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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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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>
226 lines
5.5 KiB
C
226 lines
5.5 KiB
C
/*
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* ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms
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*
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* Copyright (C) 2010 Texas Instruments, Inc
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*
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* Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
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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 as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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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/init.h>
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#include <linux/io.h>
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#include <linux/delay.h>
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#include <linux/pm.h>
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#include <linux/platform_device.h>
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#include <linux/device.h>
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#include <linux/slab.h>
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#include <linux/clk.h>
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#include <linux/mfd/davinci_voicecodec.h>
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#include <linux/spi/spi.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/soc-dai.h>
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#include <sound/soc-dapm.h>
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#include <sound/initval.h>
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#include <mach/dm365.h>
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static inline unsigned int cq93vc_read(struct snd_soc_codec *codec,
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unsigned int reg)
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{
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struct davinci_vc *davinci_vc = codec->control_data;
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return readl(davinci_vc->base + reg);
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}
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static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg,
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unsigned int value)
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{
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struct davinci_vc *davinci_vc = codec->control_data;
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writel(value, davinci_vc->base + reg);
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return 0;
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}
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static const struct snd_kcontrol_new cq93vc_snd_controls[] = {
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SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0),
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SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0),
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};
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static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
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{
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struct snd_soc_codec *codec = dai->codec;
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u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE;
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if (mute)
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cq93vc_write(codec, DAVINCI_VC_REG09,
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reg | DAVINCI_VC_REG09_MUTE);
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else
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cq93vc_write(codec, DAVINCI_VC_REG09, reg);
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return 0;
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}
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static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai,
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int clk_id, unsigned int freq, int dir)
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{
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struct snd_soc_codec *codec = codec_dai->codec;
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struct davinci_vc *davinci_vc = codec->control_data;
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switch (freq) {
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case 22579200:
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case 27000000:
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case 33868800:
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davinci_vc->cq93vc.sysclk = freq;
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return 0;
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}
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return -EINVAL;
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}
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static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
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enum snd_soc_bias_level level)
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{
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switch (level) {
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case SND_SOC_BIAS_ON:
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cq93vc_write(codec, DAVINCI_VC_REG12,
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DAVINCI_VC_REG12_POWER_ALL_ON);
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break;
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case SND_SOC_BIAS_PREPARE:
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break;
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case SND_SOC_BIAS_STANDBY:
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cq93vc_write(codec, DAVINCI_VC_REG12,
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DAVINCI_VC_REG12_POWER_ALL_OFF);
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break;
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case SND_SOC_BIAS_OFF:
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/* force all power off */
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cq93vc_write(codec, DAVINCI_VC_REG12,
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DAVINCI_VC_REG12_POWER_ALL_OFF);
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break;
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}
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codec->bias_level = level;
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return 0;
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}
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#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
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#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
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static struct snd_soc_dai_ops cq93vc_dai_ops = {
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.digital_mute = cq93vc_mute,
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.set_sysclk = cq93vc_set_dai_sysclk,
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};
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static struct snd_soc_dai_driver cq93vc_dai = {
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.name = "cq93vc-hifi",
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.playback = {
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.stream_name = "Playback",
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.channels_min = 1,
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.channels_max = 2,
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.rates = CQ93VC_RATES,
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.formats = CQ93VC_FORMATS,},
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.capture = {
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.stream_name = "Capture",
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.channels_min = 1,
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.channels_max = 2,
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.rates = CQ93VC_RATES,
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.formats = CQ93VC_FORMATS,},
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.ops = &cq93vc_dai_ops,
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};
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static int cq93vc_resume(struct snd_soc_codec *codec)
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{
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cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
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return 0;
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}
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static int cq93vc_probe(struct snd_soc_codec *codec)
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{
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struct davinci_vc *davinci_vc = codec->dev->platform_data;
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davinci_vc->cq93vc.codec = codec;
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codec->control_data = davinci_vc;
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/* Set controls */
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snd_soc_add_controls(codec, cq93vc_snd_controls,
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ARRAY_SIZE(cq93vc_snd_controls));
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/* Off, with power on */
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cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
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return 0;
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}
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static int cq93vc_remove(struct snd_soc_codec *codec)
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{
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cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF);
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return 0;
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}
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static struct snd_soc_codec_driver soc_codec_dev_cq93vc = {
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.read = cq93vc_read,
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.write = cq93vc_write,
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.set_bias_level = cq93vc_set_bias_level,
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.probe = cq93vc_probe,
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.remove = cq93vc_remove,
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.resume = cq93vc_resume,
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};
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static int cq93vc_platform_probe(struct platform_device *pdev)
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{
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return snd_soc_register_codec(&pdev->dev,
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&soc_codec_dev_cq93vc, &cq93vc_dai, 1);
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}
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static int cq93vc_platform_remove(struct platform_device *pdev)
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{
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snd_soc_unregister_codec(&pdev->dev);
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return 0;
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}
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static struct platform_driver cq93vc_codec_driver = {
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.driver = {
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.name = "cq93vc-codec",
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.owner = THIS_MODULE,
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},
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.probe = cq93vc_platform_probe,
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.remove = __devexit_p(cq93vc_platform_remove),
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};
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static int __init cq93vc_init(void)
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{
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return platform_driver_register(&cq93vc_codec_driver);
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}
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module_init(cq93vc_init);
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static void __exit cq93vc_exit(void)
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{
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platform_driver_unregister(&cq93vc_codec_driver);
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}
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module_exit(cq93vc_exit);
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MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver");
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MODULE_AUTHOR("Miguel Aguilar");
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MODULE_LICENSE("GPL");
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