#include <asm/arch/irda.h>
#include <asm/arch/usb.h>
#include <asm/arch/keypad.h>
-#include <asm/arch/dma.h>
#include <asm/arch/common.h>
+#include <asm/arch/mcbsp.h>
+#include <asm/arch/omap-alsa.h>
extern int omap_gpio_init(void);
.id = -1,
};
+static struct omap_mcbsp_reg_cfg mcbsp_regs = {
+ .spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
+ .spcr1 = RINTM(3) | RRST,
+ .rcr2 = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
+ RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(1),
+ .rcr1 = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
+ .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
+ XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(1) | XFIG,
+ .xcr1 = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
+ .srgr1 = FWID(15),
+ .srgr2 = GSYNC | CLKSP | FSGM | FPER(31),
+
+ .pcr0 = CLKXM | CLKRM | FSXP | FSRP | CLKXP | CLKRP,
+ //.pcr0 = CLKXP | CLKRP, /* mcbsp: slave */
+};
+
+static struct omap_alsa_codec_config alsa_config = {
+ .name = "H2 TSC2101",
+ .mcbsp_regs_alsa = &mcbsp_regs,
+ .codec_configure_dev = NULL, // tsc2101_configure,
+ .codec_set_samplerate = NULL, // tsc2101_set_samplerate,
+ .codec_clock_setup = NULL, // tsc2101_clock_setup,
+ .codec_clock_on = NULL, // tsc2101_clock_on,
+ .codec_clock_off = NULL, // tsc2101_clock_off,
+ .get_default_samplerate = NULL, // tsc2101_get_default_samplerate,
+};
+
+static struct platform_device h2_mcbsp1_device = {
+ .name = "omap_alsa_mcbsp",
+ .id = 1,
+ .dev = {
+ .platform_data = &alsa_config,
+ },
+};
+
static struct platform_device *h2_devices[] __initdata = {
&h2_nor_device,
&h2_nand_device,
&h2_irda_device,
&h2_kp_device,
&h2_lcd_device,
+ &h2_mcbsp1_device,
};
static void __init h2_init_smc91x(void)
#include <asm/arch/tc.h>
#include <asm/arch/keypad.h>
#include <asm/arch/common.h>
+#include <asm/arch/mcbsp.h>
+#include <asm/arch/omap-alsa.h>
static int osk_keymap[] = {
KEY(0, 0, KEY_F1),
.resource = osk5912_cf_resources,
};
+#define DEFAULT_BITPERSAMPLE 16
+
+static struct omap_mcbsp_reg_cfg mcbsp_regs = {
+ .spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
+ .spcr1 = RINTM(3) | RRST,
+ .rcr2 = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
+ RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(0),
+ .rcr1 = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
+ .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
+ XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG,
+ .xcr1 = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
+ .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1),
+ .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1),
+ /*.pcr0 = FSXM | FSRM | CLKXM | CLKRM | CLKXP | CLKRP,*/ /* mcbsp: master */
+ .pcr0 = CLKXP | CLKRP, /* mcbsp: slave */
+};
+
+static struct omap_alsa_codec_config alsa_config = {
+ .name = "OSK AIC23",
+ .mcbsp_regs_alsa = &mcbsp_regs,
+ .codec_configure_dev = NULL, // aic23_configure,
+ .codec_set_samplerate = NULL, // aic23_set_samplerate,
+ .codec_clock_setup = NULL, // aic23_clock_setup,
+ .codec_clock_on = NULL, // aic23_clock_on,
+ .codec_clock_off = NULL, // aic23_clock_off,
+ .get_default_samplerate = NULL, // aic23_get_default_samplerate,
+};
+
static struct platform_device osk5912_mcbsp1_device = {
- .name = "omap_mcbsp",
- .id = 1,
+ .name = "omap_alsa_mcbsp",
+ .id = 1,
+ .dev = {
+ .platform_data = &alsa_config,
+ },
};
static struct resource osk5912_kp_resources[] = {
/*
- * sound/arm/omap-aic23.h
+ * linux/include/asm-arm/arch-omap/omap-alsa.h
*
- * Alsa Driver for AIC23 codec on OSK5912 platform board
+ * Alsa Driver for AIC23 and TSC2101 codecs on OMAP platform boards.
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
*
* Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
* Written by Daniel Petrini, David Cohen, Anderson Briglia
* 2005/07/25 INdT-10LE Kernel Team - Alsa driver for omap osk,
* original version based in sa1100 driver
* and omap oss driver.
- *
- * 2005-12-18 Dirk Behme - Added L/R Channel Interchange fix as proposed by Ajaya Babu
*/
-#ifndef __OMAP_AIC23_H
-#define __OMAP_AIC23_H
+#ifndef __OMAP_ALSA_H
+#define __OMAP_ALSA_H
#include <sound/driver.h>
#include <asm/arch/dma.h>
#include <sound/core.h>
#include <sound/pcm.h>
+#include <asm/arch/mcbsp.h>
+#include <linux/platform_device.h>
+/*
+ * Debug functions
+ */
+#undef DEBUG
+//#define DEBUG
+
+#define ERR(ARGS...) printk(KERN_ERR "{%s}-ERROR: ", __FUNCTION__);printk(ARGS);
+
+#ifdef DEBUG
+#define DPRINTK(ARGS...) printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
+#define ADEBUG() printk("XXX Alsa debug f:%s, l:%d\n", __FUNCTION__, __LINE__)
+#define FN_IN printk(KERN_INFO "[%s]: start\n", __FUNCTION__)
+#define FN_OUT(n) printk(KERN_INFO "[%s]: end(%u)\n",__FUNCTION__, n)
+#else
+#define DPRINTK(ARGS...) /* nop */
+#define ADEBUG() /* nop */
+#define FN_IN /* nop */
+#define FN_OUT(n) /* nop */
+#endif
-#define DEFAULT_OUTPUT_VOLUME 0x60
-#define DEFAULT_INPUT_VOLUME 0x00 /* 0 ==> mute line in */
-
-#define OUTPUT_VOLUME_MIN LHV_MIN
-#define OUTPUT_VOLUME_MAX LHV_MAX
-#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN)
-#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX
-
-#define INPUT_VOLUME_MIN LIV_MIN
-#define INPUT_VOLUME_MAX LIV_MAX
-#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
-#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX
-
-#define SIDETONE_MASK 0x1c0
-#define SIDETONE_0 0x100
-#define SIDETONE_6 0x000
-#define SIDETONE_9 0x040
-#define SIDETONE_12 0x080
-#define SIDETONE_18 0x0c0
-
-#define DEFAULT_ANALOG_AUDIO_CONTROL DAC_SELECTED | STE_ENABLED | BYPASS_ON | INSEL_MIC | MICB_20DB
+#define DMA_BUF_SIZE (1024 * 8)
/*
* Buffer management for alsa and dma
/*
* Alsa card structure for aic23
*/
-struct snd_card_omap_aic23 {
+struct snd_card_omap_codec {
snd_card_t *card;
snd_pcm_t *pcm;
long samplerate;
struct audio_stream s[2]; /* playback & capture */
};
-/*********** Function Prototypes *************************/
+/* Codec specific information and function pointers.
+ * Codec (omap-alsa-aic23.c and omap-alsa-tsc2101.c)
+ * are responsible for defining the function pointers.
