Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull updates of sound stuff from Takashi Iwai:
 "Here is the first big update chunk of sound stuff for 3.4-rc1.

  In the common sound infrastructure, there are a few changes for
  dynamic PCM support (used in ASoC) and a few clean-ups.  Majority of
  changes are found, as usual, in HD-audio and ASoC.

  Some highlights of HD-audio changes:

   - All the long-standing static quirk codes for Realtek codec were
     finally removed by fixing and extending the Realtek auto-parser.

   - The mute-LED control is standardized over all HD-audio codec
     drivers using the extended vmaster hook.

   - The vmaster slave mixer elements are initialized to 0dB as default
     so that the user won't be annoyed by the silent output after
     updates, e.g.  due to the additions of new elements.

   - Other many fix-ups for the misc HD-audio devices.

  In the ASoC side, this is a very active release, including a quite a
  few framework enhancements.  Some highlights:

   - Support for widgets not associated with a CODEC, an important part
     of the dynamic PCM framework.

   - A library factoring out the common code shared by dmaengine based
     DMA drivers contributed by Lars-Peter Clausen.  This will save a
     lot of code and make it much easier to deploy enhancements to
     dmaengine.

   - Support for binary controls, used for providing runtime
     configuration of algorithm coefficients.

   - A new DAPM widget type for regulator supplies allowing drivers for
     devices that can power down unused supplies while active to do
     without any per-driver code.

   - DAPM widgets for DAIs, initially giving a speed boost for playback
     startup and shutdown and also the basis for CODEC<->CODEC DAI link
     support.

   - Support for specifying the number of significant bits on audio
     interfaces, useful for allowing applications to know how much
     effort to put into generating data for a larger sample format.

   - Conversion of the FSI driver used on some SH processors to
     DMAEngine.

   - Conversion of EP93xx drivers to DMAEngine.

   - New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
     WM2200.

   - Move audmux driver from arc/arm to sound/soc

   - McBSP move from arch/ to sound/ and updates

  Also, a few small updates and fixes for other drivers like au88x0,
  ymfpci, USB 6fire, USB usx2yaudio are included."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
  ASoC: wm8994: Provide VMID mode control and fix default sequence
  ASoC: wm8994: Add missing break in resume
  ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
  ASoC: pxa-ssp: atomically set stream active masks
  ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
  ASoC: Samsung: Added to support mono recording
  ALSA: hda - Fix build with CONFIG_PM=n
  ALSA: au88x0 - Avoid possible Oops at unbinding
  ALSA: usb-audio - Fix build error by consitification of rate list
  ASoC: core: Fix obscure leak of runtime array
  ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
  ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
  ASoC: wm8996: Add 44.1kHz support
  ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
  ASoC: mx27vis-aic32x4: Convert it to platform driver
  ALSA: hda - fix printing of high HDMI sample rates
  ALSA: ymfpci - Fix legacy registers on S3/S4 resume
  ALSA: control - Fixe a trailing white space error
  ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
  ALSA: hda - Add "Mute-LED Mode" enum control
  ...

1 files changed, 349 insertions, 0 deletions

diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
new file mode 100644
index 000000000000..93bb8eee22b3
--- /dev/null
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -0,0 +1,349 @@
+/*
+ * 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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+
+#include "omap-dmic.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+
+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 omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+	int clk_id, freq;
+	int ret;
+
+	clk_id = twl6040_get_clk_id(rtd->codec);
+	if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
+		freq = pdata->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),
+};
+
+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);
+	int hs_trim;
+	int ret = 0;
+
+	/* 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");
+
+	/*
+	 * 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 (pdata->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);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Digital Mic", NULL),
+};
+
+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;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+				ARRAY_SIZE(dmic_dapm_widgets));
+	if (ret)
+		return ret;
+
+	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 twl6040_dmic_dai[] = {
+	{
+		.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,
+	},
+};
+
+static struct snd_soc_dai_link twl6040_only_dai[] = {
+	{
+		.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,
+	},
+};
+
+/* 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 snd_soc_card *card = &omap_abe_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	if (!pdata) {
+		dev_err(&pdev->dev, "Missing pdata\n");
+		return -ENODEV;
+	}
+
+	if (pdata->card_name) {
+		card->name = pdata->card_name;
+	} else {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
+
+	if (!pdata->mclk_freq) {
+		dev_err(&pdev->dev, "MCLK frequency missing\n");
+		return -ENODEV;
+	}
+
+	if (pdata->has_dmic) {
+		card->dai_link = twl6040_dmic_dai;
+		card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
+	} else {
+		card->dai_link = twl6040_only_dai;
+		card->num_links = ARRAY_SIZE(twl6040_only_dai);
+	}
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit omap_abe_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver omap_abe_driver = {
+	.driver = {
+		.name = "omap-abe-twl6040",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.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");