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>
143 lines
3.4 KiB
C
143 lines
3.4 KiB
C
/*
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* kirkwood-t5325.c
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*
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* (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the
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* Free Software Foundation; either version 2 of the License, or (at your
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* option) any later version.
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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/interrupt.h>
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#include <linux/platform_device.h>
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#include <linux/slab.h>
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#include <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include <mach/kirkwood.h>
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#include <plat/audio.h>
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#include <asm/mach-types.h>
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#include "../codecs/alc5623.h"
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static int t5325_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 ret;
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unsigned int freq, fmt;
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fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
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ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
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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, fmt);
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if (ret < 0)
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return ret;
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freq = params_rate(params) * 256;
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return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
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}
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static struct snd_soc_ops t5325_ops = {
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.hw_params = t5325_hw_params,
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};
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static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_SPK("Speaker", 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 t5325_route[] = {
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{ "Headphone Jack", NULL, "HPL" },
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{ "Headphone Jack", NULL, "HPR" },
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{"Speaker", NULL, "SPKOUT"},
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{"Speaker", NULL, "SPKOUTN"},
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{ "MIC1", NULL, "Mic Jack" },
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{ "MIC2", NULL, "Mic Jack" },
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};
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static int t5325_dai_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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snd_soc_dapm_new_controls(dapm, t5325_dapm_widgets,
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ARRAY_SIZE(t5325_dapm_widgets));
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snd_soc_dapm_add_routes(dapm, t5325_route, ARRAY_SIZE(t5325_route));
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snd_soc_dapm_enable_pin(dapm, "Mic Jack");
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snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
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snd_soc_dapm_enable_pin(dapm, "Speaker");
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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 t5325_dai[] = {
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{
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.name = "ALC5621",
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.stream_name = "ALC5621 HiFi",
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.cpu_dai_name = "kirkwood-i2s",
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.platform_name = "kirkwood-pcm-audio",
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.codec_dai_name = "alc5621-hifi",
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.codec_name = "alc562x-codec.0-001a",
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.ops = &t5325_ops,
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.init = t5325_dai_init,
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},
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};
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static struct snd_soc_card t5325 = {
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.name = "t5325",
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.dai_link = t5325_dai,
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.num_links = ARRAY_SIZE(t5325_dai),
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};
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static struct platform_device *t5325_snd_device;
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static int __init t5325_init(void)
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{
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int ret;
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if (!machine_is_t5325())
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return 0;
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t5325_snd_device = platform_device_alloc("soc-audio", -1);
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if (!t5325_snd_device)
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return -ENOMEM;
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platform_set_drvdata(t5325_snd_device,
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&t5325);
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ret = platform_device_add(t5325_snd_device);
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if (ret) {
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printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
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platform_device_put(t5325_snd_device);
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}
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return ret;
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}
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module_init(t5325_init);
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static void __exit t5325_exit(void)
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{
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platform_device_unregister(t5325_snd_device);
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}
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module_exit(t5325_exit);
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MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
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MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
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MODULE_LICENSE("GPL");
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