+ */
+struct omap_alsa_codec_config {
+ char *name;
+ struct omap_mcbsp_reg_cfg *mcbsp_regs_alsa;
+ snd_pcm_hw_constraint_list_t *hw_constraints_rates;
+ snd_pcm_hardware_t *snd_omap_alsa_playback;
+ snd_pcm_hardware_t *snd_omap_alsa_capture;
+ void (*codec_configure_dev)(void);
+ void (*codec_set_samplerate)(long);
+ void (*codec_clock_setup)(void);
+ int (*codec_clock_on)(void);
+ int (*codec_clock_off)(void);
+ int (*get_default_samplerate)(void);
+};
-void audio_dma_callback(void *);
-int snd_omap_mixer(struct snd_card_omap_aic23 *);
+/*********** Mixer function prototypes *************************/
+int snd_omap_mixer(struct snd_card_omap_codec *);
void snd_omap_init_mixer(void);
-/* Clock functions */
-int omap_aic23_clock_on(void);
-int omap_aic23_clock_off(void);
#ifdef CONFIG_PM
void snd_omap_suspend_mixer(void);
void snd_omap_resume_mixer(void);
#endif
-/* Codec AIC23 */
-#if defined(CONFIG_SENSORS_TLV320AIC23) || defined (CONFIG_SENSORS_TLV320AIC23_MODULE)
-
-extern int tlv320aic23_write_value(u8 reg, u16 value);
-
-/* TLV320AIC23 is a write only device */
-static __inline__ void audio_aic23_write(u8 address, u16 data)
-{
- tlv320aic23_write_value(address, data);
-}
+int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_codec_config *config);
+int snd_omap_alsa_remove(struct platform_device *pdev);
+#ifdef CONFIG_PM
+int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state);
+int snd_omap_alsa_resume(struct platform_device *pdev);
+#else
+#define snd_omap_alsa_suspend NULL
+#define snd_omap_alsa_resume NULL
+#endif
-#endif /* CONFIG_SENSORS_TLV320AIC23 */
+/*********** function prototype to function called from the dma interrupt handler ******/
+void callback_omap_alsa_sound_dma(void *);
#endif
To compile this driver as a module, choose M here: the module
will be called snd-omap-aic23.
+
+config SND_OMAP_TSC2101
+ tristate "OMAP TSC2101 alsa driver"
+ depends on ARCH_OMAP && SND
+ select SND_PCM
+ select OMAP_TSC2101
+ select OMAP_UWIRE if ARCH_OMAP
+ help
+ Say Y here if you have a OMAP platform board
+ and want to use its TSC2101 audio chip. Driver has
+ been tested with H2 and iPAQ h6300.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-omap-tsc2101.
endmenu
obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o
snd-pxa2xx-ac97-objs := pxa2xx-ac97.o
-obj-$(CONFIG_SND_OMAP_AIC23) += snd-omap-aic23.o
-snd-omap-aic23-objs := omap-aic23.o omap-alsa-dma.o omap-alsa-mixer.o
+obj-$(CONFIG_SND) += omap/
+++ /dev/null
-/*
- * sound/arm/omap-aic23.c
- *
- * Alsa Driver for AIC23 codec on OSK5912 platform board
- *
- * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
- * Written by Daniel Petrini, David Cohen, Anderson Briglia
- * {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
- *
- * Based on sa11xx-uda1341.c,
- * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
- *
- * 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.
- *
- * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
- * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
- * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
- * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
- * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
- * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
- * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- * You should have received a copy of the GNU General Public License along
- * with this program; if not, write to the Free Software Foundation, Inc.,
- * 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * History:
- *
- * 2005-07-29 INdT Kernel Team - Alsa driver for omap osk. Creation of new
- * file omap-aic23.c
- *
- * 2005-12-18 Dirk Behme - Added L/R Channel Interchange fix as proposed by Ajaya Babu
- */
-
-#include <linux/config.h>
-#include <sound/driver.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/errno.h>
-#include <linux/ioctl.h>
-#include <linux/delay.h>
-#include <linux/slab.h>
-#include <linux/clk.h>
-
-#ifdef CONFIG_PM
-#include <linux/pm.h>
-#endif
-
-#include <asm/hardware.h>
-#include <asm/mach-types.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/aic23.h>
-#include <asm/arch/mcbsp.h>
-#include <asm/arch/clock.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/initval.h>
-#include <sound/memalloc.h>
-
-#include "omap-alsa-dma.h"
-#include "omap-aic23.h"
-
-#undef DEBUG
-
-#ifdef DEBUG
-#define ADEBUG() printk("XXX Alsa debug f:%s, l:%d\n", __FUNCTION__, __LINE__)
-#else
-#define ADEBUG() /* nop */
-#endif
-
-/* Define to set the AIC23 as the master w.r.t McBSP */
-#define AIC23_MASTER
-
-/*
- * AUDIO related MACROS
- */
-#define DEFAULT_BITPERSAMPLE 16
-#define AUDIO_RATE_DEFAULT 44100
-#define AUDIO_MCBSP OMAP_MCBSP1
-#define NUMBER_SAMPLE_RATES_SUPPORTED 10
-
-
-MODULE_AUTHOR("Daniel Petrini, David Cohen, Anderson Briglia - INdT");
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("OMAP AIC23 driver for ALSA");
-MODULE_SUPPORTED_DEVICE("{{AIC23,OMAP AIC23}}");
-MODULE_ALIAS("omap_mcbsp.1");
-
-static char *id = NULL;
-MODULE_PARM_DESC(id, "OMAP OSK ALSA Driver for AIC23 chip.");
-
-static struct snd_card_omap_aic23 *omap_aic23 = NULL;
-
-static struct clk *aic23_mclk = 0;
-
-struct sample_rate_rate_reg_info {
- u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
- u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
-/*
- * DAC USB-mode sampling rates (MCLK = 12 MHz)
- * The rates and rate_reg_into MUST be in the same order
- */
-static unsigned int rates[] = {
- 4000, 8000, 16000, 22050,
- 24000, 32000, 44100,
- 48000, 88200, 96000,
-};
-static const struct sample_rate_rate_reg_info
- rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
- {0x06, 1}, /* 4000 */
- {0x06, 0}, /* 8000 */
- {0x0C, 1}, /* 16000 */
- {0x11, 1}, /* 22050 */
- {0x00, 1}, /* 24000 */
- {0x0C, 0}, /* 32000 */
- {0x11, 0}, /* 44100 */
- {0x00, 0}, /* 48000 */
- {0x1F, 0}, /* 88200 */
- {0x0E, 0}, /* 96000 */
-};
-
-/*
- * mcbsp configuration structure
- */
-static struct omap_mcbsp_reg_cfg initial_config_mcbsp = {
- .spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
- .spcr1 = RINTM(3) | RRST,
- .rcr2 = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
- RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(0),
- .rcr1 = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
- .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
- XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG,
- .xcr1 = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
- .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1),
- .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1),
-#ifndef AIC23_MASTER
- /* configure McBSP to be the I2S master */
- .pcr0 = FSXM | FSRM | CLKXM | CLKRM | CLKXP | CLKRP,
-#else
- /* configure McBSP to be the I2S slave */
- .pcr0 = CLKXP | CLKRP,
-#endif /* AIC23_MASTER */
-};
-
-static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
-};
-
-/*
- * HW interface start and stop helper functions
- */
-static int audio_ifc_start(void)
-{
- omap_mcbsp_start(AUDIO_MCBSP);
- return 0;
-}
-
-static int audio_ifc_stop(void)
-{
- omap_mcbsp_stop(AUDIO_MCBSP);
- return 0;
-}
-
-/*
- * Codec/mcbsp init and configuration section
- * codec dependent code.
