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>
174 lines
4.1 KiB
C
174 lines
4.1 KiB
C
/*
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* e800-wm9712.c -- SoC audio for e800
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*
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* Copyright 2007 (c) Ian Molton <spyro@f2s.com>
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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; version 2 ONLY.
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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/gpio.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 <asm/mach-types.h>
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#include <mach/audio.h>
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#include <mach/eseries-gpio.h>
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#include "../codecs/wm9712.h"
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#include "pxa2xx-ac97.h"
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static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *kcontrol, int event)
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{
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if (event & SND_SOC_DAPM_PRE_PMU)
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gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
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else if (event & SND_SOC_DAPM_POST_PMD)
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gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
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return 0;
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}
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static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *kcontrol, int event)
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{
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if (event & SND_SOC_DAPM_PRE_PMU)
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gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
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else if (event & SND_SOC_DAPM_POST_PMD)
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gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
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return 0;
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}
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static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
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SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
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SND_SOC_DAPM_SPK("Speaker", NULL),
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SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
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e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
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SND_SOC_DAPM_POST_PMD),
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SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
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e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
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SND_SOC_DAPM_POST_PMD),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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{"Headphone Jack", NULL, "HPOUTL"},
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{"Headphone Jack", NULL, "HPOUTR"},
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{"Headphone Jack", NULL, "Headphone Amp"},
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{"Speaker Amp", NULL, "MONOOUT"},
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{"Speaker", NULL, "Speaker Amp"},
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{"MIC1", NULL, "Mic (Internal1)"},
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{"MIC2", NULL, "Mic (Internal2)"},
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};
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static int e800_ac97_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, e800_dapm_widgets,
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ARRAY_SIZE(e800_dapm_widgets));
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snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
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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 e800_dai[] = {
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{
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.name = "AC97",
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.stream_name = "AC97 HiFi",
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.cpu_dai_name = "pxa-ac97.0",
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.codec_dai_name = "wm9712-hifi",
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.platform_name = "pxa-pcm-audio",
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.codec_name = "wm9712-codec",
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.init = e800_ac97_init,
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},
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{
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.name = "AC97 Aux",
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.stream_name = "AC97 Aux",
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.cpu_dai_name = "pxa-ac97.1",
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.codec_dai_name ="wm9712-aux",
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.platform_name = "pxa-pcm-audio",
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.codec_name = "wm9712-codec",
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},
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};
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static struct snd_soc_card e800 = {
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.name = "Toshiba e800",
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.dai_link = e800_dai,
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.num_links = ARRAY_SIZE(e800_dai),
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};
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static struct platform_device *e800_snd_device;
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static int __init e800_init(void)
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{
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int ret;
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if (!machine_is_e800())
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return -ENODEV;
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ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp");
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if (ret)
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return ret;
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ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp");
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if (ret)
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goto free_hp_amp_gpio;
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ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1);
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if (ret)
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goto free_spk_amp_gpio;
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ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1);
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if (ret)
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goto free_spk_amp_gpio;
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e800_snd_device = platform_device_alloc("soc-audio", -1);
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if (!e800_snd_device)
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return -ENOMEM;
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platform_set_drvdata(e800_snd_device, &e800);
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ret = platform_device_add(e800_snd_device);
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if (!ret)
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return 0;
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/* Fail gracefully */
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platform_device_put(e800_snd_device);
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free_spk_amp_gpio:
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gpio_free(GPIO_E800_SPK_AMP_ON);
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free_hp_amp_gpio:
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gpio_free(GPIO_E800_HP_AMP_OFF);
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return ret;
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}
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static void __exit e800_exit(void)
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{
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platform_device_unregister(e800_snd_device);
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gpio_free(GPIO_E800_SPK_AMP_ON);
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gpio_free(GPIO_E800_HP_AMP_OFF);
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}
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module_init(e800_init);
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module_exit(e800_exit);
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/* Module information */
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MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
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MODULE_DESCRIPTION("ALSA SoC driver for e800");
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MODULE_LICENSE("GPL v2");
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