset Mic Bias"},
{"MAINMIC", NULL, "Main Handset Mic"},
{"Main Handset Mic", NULL, "Main Mic Bias"},
{"SUBMIC", NULL, "Sub Handset Mic"},
{"Sub Handset Mic", NULL, "Main Mic Bias"},
{"AFML", NULL, "Line In"},
{"AFMR", NULL, "Line In"},
};
static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
int connected, char *pin)
{
if (!connected)
snd_soc_dapm_disable_pin(dapm, pin);
}
static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
/*
* Configure McPDM offset cancellation based on the HSOTRIM value from
* twl6040.
*/
hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection only if it is supported */
if (priv->jack_detection) {
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET, &hs_jack);
if (ret)
return ret;
ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
hs_jack_pins);
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
}
/*
* NULL pdata means we booted with DT. In this case the routing is
* provided and the card is fully routed, no need to mark pins.
*/
if (!pdata)
return ret;
/* Disable not connected paths if not used */
twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
return ret;
}
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic"},
{"Digital Mic", NULL, "Digital Mic1 Bias"},
};
static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
.cpu_dai_name = "omap-mcpdm",
.codec_dai_name = "twl6040-legacy",
.platform_name = "omap-pcm-audio",
.codec_name = "twl6040-codec",
.init = omap_abe_twl6040_init,
.ops = &omap_abe_ops,
},
{
.name = "DMIC",
.stream_name = "DMIC Capture",
.cpu_dai_name = "omap-dmic",
.codec_dai_name = "dmic-hifi",
.platform_name = "omap-pcm-audio",
.codec_name = "dmic-codec",
.init = omap_abe_dmic_init,
.ops = &omap_abe_dmic_ops,
},
};
/* Audio machine driver */
static struct snd_soc_card omap_abe_card = {
.owner = THIS_MODULE,
.dapm_widgets = twl6040_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static __devinit int omap_abe_probe(struct platform_device *pdev)
{
struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
struct device_node *node = pdev->dev.of_node;
struct snd_soc_card *card = &omap_abe_card;
struct abe_twl6040 *priv;
int num_links = 0;
int ret = 0;
card->dev = &pdev->dev;
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
priv->dmic_codec_dev = ERR_PTR(-EINVAL);
if (node) {
struct device_node *dai_node;
if (snd_soc_of_parse_card_name(card, "ti,model")) {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
}
ret = snd_soc_of_parse_audio_routing(card,
"ti,audio-routing");
if (ret) {
dev_err(&pdev->dev,
"Error while parsing DAPM routing\n");
return ret;
}
dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
if (!dai_node) {
dev_err(&pdev->dev, "McPDM node is not provided\n");
return -EINVAL;
}
abe_twl6040_dai_links[0].cpu_dai_name = NULL;
abe_twl6040_dai_links[0].cpu_of_node = dai_node;
dai_node = of_parse_phandle(node, "ti,dmic", 0);
if (dai_node) {
num_links = 2;
abe_twl6040_dai_links[1].cpu_dai_name = NULL;
abe_twl6040_dai_links[1].cpu_of_node = dai_node;
priv->dmic_codec_dev = platform_device_register_simple(
"dmic-codec", -1, NULL, 0);
if (IS_ERR(priv->dmic_codec_dev)) {
dev_err(&pdev->dev,
"Can't instantiate dmic-codec\n");
return PTR_ERR(priv->dmic_codec_dev);
}
} else {
num_links = 1;
}
of_property_read_u32(node, "ti,jack-detection",
&priv->jack_detection);
of_property_read_u32(node, "ti,mclk-freq",
&priv->mclk_freq);
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency not provided\n");
ret = -EINVAL;
goto err_unregister;
}
omap_abe_card.fully_routed = 1;
} else if (pdata) {
if (pdata->card_name) {
card->name = pdata->card_name;
} else {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
}
if (pdata->has_dmic)
num_links = 2;
else
num_links = 1;
priv->jack_detection = pdata->jack_detection;
priv->mclk_freq = pdata->mclk_freq;
} else {
dev_err(&pdev->dev, "Missing pdata\n");
return -ENODEV;
}
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
ret = -ENODEV;
goto err_unregister;
}
card->dai_link = abe_twl6040_dai_links;
card->num_links = num_links;
snd_soc_card_set_drvdata(card, priv);
ret = snd_soc_register_card(card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
goto err_unregister;
}
return 0;
err_unregister:
if (!IS_ERR(priv->dmic_codec_dev))
platform_device_unregister(priv->dmic_codec_dev);
return ret;
}
static int __devexit omap_abe_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
snd_soc_unregister_card(card);
if (!IS_ERR(priv->dmic_codec_dev))
platform_device_unregister(priv->dmic_codec_dev);
return 0;
}
static const struct of_device_id omap_abe_of_match[] = {
{.compatible = "ti,abe-twl6040", },
{ },
};
MODULE_DEVICE_TABLE(of, omap_abe_of_match);
static struct platform_driver omap_abe_driver = {
.driver = {
.name = "omap-abe-twl6040",
.owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
.of_match_table = omap_abe_of_match,
},
.probe = omap_abe_probe,
.remove = __devexit_p(omap_abe_remove),
};
module_platform_driver(omap_abe_driver);
MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:omap-abe-twl6040");