- */
-
-/*
- * Sample rate changing
- */
-static void omap_aic23_set_samplerate(struct snd_card_omap_aic23
- *omap_aic23, long rate)
-{
- u8 count = 0;
- u16 data = 0;
-
- /* Fix the rate if it has a wrong value */
- if (rate >= 96000)
- rate = 96000;
- else if (rate >= 88200)
- rate = 88200;
- else if (rate >= 48000)
- rate = 48000;
- else if (rate >= 44100)
- rate = 44100;
- else if (rate >= 32000)
- rate = 32000;
- else if (rate >= 24000)
- rate = 24000;
- else if (rate >= 22050)
- rate = 22050;
- else if (rate >= 16000)
- rate = 16000;
- else if (rate >= 8000)
- rate = 8000;
- else
- rate = 4000;
-
- /* Search for the right sample rate */
- /* Verify what happens if the rate is not supported
- * now it goes to 96Khz */
- while ((rates[count] != rate) &&
- (count < (NUMBER_SAMPLE_RATES_SUPPORTED - 1))) {
- count++;
- }
-
- data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
- (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
-
- audio_aic23_write(SAMPLE_RATE_CONTROL_ADDR, data);
-
- omap_aic23->samplerate = rate;
-}
-
-static inline void aic23_configure(void)
-{
- /* Reset codec */
- audio_aic23_write(RESET_CONTROL_ADDR, 0);
-
- /* Initialize the AIC23 internal state */
-
- /* Analog audio path control, DAC selected, delete INSEL_MIC for line in */
- audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR, DEFAULT_ANALOG_AUDIO_CONTROL);
-
- /* Digital audio path control, de-emphasis control 44.1kHz */
- audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
-
- /* Digital audio interface, master/slave mode, I2S, 16 bit */
-#ifdef AIC23_MASTER
- audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR,
- MS_MASTER | IWL_16 | FOR_DSP);
-#else
- audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR, IWL_16 | FOR_DSP);
-#endif
-
- /* Enable digital interface */
- audio_aic23_write(DIGITAL_INTERFACE_ACT_ADDR, ACT_ON);
-
-}
-
-static void omap_aic23_audio_init(struct snd_card_omap_aic23 *omap_aic23)
-{
- /* Setup DMA stuff */
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].id = "Alsa AIC23 out";
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id =
- SNDRV_PCM_STREAM_PLAYBACK;
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev =
- OMAP_DMA_MCBSP1_TX;
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].hw_start =
- audio_ifc_start;
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].hw_stop =
- audio_ifc_stop;
-
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].id = "Alsa AIC23 in";
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].stream_id =
- SNDRV_PCM_STREAM_CAPTURE;
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev =
- OMAP_DMA_MCBSP1_RX;
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].hw_start =
- audio_ifc_start;
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].hw_stop =
- audio_ifc_stop;
-
- /* configuring the McBSP */
- omap_mcbsp_request(AUDIO_MCBSP);
-
- /* if configured, then stop mcbsp */
- omap_mcbsp_stop(AUDIO_MCBSP);
-
- omap_mcbsp_config(AUDIO_MCBSP, &initial_config_mcbsp);
- omap_mcbsp_start(AUDIO_MCBSP);
- aic23_configure();
-}
-
-/*
- * DMA functions
- * Depends on omap-aic23-dma.c functions and (omap) dma.c
- *
- */
-#define DMA_BUF_SIZE 1024 * 8
-
-static int audio_dma_request(struct audio_stream *s,
- void (*callback) (void *))
-{
- int err;
-
- err = omap_request_sound_dma(s->dma_dev, s->id, s, &s->lch);
- if (err < 0)
- printk(KERN_ERR "unable to grab audio dma 0x%x\n",
- s->dma_dev);
- return err;
-}
-
-static int audio_dma_free(struct audio_stream *s)
-{
- int err = 0;
-
- err = omap_free_sound_dma(s, &s->lch);
- if (err < 0)
- printk(KERN_ERR "Unable to free audio dma channels!\n");
- return err;
-}
-
-/*
- * This function should calculate the current position of the dma in the
- * buffer. It will help alsa middle layer to continue update the buffer.
- * Its correctness is crucial for good functioning.
- */
-static u_int audio_get_dma_pos(struct audio_stream *s)
-{
- snd_pcm_substream_t *substream = s->stream;
- snd_pcm_runtime_t *runtime = substream->runtime;
- unsigned int offset;
- unsigned long flags;
- dma_addr_t count;
- ADEBUG();
-
- /* this must be called w/ interrupts locked as requested in dma.c */
- spin_lock_irqsave(&s->dma_lock, flags);
-
- /* For the current period let's see where we are */
- count = omap_get_dma_src_addr_counter(s->lch[s->dma_q_head]);
-
- spin_unlock_irqrestore(&s->dma_lock, flags);
-
- /* Now, the position related to the end of that period */
- offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
-
- if (offset >= runtime->buffer_size || offset < 0)
- offset = 0;
-
- return offset;
-}
-
-/*
- * this stops the dma and clears the dma ptrs
- */
-static void audio_stop_dma(struct audio_stream *s)
-{
- unsigned long flags;
- ADEBUG();
-
- spin_lock_irqsave(&s->dma_lock, flags);
- s->active = 0;
- s->period = 0;
- s->periods = 0;
-
- /* this stops the dma channel and clears the buffer ptrs */
- omap_audio_stop_dma(s);
-
- omap_clear_sound_dma(s);
-
- spin_unlock_irqrestore(&s->dma_lock, flags);
-}
-
-/*
- * Main dma routine, requests dma according where you are in main alsa buffer
- */
-static void audio_process_dma(struct audio_stream *s)
-{
- snd_pcm_substream_t *substream = s->stream;
- snd_pcm_runtime_t *runtime;
- unsigned int dma_size;
- unsigned int offset;
- int ret;
-
- runtime = substream->runtime;
- if (s->active) {
- dma_size = frames_to_bytes(runtime, runtime->period_size);
- offset = dma_size * s->period;
- snd_assert(dma_size <= DMA_BUF_SIZE,);
- ret =
- omap_start_sound_dma(s,
- (dma_addr_t) runtime->dma_area +
- offset, dma_size);
- if (ret) {
- printk(KERN_ERR
- "audio_process_dma: cannot queue DMA buffer (%i)\n",
- ret);
- return;
- }
-
- s->period++;
- s->period %= runtime->periods;
- s->periods++;
- s->offset = offset;
- }
-}
-
-/*
- * This is called when dma IRQ occurs at the end of each transmited block
- */
-void audio_dma_callback(void *data)
-{
- struct audio_stream *s = data;
-
- /*
- * If we are getting a callback for an active stream then we inform
- * the PCM middle layer we've finished a period
- */
- if (s->active)
- snd_pcm_period_elapsed(s->stream);
-
- spin_lock(&s->dma_lock);
- if (s->periods > 0) {
- s->periods--;
- }
- audio_process_dma(s);
- spin_unlock(&s->dma_lock);
-}
-
-
-/*
- * Alsa section
- * PCM settings and callbacks
- */
-
-static int snd_omap_aic23_trigger(snd_pcm_substream_t * substream, int cmd)
-{
- struct snd_card_omap_aic23 *chip =
- snd_pcm_substream_chip(substream);
- int stream_id = substream->pstr->stream;
- struct audio_stream *s = &chip->s[stream_id];
- int err = 0;
- ADEBUG();
-
