mirror of
git://projects.qi-hardware.com/openwrt-xburst.git
synced 2024-09-12 09:43:40 +03:00
git-svn-id: svn://svn.openwrt.org/openwrt/trunk@13619 3c298f89-4303-0410-b956-a3cf2f4a3e73
779 lines
21 KiB
Diff
Executable File
779 lines
21 KiB
Diff
Executable File
From c03e9f46b9f15afc3e8980ae28666ff30907b173 Mon Sep 17 00:00:00 2001
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From: mokopatches <mokopatches@openmoko.org>
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Date: Wed, 16 Jul 2008 14:46:56 +0100
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Subject: [PATCH] gta02-sound.patch
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---
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include/sound/soc-dapm.h | 7 +
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sound/soc/s3c24xx/Kconfig | 9 +
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sound/soc/s3c24xx/Makefile | 3 +
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sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 667 ++++++++++++++++++++++++++++++
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sound/soc/soc-dapm.c | 24 ++
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5 files changed, 710 insertions(+), 0 deletions(-)
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create mode 100644 sound/soc/s3c24xx/neo1973_gta02_wm8753.c
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diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
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index a105b01..e867447 100644
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--- a/include/sound/soc-dapm.h
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+++ b/include/sound/soc-dapm.h
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@@ -225,6 +225,13 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
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char *pin, int status);
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int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
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+/* dapm audio endpoint control */
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+int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
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+ char *pin, int status);
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+int snd_soc_dapm_get_endpoint(struct snd_soc_codec *codec,
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+ char *pin);
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+int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
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+
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/* dapm widget types */
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enum snd_soc_dapm_type {
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snd_soc_dapm_input = 0, /* input pin */
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diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
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index 1f6dbfc..3155c43 100644
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--- a/sound/soc/s3c24xx/Kconfig
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+++ b/sound/soc/s3c24xx/Kconfig
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@@ -28,6 +28,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
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Say Y if you want to add support for SoC audio on smdk2440
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with the WM8753.
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+config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
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+ tristate "SoC I2S Audio support for NEO1973 GTA02 - WM8753"
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+ depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
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+ select SND_S3C24XX_SOC_I2S
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+ select SND_SOC_WM8753
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+ help
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+ Say Y if you want to add support for SoC audio on neo1973 gta02
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+ with the WM8753 codec
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+
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config SND_S3C24XX_SOC_SMDK2443_WM9710
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tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
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depends on SND_S3C24XX_SOC && MACH_SMDK2443
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diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
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index 0aa5fb0..f154cb1 100644
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--- a/sound/soc/s3c24xx/Makefile
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+++ b/sound/soc/s3c24xx/Makefile
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@@ -13,7 +13,10 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
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snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
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snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
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snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
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+snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
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obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
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obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
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obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
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+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
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+
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diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
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new file mode 100644
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index 0000000..f32cba3
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--- /dev/null
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+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
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@@ -0,0 +1,667 @@
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+/*
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+ * neo1973_gta02_wm8753.c -- SoC audio for Neo1973
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+ *
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+ * Copyright 2007 OpenMoko Inc
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+ * Author: Graeme Gregory <graeme@openmoko.org>
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+ * Copyright 2007 Wolfson Microelectronics PLC.
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+ * Author: Graeme Gregory <linux@wolfsonmicro.com>
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+ *
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+ * This program is free software; you can redistribute it and/or modify it
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+ * under the terms of the GNU General Public License as published by the
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+ * Free Software Foundation; either version 2 of the License, or (at your
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+ * option) any later version.
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+ *
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+ * Revision history
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+ * 06th Nov 2007 Changed from GTA01 to GTA02
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+ * 20th Jan 2007 Initial version.
