ce6120cca2
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>
160 lines
3.9 KiB
C
160 lines
3.9 KiB
C
/*
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* Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.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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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 675 Mass Ave, Cambridge, MA 02139, 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/timer.h>
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#include <linux/interrupt.h>
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#include <linux/platform_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 <linux/gpio.h>
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#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29)
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#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4)
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static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget,
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struct snd_kcontrol *ctrl, int event)
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{
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int on = 0;
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if (event & SND_SOC_DAPM_POST_PMU)
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on = 1;
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else if (event & SND_SOC_DAPM_PRE_PMD)
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on = 0;
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gpio_set_value(QI_LB60_SND_GPIO, on);
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gpio_set_value(QI_LB60_AMP_GPIO, on);
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return 0;
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}
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static const struct snd_soc_dapm_widget qi_lb60_widgets[] = {
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SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event),
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SND_SOC_DAPM_MIC("Mic", NULL),
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};
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static const struct snd_soc_dapm_route qi_lb60_routes[] = {
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{"Mic", NULL, "MIC"},
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{"Speaker", NULL, "LOUT"},
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{"Speaker", NULL, "ROUT"},
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};
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#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \
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SND_SOC_DAIFMT_NB_NF | \
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SND_SOC_DAIFMT_CBM_CFM)
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static int qi_lb60_codec_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_dai *cpu_dai = rtd->cpu_dai;
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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int ret;
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snd_soc_dapm_nc_pin(dapm, "LIN");
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snd_soc_dapm_nc_pin(dapm, "RIN");
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ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
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if (ret < 0) {
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dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret);
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return ret;
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}
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snd_soc_dapm_new_controls(dapm, qi_lb60_widgets,
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ARRAY_SIZE(qi_lb60_widgets));
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snd_soc_dapm_add_routes(dapm, qi_lb60_routes,
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ARRAY_SIZE(qi_lb60_routes));
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snd_soc_dapm_sync(dapm);
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return 0;
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}
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static struct snd_soc_dai_link qi_lb60_dai = {
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.name = "jz4740",
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.stream_name = "jz4740",
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.cpu_dai_name = "jz4740-i2s",
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.platform_name = "jz4740-pcm-audio",
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.codec_dai_name = "jz4740-hifi",
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.codec_name = "jz4740-codec",
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.init = qi_lb60_codec_init,
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};
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static struct snd_soc_card qi_lb60 = {
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.name = "QI LB60",
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.dai_link = &qi_lb60_dai,
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.num_links = 1,
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};
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static struct platform_device *qi_lb60_snd_device;
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static int __init qi_lb60_init(void)
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{
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int ret;
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qi_lb60_snd_device = platform_device_alloc("soc-audio", -1);
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if (!qi_lb60_snd_device)
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return -ENOMEM;
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ret = gpio_request(QI_LB60_SND_GPIO, "SND");
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if (ret) {
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pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n",
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QI_LB60_SND_GPIO, ret);
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goto err_device_put;
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}
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ret = gpio_request(QI_LB60_AMP_GPIO, "AMP");
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if (ret) {
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pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n",
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QI_LB60_AMP_GPIO, ret);
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goto err_gpio_free_snd;
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}
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gpio_direction_output(QI_LB60_SND_GPIO, 0);
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gpio_direction_output(QI_LB60_AMP_GPIO, 0);
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platform_set_drvdata(qi_lb60_snd_device, &qi_lb60);
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ret = platform_device_add(qi_lb60_snd_device);
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if (ret) {
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pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret);
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goto err_unset_pdata;
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}
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return 0;
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err_unset_pdata:
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platform_set_drvdata(qi_lb60_snd_device, NULL);
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/*err_gpio_free_amp:*/
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gpio_free(QI_LB60_AMP_GPIO);
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err_gpio_free_snd:
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gpio_free(QI_LB60_SND_GPIO);
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err_device_put:
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platform_device_put(qi_lb60_snd_device);
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return ret;
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}
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module_init(qi_lb60_init);
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static void __exit qi_lb60_exit(void)
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{
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gpio_free(QI_LB60_AMP_GPIO);
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gpio_free(QI_LB60_SND_GPIO);
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platform_device_unregister(qi_lb60_snd_device);
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}
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module_exit(qi_lb60_exit);
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MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
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MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support");
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MODULE_LICENSE("GPL v2");
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