- /* note local interrupts are already disabled in the midlevel code */
- spin_lock(&s->dma_lock);
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- /* requested stream startup */
- s->active = 1;
- audio_process_dma(s);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- /* requested stream shutdown */
- audio_stop_dma(s);
- break;
- default:
- err = -EINVAL;
- break;
- }
- spin_unlock(&s->dma_lock);
-
- return err;
-}
-
-static int snd_omap_aic23_prepare(snd_pcm_substream_t * substream)
-{
- struct snd_card_omap_aic23 *chip =
- snd_pcm_substream_chip(substream);
- snd_pcm_runtime_t *runtime = substream->runtime;
- struct audio_stream *s = &chip->s[substream->pstr->stream];
-
- /* set requested samplerate */
- omap_aic23_set_samplerate(chip, runtime->rate);
-
- s->period = 0;
- s->periods = 0;
-
- return 0;
-}
-
-static snd_pcm_uframes_t snd_omap_aic23_pointer(snd_pcm_substream_t *
- substream)
-{
- struct snd_card_omap_aic23 *chip =
- snd_pcm_substream_chip(substream);
-
- return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
-}
-
-/* Hardware capabilities */
-
-static snd_pcm_hardware_t snd_omap_aic23_capture = {
- .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
- .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_KNOT),
- .rate_min = 8000,
- .rate_max = 96000,
- .channels_min = 2,
- .channels_max = 2,
- .buffer_bytes_max = 128 * 1024,
- .period_bytes_min = 32,
- .period_bytes_max = 8 * 1024,
- .periods_min = 16,
- .periods_max = 255,
- .fifo_size = 0,
-};
-
-static snd_pcm_hardware_t snd_omap_aic23_playback = {
- .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
- .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_KNOT),
- .rate_min = 8000,
- .rate_max = 96000,
- .channels_min = 2,
- .channels_max = 2,
- .buffer_bytes_max = 128 * 1024,
- .period_bytes_min = 32,
- .period_bytes_max = 8 * 1024,
- .periods_min = 16,
- .periods_max = 255,
- .fifo_size = 0,
-};
-
-static int snd_card_omap_aic23_open(snd_pcm_substream_t * substream)
-{
- struct snd_card_omap_aic23 *chip =
- snd_pcm_substream_chip(substream);
- snd_pcm_runtime_t *runtime = substream->runtime;
- int stream_id = substream->pstr->stream;
- int err;
- ADEBUG();
-
- chip->s[stream_id].stream = substream;
-
- omap_aic23_clock_on();
-
- if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
- runtime->hw = snd_omap_aic23_playback;
- else
- runtime->hw = snd_omap_aic23_capture;
- if ((err =
- snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS)) <
- 0)
- return err;
- if ((err =
- snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &hw_constraints_rates)) < 0)
- return err;
-
- return 0;
-}
-
-static int snd_card_omap_aic23_close(snd_pcm_substream_t * substream)
-{
- struct snd_card_omap_aic23 *chip =
- snd_pcm_substream_chip(substream);
- ADEBUG();
-
- omap_aic23_clock_off();
- chip->s[substream->pstr->stream].stream = NULL;
-
- return 0;
-}
-
-/* HW params & free */
-
-static int snd_omap_aic23_hw_params(snd_pcm_substream_t * substream,
- snd_pcm_hw_params_t * hw_params)
-{
- return snd_pcm_lib_malloc_pages(substream,
- params_buffer_bytes(hw_params));
-}
-
-static int snd_omap_aic23_hw_free(snd_pcm_substream_t * substream)
-{
- return snd_pcm_lib_free_pages(substream);
-}
-
-/* pcm operations */
-
-static snd_pcm_ops_t snd_card_omap_aic23_playback_ops = {
- .open = snd_card_omap_aic23_open,
- .close = snd_card_omap_aic23_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_omap_aic23_hw_params,
- .hw_free = snd_omap_aic23_hw_free,
- .prepare = snd_omap_aic23_prepare,
- .trigger = snd_omap_aic23_trigger,
- .pointer = snd_omap_aic23_pointer,
-};
-
-static snd_pcm_ops_t snd_card_omap_aic23_capture_ops = {
- .open = snd_card_omap_aic23_open,
- .close = snd_card_omap_aic23_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_omap_aic23_hw_params,
- .hw_free = snd_omap_aic23_hw_free,
- .prepare = snd_omap_aic23_prepare,
- .trigger = snd_omap_aic23_trigger,
- .pointer = snd_omap_aic23_pointer,
-};
-
-/*
- * Alsa init and exit section
- *
- * Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
- */
-static int __init snd_card_omap_aic23_pcm(struct snd_card_omap_aic23
- *omap_aic23, int device)
-{
- snd_pcm_t *pcm;
- int err;
- ADEBUG();
-
- if ((err =
- snd_pcm_new(omap_aic23->card, "AIC23 PCM", device, 1, 1,
- &pcm)) < 0)
- return err;
-
- /* sets up initial buffer with continuous allocation */
- snd_pcm_lib_preallocate_pages_for_all(pcm,
- SNDRV_DMA_TYPE_CONTINUOUS,
- snd_dma_continuous_data
- (GFP_KERNEL),
- 128 * 1024, 128 * 1024);
-
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
- &snd_card_omap_aic23_playback_ops);
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
- &snd_card_omap_aic23_capture_ops);
- pcm->private_data = omap_aic23;
- pcm->info_flags = 0;
- strcpy(pcm->name, "omap aic23 pcm");
-
- omap_aic23_audio_init(omap_aic23);
-
- /* setup DMA controller */
- audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK],
- audio_dma_callback);
- audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE],
- audio_dma_callback);
-
- omap_aic23->pcm = pcm;
-
- return 0;
-}
-
-
-#ifdef CONFIG_PM
-
-static int snd_omap_aic23_suspend(snd_card_t * card, pm_message_t state)
-{
- struct snd_card_omap_aic23 *chip = card->private_data;
- ADEBUG();
-
- if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
- snd_pcm_suspend_all(chip->pcm);
- /* Mutes and turn clock off */
- omap_aic23_clock_off();
- snd_omap_suspend_mixer();
- }
-
- return 0;
-}
-
-/*
- * Prepare hardware for resume
- */
-static int snd_omap_aic23_resume(snd_card_t * card)
-{
- struct snd_card_omap_aic23 *chip = card->private_data;
- ADEBUG();
-
- if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
- omap_aic23_clock_on();
- snd_omap_resume_mixer();
- }
-
- return 0;
-}
-
-/*
- * Driver suspend/resume - calls alsa functions. Some hints from aaci.c
- */
-static int omap_aic23_suspend(struct platform_device *pdev, pm_message_t state)
-{
- snd_card_t *card = platform_get_drvdata(pdev);
-
- if (card->power_state != SNDRV_CTL_POWER_D3hot) {
- snd_omap_aic23_suspend(card, PMSG_SUSPEND);
- }
- return 0;
-}
-
-static int omap_aic23_resume(struct platform_device *pdev)
-{
- snd_card_t *card = platform_get_drvdata(pdev);
-
- if (card->power_state != SNDRV_CTL_POWER_D0) {
- snd_omap_aic23_resume(card);
- }
- return 0;
-}
-
-#else
-#define snd_omap_aic23_suspend NULL
-#define snd_omap_aic23_resume NULL
-#define omap_aic23_suspend NULL
-#define omap_aic23_resume NULL
-
-#endif /* CONFIG_PM */
-
-/*
- */
-void snd_omap_aic23_free(snd_card_t * card)
-{
- struct snd_card_omap_aic23 *chip = card->private_data;
- ADEBUG();
-
- /*
- * Turn off codec after it is done.
