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>
147 lines
3.8 KiB
C
147 lines
3.8 KiB
C
/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
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*
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* Copyright 2009 Simtec Electronics
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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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#include <linux/module.h>
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#include <linux/clk.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 <plat/audio-simtec.h>
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#include "s3c-dma.h"
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#include "s3c24xx-i2s.h"
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#include "s3c24xx_simtec.h"
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#include "../codecs/tlv320aic3x.h"
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static const struct snd_soc_dapm_widget dapm_widgets[] = {
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SND_SOC_DAPM_LINE("GSM Out", NULL),
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SND_SOC_DAPM_LINE("GSM In", NULL),
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SND_SOC_DAPM_LINE("Line In", NULL),
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SND_SOC_DAPM_LINE("Line Out", NULL),
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SND_SOC_DAPM_LINE("ZV", NULL),
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SND_SOC_DAPM_MIC("Mic Jack", NULL),
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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};
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static const struct snd_soc_dapm_route base_map[] = {
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/* Headphone connected to HP{L,R}OUT and HP{L,R}COM */
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{ "Headphone Jack", NULL, "HPLOUT" },
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{ "Headphone Jack", NULL, "HPLCOM" },
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{ "Headphone Jack", NULL, "HPROUT" },
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{ "Headphone Jack", NULL, "HPRCOM" },
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/* ZV connected to Line1 */
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{ "LINE1L", NULL, "ZV" },
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{ "LINE1R", NULL, "ZV" },
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/* Line In connected to Line2 */
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{ "LINE2L", NULL, "Line In" },
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{ "LINE2R", NULL, "Line In" },
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/* Microphone connected to MIC3R and MIC_BIAS */
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{ "MIC3L", NULL, "Mic Jack" },
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/* GSM connected to MONO_LOUT and MIC3L (in) */
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{ "GSM Out", NULL, "MONO_LOUT" },
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{ "MIC3L", NULL, "GSM In" },
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/* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are
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* not using the DAPM to power it up and down as there it makes
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* a click when powering up. */
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};
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/**
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* simtec_hermes_init - initialise and add controls
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* @codec; The codec instance to attach to.
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*
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* Attach our controls and configure the necessary codec
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* mappings for our sound card instance.
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*/
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static int simtec_hermes_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, dapm_widgets,
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ARRAY_SIZE(dapm_widgets));
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snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map));
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snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
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snd_soc_dapm_enable_pin(dapm, "Line In");
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snd_soc_dapm_enable_pin(dapm, "Line Out");
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snd_soc_dapm_enable_pin(dapm, "Mic Jack");
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simtec_audio_init(rtd);
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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 simtec_dai_aic33 = {
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.name = "tlv320aic33",
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.stream_name = "TLV320AIC33",
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.codec_name = "tlv320aic3x-codec.0-0x1a",
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.cpu_dai_name = "s3c24xx-i2s",
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.codec_dai_name = "tlv320aic3x-hifi",
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.platform_name = "s3c24xx-pcm-audio",
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.init = simtec_hermes_init,
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};
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/* simtec audio machine driver */
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static struct snd_soc_card snd_soc_machine_simtec_aic33 = {
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.name = "Simtec-Hermes",
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.dai_link = &simtec_dai_aic33,
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.num_links = 1,
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};
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static int __devinit simtec_audio_hermes_probe(struct platform_device *pd)
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{
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dev_info(&pd->dev, "probing....\n");
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return simtec_audio_core_probe(pd, &snd_soc_machine_simtec_aic33);
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}
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static struct platform_driver simtec_audio_hermes_platdrv = {
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.driver = {
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.owner = THIS_MODULE,
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.name = "s3c24xx-simtec-hermes-snd",
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.pm = simtec_audio_pm,
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},
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.probe = simtec_audio_hermes_probe,
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.remove = __devexit_p(simtec_audio_remove),
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};
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MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd");
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static int __init simtec_hermes_modinit(void)
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{
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return platform_driver_register(&simtec_audio_hermes_platdrv);
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}
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static void __exit simtec_hermes_modexit(void)
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{
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platform_driver_unregister(&simtec_audio_hermes_platdrv);
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}
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module_init(simtec_hermes_modinit);
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module_exit(simtec_hermes_modexit);
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MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
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MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
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MODULE_LICENSE("GPL");
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