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Diffstat (limited to 'sound/soc/s3c24xx/om_gta03_wm8753.c')
-rw-r--r--sound/soc/s3c24xx/om_gta03_wm8753.c568
1 files changed, 568 insertions, 0 deletions
diff --git a/sound/soc/s3c24xx/om_gta03_wm8753.c b/sound/soc/s3c24xx/om_gta03_wm8753.c
new file mode 100644
index 00000000000..60879b8bc80
--- /dev/null
+++ b/sound/soc/s3c24xx/om_gta03_wm8753.c
@@ -0,0 +1,568 @@
+/*
+ * om_gta03_wm8753.c -- SoC audio for GTA03
+ *
+ * Based on neo1973_gta02_wm8753
+ *
+ * Copyright 2009 Openmoko Inc
+ * Author: Ben Dooks <ben@simtec.co.uk>
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory <linux@wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+
+#include <plat/regs-s3c2412-iis.h>
+
+#include "../codecs/wm8753.h"
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-i2s.h"
+
+/* define the scenarios */
+#define NEO_AUDIO_OFF 0
+#define NEO_GSM_CALL_AUDIO_HANDSET 1
+#define NEO_GSM_CALL_AUDIO_HEADSET 2
+#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
+#define NEO_STEREO_TO_SPEAKERS 4
+#define NEO_STEREO_TO_HEADPHONES 5
+#define NEO_CAPTURE_HANDSET 6
+#define NEO_CAPTURE_HEADSET 7
+#define NEO_CAPTURE_BLUETOOTH 8
+#define NEO_STEREO_TO_HANDSET_SPK 9
+
+static struct snd_soc_card om_gta03;
+
+static int om_gta03_hifi_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->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pll_out = 0, bclk = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ iis_clkrate = s3c64xx_i2s_get_clockrate(cpu_dai);
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ pll_out = 12288000;
+ break;
+ case 48000:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 12288000;
+ break;
+ case 96000:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 12288000;
+ break;
+ case 11025:
+ bclk = WM8753_BCLK_DIV_16;
+ pll_out = 11289600;
+ break;
+ case 22050:
+ bclk = WM8753_BCLK_DIV_8;
+ pll_out = 11289600;
+ break;
+ case 44100:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 11289600;
+ break;
+ case 88200:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+#if 0
+ /* do not think we need to set this if the cpu is not the bitclk
+ * master */
+ /* set MCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+#endif
+
+ /* set codec BCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C64XX_DIV_PRESCALER, 4);
+ if (ret < 0)
+ return ret;
+
+ /* codec PLL input is PCLK/4 */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ iis_clkrate / 4, pll_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int om_gta03_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * GTA03 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops om_gta03_hifi_ops = {
+ .hw_params = om_gta03_hifi_hw_params,
+ .hw_free = om_gta03_hifi_hw_free,
+};
+
+static int om_gta03_voice_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->dai->codec_dai;
+ unsigned int pcmdiv = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ iis_clkrate = s3c64xx_i2s_get_clockrate(rtd->dai->cpu_dai);
+
+ if (params_rate(params) != 8000)
+ return -EINVAL;
+ if (params_channels(params) != 1)
+ return -EINVAL;
+
+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
+ /* todo: gg check mode (DSP_B) against CSR datasheet */
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, (SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS));
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
+ 12288000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set codec PCM division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
+ if (ret < 0)
+ return ret;
+
+ /* configue and enable PLL for 12.288MHz output */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+ iis_clkrate / 4, 12288000);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int om_gta03_voice_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops om_gta03_voice_ops = {
+ .hw_params = om_gta03_voice_hw_params,
+ .hw_free = om_gta03_voice_hw_free,
+};
+
+static int om_gta03_set_stereo_out(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, "Stereo Out", val);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static int om_gta03_get_stereo_out(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, "Stereo Out");
+
+ return 0;
+}
+
+