- * Can't do it immediately, since it may still have
- * buffered data.
- */
- set_current_state(TASK_INTERRUPTIBLE);
- schedule_timeout(2);
-
- omap_mcbsp_stop(AUDIO_MCBSP);
- omap_mcbsp_free(AUDIO_MCBSP);
-
- audio_aic23_write(RESET_CONTROL_ADDR, 0);
- audio_aic23_write(POWER_DOWN_CONTROL_ADDR, 0xff);
-
- audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
- audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
-}
-
-/*
- * Omap MCBSP clock configuration
- *
- * Here we have some functions that allows clock to be enabled and
- * disabled only when needed. Besides doing clock configuration
- * it allows turn on/turn off audio when necessary.
- */
-#define CODEC_CLOCK 12000000
-#define AUDIO_RATE_DEFAULT 44100
-
-/*
- * Do clock framework mclk search
- */
-static __init void omap_aic23_clock_setup(void)
-{
- aic23_mclk = clk_get(0, "mclk");
-}
-
-/*
- * Do some sanity check, set clock rate, starts it and
- * turn codec audio on
- */
-int omap_aic23_clock_on(void)
-{
- if (clk_get_usecount(aic23_mclk) > 0) {
- /* MCLK is already in use */
- printk(KERN_WARNING
- "MCLK in use at %d Hz. We change it to %d Hz\n",
- (uint) clk_get_rate(aic23_mclk),
- CODEC_CLOCK);
- }
-
- if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) {
- printk(KERN_ERR
- "Cannot set MCLK for AIC23 CODEC\n");
- return -ECANCELED;
- }
-
- clk_enable(aic23_mclk);
-
- printk(KERN_DEBUG
- "MCLK = %d [%d], usecount = %d\n",
- (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK,
- clk_get_usecount(aic23_mclk));
-
- /* Now turn the audio on */
- audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
- ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
- ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);
-
- return 0;
-}
-/*
- * Do some sanity check, turn clock off and then turn
- * codec audio off
- */
-int omap_aic23_clock_off(void)
-{
- if (clk_get_usecount(aic23_mclk) > 0) {
- if (clk_get_rate(aic23_mclk) != CODEC_CLOCK) {
- printk(KERN_WARNING
- "MCLK for audio should be %d Hz. But is %d Hz\n",
- (uint) clk_get_rate(aic23_mclk),
- CODEC_CLOCK);
- }
-
- clk_disable(aic23_mclk);
- }
-
- audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
- DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
- ADC_OFF | MIC_OFF | LINE_OFF);
- return 0;
-}
-
-/* module init & exit */
-
-/*
- * Inits alsa soudcard structure
- */
-static int __init snd_omap_aic23_probe(struct platform_device *pdev)
-{
- int err = 0;
- snd_card_t *card;
- ADEBUG();
-
- /* gets clock from clock framework */
- omap_aic23_clock_setup();
-
- /* register the soundcard */
- card = snd_card_new(-1, id, THIS_MODULE, sizeof(omap_aic23));
- if (card == NULL)
- return -ENOMEM;
-
- omap_aic23 = kcalloc(1, sizeof(*omap_aic23), GFP_KERNEL);
- if (omap_aic23 == NULL)
- return -ENOMEM;
-
- card->private_data = (void *) omap_aic23;
- card->private_free = snd_omap_aic23_free;
-
- omap_aic23->card = card;
- omap_aic23->samplerate = AUDIO_RATE_DEFAULT;
-
- spin_lock_init(&omap_aic23->s[0].dma_lock);
- spin_lock_init(&omap_aic23->s[1].dma_lock);
-
- /* mixer */
- if ((err = snd_omap_mixer(omap_aic23)) < 0)
- goto nodev;
-
- /* PCM */
- if ((err = snd_card_omap_aic23_pcm(omap_aic23, 0)) < 0)
- goto nodev;
-
- strcpy(card->driver, "AIC23");
- strcpy(card->shortname, "OSK AIC23");
- sprintf(card->longname, "OMAP OSK with AIC23");
-
- snd_omap_init_mixer();
-
- snd_card_set_dev(card, &pdev->dev);
-
- if ((err = snd_card_register(card)) == 0) {
- printk(KERN_INFO "OSK audio support initialized\n");
- platform_set_drvdata(pdev, card);
- return 0;
- }
-
-nodev:
- snd_omap_aic23_free(card);
-
- return err;
-}
-
-static int snd_omap_aic23_remove(struct platform_device *pdev)
-{
- snd_card_t *card = platform_get_drvdata(pdev);
- struct snd_card_omap_aic23 *chip = card->private_data;
-
- snd_card_free(card);
-
- omap_aic23 = NULL;
- card->private_data = NULL;
- kfree(chip);
-
- platform_set_drvdata(pdev, NULL);
-
- return 0;
-
-}
-
-static struct platform_driver omap_alsa_driver = {
- .probe = snd_omap_aic23_probe,
- .remove = snd_omap_aic23_remove,
- .suspend = omap_aic23_suspend,
- .resume = omap_aic23_resume,
- .driver = {
- .name = "omap_mcbsp",
- },
-};
-
-static int __init omap_aic23_init(void)
-{
- int err;
- ADEBUG();
-
- err = platform_driver_register(&omap_alsa_driver);
-
- return err;
-}
-
-static void __exit omap_aic23_exit(void)
-{
- ADEBUG();
-
- platform_driver_unregister(&omap_alsa_driver);
-}
-
-module_init(omap_aic23_init);
-module_exit(omap_aic23_exit);
--- /dev/null
+#
+## Makefile for ALSA OMAP
+#
+#
+obj-$(CONFIG_SND_OMAP_AIC23) += snd-omap-alsa-aic23.o
+snd-omap-alsa-aic23-objs := omap-alsa.o omap-alsa-dma.o omap-alsa-aic23.o omap-alsa-aic23-mixer.o
+
+obj-$(CONFIG_SND_OMAP_TSC2101) += snd-omap-alsa-tsc2101.o
+snd-omap-alsa-tsc2101-objs := omap-alsa.o omap-alsa-dma.o omap-alsa-tsc2101.o omap-alsa-tsc2101-mixer.o
/*
- * sound/arm/omap-alsa-mixer.c
+ * sound/arm/omap/omap-alsa-aic23-mixer.c
*
* Alsa Driver Mixer for generic codecs for omap boards
*
#include <linux/config.h>
#include <sound/driver.h>
-#include <linux/module.h>
-#include <linux/device.h>
-#include <linux/init.h>
-#include <linux/errno.h>
-#include <linux/ioctl.h>
-#include <linux/delay.h>
-#include <linux/slab.h>
-
-#include <asm/hardware.h>
-#include <asm/mach-types.h>
-#include <asm/arch/dma.h>
#include <asm/arch/aic23.h>
-#include "omap-aic23.h"
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-aic23.h"
#include <sound/initval.h>
#include <sound/control.h>
/* Codec AIC23 */
#if defined(CONFIG_SENSORS_TLV320AIC23) || defined (CONFIG_SENSORS_TLV320AIC23_MODULE)
-extern __inline__ void audio_aic23_write(u8, u16);
+extern void audio_aic23_write(u8, u16);
#define MIXER_NAME "Mixer AIC23"
#define SND_OMAP_WRITE(reg, val) audio_aic23_write(reg, val)
OMAP_MUX("Capture Source", ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, INSEL_MIC),
};
-void snd_omap_init_mixer(void)
-{
- u16 vol_reg;
-
- /* Line's default values */
- omap_regs[LINE_INDEX].l_reg = DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK;
