]> www.pilppa.org Git - linux-2.6-omap-h63xx.git/blobdiff - sound/soc/pxa/magician.c
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
[linux-2.6-omap-h63xx.git] / sound / soc / pxa / magician.c
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644 (file)
index 0000000..f7c4544
--- /dev/null
@@ -0,0 +1,560 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC       0
+#define MAGICIAN_MIC_EXT   1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+       if (magician_spk_switch)
+               snd_soc_dapm_enable_pin(codec, "Speaker");
+       else
+               snd_soc_dapm_disable_pin(codec, "Speaker");
+       if (magician_hp_switch)
+               snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+       else
+               snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+       switch (magician_in_sel) {
+       case MAGICIAN_MIC:
+               snd_soc_dapm_disable_pin(codec, "Headset Mic");
+               snd_soc_dapm_enable_pin(codec, "Call Mic");
+               break;
+       case MAGICIAN_MIC_EXT:
+               snd_soc_dapm_disable_pin(codec, "Call Mic");
+               snd_soc_dapm_enable_pin(codec, "Headset Mic");
+               break;
+       }
+
+       snd_soc_dapm_sync(codec);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+       struct snd_soc_pcm_runtime *rtd = substream->private_data;
+       struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+       /* check the jack status at stream startup */
+       magician_ext_control(codec);
+
+       return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_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 acps, acds, width, rate;
+       unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+       int ret = 0;
+
+       rate = params_rate(params);
+       width = snd_pcm_format_physical_width(params_format(params));
+
+       /*
+        * rate = SSPSCLK / (2 * width(16 or 32))
+        * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+        */
+       switch (params_rate(params)) {
+       case 8000:
+               /* off by a factor of 2: bug in the PXA27x audio clock? */
+               acps = 32842000;
+               switch (width) {
+               case 16:
+                       /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_16;
+                       break;
+               case 32:
+                       /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_8;
+               }
+               break;
+       case 11025:
+               acps = 5622000;
+               switch (width) {
+               case 16:
+                       /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_4;
+                       break;
+               case 32:
+                       /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_2;
+               }
+               break;
+       case 22050:
+               acps = 5622000;
+               switch (width) {
+               case 16:
+                       /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                       break;
+               case 32:
+                       /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_1;
+               }
+               break;
+       case 44100:
+               acps = 5622000;
+               switch (width) {
+               case 16:
+                       /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                       break;
+               case 32:
+                       /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_1;
+               }
+               break;
+       case 48000:
+               acps = 12235000;
+               switch (width) {
+               case 16:
+                       /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                       break;
+               case 32:
+                       /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_1;
+               }
+               break;
+       case 96000:
+               acps = 12235000;
+               switch (width) {
+               case 16:
+                       /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_1;
+                       break;
+               case 32:
+                       /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+                       acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                       div4 = PXA_SSP_CLK_SCDB_1;
+                       break;
+               }
+               break;
+       }
+
+       /* set codec DAI configuration */
+       ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+                       SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+       if (ret < 0)
+               return ret;
+
+       /* set cpu DAI configuration */
+       ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+                       SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
+       if (ret < 0)
+               return ret;
+
+       ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+       if (ret < 0)
+               return ret;
+
+       /* set audio clock as clock source */
+       ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+                       SND_SOC_CLOCK_OUT);
+       if (ret < 0)
+               return ret;
+
+       /* set the SSP audio system clock ACDS divider */
+       ret = snd_soc_dai_set_clkdiv(cpu_dai,
+                       PXA_SSP_AUDIO_DIV_ACDS, acds);
+       if (ret < 0)
+               return ret;
+
+       /* set the SSP audio system clock SCDB divider4 */
+       ret = snd_soc_dai_set_clkdiv(cpu_dai,
+                       PXA_SSP_AUDIO_DIV_SCDB, div4);
+       if (ret < 0)
+               return ret;
+
+       /* set SSP audio pll clock */
+       ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+       if (ret < 0)
+               return ret;
+
+       return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_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;
+       int ret = 0;
+
+       /* set codec DAI configuration */
+       ret = snd_soc_dai_set_fmt(codec_dai,
+                       SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+                       SND_SOC_DAIFMT_CBS_CFS);
+       if (ret < 0)
+               return ret;
+
+       /* set cpu DAI configuration */
+       ret = snd_soc_dai_set_fmt(cpu_dai,
+                       SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+                       SND_SOC_DAIFMT_CBS_CFS);
+       if (ret < 0)
+               return ret;
+
+       /* set the I2S system clock as output */
+       ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+                       SND_SOC_CLOCK_OUT);
+       if (ret < 0)
+               return ret;
+
+       return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+       .startup = magician_startup,
+       .hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+       .startup = magician_startup,
+       .hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+       ucontrol->value.integer.value[0] = magician_hp_switch;
+       return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+       struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+       if (magician_hp_switch == ucontrol->value.integer.value[0])
+               return 0;
+
+       magician_hp_switch = ucontrol->value.integer.value[0];
+       magician_ext_control(codec);
+       return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
+{
+       ucontrol->value.integer.value[0] = magician_spk_switch;
+       return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
+{
+       struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+       if (magician_spk_switch == ucontrol->value.integer.value[0])
+               return 0;
+