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+ * 05th Feb 2007 Rename all to Neo1973
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+ *
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+ */
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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/timer.h>
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+#include <linux/interrupt.h>
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+#include <linux/platform_device.h>
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+#include <linux/i2c.h>
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+#include <sound/driver.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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+
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+#include <asm/mach-types.h>
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+#include <asm/hardware/scoop.h>
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+#include <asm/plat-s3c24xx/regs-iis.h>
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+#include <asm/arch/regs-clock.h>
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+#include <asm/arch/regs-gpio.h>
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+#include <asm/hardware.h>
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+#include <asm/arch/audio.h>
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+#include <asm/io.h>
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+#include <asm/arch/spi-gpio.h>
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+#include <asm/arch/regs-gpioj.h>
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+#include <asm/arch/gta02.h>
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+#include "../codecs/wm8753.h"
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+#include "s3c24xx-pcm.h"
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+#include "s3c24xx-i2s.h"
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+
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+/* define the scenarios */
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+#define NEO_AUDIO_OFF 0
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+#define NEO_GSM_CALL_AUDIO_HANDSET 1
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+#define NEO_GSM_CALL_AUDIO_HEADSET 2
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+#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
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+#define NEO_STEREO_TO_SPEAKERS 4
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+#define NEO_STEREO_TO_HEADPHONES 5
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+#define NEO_CAPTURE_HANDSET 6
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+#define NEO_CAPTURE_HEADSET 7
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+#define NEO_CAPTURE_BLUETOOTH 8
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+#define NEO_STEREO_TO_HANDSET_SPK 9
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+
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+static struct snd_soc_machine neo1973_gta02;
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+
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+static int neo1973_gta02_hifi_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_codec_dai *codec_dai = rtd->dai->codec_dai;
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+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
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+ unsigned int pll_out = 0, bclk = 0;
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+ int ret = 0;
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+ unsigned long iis_clkrate;
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+
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+ iis_clkrate = s3c24xx_i2s_get_clockrate();
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+
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+ switch (params_rate(params)) {
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+ case 8000:
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+ case 16000:
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+ pll_out = 12288000;
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+ break;
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+ case 48000:
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+ bclk = WM8753_BCLK_DIV_4;
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+ pll_out = 12288000;
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+ break;
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+ case 96000:
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+ bclk = WM8753_BCLK_DIV_2;
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+ pll_out = 12288000;
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+ break;
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+ case 11025:
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+ bclk = WM8753_BCLK_DIV_16;
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+ pll_out = 11289600;
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+ break;
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+ case 22050:
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+ bclk = WM8753_BCLK_DIV_8;
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+ pll_out = 11289600;
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+ break;
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+ case 44100:
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+ bclk = WM8753_BCLK_DIV_4;
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+ pll_out = 11289600;
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+ break;
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+ case 88200:
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+ bclk = WM8753_BCLK_DIV_2;
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+ pll_out = 11289600;
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+ break;
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+ }
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+
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+ /* set codec DAI configuration */
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+ ret = codec_dai->dai_ops.set_fmt(codec_dai,
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+ SND_SOC_DAIFMT_I2S | 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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+ return ret;
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+
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+ /* set cpu DAI configuration */
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+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
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+ SND_SOC_DAIFMT_I2S | 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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+ return ret;
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+
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+ /* set the codec system clock for DAC and ADC */
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+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
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+ SND_SOC_CLOCK_IN);
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+ if (ret < 0)
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+ return ret;
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+
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+ /* set MCLK division for sample rate */
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+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
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+ S3C2410_IISMOD_32FS );
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+ if (ret < 0)
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+ return ret;
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+
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+ /* set codec BCLK division for sample rate */
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+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai,
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+ WM8753_BCLKDIV, bclk);
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+ if (ret < 0)
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+ return ret;
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+
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+ /* set prescaler division for sample rate */
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+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
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+ S3C24XX_PRESCALE(4,4));
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+ if (ret < 0)
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+ return ret;
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+
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+ /* codec PLL input is PCLK/4 */
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+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1,
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+ iis_clkrate / 4, pll_out);
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+ if (ret < 0)
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+ return ret;
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+
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+ return 0;
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+}
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+
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+static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
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+{
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+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
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+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
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+
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+ /* disable the PLL */
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+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
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+}
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+
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+/*
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+ * Neo1973 WM8753 HiFi DAI opserations.
