Re: [alsa-devel] [PATCH] Add ASoC Magician machine support.

At Tue, 20 Nov 2007 09:25:43 +0000, Mark Brown wrote:

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Philipp Zabel philipp.zabel@gmail.com Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com

Hmm... Some patches seem to have inconsistent From: and the author attribute. For exmaple, this module has

MODULE_AUTHOR("Philipp Zabel");

while Liam is in From header here (and it's a broken address :)

Otherwise it looks OK except for a few coding style issues. To be sure, try checkpatch.pl.

Takashi

---

sound/soc/pxa/Kconfig | 11 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 539 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 552 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/magician.c

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index bcb3aa0..3682f38 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -58,6 +58,17 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa).

+config SND_PXA2XX_SOC_MAGICIAN

• tristate "SoC Audio support for HTC Magician"
• depends on SND_PXA2XX_SOC
• select SND_PXA2XX_SOC_I2S
• select SND_PXA2XX_SOC_SSP
• select SND_SOC_UDA1380
• help

•  Say Y if you want to add support for SoC audio on the
  

•  HTC Magician.
  

config SND_PXA2XX_SOC_AMESOM_TLV320 tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k" depends on SND_PXA2XX_SOC && MACH_AMESOM diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 931bdc7..1faa751 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -15,9 +15,11 @@ snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-spitz-objs := spitz.o snd-soc-amesom-tlv320-objs := amesom_tlv320.o +snd-soc-magician-objs := magician.o

obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o \ No newline at end of file diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c new file mode 100644 index 0000000..7eb671c --- /dev/null +++ b/sound/soc/pxa/magician.c @@ -0,0 +1,539 @@ +/*

• • SoC audio for HTC Magician
• • Copyright (c) 2006 Philipp Zabel philipp.zabel@gmail.com
• • based on spitz.c,
• • Authors: Liam Girdwood liam.girdwood@wolfsonmicro.com

• •      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 <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h>

+#include <asm/hardware/scoop.h> +#include <asm/arch/pxa-regs.h> +#include <asm/arch/hardware.h> +#include <asm/arch/magician.h> +#include <asm/arch/magician_cpld.h> +#include <asm/mach-types.h> +#include "../codecs/uda1380.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-ssp.h"

+#define MAGICIAN_HP_ON 0 +#define MAGICIAN_HP_OFF 1

+#define MAGICIAN_SPK_ON 0 +#define MAGICIAN_SPK_OFF 1

+#define MAGICIAN_MIC 0 +#define MAGICIAN_MIC_EXT 1

+/*

• • SSP GPIO's
• */

+#define GPIO23_SSPSCLK_MD (23 | GPIO_ALT_FN_2_OUT) +#define GPIO24_SSPSFRM_MD (24 | GPIO_ALT_FN_2_OUT) +#define GPIO25_SSPTXD_MD (25 | GPIO_ALT_FN_2_OUT)

+static int magician_hp_func = MAGICIAN_HP_OFF; +static int magician_spk_func = MAGICIAN_SPK_ON; +static int magician_in_sel = MAGICIAN_MIC;

+extern struct platform_device magician_cpld;

+static void magician_ext_control(struct snd_soc_codec *codec) +{

• snd_soc_dapm_set_endpoint(codec, "Speaker",

• 	(magician_spk_func == MAGICIAN_SPK_ON));
  

• snd_soc_dapm_set_endpoint(codec, "Headphone Jack",

• 	(magician_hp_func == MAGICIAN_HP_ON));
  

• switch (magician_in_sel) {
• case MAGICIAN_MIC:

• snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
  

• snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
  

• break;
  

• case MAGICIAN_MIC_EXT:

• snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
  

• snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
  

• break;
  

• }
• snd_soc_dapm_sync_endpoints(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->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_codec_dai *codec_dai = rtd->dai->codec_dai;
• struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
• unsigned int acps, acds, div4;
• int ret = 0;
• /*

• * Rate = SSPSCLK / (word size(16))
  

• * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
  

• */
  

• switch (params_rate(params)) {
• case 8000:

• acps = 32842000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_32;	/* wrong - 32 bits/sample */
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 11025:

• acps = 5622000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_8;	/* 16 bits/sample, 1 slot */
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 22050:

• acps = 5622000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 44100:

• acps = 11345000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 48000:

• acps = 12235000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• }
• /* set codec DAI configuration */
• ret = codec_dai->dai_ops.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 = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |

• 	SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
  

• if (ret < 0)

• return ret;
  

• /* set audio clock as clock source */
• ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,

• 	SND_SOC_CLOCK_OUT);
  

• if (ret < 0)

• return ret;
  

• /* set the SSP audio system clock ACDS divider */
• ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,

• 	PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
  

• if (ret < 0)

• return ret;
  

• /* set the SSP audio system clock SCDB divider4 */
• ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,

• 	PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
  

• if (ret < 0)

• return ret;
  

• /* set SSP audio pll clock */
• ret = cpu_dai->dai_ops.set_pll(cpu_dai, 0, 0, acps);
• if (ret < 0)

• return ret;
  

• return 0;

+}

+/*

• • We have to enable the SSP port early so the UDA1380 can flush
• • it's register cache. The UDA1380 can only write it's interpolator and
• • decimator registers when the link is running.
• */

+static int magician_playback_prepare(struct snd_pcm_substream *substream) +{

• /* enable SSP clock - is this needed ? */
• SSCR0_P(1) |= SSCR0_SSE;
• /* FIXME: ENABLE I2S */
• SACR0 |= SACR0_BCKD;
• SACR0 |= SACR0_ENB;
• pxa_set_cken(CKEN8_I2S, 1);
• return 0;

+}

+static int magician_playback_hw_free(struct snd_pcm_substream *substream) +{

• /* FIXME: DISABLE I2S */
• SACR0 &= ~SACR0_ENB;
• SACR0 &= ~SACR0_BCKD;
• pxa_set_cken(CKEN8_I2S, 0);
• 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_codec_dai *codec_dai = rtd->dai->codec_dai;
• struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
• int ret = 0;
• /* set codec DAI configuration */
• ret = codec_dai->dai_ops.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 = cpu_dai->dai_ops.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 = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,

• 	SND_SOC_CLOCK_OUT);
  

• if (ret < 0)

• return ret;
  

• return 0;

+}

+/*

• • We have to enable the I2S port early so the UDA1380 can flush
• • it's register cache. The UDA1380 can only write it's interpolator and
• • decimator registers when the link is running.
• */

+static int magician_capture_prepare(struct snd_pcm_substream *substream) +{

• SACR0 |= SACR0_ENB;
• return 0;

+}

+static struct snd_soc_ops magician_capture_ops = {

• .startup = magician_startup,
• .hw_params = magician_capture_hw_params,
• .prepare = magician_capture_prepare,

+};

+static struct snd_soc_ops magician_playback_ops = {

• .startup = magician_startup,
• .hw_params = magician_playback_hw_params,
• .prepare = magician_playback_prepare,
• .hw_free = magician_playback_hw_free,

+};

+static int magician_get_jack(struct snd_kcontrol * kcontrol,

• 	     struct snd_ctl_elem_value * ucontrol)
  

+{

• ucontrol->value.integer.value[0] = magician_hp_func;
• 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_func == ucontrol->value.integer.value[0])

• return 0;
  

• magician_hp_func = 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_func;
• 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_func == ucontrol->value.integer.value[0])

• return 0;
  

• magician_spk_func = 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)
  

+{

• struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
• 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:

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_IN_SEL0);
  

• magician_egpio_enable(&magician_cpld,
  

• 		      EGPIO_NR_MAGICIAN_IN_SEL1);
  

• break;
  

• case MAGICIAN_MIC_EXT:

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_IN_SEL0);
  

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_IN_SEL1);
  

• }
• return 1;

