mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-02 15:56:42 +07:00
Merge branch 'for-2.6.31' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc
This commit is contained in:
Takashi Iwai
commit
dd4a416442
@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
|
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switch (resetgpio_action) {
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case RESETGPIO_NORMAL_ALTFUNC:
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if (reset_gpio == 113)
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mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
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mode = 113 | GPIO_ALT_FN_2_OUT;
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if (reset_gpio == 95)
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mode = 95 | GPIO_ALT_FN_1_OUT;
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break;
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@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
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and FRAME signals on the PlayPaq. Unless you want to play
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with the AT32 as the SSC master, you probably want to say N here,
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as this will give you better sound quality.
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config SND_AT91_SOC_AFEB9260
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tristate "SoC Audio support for AFEB9260 board"
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depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
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select SND_ATMEL_SOC_SSC
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select SND_SOC_TLV320AIC23
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help
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Say Y here to support sound on AFEB9260 board.
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@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
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obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
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obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
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obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
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203
sound/soc/atmel/snd-soc-afeb9260.c
Normal file
203
sound/soc/atmel/snd-soc-afeb9260.c
Normal file
@ -0,0 +1,203 @@
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/*
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* afeb9260.c -- SoC audio for AFEB9260
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*
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* Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* 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, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*
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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/kernel.h>
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#include <linux/clk.h>
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#include <linux/platform_device.h>
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#include <linux/atmel-ssc.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 <asm/mach-types.h>
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#include <mach/hardware.h>
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#include <linux/gpio.h>
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#include "../codecs/tlv320aic23.h"
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#include "atmel-pcm.h"
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#include "atmel_ssc_dai.h"
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#define CODEC_CLOCK 12000000
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static int afeb9260_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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int err;
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/* Set codec DAI configuration */
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err = snd_soc_dai_set_fmt(codec_dai,
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SND_SOC_DAIFMT_I2S|
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SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (err < 0) {
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printk(KERN_ERR "can't set codec DAI configuration\n");
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return err;
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}
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/* Set cpu DAI configuration */
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err = snd_soc_dai_set_fmt(cpu_dai,
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SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (err < 0) {
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printk(KERN_ERR "can't set cpu DAI configuration\n");
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return err;
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}
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||||
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/* Set the codec system clock for DAC and ADC */
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err =
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snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
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if (err < 0) {
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printk(KERN_ERR "can't set codec system clock\n");
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return err;
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}
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||||
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||||
return err;
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||||
}
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static struct snd_soc_ops afeb9260_ops = {
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.hw_params = afeb9260_hw_params,
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};
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static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_LINE("Line In", NULL),
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SND_SOC_DAPM_MIC("Mic Jack", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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{"Headphone Jack", NULL, "LHPOUT"},
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{"Headphone Jack", NULL, "RHPOUT"},
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{"LLINEIN", NULL, "Line In"},
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{"RLINEIN", NULL, "Line In"},
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{"MICIN", NULL, "Mic Jack"},
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};
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static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
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{
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/* Add afeb9260 specific widgets */
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snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
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ARRAY_SIZE(tlv320aic23_dapm_widgets));
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/* Set up afeb9260 specific audio path audio_map */
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snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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snd_soc_dapm_enable_pin(codec, "Headphone Jack");
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snd_soc_dapm_enable_pin(codec, "Line In");
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snd_soc_dapm_enable_pin(codec, "Mic Jack");
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snd_soc_dapm_sync(codec);
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return 0;
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}
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/* Digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link afeb9260_dai = {
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.name = "TLV320AIC23",
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.stream_name = "AIC23",
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.cpu_dai = &atmel_ssc_dai[0],
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.codec_dai = &tlv320aic23_dai,
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.init = afeb9260_tlv320aic23_init,
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.ops = &afeb9260_ops,
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_machine_afeb9260 = {
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.name = "AFEB9260",
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.platform = &atmel_soc_platform,
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.dai_link = &afeb9260_dai,
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.num_links = 1,
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};
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/* Audio subsystem */
