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>
299 lines
7 KiB
C
299 lines
7 KiB
C
/*
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* tosa.c -- SoC audio for Tosa
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*
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* Copyright 2005 Wolfson Microelectronics PLC.
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* Copyright 2005 Openedhand Ltd.
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*
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* Authors: Liam Girdwood <lrg@slimlogic.co.uk>
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* Richard Purdie <richard@openedhand.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; 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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* GPIO's
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* 1 - Jack Insertion
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* 5 - Hookswitch (headset answer/hang up switch)
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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/device.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/tosa.h>
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#include <mach/audio.h>
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#include "../codecs/wm9712.h"
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#include "pxa2xx-ac97.h"
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static struct snd_soc_card tosa;
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#define TOSA_HP 0
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#define TOSA_MIC_INT 1
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#define TOSA_HEADSET 2
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#define TOSA_HP_OFF 3
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#define TOSA_SPK_ON 0
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#define TOSA_SPK_OFF 1
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static int tosa_jack_func;
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static int tosa_spk_func;
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static void tosa_ext_control(struct snd_soc_codec *codec)
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{
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/* set up jack connection */
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switch (tosa_jack_func) {
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case TOSA_HP:
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snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
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snd_soc_dapm_enable_pin(codec, "Headphone Jack");
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snd_soc_dapm_disable_pin(codec, "Headset Jack");
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break;
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case TOSA_MIC_INT:
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snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
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snd_soc_dapm_disable_pin(codec, "Headphone Jack");
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snd_soc_dapm_disable_pin(codec, "Headset Jack");
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break;
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case TOSA_HEADSET:
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snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
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snd_soc_dapm_disable_pin(codec, "Headphone Jack");
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snd_soc_dapm_enable_pin(codec, "Headset Jack");
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break;
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}
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if (tosa_spk_func == TOSA_SPK_ON)
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snd_soc_dapm_enable_pin(codec, "Speaker");
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else
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snd_soc_dapm_disable_pin(codec, "Speaker");
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snd_soc_dapm_sync(codec);
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}
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static int tosa_startup(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_codec *codec = rtd->card->codec;
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/* check the jack status at stream startup */
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tosa_ext_control(codec);
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return 0;
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}
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static struct snd_soc_ops tosa_ops = {
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.startup = tosa_startup,
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};
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static int tosa_get_jack(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.integer.value[0] = tosa_jack_func;
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return 0;
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}
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static int tosa_set_jack(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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if (tosa_jack_func == ucontrol->value.integer.value[0])
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return 0;
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tosa_jack_func = ucontrol->value.integer.value[0];
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tosa_ext_control(codec);
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return 1;
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}
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static int tosa_get_spk(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.integer.value[0] = tosa_spk_func;
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return 0;
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}
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static int tosa_set_spk(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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if (tosa_spk_func == ucontrol->value.integer.value[0])
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return 0;
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tosa_spk_func = ucontrol->value.integer.value[0];
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tosa_ext_control(codec);
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return 1;
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}
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/* tosa dapm event handlers */
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static int tosa_hp_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0);
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return 0;
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}
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/* tosa machine dapm widgets */
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static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
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SND_SOC_DAPM_HP("Headset Jack", NULL),
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SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
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SND_SOC_DAPM_SPK("Speaker", NULL),
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};
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/* tosa audio map */
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static const struct snd_soc_dapm_route audio_map[] = {
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/* headphone connected to HPOUTL, HPOUTR */
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{"Headphone Jack", NULL, "HPOUTL"},
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{"Headphone Jack", NULL, "HPOUTR"},
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/* ext speaker connected to LOUT2, ROUT2 */
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{"Speaker", NULL, "LOUT2"},
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{"Speaker", NULL, "ROUT2"},
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/* internal mic is connected to mic1, mic2 differential - with bias */
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{"MIC1", NULL, "Mic Bias"},
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{"MIC2", NULL, "Mic Bias"},
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{"Mic Bias", NULL, "Mic (Internal)"},
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/* headset is connected to HPOUTR, and LINEINR with bias */
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{"Headset Jack", NULL, "HPOUTR"},
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{"LINEINR", NULL, "Mic Bias"},
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{"Mic Bias", NULL, "Headset Jack"},
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};
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static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
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"Off"};
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static const char *spk_function[] = {"On", "Off"};
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static const struct soc_enum tosa_enum[] = {
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SOC_ENUM_SINGLE_EXT(5, jack_function),
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SOC_ENUM_SINGLE_EXT(2, spk_function),
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};
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static const struct snd_kcontrol_new tosa_controls[] = {
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SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
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tosa_set_jack),
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SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
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tosa_set_spk),
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};
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static int tosa_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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int err;
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snd_soc_dapm_nc_pin(codec, "OUT3");
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snd_soc_dapm_nc_pin(codec, "MONOOUT");
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/* add tosa specific controls */
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err = snd_soc_add_controls(codec, tosa_controls,
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ARRAY_SIZE(tosa_controls));
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if (err < 0)
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return err;
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/* add tosa specific widgets */
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snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
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ARRAY_SIZE(tosa_dapm_widgets));
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/* set up tosa specific audio path audio_map */
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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 tosa_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 = tosa_ac97_init,
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.ops = &tosa_ops,
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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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.ops = &tosa_ops,
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},
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};
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static int tosa_probe(struct platform_device *dev)
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{
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int ret;
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ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
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if (ret)
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return ret;
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ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
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if (ret)
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gpio_free(TOSA_GPIO_L_MUTE);
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return ret;
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}
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static int tosa_remove(struct platform_device *dev)
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{
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gpio_free(TOSA_GPIO_L_MUTE);
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return 0;
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}
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static struct snd_soc_card tosa = {
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.name = "Tosa",
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.dai_link = tosa_dai,
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.num_links = ARRAY_SIZE(tosa_dai),
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.probe = tosa_probe,
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.remove = tosa_remove,
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};
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static struct platform_device *tosa_snd_device;
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static int __init tosa_init(void)
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{
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int ret;
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if (!machine_is_tosa())
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return -ENODEV;
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tosa_snd_device = platform_device_alloc("soc-audio", -1);
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if (!tosa_snd_device) {
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ret = -ENOMEM;
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goto err_alloc;
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}
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platform_set_drvdata(tosa_snd_device, &tosa);
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ret = platform_device_add(tosa_snd_device);
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if (!ret)
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return 0;
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platform_device_put(tosa_snd_device);
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err_alloc:
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return ret;
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}
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static void __exit tosa_exit(void)
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{
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platform_device_unregister(tosa_snd_device);
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}
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module_init(tosa_init);
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module_exit(tosa_exit);
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/* Module information */
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MODULE_AUTHOR("Richard Purdie");
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MODULE_DESCRIPTION("ALSA SoC Tosa");
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MODULE_LICENSE("GPL");
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