+static int om_gta03_set_gsm_out(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, "GSM Line Out", val);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static int om_gta03_get_gsm_out(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 Out");
+
+ return 0;
+}
+
+static int om_gta03_set_gsm_in(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, "GSM Line In", val);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+ return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+ return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+ return 0;
+}
+
+static int om_gta03_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 struct snd_soc_dapm_route audio_map[] = {
+
+ {"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"},
+};
+
+static const struct snd_kcontrol_new wm8753_om_gta03_controls[] = {
+ SOC_SINGLE_EXT("DAPM Stereo Out Switch", 0, 0, 1, 0,
+ om_gta03_get_stereo_out,
+ om_gta03_set_stereo_out),
+ SOC_SINGLE_EXT("DAPM GSM Line Out Switch", 1, 0, 1, 0,
+ om_gta03_get_gsm_out,
+ om_gta03_set_gsm_out),
+ SOC_SINGLE_EXT("DAPM GSM Line In Switch", 2, 0, 1, 0,
+ om_gta03_get_gsm_in,
+ om_gta03_set_gsm_in),
+ SOC_SINGLE_EXT("DAPM Headset Mic Switch", 3, 0, 1, 0,
+ om_gta03_get_headset_mic,
+ om_gta03_set_headset_mic),
+ SOC_SINGLE_EXT("DAPM Handset Mic Switch", 4, 0, 1, 0,
+ om_gta03_get_handset_mic,
+ om_gta03_set_handset_mic),
+ SOC_SINGLE_EXT("DAPM Handset Spk Switch", 5, 0, 1, 0,
+ om_gta03_get_handset_spk,
+ om_gta03_set_handset_spk),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 GTA02.
+ */
+static int om_gta03_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 */
+ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ ARRAY_SIZE(wm8753_dapm_widgets));
+
+ /* add neo1973 gta02 specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8753_om_gta03_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8753_om_gta03_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* set up neo1973 gta02 specific audio path audio_mapnects */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* 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(codec);
+
+ return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_dai bt_dai =
+{ .name = "Bluetooth",
+ .id = 0,
+ .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 om_gta03_dai[] = {
+ { /* Hifi Playback - for similatious use with voice below */
+ .name = "WM8753",
+ .stream_name = "WM8753 HiFi",
+ .cpu_dai = &s3c64xx_i2s_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
+ .init = om_gta03_wm8753_init,
+ .ops = &om_gta03_hifi_ops,
+ },
+ { /* Voice via BT */
+ .name = "Bluetooth",
+ .stream_name = "Voice",
+ .cpu_dai = &bt_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+ .ops = &om_gta03_voice_ops,
+ },
+};
+
+static struct snd_soc_card om_gta03 = {
+ .name = "om-gta03",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = om_gta03_dai,
+ .num_links = ARRAY_SIZE(om_gta03_dai),
+};
+
+/* Audio private data */
+static struct wm8753_setup_data soc_codec_data_wm8753_gta02 = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1a,
+};
+
+static struct snd_soc_device om_gta03_snd_devdata = {
+ .card = &om_gta03,
+ .codec_dev = &soc_codec_dev_wm8753,
+ .codec_data = &soc_codec_data_wm8753_gta02,
+};
+
+static struct platform_device *om_gta03_snd_device;
+
+static int __init om_gta03_init(void)
+{
+ int ret;
+
+ if (!machine_is_openmoko_gta03()) {
+ printk(KERN_INFO "Only GTA03 supported by ASoC driver\n");
+ return -ENODEV;
+ }
+
+ /* register bluetooth DAI here */
+ ret = snd_soc_register_dai(&bt_dai);
+ if (ret)
+ return ret;
+
+ om_gta03_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!om_gta03_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(om_gta03_snd_device, &om_gta03_snd_devdata);
+ om_gta03_snd_devdata.dev = &om_gta03_snd_device->dev;
+ ret = platform_device_add(om_gta03_snd_device);
+
+ if (ret) {
+ platform_device_put(om_gta03_snd_device);
+ return ret;
+ }
+
+ return ret;
+}
+
+static void __exit om_gta03_exit(void)
+{
+ platform_device_unregister(om_gta03_snd_device);
+}
+
+module_init(om_gta03_init);
+module_exit(om_gta03_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org; Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC WM8753 OM GTA03");
+MODULE_LICENSE("GPL");
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-03 17:32:38 +0000