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static struct snd_soc_device afeb9260_snd_devdata = {
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.card = &snd_soc_machine_afeb9260,
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.codec_dev = &soc_codec_dev_tlv320aic23,
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};
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static struct platform_device *afeb9260_snd_device;
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static int __init afeb9260_soc_init(void)
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{
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int err;
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struct device *dev;
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struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
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struct ssc_device *ssc = NULL;
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if (!(machine_is_afeb9260()))
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return -ENODEV;
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ssc = ssc_request(0);
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if (IS_ERR(ssc)) {
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printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
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err = PTR_ERR(ssc);
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ssc = NULL;
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goto err_ssc;
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}
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ssc_p->ssc = ssc;
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afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
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if (!afeb9260_snd_device) {
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printk(KERN_ERR "ASoC: Platform device allocation failed\n");
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return -ENOMEM;
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}
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platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
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afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
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err = platform_device_add(afeb9260_snd_device);
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if (err)
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goto err1;
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dev = &afeb9260_snd_device->dev;
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return 0;
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err1:
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platform_device_del(afeb9260_snd_device);
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platform_device_put(afeb9260_snd_device);
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err_ssc:
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return err;
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}
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static void __exit afeb9260_soc_exit(void)
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{
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platform_device_unregister(afeb9260_snd_device);
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}
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module_init(afeb9260_soc_init);
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module_exit(afeb9260_soc_exit);
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MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
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MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
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MODULE_LICENSE("GPL");
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@ -422,36 +422,18 @@ static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
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SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
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/* Left analog microphone selection */
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static const char *twl4030_analoglmic_texts[] =
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{"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
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static const unsigned int twl4030_analoglmic_values[] =
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{0x0, 0x1, 0x2, 0x4, 0x8};
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static const struct soc_enum twl4030_analoglmic_enum =
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SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
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ARRAY_SIZE(twl4030_analoglmic_texts),
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twl4030_analoglmic_texts,
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twl4030_analoglmic_values);
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static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
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SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
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static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
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SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
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SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
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SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
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SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
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};
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||||
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||||
/* Right analog microphone selection */
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static const char *twl4030_analogrmic_texts[] =
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{"Off", "Sub mic", "AUXR"};
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static const unsigned int twl4030_analogrmic_values[] =
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{0x0, 0x1, 0x4};
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||||
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static const struct soc_enum twl4030_analogrmic_enum =
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||||
SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
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||||
ARRAY_SIZE(twl4030_analogrmic_texts),
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twl4030_analogrmic_texts,
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||||
twl4030_analogrmic_values);
|
||||
|
||||
static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
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SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
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||||
static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
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||||
SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
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||||
SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0),
|
||||
};
|
||||
|
||||
/* TX1 L/R Analog/Digital microphone selection */
|
||||
static const char *twl4030_micpathtx1_texts[] =
|
||||
@ -1138,11 +1120,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
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||||
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
|
||||
SND_SOC_DAPM_POST_REG),
|
||||
|
||||
/* Analog input muxes with switch for the capture amplifiers */
|
||||
SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
|
||||
TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
|
||||
SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
|
||||
TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
|
||||
/* Analog input mixers for the capture amplifiers */
|
||||
SND_SOC_DAPM_MIXER("Analog Left Capture Route",
|
||||
TWL4030_REG_ANAMICL, 4, 0,
|
||||
&twl4030_dapm_analoglmic_controls[0],
|
||||
ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
|
||||
SND_SOC_DAPM_MIXER("Analog Right Capture Route",
|
||||
TWL4030_REG_ANAMICR, 4, 0,
|
||||
&twl4030_dapm_analogrmic_controls[0],
|
||||
ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
|
||||
|
||||
SND_SOC_DAPM_PGA("ADC Physical Left",
|
||||
TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
|
||||
|
||||
@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800
|
||||
Toshiba e800 PDA
|
||||
|
||||
config SND_PXA2XX_SOC_EM_X270
|
||||
tristate "SoC Audio support for CompuLab EM-x270"
|
||||
tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
|
||||
depends on SND_PXA2XX_SOC && MACH_EM_X270
|
||||
select SND_PXA2XX_SOC_AC97
|
||||
select SND_SOC_WM9712
|
||||
help
|
||||
Say Y if you want to add support for SoC audio on
|
||||
CompuLab EM-x270.
|
||||
CompuLab EM-x270, eXeda and CM-X300 machines.
