forked from luck/tmp_suning_uos_patched
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>
294 lines
7.6 KiB
C
294 lines
7.6 KiB
C
/*
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* zoom2.c -- SoC audio for Zoom2
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*
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* Author: Misael Lopez Cruz <x0052729@ti.com>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* 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., 51 Franklin St, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*
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*/
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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 <asm/mach-types.h>
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#include <mach/hardware.h>
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#include <mach/gpio.h>
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#include <mach/board-zoom.h>
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#include <plat/mcbsp.h>
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/* Register descriptions for twl4030 codec part */
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#include <linux/mfd/twl4030-codec.h>
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#include "omap-mcbsp.h"
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#include "omap-pcm.h"
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#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15)
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static int zoom2_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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int ret;
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/* Set codec DAI configuration */
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ret = snd_soc_dai_set_fmt(codec_dai,
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SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (ret < 0) {
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printk(KERN_ERR "can't set codec DAI configuration\n");
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return ret;
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}
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/* Set cpu DAI configuration */
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ret = snd_soc_dai_set_fmt(cpu_dai,
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SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (ret < 0) {
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printk(KERN_ERR "can't set cpu DAI configuration\n");
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return ret;
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}
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/* Set the codec system clock for DAC and ADC */
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ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
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SND_SOC_CLOCK_IN);
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if (ret < 0) {
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printk(KERN_ERR "can't set codec system clock\n");
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return ret;
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}
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return 0;
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}
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static struct snd_soc_ops zoom2_ops = {
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.hw_params = zoom2_hw_params,
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};
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static int zoom2_hw_voice_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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int ret;
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/* Set codec DAI configuration */
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ret = snd_soc_dai_set_fmt(codec_dai,
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SND_SOC_DAIFMT_DSP_A |
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SND_SOC_DAIFMT_IB_NF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (ret) {
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printk(KERN_ERR "can't set codec DAI configuration\n");
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return ret;
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}
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/* Set cpu DAI configuration */
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ret = snd_soc_dai_set_fmt(cpu_dai,
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SND_SOC_DAIFMT_DSP_A |
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SND_SOC_DAIFMT_IB_NF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (ret < 0) {
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printk(KERN_ERR "can't set cpu DAI configuration\n");
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return ret;
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}
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/* Set the codec system clock for DAC and ADC */
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ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
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SND_SOC_CLOCK_IN);
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if (ret < 0) {
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printk(KERN_ERR "can't set codec system clock\n");
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return ret;
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}
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return 0;
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}
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static struct snd_soc_ops zoom2_voice_ops = {
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.hw_params = zoom2_hw_voice_params,
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};
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/* Zoom2 machine DAPM */
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static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
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SND_SOC_DAPM_MIC("Ext Mic", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_HP("Headset Stereophone", NULL),
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SND_SOC_DAPM_LINE("Aux In", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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/* External Mics: MAINMIC, SUBMIC with bias*/
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{"MAINMIC", NULL, "Mic Bias 1"},
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{"SUBMIC", NULL, "Mic Bias 2"},
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{"Mic Bias 1", NULL, "Ext Mic"},
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{"Mic Bias 2", NULL, "Ext Mic"},
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/* External Speakers: HFL, HFR */
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{"Ext Spk", NULL, "HFL"},
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{"Ext Spk", NULL, "HFR"},
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/* Headset Stereophone: HSOL, HSOR */
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{"Headset Stereophone", NULL, "HSOL"},
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{"Headset Stereophone", NULL, "HSOR"},
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/* Headset Mic: HSMIC with bias */
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{"HSMIC", NULL, "Headset Mic Bias"},
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{"Headset Mic Bias", NULL, "Headset Mic"},
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/* Aux In: AUXL, AUXR */
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{"Aux In", NULL, "AUXL"},
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{"Aux In", NULL, "AUXR"},
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};
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static int zoom2_twl4030_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 ret;
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/* Add Zoom2 specific widgets */
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ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
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ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
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if (ret)
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return ret;
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/* Set up Zoom2 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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/* Zoom2 connected pins */
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snd_soc_dapm_enable_pin(codec, "Ext Mic");
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snd_soc_dapm_enable_pin(codec, "Ext Spk");
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snd_soc_dapm_enable_pin(codec, "Headset Mic");
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snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
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snd_soc_dapm_enable_pin(codec, "Aux In");
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/* TWL4030 not connected pins */
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snd_soc_dapm_nc_pin(codec, "CARKITMIC");
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snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
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snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
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snd_soc_dapm_nc_pin(codec, "EARPIECE");
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snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
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snd_soc_dapm_nc_pin(codec, "PREDRIVER");
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snd_soc_dapm_nc_pin(codec, "CARKITL");
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snd_soc_dapm_nc_pin(codec, "CARKITR");
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ret = snd_soc_dapm_sync(codec);
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return ret;
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}
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static int zoom2_twl4030_voice_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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unsigned short reg;
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/* Enable voice interface */
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reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
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reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
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codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg);
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return 0;
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}
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/* Digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link zoom2_dai[] = {
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{
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.name = "TWL4030 I2S",
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.stream_name = "TWL4030 Audio",
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.cpu_dai_name = "omap-mcbsp-dai.1",
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.codec_dai_name = "twl4030-hifi",
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.platform_name = "omap-pcm-audio",
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.codec_name = "twl4030-codec",
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.init = zoom2_twl4030_init,
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.ops = &zoom2_ops,
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},
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{
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.name = "TWL4030 PCM",
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.stream_name = "TWL4030 Voice",
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.cpu_dai_name = "omap-mcbsp-dai.2",
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.codec_dai_name = "twl4030-voice",
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.platform_name = "omap-pcm-audio",
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.codec_name = "twl4030-codec",
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.init = zoom2_twl4030_voice_init,
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.ops = &zoom2_voice_ops,
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},
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_zoom2 = {
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.name = "Zoom2",
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.dai_link = zoom2_dai,
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.num_links = ARRAY_SIZE(zoom2_dai),
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};
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static struct platform_device *zoom2_snd_device;
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static int __init zoom2_soc_init(void)
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{
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int ret;
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if (!machine_is_omap_zoom2()) {
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pr_debug("Not Zoom2!\n");
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return -ENODEV;
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}
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printk(KERN_INFO "Zoom2 SoC init\n");
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zoom2_snd_device = platform_device_alloc("soc-audio", -1);
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if (!zoom2_snd_device) {
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printk(KERN_ERR "Platform device allocation failed\n");
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return -ENOMEM;
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}
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platform_set_drvdata(zoom2_snd_device, &snd_soc_zoom2);
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ret = platform_device_add(zoom2_snd_device);
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if (ret)
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goto err1;
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BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
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gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
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BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
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gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
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return 0;
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err1:
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printk(KERN_ERR "Unable to add platform device\n");
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platform_device_put(zoom2_snd_device);
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return ret;
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}
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module_init(zoom2_soc_init);
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static void __exit zoom2_soc_exit(void)
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{
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gpio_free(ZOOM2_HEADSET_MUX_GPIO);
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gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
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platform_device_unregister(zoom2_snd_device);
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}
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module_exit(zoom2_soc_exit);
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MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
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MODULE_DESCRIPTION("ALSA SoC Zoom2");
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MODULE_LICENSE("GPL");
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