- omap_regs[LINE_INDEX].r_reg = DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK;
- omap_regs[LINE_INDEX].sw = 0;
- SND_OMAP_WRITE(LEFT_LINE_VOLUME_ADDR, DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
- SND_OMAP_WRITE(RIGHT_LINE_VOLUME_ADDR, DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
-
- /* Analog Audio Control's default values */
- omap_regs[AAC_INDEX].l_reg = DEFAULT_ANALOG_AUDIO_CONTROL;
-
- /* Headphone's default values */
- vol_reg = LZC_ON;
- vol_reg &= ~OUTPUT_VOLUME_MASK;
- vol_reg |= DEFAULT_OUTPUT_VOLUME;
- omap_regs[PCM_INDEX].l_reg = DEFAULT_OUTPUT_VOLUME;
- omap_regs[PCM_INDEX].r_reg = DEFAULT_OUTPUT_VOLUME;
- omap_regs[PCM_INDEX].sw = 1;
- SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR, vol_reg);
- SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR, vol_reg);
-}
-
#ifdef CONFIG_PM
void snd_omap_suspend_mixer(void)
}
#endif
-int snd_omap_mixer(struct snd_card_omap_aic23 *chip)
+void snd_omap_init_mixer(void)
+{
+ u16 vol_reg;
+
+ /* Line's default values */
+ omap_regs[LINE_INDEX].l_reg = DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK;
+ omap_regs[LINE_INDEX].r_reg = DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK;
+ omap_regs[LINE_INDEX].sw = 0;
+ SND_OMAP_WRITE(LEFT_LINE_VOLUME_ADDR, DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
+ SND_OMAP_WRITE(RIGHT_LINE_VOLUME_ADDR, DEFAULT_INPUT_VOLUME & INPUT_VOLUME_MASK);
+
+ /* Analog Audio Control's default values */
+ omap_regs[AAC_INDEX].l_reg = DEFAULT_ANALOG_AUDIO_CONTROL;
+
+ /* Headphone's default values */
+ vol_reg = LZC_ON;
+ vol_reg &= ~OUTPUT_VOLUME_MASK;
+ vol_reg |= DEFAULT_OUTPUT_VOLUME;
+ omap_regs[PCM_INDEX].l_reg = DEFAULT_OUTPUT_VOLUME;
+ omap_regs[PCM_INDEX].r_reg = DEFAULT_OUTPUT_VOLUME;
+ omap_regs[PCM_INDEX].sw = 1;
+ SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR, vol_reg);
+ SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR, vol_reg);
+}
+
+int snd_omap_mixer(struct snd_card_omap_codec *chip)
{
snd_card_t *card;
unsigned int idx;
return 0;
}
-
--- /dev/null
+/*
+ * arch/arm/mach-omap1/omap-alsa-aic23.c
+ *
+ * Alsa codec Driver for AIC23 chip on OSK5912 platform board
+ *
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Written by Daniel Petrini, David Cohen, Anderson Briglia
+ * {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * Based in former alsa driver for osk and oss driver
+ *
+ * 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 <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <linux/clk.h>
+#include <asm/arch/clock.h>
+#include <asm/arch/aic23.h>
+
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-aic23.h"
+
+static struct clk *aic23_mclk = 0;
+
+/* aic23 related */
+static const struct aic23_samplerate_reg_info
+ rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
+ {4000, 0x06, 1}, /* 4000 */
+ {8000, 0x06, 0}, /* 8000 */
+ {16000, 0x0C, 1}, /* 16000 */
+ {22050, 0x11, 1}, /* 22050 */
+ {24000, 0x00, 1}, /* 24000 */
+ {32000, 0x0C, 0}, /* 32000 */
+ {44100, 0x11, 0}, /* 44100 */
+ {48000, 0x00, 0}, /* 48000 */
+ {88200, 0x1F, 0}, /* 88200 */
+ {96000, 0x0E, 0}, /* 96000 */
+};
+
+/*
+ * Hardware capabilities
+ */
+
+ /*
+ * DAC USB-mode sampling rates (MCLK = 12 MHz)
+ * The rates and rate_reg_into MUST be in the same order
+ */
+static unsigned int rates[] = {
+ 4000, 8000, 16000, 22050,
+ 24000, 32000, 44100,
+ 48000, 88200, 96000,
+};
+
+static snd_pcm_hw_constraint_list_t aic23_hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static snd_pcm_hardware_t aic23_snd_omap_alsa_playback = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static snd_pcm_hardware_t aic23_snd_omap_alsa_capture = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+/*
+ * Codec/mcbsp init and configuration section
+ * codec dependent code.
+ */
+
+extern int tlv320aic23_write_value(u8 reg, u16 value);
+
+/* TLV320AIC23 is a write only device */
+void audio_aic23_write(u8 address, u16 data)
+{
+ tlv320aic23_write_value(address, data);
+}
+EXPORT_SYMBOL_GPL(audio_aic23_write);
+
+/*
+ * Sample rate changing
+ */
+void aic23_set_samplerate(long rate)
+{
+ u8 count = 0;
+ u16 data = 0;
+
+ /* Fix the rate if it has a wrong value */
+ if (rate >= 96000)
+ rate = 96000;
+ else if (rate >= 88200)
+ rate = 88200;
+ else if (rate >= 48000)
+ rate = 48000;
+ else if (rate >= 44100)
+ rate = 44100;
+ else if (rate >= 32000)
+ rate = 32000;
+ else if (rate >= 24000)
+ rate = 24000;
+ else if (rate >= 22050)
+ rate = 22050;
+ else if (rate >= 16000)
+ rate = 16000;
+ else if (rate >= 8000)
+ rate = 8000;
+ else
+ rate = 4000;
+
+ /* Search for the right sample rate */
+ /* Verify what happens if the rate is not supported
+ * now it goes to 96Khz */
+ while ((rate_reg_info[count].sample_rate != rate) &&
+ (count < (NUMBER_SAMPLE_RATES_SUPPORTED - 1))) {
+ count++;
+ }
+
+ data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
+ (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
+
+ audio_aic23_write(SAMPLE_RATE_CONTROL_ADDR, data);
+}
+
+inline void aic23_configure(void)
+{
+ /* Reset codec */
+ audio_aic23_write(RESET_CONTROL_ADDR, 0);
+
+ /* Initialize the AIC23 internal state */
+
+ /* Analog audio path control, DAC selected, delete INSEL_MIC for line in */
+ audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR, DEFAULT_ANALOG_AUDIO_CONTROL);
+
+ /* Digital audio path control, de-emphasis control 44.1kHz */
+ audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
+
+ /* Digital audio interface, master/slave mode, I2S, 16 bit */
+#ifdef AIC23_MASTER
+ audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR,
+ MS_MASTER | IWL_16 | FOR_DSP);
+#else
+ audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR, IWL_16 | FOR_DSP);
+#endif
+
+ /* Enable digital interface */
+ audio_aic23_write(DIGITAL_INTERFACE_ACT_ADDR, ACT_ON);
+}
+
+/*
+ * Omap MCBSP clock configuration and Power Management
+ *
+ * Here we have some functions that allows clock to be enabled and
+ * disabled only when needed. Besides doing clock configuration
+ * it allows turn on/turn off audio when necessary.