|
||||
|
||||
config SND_PXA2XX_SOC_PALM27X
|
||||
bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
|
||||
|
||||
@ -1,7 +1,7 @@
|
||||
/*
|
||||
* em-x270.c -- SoC audio for EM-X270
|
||||
* SoC audio driver for EM-X270, eXeda and CM-X300
|
||||
*
|
||||
* Copyright 2007 CompuLab, Ltd.
|
||||
* Copyright 2007, 2009 CompuLab, Ltd.
|
||||
*
|
||||
* Author: Mike Rapoport <mike@compulab.co.il>
|
||||
*
|
||||
@ -68,7 +68,8 @@ static int __init em_x270_init(void)
|
||||
{
|
||||
int ret;
|
||||
|
||||
if (!machine_is_em_x270())
|
||||
if (!(machine_is_em_x270() || machine_is_exeda()
|
||||
|| machine_is_cm_x300()))
|
||||
return -ENODEV;
|
||||
|
||||
em_x270_snd_device = platform_device_alloc("soc-audio", -1);
|
||||
@ -95,5 +96,5 @@ module_exit(em_x270_exit);
|
||||
|
||||
/* Module information */
|
||||
MODULE_AUTHOR("Mike Rapoport");
|
||||
MODULE_DESCRIPTION("ALSA SoC EM-X270");
|
||||
MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
|
||||
MODULE_LICENSE("GPL");
|
||||
|
||||
@ -329,6 +329,7 @@ struct snd_soc_dai pxa_i2s_dai = {
|
||||
.rates = PXA2XX_I2S_RATES,
|
||||
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
|
||||
.ops = &pxa_i2s_dai_ops,
|
||||
.symmetric_rates = 1,
|
||||
};
|
||||
|
||||
EXPORT_SYMBOL_GPL(pxa_i2s_dai);
|
||||
|
||||
@ -992,6 +992,9 @@ static int soc_remove(struct platform_device *pdev)
|
||||
struct snd_soc_platform *platform = card->platform;
|
||||
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
|
||||
|
||||
if (!card->instantiated)
|
||||
return 0;
|
||||
|
||||
run_delayed_work(&card->delayed_work);
|
||||
|
||||
if (platform->remove)
|
||||
@ -2387,6 +2390,39 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform)
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
|
||||
|
||||
static u64 codec_format_map[] = {
|
||||
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
|
||||
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
|
||||
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
|
||||
SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
|
||||
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
|
||||
SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
|
||||
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
|
||||
SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
|
||||
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
|
||||
SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
|
||||
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
|
||||
SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
|
||||
SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
|
||||
SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
|
||||
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
|
||||
| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
|
||||
};
|
||||
|
||||
/* Fix up the DAI formats for endianness: codecs don't actually see
|
||||
* the endianness of the data but we're using the CPU format
|
||||
* definitions which do need to include endianness so we ensure that
|
||||
* codec DAIs always have both big and little endian variants set.
|
||||
*/
|
||||
static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
|
||||
{
|
||||
int i;
|
||||
|
||||
for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
|
||||
if (stream->formats & codec_format_map[i])
|
||||
stream->formats |= codec_format_map[i];
|
||||
}
|
||||
|
||||
/**
|
||||
* snd_soc_register_codec - Register a codec with the ASoC core
|
||||
*
|
||||
@ -2394,6 +2430,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
|
||||
*/
|
||||
int snd_soc_register_codec(struct snd_soc_codec *codec)
|
||||
{
|
||||
int i;
|
||||
|
||||
if (!codec->name)
|
||||
return -EINVAL;
|
||||
|
||||
@ -2403,6 +2441,11 @@ int snd_soc_register_codec(struct snd_soc_codec *codec)
|
||||
|
||||
INIT_LIST_HEAD(&codec->list);
|
||||
|
||||
for (i = 0; i < codec->num_dai; i++) {
|
||||
fixup_codec_formats(&codec->dai[i].playback);
|
||||
fixup_codec_formats(&codec->dai[i].capture);
|
||||
}
|
||||
|
||||
mutex_lock(&client_mutex);
|
||||
list_add(&codec->list, &codec_list);
|
||||
snd_soc_instantiate_cards();
|
||||
|
||||
Loading…
Reference in New Issue
Block a user