Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623',... (1f491557) · Commits · e / devices / android_kernel_fairphone_FP4

sound/soc/atmel/Kconfig

+0 −9

Original line number	Diff line number	Diff line
		@@ -52,12 +52,3 @@ config SND_AT91_SOC_SAM9X5_WM8731
		help
		Say Y if you want to add support for audio SoC on an
		at91sam9x5 based board that is using WM8731 codec.

		config SND_AT91_SOC_AFEB9260
		tristate "SoC Audio support for AFEB9260 board"
		depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
		select SND_ATMEL_SOC_PDC
		select SND_ATMEL_SOC_SSC
		select SND_SOC_TLV320AIC23_I2C
		help
		Say Y here to support sound on AFEB9260 board.

sound/soc/atmel/Makefile

+0 −1

Original line number	Diff line number	Diff line
		@@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
		obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
		obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
		obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
		obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o

sound/soc/atmel/atmel_ssc_dai.c

+4 −1

Original line number	Diff line number	Diff line
		@@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
		* transmit and receive, so if a value has already
		* been set, it must match this value.
		*/
		if (ssc_p->cmr_div == 0)
		if (ssc_p->dir_mask !=
		(SSC_DIR_MASK_PLAYBACK \| SSC_DIR_MASK_CAPTURE))
		ssc_p->cmr_div = div;
		else if (ssc_p->cmr_div == 0)
		ssc_p->cmr_div = div;
		else
		if (div != ssc_p->cmr_div)

sound/soc/atmel/snd-soc-afeb9260.c

deleted100644 → 0

+0 −151

Original line number	Diff line number	Diff line
		/*
		* afeb9260.c -- SoC audio for AFEB9260
		*
		* Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
		*
		* This program is free software; you can redistribute it and/or
		* modify it under the terms of the GNU General Public License
		* version 2 as published by the Free Software Foundation.
		*
		* This program is distributed in the hope that it will be useful, but
		* WITHOUT ANY WARRANTY; without even the implied warranty of
		* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
		* General Public License for more details.
		*
		* You should have received a copy of the GNU General Public License
		* along with this program; if not, write to the Free Software
		* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
		* 02110-1301 USA
		*
		*/

		#include <linux/module.h>
		#include <linux/moduleparam.h>
		#include <linux/kernel.h>
		#include <linux/clk.h>
		#include <linux/platform_device.h>

		#include <linux/atmel-ssc.h>
		#include <sound/core.h>
		#include <sound/pcm.h>
		#include <sound/pcm_params.h>
		#include <sound/soc.h>

		#include <asm/mach-types.h>
		#include <mach/hardware.h>
		#include <linux/gpio.h>

		#include "../codecs/tlv320aic23.h"
		#include "atmel-pcm.h"
		#include "atmel_ssc_dai.h"

		#define CODEC_CLOCK 12000000

		static int afeb9260_hw_params(struct snd_pcm_substream *substream,
		struct snd_pcm_hw_params *params)
		{
		struct snd_soc_pcm_runtime *rtd = substream->private_data;
		struct snd_soc_dai *codec_dai = rtd->codec_dai;
		int err;

		/* Set the codec system clock for DAC and ADC */
		err =
		snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);

		if (err < 0) {
		printk(KERN_ERR "can't set codec system clock\n");
		return err;
		}

		return err;
		}

		static struct snd_soc_ops afeb9260_ops = {
		.hw_params = afeb9260_hw_params,
		};

		static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
		SND_SOC_DAPM_HP("Headphone Jack", NULL),
		SND_SOC_DAPM_LINE("Line In", NULL),
		SND_SOC_DAPM_MIC("Mic Jack", NULL),
		};

		static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
		{"Headphone Jack", NULL, "LHPOUT"},
		{"Headphone Jack", NULL, "RHPOUT"},

		{"LLINEIN", NULL, "Line In"},
		{"RLINEIN", NULL, "Line In"},

		{"MICIN", NULL, "Mic Jack"},
		};


		/* Digital audio interface glue - connects codec <--> CPU */
		static struct snd_soc_dai_link afeb9260_dai = {
		.name = "TLV320AIC23",
		.stream_name = "AIC23",
		.cpu_dai_name = "atmel-ssc-dai.0",
		.codec_dai_name = "tlv320aic23-hifi",
		.platform_name = "atmel_pcm-audio",
		.codec_name = "tlv320aic23-codec.0-001a",
		.dai_fmt = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_IF \|
		SND_SOC_DAIFMT_CBM_CFM,
		.ops = &afeb9260_ops,
		};

		/* Audio machine driver */
		static struct snd_soc_card snd_soc_machine_afeb9260 = {
		.name = "AFEB9260",
		.owner = THIS_MODULE,
		.dai_link = &afeb9260_dai,
		.num_links = 1,

		.dapm_widgets = tlv320aic23_dapm_widgets,
		.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
		.dapm_routes = afeb9260_audio_map,
		.num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
		};

		static struct platform_device *afeb9260_snd_device;

		static int __init afeb9260_soc_init(void)
		{
		int err;
		struct device *dev;

		if (!(machine_is_afeb9260()))
		return -ENODEV;


		afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
		if (!afeb9260_snd_device) {
		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
		return -ENOMEM;
		}

		platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260);
		err = platform_device_add(afeb9260_snd_device);
		if (err)
		goto err1;

		dev = &afeb9260_snd_device->dev;

		return 0;
		err1:
		platform_device_put(afeb9260_snd_device);
		return err;
		}

		static void __exit afeb9260_soc_exit(void)
		{
		platform_device_unregister(afeb9260_snd_device);
		}

		module_init(afeb9260_soc_init);
		module_exit(afeb9260_soc_exit);

		MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
		MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
		MODULE_LICENSE("GPL");

sound/soc/codecs/ak4671.c

+0 −13

Original line number	Diff line number	Diff line
		@@ -611,20 +611,7 @@ static struct snd_soc_dai_driver ak4671_dai = {
		.ops = &ak4671_dai_ops,
		};

		static int ak4671_probe(struct snd_soc_codec *codec)
		{
		return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
		}

		static int ak4671_remove(struct snd_soc_codec *codec)
		{
		ak4671_set_bias_level(codec, SND_SOC_BIAS_OFF);
		return 0;
		}

		static struct snd_soc_codec_driver soc_codec_dev_ak4671 = {
		.probe = ak4671_probe,
		.remove = ak4671_remove,
		.set_bias_level = ak4671_set_bias_level,
		.controls = ak4671_snd_controls,
		.num_controls = ARRAY_SIZE(ak4671_snd_controls),