--- /dev/null
+/*
+ * 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);
+
+ /* 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);
+
+ /* TODO: The SSI driver should figure this out for us */
+ switch (channels) {
+ case 2:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
+ break;
+ case 1:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 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;
+
+ /* Headphone jack detection */
+ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
+ &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
+ wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
+
+ /* Microphone jack detection */
+ snd_soc_card_jack_new(rtd->card, "Microphone",
+ SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
+ mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
+ wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
+ SND_JACK_BTN_0);
+
+ snd_soc_dapm_force_enable_pin(&rtd->card->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-ssi.0",
+ .codec_name = "wm8350-codec.0-0x1a",
+ .init = wm1133_ev1_init,
+ .ops = &wm1133_ev1_ops,
+ .symmetric_rates = 1,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static struct snd_soc_card wm1133_ev1 = {
+ .name = "WM1133-EV1",
+ .owner = THIS_MODULE,
+ .dai_link = &wm1133_ev1_dai,
+ .num_links = 1,
+
+ .dapm_widgets = wm1133_ev1_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
+ .dapm_routes = wm1133_ev1_map,
+ .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
+};
+
+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");
Code Review / kvmfornfv.git / blobdiff