forked from luck/tmp_suning_uos_patched
875065491f
One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
187 lines
4.7 KiB
C
187 lines
4.7 KiB
C
/*
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* File: sound/soc/blackfin/bf5xx-ssm2602.c
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* Author: Cliff Cai <Cliff.Cai@analog.com>
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*
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* Created: Tue June 06 2008
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* Description: board driver for SSM2602 sound chip
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*
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* Modified:
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* Copyright 2008 Analog Devices Inc.
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*
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* Bugs: Enter bugs at http://blackfin.uclinux.org/
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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 as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, see the file COPYING, or write
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* to the Free Software Foundation, Inc.,
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* 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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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 <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 <sound/pcm_params.h>
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#include <asm/dma.h>
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#include <asm/portmux.h>
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#include <linux/gpio.h>
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#include "../codecs/ssm2602.h"
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#include "bf5xx-sport.h"
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#include "bf5xx-i2s-pcm.h"
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#include "bf5xx-i2s.h"
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static struct snd_soc_card bf5xx_ssm2602;
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static int bf5xx_ssm2602_startup(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 *cpu_dai = rtd->dai->cpu_dai;
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pr_debug("%s enter\n", __func__);
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cpu_dai->private_data = sport_handle;
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return 0;
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}
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static int bf5xx_ssm2602_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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struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
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unsigned int clk = 0;
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int ret = 0;
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pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
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params_format(params));
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/*
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* If you are using a crystal source which frequency is not 12MHz
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* then modify the below case statement with frequency of the crystal.
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*
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* If you are using the SPORT to generate clocking then this is
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* where to do it.
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*/
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switch (params_rate(params)) {
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case 8000:
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case 16000:
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case 48000:
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case 96000:
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case 11025:
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case 22050:
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case 44100:
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clk = 12000000;
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break;
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}
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/*
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* CODEC is master for BCLK and LRC in this configuration.
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*/
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/* set codec DAI configuration */
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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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/* set cpu DAI configuration */
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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_sysclk(codec_dai, SSM2602_SYSCLK, clk,
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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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return 0;
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}
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static struct snd_soc_ops bf5xx_ssm2602_ops = {
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.startup = bf5xx_ssm2602_startup,
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.hw_params = bf5xx_ssm2602_hw_params,
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};
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static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
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.name = "ssm2602",
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.stream_name = "SSM2602",
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.cpu_dai = &bf5xx_i2s_dai,
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.codec_dai = &ssm2602_dai,
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.ops = &bf5xx_ssm2602_ops,
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};
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/*
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* SSM2602 2 wire address is determined by CSB
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* state during powerup.
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* low = 0x1a
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* high = 0x1b
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*/
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static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
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.i2c_bus = 0,
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.i2c_address = 0x1b,
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};
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static struct snd_soc_card bf5xx_ssm2602 = {
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.name = "bf5xx_ssm2602",
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.dai_link = &bf5xx_ssm2602_dai,
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.num_links = 1,
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};
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static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
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.card = &bf5xx_ssm2602,
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.platform = &bf5xx_i2s_soc_platform,
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.codec_dev = &soc_codec_dev_ssm2602,
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.codec_data = &bf5xx_ssm2602_setup,
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};
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static struct platform_device *bf52x_ssm2602_snd_device;
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static int __init bf5xx_ssm2602_init(void)
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{
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int ret;
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pr_debug("%s enter\n", __func__);
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bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1);
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if (!bf52x_ssm2602_snd_device)
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return -ENOMEM;
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platform_set_drvdata(bf52x_ssm2602_snd_device,
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&bf5xx_ssm2602_snd_devdata);
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bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev;
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ret = platform_device_add(bf52x_ssm2602_snd_device);
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if (ret)
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platform_device_put(bf52x_ssm2602_snd_device);
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return ret;
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}
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static void __exit bf5xx_ssm2602_exit(void)
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{
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pr_debug("%s enter\n", __func__);
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platform_device_unregister(bf52x_ssm2602_snd_device);
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}
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module_init(bf5xx_ssm2602_init);
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module_exit(bf5xx_ssm2602_exit);
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/* Module information */
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MODULE_AUTHOR("Cliff Cai");
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MODULE_DESCRIPTION("ALSA SoC SSM2602 BF527-EZKIT");
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MODULE_LICENSE("GPL");
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