Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This is the first big chunk for 3.5 merges of sound stuff.

  There are a few big changes in different areas.  First off, the
  streaming logic of USB-audio endpoints has been largely rewritten for
  the better support of "implicit feedback".  If anything about USB got
  broken, this change has to be checked.

  For HD-audio, the resume procedure was changed; instead of delaying
  the resume of the hardware until the first use, now waking up
  immediately at resume.  This is for buggy BIOS.

  For ASoC, dynamic PCM support and the improved support for digital
  links between off-SoC devices are major framework changes.

  Some highlights are below:

  * HD-audio
   - Avoid accesses of invalid pin-control bits that may stall the codec
   - V-ref setup cleanups
   - Fix the races in power-saving code
   - Fix the races in codec cache hashes and connection lists
   - Split some common codes for BIOS auto-parser to hda_auto_parser.c
   - Changed the PM resume code to wake up immediately for buggy BIOS
   - Creative SoundCore3D support
   - Add Conexant CX20751/2/3/4 codec support

  * ASoC
   - Dynamic PCM support, allowing support for SoCs with internal
     routing through components with tight sequencing and formatting
     constraints within their internal paths or where there are multiple
     components connected with CPU managed DMA controllers inside the
     SoC.
   - Greatly improved support for direct digital links between off-SoC
     devices, providing a much simpler way of connecting things like
     digital basebands to CODECs.
   - Much more fine grained and robust locking, cleaning up some of the
     confusion that crept in with multi-component.
   - CPU support for nVidia Tegra 30 I2S and audio hub controllers and
     ST-Ericsson MSP I2S controolers
   - New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
     Texas Instruments LM49453.
   - Some regmap changes needed by the Tegra I2S driver.
   - mc13783 audio support.

  * Misc
   - Rewrite with module_pci_driver()
   - Xonar DGX support for snd-oxygen
   - Improvement of packet handling in snd-firewire driver
   - New USB-endpoint streaming logic
   - Enhanced M-audio FTU quirks and relevant cleanups
   - Increment the support of OSS devices to 256
   - snd-aloop accuracy improvement

  There are a few more pending changes for 3.5, but they will be sent
  slightly later as partly depending on the changes of DRM."

Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree).  Also some
fairly trivial context conflicts in the imx and mcx soc drivers.

* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
  ALSA: snd-usb: fix stream info output in /proc
  ALSA: pcm - Add proper state checks to snd_pcm_drain()
  ALSA: sh: Fix up namespace collision in sh_dac_audio.
  ALSA: hda/realtek - Fix unused variable compile warning
  ASoC: sh: fsi: enable chip specific data transfer mode
  ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
  ASoC: sh: fsi: use same format for IN/OUT
  ASoC: sh: fsi: add fsi_version() and removed meaningless version check
  ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
  ASoC: tegra: Add machine driver for WM8753 codec
  ALSA: hda - Fix possible races of accesses to connection list array
  ASoC: OMAP: HDMI: Introduce codec
  ARM: mx31_3ds: Add sound support
  ASoC: imx-mc13783 cleanup
  mx31moboard: Add sound support
  ASoC: mc13783 codec cleanups
  ASoC: add imx-mc13783 sound support
  ASoC: Add mc13783 codec
  mfd: mc13xxx: add codec platform data
  ASoC: don't flip master of DT-instantiated DAI links
  ...

1 files changed, 304 insertions, 0 deletions

diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
new file mode 100644
index 000000000000..fe54a69073e5
--- /dev/null
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -0,0 +1,304 @@
+/*
+ *  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 <linux/module.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "imx-ssi.h"
+#include "../codecs/wm8350.h"
+#include "imx-audmux.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->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->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_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
+				  ARRAY_SIZE(wm1133_ev1_widgets));
+
+	snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
+				ARRAY_SIZE(wm1133_ev1_map));
+
+	/* Headphone jack detection */
+	snd_soc_jack_new(codec, "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(codec, "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(dapm, "Mic Bias");
+
+	return 0;
+}
+
+
+static struct snd_soc_dai_link wm1133_ev1_dai = {
+	.name = "WM1133-EV1",
+	.stream_name = "Audio",
+	.cpu_dai_name = "imx-ssi.0",
+	.codec_dai_name = "wm8350-hifi",
+	.platform_name = "imx-fiq-pcm-audio.0",
+	.codec_name = "wm8350-codec.0-0x1a",
+	.init = wm1133_ev1_init,
+	.ops = &wm1133_ev1_ops,
+	.symmetric_rates = 1,
+};
+
+static struct snd_soc_card wm1133_ev1 = {
+	.name = "WM1133-EV1",
+	.owner = THIS_MODULE,
+	.dai_link = &wm1133_ev1_dai,
+	.num_links = 1,
+};
+
+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 = IMX_AUDMUX_V2_PTCR_SYN |
+		IMX_AUDMUX_V2_PTCR_TFSDIR |
+		IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
+		IMX_AUDMUX_V2_PTCR_TCLKDIR |
+		IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+	pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
+
+	ptcr = IMX_AUDMUX_V2_PTCR_SYN;
+	pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
+	imx_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);
+	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");