+ */
+/*
+ * Do clock framework mclk search
+ */
+void aic23_clock_setup(void)
+{
+ aic23_mclk = clk_get(0, "mclk");
+}
+
+/*
+ * Do some sanity check, set clock rate, starts it and
+ * turn codec audio on
+ */
+int aic23_clock_on(void)
+{
+ if (clk_get_usecount(aic23_mclk) > 0) {
+ /* MCLK is already in use */
+ printk(KERN_WARNING
+ "MCLK in use at %d Hz. We change it to %d Hz\n",
+ (uint) clk_get_rate(aic23_mclk),
+ CODEC_CLOCK);
+ }
+
+ if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) {
+ printk(KERN_ERR
+ "Cannot set MCLK for AIC23 CODEC\n");
+ return -ECANCELED;
+ }
+
+ clk_enable(aic23_mclk);
+
+ printk(KERN_DEBUG
+ "MCLK = %d [%d], usecount = %d\n",
+ (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK,
+ clk_get_usecount(aic23_mclk));
+
+ /* Now turn the audio on */
+ audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
+ ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
+ ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);
+ return 0;
+}
+/*
+ * Do some sanity check, turn clock off and then turn
+ * codec audio off
+ */
+int aic23_clock_off(void)
+{
+ if (clk_get_usecount(aic23_mclk) > 0) {
+ if (clk_get_rate(aic23_mclk) != CODEC_CLOCK) {
+ printk(KERN_WARNING
+ "MCLK for audio should be %d Hz. But is %d Hz\n",
+ (uint) clk_get_rate(aic23_mclk),
+ CODEC_CLOCK);
+ }
+
+ clk_disable(aic23_mclk);
+ }
+
+ audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
+ DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
+ ADC_OFF | MIC_OFF | LINE_OFF);
+ return 0;
+}
+
+int aic23_get_default_samplerate(void)
+{
+ return DEFAULT_SAMPLE_RATE;
+}
+
+static int __init snd_omap_alsa_aic23_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct omap_alsa_codec_config *codec_cfg;
+
+ codec_cfg = pdev->dev.platform_data;
+ if (codec_cfg != NULL) {
+ codec_cfg->hw_constraints_rates = &aic23_hw_constraints_rates;
+ codec_cfg->snd_omap_alsa_playback = &aic23_snd_omap_alsa_playback;
+ codec_cfg->snd_omap_alsa_capture = &aic23_snd_omap_alsa_capture;
+ codec_cfg->codec_configure_dev = aic23_configure;
+ codec_cfg->codec_set_samplerate = aic23_set_samplerate;
+ codec_cfg->codec_clock_setup = aic23_clock_setup;
+ codec_cfg->codec_clock_on = aic23_clock_on;
+ codec_cfg->codec_clock_off = aic23_clock_off;
+ codec_cfg->get_default_samplerate = aic23_get_default_samplerate;
+ ret = snd_omap_alsa_post_probe(pdev, codec_cfg);
+ }
+ else
+ ret = -ENODEV;
+ return ret;
+}
+
+static struct platform_driver omap_alsa_driver = {
+ .probe = snd_omap_alsa_aic23_probe,
+ .remove = snd_omap_alsa_remove,
+ .suspend = snd_omap_alsa_suspend,
+ .resume = snd_omap_alsa_resume,
+ .driver = {
+ .name = "omap_alsa_mcbsp",
+ },
+};
+
+static int __init omap_alsa_aic23_init(void)
+{
+ int err;
+
+ ADEBUG();
+ err = platform_driver_register(&omap_alsa_driver);
+
+ return err;
+}
+
+static void __exit omap_alsa_aic23_exit(void)
+{
+ ADEBUG();
+
+ platform_driver_unregister(&omap_alsa_driver);
+}
+
+module_init(omap_alsa_aic23_init);
+module_exit(omap_alsa_aic23_exit);
--- /dev/null
+/*
+ * sound/arm/omap-alsa-aic23.h
+ *
+ * Alsa Driver for AIC23 codec on OSK5912 platform board
+ *
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Written by Daniel Petrini, David Cohen, Anderson Briglia
+ * {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * 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.
+ */
+
+#ifndef __OMAP_ALSA_AIC23_H
+#define __OMAP_ALSA_AIC23_H
+
+#include <sound/driver.h>
+#include <asm/arch/dma.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <asm/arch/mcbsp.h>
+
+/* Define to set the AIC23 as the master w.r.t McBSP */
+#define AIC23_MASTER
+
+#define NUMBER_SAMPLE_RATES_SUPPORTED 10
+
+/*
+ * AUDIO related MACROS
+ */
+#ifndef DEFAULT_BITPERSAMPLE
+#define DEFAULT_BITPERSAMPLE 16
+#endif
+
+#define DEFAULT_SAMPLE_RATE 44100
+#define CODEC_CLOCK 12000000
+#define AUDIO_MCBSP OMAP_MCBSP1
+
+#define DEFAULT_OUTPUT_VOLUME 0x60
+#define DEFAULT_INPUT_VOLUME 0x00 /* 0 ==> mute line in */
+
+#define OUTPUT_VOLUME_MIN LHV_MIN
+#define OUTPUT_VOLUME_MAX LHV_MAX
+#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN)
+#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX
+
+#define INPUT_VOLUME_MIN LIV_MIN
+#define INPUT_VOLUME_MAX LIV_MAX
+#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
+#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX
+
+#define SIDETONE_MASK 0x1c0
+#define SIDETONE_0 0x100
+#define SIDETONE_6 0x000
+#define SIDETONE_9 0x040
+#define SIDETONE_12 0x080
+#define SIDETONE_18 0x0c0
+
+#define DEFAULT_ANALOG_AUDIO_CONTROL DAC_SELECTED | STE_ENABLED | BYPASS_ON | INSEL_MIC | MICB_20DB
+
+struct aic23_samplerate_reg_info {
+ u32 sample_rate;
+ u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
+ u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+/*
+ * Defines codec specific functions pointers that can be used from the
+ * common omap-alse base driver for all omap codecs. (tsc2101 and aic23)
+ */
+void define_codec_functions(struct omap_alsa_codec_config *codec_config);
+inline void aic23_configure(void);
+void aic23_set_samplerate(long rate);
+void aic23_clock_setup(void);
+int aic23_clock_on(void);
+int aic23_clock_off(void);
+int aic23_get_default_samplerate(void);
+
+#endif
/*
- * sound/arm/omap-alsa-dma.c
+ * sound/arm/omap/omap-alsa-dma.c
*
* Common audio DMA handling for the OMAP processors
*
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
* Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
*
* Copyright (C) 2004 Texas Instruments, Inc.
