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author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-05-20 09:41:44 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-05-20 09:41:44 -0700 |
commit | 7f06a8b26aba1dc03b42272dc0089a800372c575 (patch) | |
tree | 7c67198f83d069eb13fd417e022d111b7e4c82a1 /sound/soc/imx | |
parent | c3ad33c9bcb6616999953af76f16318120fe3691 (diff) | |
parent | d71f4cece4bd97d05592836202fc04ff2e7817e3 (diff) | |
download | linux-7f06a8b26aba1dc03b42272dc0089a800372c575.tar.bz2 |
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
...
Diffstat (limited to 'sound/soc/imx')
-rw-r--r-- | sound/soc/imx/Kconfig | 8 | ||||
-rw-r--r-- | sound/soc/imx/Makefile | 3 | ||||
-rw-r--r-- | sound/soc/imx/wm1133-ev1.c | 308 |
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"); 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