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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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1c0f3edbca
ASoC is now supporting modern style dai_link (= snd_soc_dai_link_component) for CPU/Codec/Platform. This patch switches to use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
113 lines
2.9 KiB
C
113 lines
2.9 KiB
C
// SPDX-License-Identifier: GPL-2.0
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//
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// Copyright 2009 Simtec Electronics
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#include <linux/module.h>
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#include <sound/soc.h>
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#include "s3c24xx_simtec.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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simtec_audio_init(rtd);
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return 0;
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}
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SND_SOC_DAILINK_DEFS(tlv320aic33,
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DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")),
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DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.0-001a",
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"tlv320aic3x-hifi")),
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DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis")));
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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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.init = simtec_hermes_init,
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SND_SOC_DAILINK_REG(tlv320aic33),
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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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.owner = THIS_MODULE,
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.dai_link = &simtec_dai_aic33,
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.num_links = 1,
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.dapm_widgets = dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
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.dapm_routes = base_map,
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.num_dapm_routes = ARRAY_SIZE(base_map),
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};
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static int 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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.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 = simtec_audio_remove,
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};
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module_platform_driver(simtec_audio_hermes_platdrv);
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MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd");
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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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