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+ */
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+static struct snd_soc_ops neo1973_gta02_hifi_ops = {
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+ .hw_params = neo1973_gta02_hifi_hw_params,
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+ .hw_free = neo1973_gta02_hifi_hw_free,
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+};
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+
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+static int neo1973_gta02_voice_hw_params(
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+ 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_codec_dai *codec_dai = rtd->dai->codec_dai;
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+ unsigned int pcmdiv = 0;
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+ int ret = 0;
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+ unsigned long iis_clkrate;
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+
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+ iis_clkrate = s3c24xx_i2s_get_clockrate();
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+
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+ if (params_rate(params) != 8000)
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+ return -EINVAL;
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+ if (params_channels(params) != 1)
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+ return -EINVAL;
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+
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+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
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+
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+ /* todo: gg check mode (DSP_B) against CSR datasheet */
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+ /* set codec DAI configuration */
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+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
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+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
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+ if (ret < 0)
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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 = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK,
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+ 12288000, SND_SOC_CLOCK_IN);
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+ if (ret < 0)
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+ return ret;
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+
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+ /* set codec PCM division for sample rate */
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+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV,
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+ pcmdiv);
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+ if (ret < 0)
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+ return ret;
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+
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+ /* configue and enable PLL for 12.288MHz output */
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+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2,
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+ iis_clkrate / 4, 12288000);
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+ if (ret < 0)
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+ return ret;
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+
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+ return 0;
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+}
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+
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+static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
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+{
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+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
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+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
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+
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+ /* disable the PLL */
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+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
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+}
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+
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+static struct snd_soc_ops neo1973_gta02_voice_ops = {
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+ .hw_params = neo1973_gta02_voice_hw_params,
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+ .hw_free = neo1973_gta02_voice_hw_free,
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+};
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+
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+#define LM4853_AMP 1
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+#define LM4853_SPK 2
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+
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+static u8 lm4853_state=0;
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+
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+static int lm4853_set_state(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ int val = ucontrol->value.integer.value[0];
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+
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+ if(val) {
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+ lm4853_state |= LM4853_AMP;
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+ s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT,0);
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+ } else {
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+ lm4853_state &= ~LM4853_AMP;
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+ s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT,1);
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+ }
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+
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+ return 0;
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+}
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+
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+static int lm4853_get_state(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
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+
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+ return 0;
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+}
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+
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+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ int val = ucontrol->value.integer.value[0];
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+
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+ if(val) {
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+ lm4853_state |= LM4853_SPK;
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+ s3c2410_gpio_setpin(GTA02_GPIO_HP_IN,0);
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+ } else {
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+ lm4853_state &= ~LM4853_SPK;
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+ s3c2410_gpio_setpin(GTA02_GPIO_HP_IN,1);
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+ }
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+
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+ return 0;
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+}
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+
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+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
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+
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+ return 0;
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+}