+}

+static int magician_spk_power(struct snd_soc_dapm_widget *w, int event) +{

• if (SND_SOC_DAPM_EVENT_ON(event))

• magician_egpio_enable(&magician_cpld,
  

• 		      EGPIO_NR_MAGICIAN_SPK_POWER);
  

• else

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_SPK_POWER);
  

• return 0;

+}

+static int magician_hp_power(struct snd_soc_dapm_widget *w, int event) +{

• if (SND_SOC_DAPM_EVENT_ON(event))

• magician_egpio_enable(&magician_cpld,
  

• 		      EGPIO_NR_MAGICIAN_EP_POWER);
  

• else

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_EP_POWER);
  

• return 0;

+}

+static int magician_mic_bias(struct snd_soc_dapm_widget *w, int event) +{ +// if (SND_SOC_DAPM_EVENT_ON(event)) +// magician_egpio_enable(&magician_cpld, +// EGPIO_NR_MAGICIAN_MIC_POWER); +// else +// magician_egpio_disable(&magician_cpld, +// EGPIO_NR_MAGICIAN_MIC_POWER);

• 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 char *audio_map[][3] = {

• /* 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"},
• {NULL, NULL, NULL},

+};

+static const char *hp_function[] = { "On", "Off" }; +static const char *spk_function[] = { "On", "Off" }; +static const char *input_select[] = { "Call Mic", "Headset Mic" }; +static const struct soc_enum magician_enum[] = {

• SOC_ENUM_SINGLE_EXT(4, hp_function),
• SOC_ENUM_SINGLE_EXT(2, spk_function),
• SOC_ENUM_SINGLE_EXT(2, input_select),

+};

+static const struct snd_kcontrol_new uda1380_magician_controls[] = {

• SOC_ENUM_EXT("Headphone Switch", magician_enum[0], magician_get_jack,

• 	magician_set_hp),
  

• SOC_ENUM_EXT("Speaker Switch", magician_enum[1], magician_get_spk,

• 	magician_set_spk),
  

• SOC_ENUM_EXT("Input Select", magician_enum[2], 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 i, err;
• /* NC codec pins */
• snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
• snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
• /* FIXME: is anything connected here? */
• snd_soc_dapm_set_endpoint(codec, "VINL", 0);
• snd_soc_dapm_set_endpoint(codec, "VINR", 0);
• /* Add magician specific controls */
• for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {

• if ((err = snd_ctl_add(codec->card,
  

• 		snd_soc_cnew(&uda1380_magician_controls[i],
  

• 		codec, NULL))) < 0)
  

• 	return err;
  

• }
• /* Add magician specific widgets */
• for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {

• snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
  

• }
• /* Set up magician specific audio path interconnects */
• 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]);
  

• }
• snd_soc_dapm_sync_endpoints(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[0],
• .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_machine snd_soc_machine_magician = {

• .name = "Magician",
• .dai_link = magician_dai,
• .num_links = ARRAY_SIZE(magician_dai),

+};

+/* 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 = {

• .machine = &snd_soc_machine_magician,
• .platform = &pxa2xx_soc_platform,
• .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;
  

• magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
• /* we may need to have the clock running here - pH5 */
• magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
• udelay(5);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
• /* correct place? we'll need it to talk to the uda1380 */
• request_module("i2c-pxa");
• magician_snd_device = platform_device_alloc("soc-audio", -1);
• if (!magician_snd_device)

• return -ENOMEM;
  

• 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);
  

• pxa_gpio_mode(GPIO23_SSPSCLK_MD);
• pxa_gpio_mode(GPIO24_SSPSFRM_MD);
• pxa_gpio_mode(GPIO25_SSPTXD_MD);
• return ret;

+}

+static void __exit magician_exit(void) +{

• platform_device_unregister(magician_snd_device);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);

+}

+module_init(magician_init); +module_exit(magician_exit);

+MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC Magician");

+MODULE_LICENSE("GPL");

1.5.3.5

---

Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel