forked from luck/tmp_suning_uos_patched
b2c3e92311
wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
219 lines
5.1 KiB
C
219 lines
5.1 KiB
C
/*
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* ALSA SoC driver for Migo-R
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*
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* Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
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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/device.h>
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#include <linux/firmware.h>
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#include <linux/module.h>
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#include <asm/clock.h>
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#include <cpu/sh7722.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include "../codecs/wm8978.h"
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#include "siu.h"
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/* Default 8000Hz sampling frequency */
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static unsigned long codec_freq = 8000 * 512;
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static unsigned int use_count;
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/* External clock, sourced from the codec at the SIUMCKB pin */
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static unsigned long siumckb_recalc(struct clk *clk)
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{
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return codec_freq;
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}
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static struct clk_ops siumckb_clk_ops = {
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.recalc = siumckb_recalc,
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};
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static struct clk siumckb_clk = {
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.name = "siumckb_clk",
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.id = -1,
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.ops = &siumckb_clk_ops,
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.rate = 0, /* initialised at run-time */
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};
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static int migor_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->dai->codec_dai;
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int ret;
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unsigned int rate = params_rate(params);
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ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000,
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SND_SOC_CLOCK_IN);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512);
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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_NB_IF |
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SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS);
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if (ret < 0)
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return ret;
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codec_freq = rate * 512;
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/*
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* This propagates the parent frequency change to children and
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* recalculates the frequency table
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*/
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clk_set_rate(&siumckb_clk, codec_freq);
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dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq);
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ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT,
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codec_freq / 2, SND_SOC_CLOCK_IN);
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if (!ret)
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use_count++;
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return ret;
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}
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static int migor_hw_free(struct snd_pcm_substream *substream)
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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->dai->codec_dai;
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if (use_count) {
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use_count--;
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if (!use_count)
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snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0,
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SND_SOC_CLOCK_IN);
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} else {
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dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n");
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}
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return 0;
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}
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static struct snd_soc_ops migor_dai_ops = {
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.hw_params = migor_hw_params,
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.hw_free = migor_hw_free,
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};
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static const struct snd_soc_dapm_widget migor_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone", NULL),
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SND_SOC_DAPM_MIC("Onboard Microphone", NULL),
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SND_SOC_DAPM_MIC("External Microphone", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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/* Headphone output connected to LHP/RHP, enable OUT4 for VMID */
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{ "Headphone", NULL, "OUT4 VMID" },
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{ "OUT4 VMID", NULL, "LHP" },
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{ "OUT4 VMID", NULL, "RHP" },
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/* On-board microphone */
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{ "RMICN", NULL, "Mic Bias" },
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{ "RMICP", NULL, "Mic Bias" },
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{ "Mic Bias", NULL, "Onboard Microphone" },
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/* External microphone */
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{ "LMICN", NULL, "Mic Bias" },
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{ "LMICP", NULL, "Mic Bias" },
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{ "Mic Bias", NULL, "External Microphone" },
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};
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static int migor_dai_init(struct snd_soc_codec *codec)
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{
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snd_soc_dapm_new_controls(codec, migor_dapm_widgets,
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ARRAY_SIZE(migor_dapm_widgets));
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snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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return 0;
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}
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/* migor digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link migor_dai = {
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.name = "wm8978",
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.stream_name = "WM8978",
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.cpu_dai = &siu_i2s_dai,
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.codec_dai = &wm8978_dai,
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.ops = &migor_dai_ops,
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.init = migor_dai_init,
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};
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/* migor audio machine driver */
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static struct snd_soc_card snd_soc_migor = {
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.name = "Migo-R",
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.platform = &siu_platform,
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.dai_link = &migor_dai,
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.num_links = 1,
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};
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/* migor audio subsystem */
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static struct snd_soc_device migor_snd_devdata = {
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.card = &snd_soc_migor,
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.codec_dev = &soc_codec_dev_wm8978,
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};
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static struct platform_device *migor_snd_device;
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static int __init migor_init(void)
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{
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int ret;
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ret = clk_register(&siumckb_clk);
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if (ret < 0)
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return ret;
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/* Port number used on this machine: port B */
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migor_snd_device = platform_device_alloc("soc-audio", 1);
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if (!migor_snd_device) {
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ret = -ENOMEM;
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goto epdevalloc;
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}
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platform_set_drvdata(migor_snd_device, &migor_snd_devdata);
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migor_snd_devdata.dev = &migor_snd_device->dev;
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ret = platform_device_add(migor_snd_device);
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if (ret)
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goto epdevadd;
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return 0;
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epdevadd:
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platform_device_put(migor_snd_device);
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epdevalloc:
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clk_unregister(&siumckb_clk);
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return ret;
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}
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static void __exit migor_exit(void)
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{
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clk_unregister(&siumckb_clk);
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platform_device_unregister(migor_snd_device);
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}
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module_init(migor_init);
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module_exit(migor_exit);
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MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski@gmx.de>");
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MODULE_DESCRIPTION("ALSA SoC Migor");
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MODULE_LICENSE("GPL v2");
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