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+
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+static int neo1973_gta02_set_stereo_out(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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+ int val = ucontrol->value.integer.value[0];
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+
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+ snd_soc_dapm_set_endpoint(codec, "Stereo Out", val);
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+
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+ snd_soc_dapm_sync_endpoints(codec);
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+
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+ return 0;
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+}
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+
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+static int neo1973_gta02_get_stereo_out(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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+
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+ ucontrol->value.integer.value[0] =
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+ snd_soc_dapm_get_endpoint(codec, "Stereo Out");
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+
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+ return 0;
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+}
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+
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+
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+static int neo1973_gta02_set_gsm_out(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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+ int val = ucontrol->value.integer.value[0];
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+
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+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", val);
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+
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+ snd_soc_dapm_sync_endpoints(codec);
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+
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+ return 0;
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+}
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+
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+static int neo1973_gta02_get_gsm_out(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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+
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+ ucontrol->value.integer.value[0] =
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+ snd_soc_dapm_get_endpoint(codec, "GSM Line Out");
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+
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+ return 0;
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+}
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+
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+static int neo1973_gta02_set_gsm_in(struct snd_kcontrol *kcontrol,
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+ struct snd_ctl_elem_value *ucontrol)
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+{
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+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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+ int val = ucontrol->value.integer.value[0];
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+
|
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+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", val);
|
|
+
|
|
+ snd_soc_dapm_sync_endpoints(codec);
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_get_gsm_in(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+
|
|
+ ucontrol->value.integer.value[0] =
|
|
+ snd_soc_dapm_get_endpoint(codec, "GSM Line In");
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_set_headset_mic(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+ int val = ucontrol->value.integer.value[0];
|
|
+
|
|
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", val);
|
|
+
|
|
+ snd_soc_dapm_sync_endpoints(codec);
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_get_headset_mic(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+
|
|
+ ucontrol->value.integer.value[0] =
|
|
+ snd_soc_dapm_get_endpoint(codec, "Headset Mic");
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_set_handset_mic(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+ int val = ucontrol->value.integer.value[0];
|
|
+
|
|
+ snd_soc_dapm_set_endpoint(codec, "Handset Mic", val);
|
|
+
|
|
+ snd_soc_dapm_sync_endpoints(codec);
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_get_handset_mic(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+
|
|
+ ucontrol->value.integer.value[0] =
|
|
+ snd_soc_dapm_get_endpoint(codec, "Handset Mic");
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_set_handset_spk(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+ int val = ucontrol->value.integer.value[0];
|
|
+
|
|
+ snd_soc_dapm_set_endpoint(codec, "Handset Spk", val);
|
|
+
|
|
+ snd_soc_dapm_sync_endpoints(codec);
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static int neo1973_gta02_get_handset_spk(struct snd_kcontrol *kcontrol,
|
|
+ struct snd_ctl_elem_value *ucontrol)
|
|
+{
|
|
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
+
|
|
+ ucontrol->value.integer.value[0] =
|
|
+ snd_soc_dapm_get_endpoint(codec, "Handset Spk");
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
|
|
+ SND_SOC_DAPM_LINE("Stereo Out", NULL),
|
|
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
|
|
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
|
|
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
|
|
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
|
|
+ SND_SOC_DAPM_SPK("Handset Spk", NULL),
|
|
+};
|
|
+
|
|
+
|
|
+/* example machine audio_mapnections */
|
|
+static const char* audio_map[][3] = {
|
|
+
|
|
+ /* Connections to the lm4853 amp */
|
|
+ {"Stereo Out", NULL, "LOUT1"},
|
|
+ {"Stereo Out", NULL, "ROUT1"},
|
|
+
|
|
+ /* Connections to the GSM Module */
|
|
+ {"GSM Line Out", NULL, "MONO1"},
|
|
+ {"GSM Line Out", NULL, "MONO2"},
|
|
+ {"RXP", NULL, "GSM Line In"},
|
|
+ {"RXN", NULL, "GSM Line In"},
|
|
+
|
|
+ /* Connections to Headset */
|
|
+ {"MIC1", NULL, "Mic Bias"},
|
|
+ {"Mic Bias", NULL, "Headset Mic"},
|
|
+
|
|
+ /* Call Mic */
|
|
+ {"MIC2", NULL, "Mic Bias"},
|
|
+ {"MIC2N", NULL, "Mic Bias"},
|
|
+ {"Mic Bias", NULL, "Handset Mic"},
|
|
+
|
|
+ /* Call Speaker */
|
|
+ {"Handset Spk", NULL, "LOUT2"},
|
|
+ {"Handset Spk", NULL, "ROUT2"},
|
|
+
|
|
+ /* Connect the ALC pins */
|
|
+ {"ACIN", NULL, "ACOP"},
|
|
+
|
|
+ {NULL, NULL, NULL},
|
|
+};
|
|
+
|
|
+static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
|
|
+ SOC_SINGLE_EXT("DAPM Stereo Out Switch", 0, 0, 1, 0,
|
|
+ neo1973_gta02_get_stereo_out,
|
|
+ neo1973_gta02_set_stereo_out),
|
|
+ SOC_SINGLE_EXT("DAPM GSM Line Out Switch", 1, 0, 1, 0,
|
|
+ neo1973_gta02_get_gsm_out,
|
|
+ neo1973_gta02_set_gsm_out),
|
|
+ SOC_SINGLE_EXT("DAPM GSM Line In Switch", 2, 0, 1, 0,
|
|
+ neo1973_gta02_get_gsm_in,
|
|
+ neo1973_gta02_set_gsm_in),
|
|
+ SOC_SINGLE_EXT("DAPM Headset Mic Switch", 3, 0, 1, 0,
|
|
+ neo1973_gta02_get_headset_mic,
|
|
+ neo1973_gta02_set_headset_mic),
|
|
+ SOC_SINGLE_EXT("DAPM Handset Mic Switch", 4, 0, 1, 0,
|
|
+ neo1973_gta02_get_handset_mic,
|
|
+ neo1973_gta02_set_handset_mic),
|
|
+ SOC_SINGLE_EXT("DAPM Handset Spk Switch", 5, 0, 1, 0,
|
|
+ neo1973_gta02_get_handset_spk,
|
|
+ neo1973_gta02_set_handset_spk),
|
|
+ SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
|
|
+ lm4853_get_state,
|
|
+ lm4853_set_state),
|
|
+ SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
|
|
+ lm4853_get_spk,
|
|
+ lm4853_set_spk),
|
|
+};
|
|
+
|
|
+/*
|
|
+ * This is an example machine initialisation for a wm8753 connected to a
|
|
+ * neo1973 GTA02.