+       magician_spk_switch = ucontrol->value.integer.value[0];
+       magician_ext_control(codec);
+       return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
+{
+       ucontrol->value.integer.value[0] = magician_in_sel;
+       return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
+{
+       if (magician_in_sel == ucontrol->value.integer.value[0])
+               return 0;
+
+       magician_in_sel = ucontrol->value.integer.value[0];
+
+       switch (magician_in_sel) {
+       case MAGICIAN_MIC:
+               gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+               break;
+       case MAGICIAN_MIC_EXT:
+               gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+       }
+
+       return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+                               struct snd_kcontrol *k, int event)
+{
+       gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+       return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+                               struct snd_kcontrol *k, int event)
+{
+       gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+       return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+                               struct snd_kcontrol *k, int event)
+{
+       gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+       return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+       SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+       SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+       SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+       SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+       /* Headphone connected to VOUTL, VOUTR */
+       {"Headphone Jack", NULL, "VOUTL"},
+       {"Headphone Jack", NULL, "VOUTR"},
+
+       /* Speaker connected to VOUTL, VOUTR */
+       {"Speaker", NULL, "VOUTL"},
+       {"Speaker", NULL, "VOUTR"},
+
+       /* Mics are connected to VINM */
+       {"VINM", NULL, "Headset Mic"},
+       {"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+       SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+       SOC_SINGLE_BOOL_EXT("Headphone Switch",
+                       (unsigned long)&magician_hp_switch,
+                       magician_get_hp, magician_set_hp),
+       SOC_SINGLE_BOOL_EXT("Speaker Switch",
+                       (unsigned long)&magician_spk_switch,
+                       magician_get_spk, magician_set_spk),
+       SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+                       magician_get_input, magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_codec *codec)
+{
+       int err;
+
+       /* NC codec pins */
+       snd_soc_dapm_nc_pin(codec, "VOUTLHP");
+       snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+
+       /* FIXME: is anything connected here? */
+       snd_soc_dapm_nc_pin(codec, "VINL");
+       snd_soc_dapm_nc_pin(codec, "VINR");
+
+       /* Add magician specific controls */
+       err = snd_soc_add_controls(codec, uda1380_magician_controls,
+                               ARRAY_SIZE(uda1380_magician_controls));
+       if (err < 0)
+               return err;
+
+       /* Add magician specific widgets */
+       snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+                                 ARRAY_SIZE(uda1380_dapm_widgets));
+
+       /* Set up magician specific audio path interconnects */
+       snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+       snd_soc_dapm_sync(codec);
+       return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+       .name = "uda1380",
+       .stream_name = "UDA1380 Playback",
+       .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
+       .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+       .init = magician_uda1380_init,
+       .ops = &magician_playback_ops,
+},
+{
+       .name = "uda1380",
+       .stream_name = "UDA1380 Capture",
+       .cpu_dai = &pxa_i2s_dai,
+       .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+       .ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+       .name = "Magician",
+       .dai_link = magician_dai,
+       .num_links = ARRAY_SIZE(magician_dai),
+       .platform = &pxa2xx_soc_platform,
+};
+
+/* magician audio private data */
+static struct uda1380_setup_data magician_uda1380_setup = {
+       .i2c_address = 0x18,
+       .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+/* magician audio subsystem */
+static struct snd_soc_device magician_snd_devdata = {
+       .card = &snd_soc_card_magician,
+       .codec_dev = &soc_codec_dev_uda1380,
+       .codec_data = &magician_uda1380_setup,
+};
+
+static struct platform_device *magician_snd_device;
+
+static int __init magician_init(void)
+{
+       int ret;
+
+       if (!machine_is_magician())
+               return -ENODEV;
+
+       ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
+       if (ret)
+               goto err_request_power;
+       ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
+       if (ret)
+               goto err_request_reset;
+       ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+       if (ret)
+               goto err_request_spk;
+       ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+       if (ret)
+               goto err_request_ep;
+       ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+       if (ret)
+               goto err_request_mic;
+       ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+       if (ret)
+               goto err_request_in_sel0;
+       ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+       if (ret)
+               goto err_request_in_sel1;
+
+       gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
+       gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+       /* we may need to have the clock running here - pH5 */
+       gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
+       udelay(5);
+       gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
+
+       magician_snd_device = platform_device_alloc("soc-audio", -1);
+       if (!magician_snd_device) {
+               ret = -ENOMEM;
+               goto err_pdev;
+       }
+
+       platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
+       magician_snd_devdata.dev = &magician_snd_device->dev;
+       ret = platform_device_add(magician_snd_device);
+       if (ret) {
+               platform_device_put(magician_snd_device);
+               goto err_pdev;
+       }
+
+       return 0;
+
+err_pdev:
+       gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+       gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+       gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+       gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+       gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+       gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+err_request_reset:
+       gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+err_request_power:
+       return ret;
+}
+
+static void __exit magician_exit(void)
+{
+       platform_device_unregister(magician_snd_device);
+
+       gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+       gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+       gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+       gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
+
+       gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+       gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+       gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+       gpio_free(EGPIO_MAGICIAN_EP_POWER);
+       gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+       gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+       gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");