Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
  ALSA: hda: Storage class should be before const qualifier
  ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
  ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
  ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
  ASoC: TWL6040: Enable earphone path in codec
  ASoC: SDP4430: Add support for Earphone speaker
  ASoC: SDP4430: Add sdp4430 machine driver
  ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
  ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
  ALSA: sound/pci/asihpi: Use kzalloc
  ALSA: hdmi - dont fail on extra nodes
  ALSA: intelhdmi - add id for the CougarPoint chipset
  ALSA: intelhdmi - user friendly codec name
  ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
  ALSA: asihpi: incorrect range check
  ALSA: asihpi: testing the wrong variable
  ALSA: es1688: add pedantic range checks
  ARM: McBSP: Add support for omap4 in McBSP driver
  ARM: McBSP: Fix request for irq in OMAP4
  OMAP: McBSP: Add 32-bit mode support
  ...

3 files changed, 319 insertions, 0 deletions

diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index 7174b4c710de..eba9b9d257a1 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,3 +11,11 @@ config SND_IMX_SOC
 config SND_MXC_SOC_SSI
         tristate
 
+config SND_MXC_SOC_WM1133_EV1
+        tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
+        depends on SND_IMX_SOC && EXPERIMENTAL
+        select SND_SOC_WM8350
+        select SND_MXC_SOC_SSI
+        help
+          Enable support for audio on the i.MX31ADS with the WM1133-EV1
+          PMIC board with WM8835x fitted.
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index 9f8bb92ddfcc..2d203635ac11 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -9,4 +9,7 @@ obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o
 