|
|
+ */
|
|
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
|
|
+{
|
|
+ int i, err;
|
|
+
|
|
+ /* set up NC codec pins */
|
|
+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
|
|
+
|
|
+ /* Add neo1973 gta02 specific widgets */
|
|
+ for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
|
|
+ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
|
|
+
|
|
+ /* add neo1973 gta02 specific controls */
|
|
+ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_gta02_controls); i++) {
|
|
+ err = snd_ctl_add(codec->card,
|
|
+ snd_soc_cnew(&wm8753_neo1973_gta02_controls[i],
|
|
+ codec, NULL));
|
|
+ if (err < 0)
|
|
+ return err;
|
|
+ }
|
|
+
|
|
+ /* set up neo1973 gta02 specific audio path audio_mapnects */
|
|
+ for (i = 0; audio_map[i][0] != NULL; i++) {
|
|
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
|
|
+ audio_map[i][1], audio_map[i][2]);
|
|
+ }
|
|
+
|
|
+ /* set endpoints to default off mode */
|
|
+ snd_soc_dapm_set_endpoint(codec, "Stereo Out", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out",0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "Handset Mic", 0);
|
|
+ snd_soc_dapm_set_endpoint(codec, "Handset Spk", 0);
|
|
+
|
|
+ snd_soc_dapm_sync_endpoints(codec);
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+/*
|
|
+ * BT Codec DAI
|
|
+ */
|
|
+static struct snd_soc_cpu_dai bt_dai =
|
|
+{ .name = "Bluetooth",
|
|
+ .id = 0,
|
|
+ .type = SND_SOC_DAI_PCM,
|
|
+ .playback = {
|
|
+ .channels_min = 1,
|
|
+ .channels_max = 1,
|
|
+ .rates = SNDRV_PCM_RATE_8000,
|
|
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
|
|
+ .capture = {
|
|
+ .channels_min = 1,
|
|
+ .channels_max = 1,
|
|
+ .rates = SNDRV_PCM_RATE_8000,
|
|
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
|
|
+};
|
|
+
|
|
+static struct snd_soc_dai_link neo1973_gta02_dai[] = {
|
|
+{ /* Hifi Playback - for similatious use with voice below */
|
|
+ .name = "WM8753",
|
|
+ .stream_name = "WM8753 HiFi",
|
|
+ .cpu_dai = &s3c24xx_i2s_dai,
|
|
+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
|
|
+ .init = neo1973_gta02_wm8753_init,
|
|
+ .ops = &neo1973_gta02_hifi_ops,
|
|
+},
|
|
+{ /* Voice via BT */
|
|
+ .name = "Bluetooth",
|
|
+ .stream_name = "Voice",
|
|
+ .cpu_dai = &bt_dai,
|
|
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
|
|
+ .ops = &neo1973_gta02_voice_ops,
|
|
+},
|
|
+};
|
|
+
|
|
+#ifdef CONFIG_PM
|
|
+int neo1973_gta02_suspend(struct platform_device *pdev, pm_message_t state)
|
|
+{
|
|
+ s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 1);
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+
|
|
+int neo1973_gta02_resume(struct platform_device *pdev)
|
|
+{
|
|
+ if(lm4853_state & LM4853_AMP)
|
|
+ s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 0);
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+#else
|
|
+#define neo1973_gta02_suspend NULL
|
|
+#define neo1973_gta02_resume NULL
|
|
+#endif
|
|
+
|
|
+static struct snd_soc_machine neo1973_gta02 = {
|
|
+ .name = "neo1973-gta02",
|
|
+ .suspend_pre = neo1973_gta02_suspend,
|
