Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
188 lines
4.7 KiB
C
188 lines
4.7 KiB
C
/*
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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->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->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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/* 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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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_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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/* Add afeb9260 specific widgets */
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snd_soc_dapm_new_controls(dapm, 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(dapm, audio_map, ARRAY_SIZE(audio_map));
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snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
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snd_soc_dapm_enable_pin(dapm, "Line In");
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snd_soc_dapm_enable_pin(dapm, "Mic Jack");
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snd_soc_dapm_sync(dapm);
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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_name = "atmel-ssc-dai.0",
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.codec_dai_name = "tlv320aic23-hifi",
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.platform_name = "atmel_pcm-audio",
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.codec_name = "tlv320aic23-codec.0-0x1a",
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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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.dai_link = &afeb9260_dai,
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.num_links = 1,
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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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if (!(machine_is_afeb9260()))
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return -ENODEV;
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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, &snd_soc_machine_afeb9260);
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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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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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