 # i.MX Machine Support
 snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
new file mode 100644
index 000000000000..a6e7d9497639
--- /dev/null
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -0,0 +1,308 @@
+/*
+ *  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
+ *
+ *  Copyright (c) 2010 Wolfson Microelectronics plc
+ *  Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  Based on an earlier driver for the same hardware by Liam Girdwood.
+ *
+ *  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/platform_device.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audmux.h>
+
+#include "imx-ssi.h"
+#include "../codecs/wm8350.h"
+
+/* There is a silicon mic on the board optionally connected via a solder pad
+ * SP1.  Define this to enable it.
+ */
+#undef USE_SIMIC
+
+struct _wm8350_audio {
+        unsigned int channels;
+        snd_pcm_format_t format;
+        unsigned int rate;
+        unsigned int sysclk;
+        unsigned int bclkdiv;
+        unsigned int clkdiv;
+        unsigned int lr_rate;
+};
+
+/* in order of power consumption per rate (lowest first) */
+static const struct _wm8350_audio wm8350_audio[] = {
+        /* 16bit mono modes */
+        {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
+         WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
+
+        /* 16 bit stereo modes */
+        {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
+         WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
+         WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
+         WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
+         WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
+         WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
+         WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
+         WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
+         WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+        {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
+         WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+
+        /* 24bit stereo modes */
+        {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
+         WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+        {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
+         WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+        {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
+         WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+        {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
+         WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+};
+
+static int wm1133_ev1_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 i, found = 0;
+        snd_pcm_format_t format = params_format(params);
+        unsigned int rate = params_rate(params);
+        unsigned int channels = params_channels(params);
+        u32 dai_format;
+
+        /* find the correct audio parameters */
+        for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
+                if (rate == wm8350_audio[i].rate &&
+                    format == wm8350_audio[i].format &&
+                    channels == wm8350_audio[i].channels) {
+                        found = 1;
+                        break;
+                }
+        }
+        if (!found)
+                return -EINVAL;
+
+        /* codec FLL input is 14.75 MHz from MCLK */
+        snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
+
+        dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+                SND_SOC_DAIFMT_CBM_CFM;
+
+        /* set codec DAI configuration */
+        snd_soc_dai_set_fmt(codec_dai, dai_format);
+
+        /* set cpu DAI configuration */
+        snd_soc_dai_set_fmt(cpu_dai, dai_format);
+
+        /* TODO: The SSI driver should figure this out for us */
+        switch (channels) {
+        case 2:
+                snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+                break;
+        case 1:
+                snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* set MCLK as the codec system clock for DAC and ADC */
+        snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
+                               wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
+
+        /* set codec BCLK division for sample rate */
+        snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
+                               wm8350_audio[i].bclkdiv);
+
+        /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
+        snd_soc_dai_set_clkdiv(codec_dai,
+                               WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
+        snd_soc_dai_set_clkdiv(codec_dai,
+                               WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
+
+        /* now configure DAC and ADC clocks */
+        snd_soc_dai_set_clkdiv(codec_dai,
+                               WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
+
+        snd_soc_dai_set_clkdiv(codec_dai,
+                               WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
+
+        return 0;
+}
+
+static struct snd_soc_ops wm1133_ev1_ops = {
+        .hw_params = wm1133_ev1_hw_params,
+};
+
+static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
+#ifdef USE_SIMIC
+        SND_SOC_DAPM_MIC("SiMIC", NULL),
+#endif
+        SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
+        SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
+        SND_SOC_DAPM_LINE("Line In Jack", NULL),
+        SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+/* imx32ads soc_card audio map */
+static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
+
+#ifdef USE_SIMIC
+        /* SiMIC --> IN1LN (with automatic bias) via SP1 */
+        { "IN1LN", NULL, "Mic Bias" },
+        { "Mic Bias", NULL, "SiMIC" },
+#endif
+
+        /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
+        { "IN1LN", NULL, "Mic Bias" },
+        { "IN1LP", NULL, "Mic1 Jack" },
+        { "Mic Bias", NULL, "Mic1 Jack" },
+
+        /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
+        { "IN1RN", NULL, "Mic Bias" },
+        { "IN1RP", NULL, "Mic2 Jack" },
+        { "Mic Bias", NULL, "Mic2 Jack" },
+
+        /* Line in Jack --> AUX (L+R) */
+        { "IN3R", NULL, "Line In Jack" },
+        { "IN3L", NULL, "Line In Jack" },
+
+        /* Out1 --> Headphone Jack */
+        { "Headphone Jack", NULL, "OUT1R" },
+        { "Headphone Jack", NULL, "OUT1L" },
+
+        /* Out1 --> Line Out Jack */
+        { "Line Out Jack", NULL, "OUT2R" },
+        { "Line Out Jack", NULL, "OUT2L" },
+};
+
+static struct snd_soc_jack hp_jack;
+
+static struct snd_soc_jack_pin hp_jack_pins[] = {
+        { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
+};
+
+static struct snd_soc_jack mic_jack;
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+        { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
+        { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
+};
+
+static int wm1133_ev1_init(struct snd_soc_codec *codec)
+{
+        struct snd_soc_card *card = codec->socdev->card;
+
+        snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets,
+                                  ARRAY_SIZE(wm1133_ev1_widgets));
+
+        snd_soc_dapm_add_routes(codec, wm1133_ev1_map,
+                                ARRAY_SIZE(wm1133_ev1_map));
+
+        /* Headphone jack detection */
+        snd_soc_jack_new(card, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
+        snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
+                              hp_jack_pins);
+        wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
+
+        /* Microphone jack detection */
+        snd_soc_jack_new(card, "Microphone",
+                         SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
+        snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+                              mic_jack_pins);
+        wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
+                               SND_JACK_BTN_0);
+
+        snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+
+        return 0;
+}
+
+
+static struct snd_soc_dai_link wm1133_ev1_dai = {
+        .name = "WM1133-EV1",
+        .stream_name = "Audio",
+        .cpu_dai = &imx_ssi_pcm_dai[0],
+        .codec_dai = &wm8350_dai,
+        .init = wm1133_ev1_init,
+        .ops = &wm1133_ev1_ops,
+        .symmetric_rates = 1,
+};
+
+static struct snd_soc_card wm1133_ev1 = {
+        .name = "WM1133-EV1",
+        .platform = &imx_soc_platform,
+        .dai_link = &wm1133_ev1_dai,
+        .num_links = 1,
+};
+
+static struct snd_soc_device wm1133_ev1_snd_devdata = {
+        .card = &wm1133_ev1,
+        .codec_dev = &soc_codec_dev_wm8350,
+};
+
+static struct platform_device *wm1133_ev1_snd_device;
+
+static int __init wm1133_ev1_audio_init(void)
+{
+        int ret;
+        unsigned int ptcr, pdcr;
+
+        /* SSI0 mastered by port 5 */
+        ptcr = MXC_AUDMUX_V2_PTCR_SYN |
+                MXC_AUDMUX_V2_PTCR_TFSDIR |
+                MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
+                MXC_AUDMUX_V2_PTCR_TCLKDIR |
+                MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+        pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+        mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
+
+        ptcr = MXC_AUDMUX_V2_PTCR_SYN;
+        pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
+        mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
+
+        wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!wm1133_ev1_snd_device)
+                return -ENOMEM;
+
+        platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1_snd_devdata);
+        wm1133_ev1_snd_devdata.dev = &wm1133_ev1_snd_device->dev;
+        ret = platform_device_add(wm1133_ev1_snd_device);
+
+        if (ret)
+                platform_device_put(wm1133_ev1_snd_device);
+
+        return ret;
+}
+module_init(wm1133_ev1_audio_init);
+
+static void __exit wm1133_ev1_audio_exit(void)
+{
+        platform_device_unregister(wm1133_ev1_snd_device);
+}
+module_exit(wm1133_ev1_audio_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
+MODULE_LICENSE("GPL");