b6905d0b16
88PM860x codec is used in Marvell saarb development board. 88PM860x codec is used as master mode for SSP communication. Only I2S format is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
201 lines
5.3 KiB
C
201 lines
5.3 KiB
C
/*
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* saarb.c -- SoC audio for saarb
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*
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* Copyright (C) 2010 Marvell International Ltd.
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* Haojian Zhuang <haojian.zhuang@marvell.com>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2 as
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* published by the Free Software Foundation.
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*/
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/device.h>
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#include <linux/clk.h>
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#include <linux/i2c.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include <sound/jack.h>
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#include <asm/mach-types.h>
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#include "../codecs/88pm860x-codec.h"
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#include "pxa-ssp.h"
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static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
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static struct platform_device *saarb_snd_device;
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static struct snd_soc_jack hs_jack, mic_jack;
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static struct snd_soc_jack_pin hs_jack_pins[] = {
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{ .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
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};
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static struct snd_soc_jack_pin mic_jack_pins[] = {
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{ .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
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};
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/* saarb machine dapm widgets */
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static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
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SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
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SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
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SND_SOC_DAPM_SPK("Ext Speaker", NULL),
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SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
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};
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/* saarb machine audio map */
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static const struct snd_soc_dapm_route audio_map[] = {
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{"Headset Stereophone", NULL, "HS1"},
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{"Headset Stereophone", NULL, "HS2"},
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{"Ext Speaker", NULL, "LSP"},
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{"Ext Speaker", NULL, "LSN"},
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{"Lineout Out 1", NULL, "LINEOUT1"},
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{"Lineout Out 2", NULL, "LINEOUT2"},
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{"MIC1P", NULL, "Mic1 Bias"},
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{"MIC1N", NULL, "Mic1 Bias"},
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{"Mic1 Bias", NULL, "Ext Mic 1"},
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{"MIC2P", NULL, "Mic1 Bias"},
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{"MIC2N", NULL, "Mic1 Bias"},
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{"Mic1 Bias", NULL, "Headset Mic 2"},
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{"MIC3P", NULL, "Mic3 Bias"},
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{"MIC3N", NULL, "Mic3 Bias"},
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{"Mic3 Bias", NULL, "Ext Mic 3"},
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};
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static int saarb_i2s_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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int width = snd_pcm_format_physical_width(params_format(params));
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int ret;
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ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
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PM860X_CLK_DIR_OUT);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
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return ret;
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}
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static struct snd_soc_ops saarb_i2s_ops = {
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.hw_params = saarb_i2s_hw_params,
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};
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static struct snd_soc_dai_link saarb_dai[] = {
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{
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.name = "88PM860x I2S",
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.stream_name = "I2S Audio",
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.cpu_dai_name = "pxa-ssp-dai.1",
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.codec_dai_name = "88pm860x-i2s",
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.platform_name = "pxa-pcm-audio",
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.codec_name = "88pm860x-codec",
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.init = saarb_pm860x_init,
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.ops = &saarb_i2s_ops,
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},
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};
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static struct snd_soc_card snd_soc_card_saarb = {
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.name = "Saarb",
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.dai_link = saarb_dai,
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.num_links = ARRAY_SIZE(saarb_dai),
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};
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static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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int ret;
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snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
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ARRAY_SIZE(saarb_dapm_widgets));
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snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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/* connected pins */
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snd_soc_dapm_enable_pin(codec, "Ext Speaker");
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snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
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snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
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snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
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snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
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ret = snd_soc_dapm_sync(codec);
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if (ret)
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return ret;
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/* Headset jack detection */
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snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
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| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
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&hs_jack);
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snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
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hs_jack_pins);
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snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
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&mic_jack);
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snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
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mic_jack_pins);
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/* headphone, microphone detection & headset short detection */
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pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
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SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
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pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
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return 0;
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}
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static int __init saarb_init(void)
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{
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int ret;
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if (!machine_is_saarb())
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return -ENODEV;
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saarb_snd_device = platform_device_alloc("soc-audio", -1);
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if (!saarb_snd_device)
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return -ENOMEM;
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platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
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ret = platform_device_add(saarb_snd_device);
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if (ret)
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platform_device_put(saarb_snd_device);
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return ret;
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}
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static void __exit saarb_exit(void)
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{
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platform_device_unregister(saarb_snd_device);
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}
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module_init(saarb_init);
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module_exit(saarb_exit);
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MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
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MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
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MODULE_LICENSE("GPL");
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