f0fba2ad1b
This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
172 lines
4.1 KiB
C
172 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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snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
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ARRAY_SIZE(e800_dapm_widgets));
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snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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snd_soc_dapm_sync(codec);
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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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