Merge branch 'topic/asoc' into for-linus

Conflicts:
	sound/soc/codecs/ad1938.c

1 files changed, 308 insertions, 0 deletions

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");