|
+ .resume_post = neo1973_gta02_resume,
|
|
+ .dai_link = neo1973_gta02_dai,
|
|
+ .num_links = ARRAY_SIZE(neo1973_gta02_dai),
|
|
+};
|
|
+
|
|
+static struct wm8753_setup_data neo1973_gta02_wm8753_setup = {
|
|
+ .i2c_address = 0x1a,
|
|
+};
|
|
+
|
|
+static struct snd_soc_device neo1973_gta02_snd_devdata = {
|
|
+ .machine = &neo1973_gta02,
|
|
+ .platform = &s3c24xx_soc_platform,
|
|
+ .codec_dev = &soc_codec_dev_wm8753,
|
|
+ .codec_data = &neo1973_gta02_wm8753_setup,
|
|
+};
|
|
+
|
|
+static struct platform_device *neo1973_gta02_snd_device;
|
|
+
|
|
+static int __init neo1973_gta02_init(void)
|
|
+{
|
|
+ int ret;
|
|
+
|
|
+ if (!machine_is_neo1973_gta02()) {
|
|
+ printk(KERN_INFO
|
|
+ "Only GTA02 hardware supported by ASoc driver\n");
|
|
+ return -ENODEV;
|
|
+ }
|
|
+
|
|
+ neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
|
|
+ if (!neo1973_gta02_snd_device)
|
|
+ return -ENOMEM;
|
|
+
|
|
+ platform_set_drvdata(neo1973_gta02_snd_device,
|
|
+ &neo1973_gta02_snd_devdata);
|
|
+ neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev;
|
|
+ ret = platform_device_add(neo1973_gta02_snd_device);
|
|
+
|
|
+ if (ret)
|
|
+ platform_device_put(neo1973_gta02_snd_device);
|
|
+
|
|
+ /* Initialise GPIOs used by amp */
|
|
+ s3c2410_gpio_cfgpin(GTA02_GPIO_HP_IN, S3C2410_GPIO_OUTPUT);
|
|
+ s3c2410_gpio_cfgpin(GTA02_GPIO_AMP_SHUT, S3C2410_GPIO_OUTPUT);
|
|
+
|
|
+ /* Amp off by default */
|
|
+ s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 1);
|
|
+
|
|
+ /* Speaker off by default */
|
|
+ s3c2410_gpio_setpin(GTA02_GPIO_HP_IN, 1);
|
|
+
|
|
+ return ret;
|
|
+}
|
|
+
|
|
+static void __exit neo1973_gta02_exit(void)
|
|
+{
|
|
+ platform_device_unregister(neo1973_gta02_snd_device);
|
|
+}
|
|
+
|
|
+module_init(neo1973_gta02_init);
|
|
+module_exit(neo1973_gta02_exit);
|
|
+
|
|
+/* Module information */
|
|
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
|
|
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
|
|
+MODULE_LICENSE("GPL");
|
|
+
|
|
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
|
|
index af3326c..95df1ff 100644
|
|
--- a/sound/soc/soc-dapm.c
|
|
+++ b/sound/soc/soc-dapm.c
|
|
@@ -1343,6 +1343,30 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
|
|
EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
|
|
|
|
/**
|
|
+ * snd_soc_dapm_get_endpoint - get audio endpoint status
|
|
+ * @codec: audio codec
|
|
+ * @endpoint: audio signal endpoint (or start point)
|
|
+ *
|
|
+ * Get audio endpoint status - connected or disconnected.
|
|
+ *
|
|
+ * Returns status
|
|
+ */
|
|
+int snd_soc_dapm_get_endpoint(struct snd_soc_codec *codec,
|
|
+ char *endpoint)
|
|
+{
|
|
+ struct snd_soc_dapm_widget *w;
|
|
+
|
|
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
|
|
+ if (!strcmp(w->name, endpoint)) {
|
|
+ return w->connected;
|
|
+ }
|
|
+ }
|
|
+
|
|
+ return 0;
|
|
+}
|
|
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint);
|
|
+
|
|
+/**
|
|
* snd_soc_dapm_free - free dapm resources
|
|
* @socdev: SoC device
|
|
*
|
|
--
|
|
1.5.6.3
|
|
|