#include <asm/arch/mcbsp.h>
-#include "omap-aic23.h"
+#include <asm/arch/omap-alsa.h>
#undef DEBUG
-//#define DEBUG
-#ifdef DEBUG
-#define DPRINTK(ARGS...) printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
-#define FN_IN printk(KERN_INFO "[%s]: start\n", __FUNCTION__)
-#define FN_OUT(n) printk(KERN_INFO "[%s]: end(%u)\n",__FUNCTION__, n)
-#else
-
-#define DPRINTK( x... )
-#define FN_IN
-#define FN_OUT(x)
-#endif
#define ERR(ARGS...) printk(KERN_ERR "{%s}-ERROR: ", __FUNCTION__);printk(ARGS);
FN_OUT(0);
}
-int omap_request_sound_dma(int device_id, const char *device_name,
+int omap_request_alsa_sound_dma(int device_id, const char *device_name,
void *data, int **channels)
{
int i, err = 0;
}
spin_lock(&dma_list_lock);
for (i = 0; i < nr_linked_channels; i++) {
- err =
- omap_request_dma(device_id, device_name,
- sound_dma_irq_handler, data,
- &chan[i]);
+ err = omap_request_dma(device_id,
+ device_name,
+ sound_dma_irq_handler,
+ data,
+ &chan[i]);
/* Handle Failure condition here */
if (err < 0) {
**************************************************************************************/
static void omap_sound_dma_unlink_lch(void *data)
{
- struct audio_stream *s = (struct audio_stream *) data;
+ struct audio_stream *s = (struct audio_stream *)data;
int *chan = s->lch;
int i;
FN_OUT(0);
}
-int omap_free_sound_dma(void *data, int **channels)
+int omap_free_alsa_sound_dma(void *data, int **channels)
{
-
int i;
int *chan = NULL;
+
FN_IN;
if (unlikely(NULL == channels)) {
BUG();
* Stop all the DMA channels of the stream
*
**************************************************************************************/
-void omap_audio_stop_dma(struct audio_stream *s)
+void omap_stop_alsa_sound_dma(struct audio_stream *s)
{
int *chan = s->lch;
int i;
+
FN_IN;
if (unlikely(NULL == chan)) {
BUG();
* Clear any pending transfers
*
**************************************************************************************/
-void omap_clear_sound_dma(struct audio_stream * s)
+void omap_clear_alsa_sound_dma(struct audio_stream * s)
{
FN_IN;
omap_clear_dma(s->lch[s->dma_q_head]);
return;
}
-/*********************************** MODULE FUNCTIONS DEFINTIONS ***********************/
-
-#ifdef OMAP1610_MCBSP1_BASE
-#undef OMAP1610_MCBSP1_BASE
-#endif
-#define OMAP1610_MCBSP1_BASE 0xE1011000
-
/***************************************************************************************
*
* DMA related functions
int dt = 0x1; /* data type 16 */
int cen = 32; /* Stereo */
int cfn = dma_size / (2 * cen);
+
FN_IN;
omap_set_dma_dest_params(channel, 0x05, 0x00,
- (OMAP1610_MCBSP1_BASE + 0x806),
+ (OMAP1510_MCBSP1_BASE + 0x06),
0, 0);
omap_set_dma_src_params(channel, 0x00, 0x01, dma_ptr,
0, 0);
{
int dt = 0x1; /* data type 16 */
int cen = 32; /* stereo */
-
int cfn = dma_size / (2 * cen);
+
FN_IN;
omap_set_dma_src_params(channel, 0x05, 0x00,
- (OMAP1610_MCBSP1_BASE + 0x802),
+ (OMAP1510_MCBSP1_BASE + 0x02),
0, 0);
omap_set_dma_dest_params(channel, 0x00, 0x01, dma_ptr, 0, 0);
omap_set_dma_transfer_params(channel, dt, cen, cfn, 0x00, 0, 0);
int channel = s->lch[s->dma_q_head];
FN_IN;
if (!s->started) {
- s->hw_stop(); /* stops McBSP Interface */
+ s->hw_stop(); /* stops McBSP Interface */
omap_start_dma(channel);
s->started = 1;
s->hw_start(); /* start McBSP interface */
* Do the initial set of work to initialize all the channels as required.
* We shall then initate a transfer
*/
-int omap_start_sound_dma(struct audio_stream *s, dma_addr_t dma_ptr,
- u_int dma_size)
+int omap_start_alsa_sound_dma(struct audio_stream *s,
+ dma_addr_t dma_ptr,
+ u_int dma_size)
{
int ret = -EPERM;
}
if (ch_status & DCSR_END_BLOCK)
- audio_dma_callback(s);
+ callback_omap_alsa_sound_dma(s);
FN_OUT(0);
return;
}
MODULE_AUTHOR("Texas Instruments");
-MODULE_DESCRIPTION
- ("Common DMA handling for Audio driver on OMAP processors");
+MODULE_DESCRIPTION("Common DMA handling for Audio driver on OMAP processors");
MODULE_LICENSE("GPL");
-EXPORT_SYMBOL(omap_start_sound_dma);
-EXPORT_SYMBOL(omap_clear_sound_dma);
-EXPORT_SYMBOL(omap_request_sound_dma);
-EXPORT_SYMBOL(omap_free_sound_dma);
-EXPORT_SYMBOL(omap_audio_stop_dma);
+EXPORT_SYMBOL(omap_start_alsa_sound_dma);
+EXPORT_SYMBOL(omap_clear_alsa_sound_dma);
+EXPORT_SYMBOL(omap_request_alsa_sound_dma);
+EXPORT_SYMBOL(omap_free_alsa_sound_dma);
+EXPORT_SYMBOL(omap_stop_alsa_sound_dma);
/*
- * linux/sound/arm/omap-alsa-dma.h
+ * linux/sound/arm/omap/omap-alsa-dma.h
*
* Common audio DMA handling for the OMAP processors
*
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
* Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
*
* Copyright (C) 2004 Texas Instruments, Inc.
/************************** INCLUDES *************************************/
-#include "omap-aic23.h"
-
-/************************** GLOBAL MACROS *************************************/
-
-/* Provide the Macro interfaces common across platforms */
-#define DMA_REQUEST(e,s, cb) {e=omap_request_sound_dma(s->dma_dev, s->id, s, &s->lch);}
-#define DMA_FREE(s) omap_free_sound_dma(s, &s->lch)
-#define DMA_CLEAR(s) omap_clear_sound_dma(s)
+#include <asm/arch/omap-alsa.h>
/************************** GLOBAL DATA STRUCTURES *********************************/
/**************** ARCH SPECIFIC FUNCIONS *******************************************/
-void omap_clear_sound_dma(struct audio_stream * s);
+void omap_clear_alsa_sound_dma(struct audio_stream * s);
-int omap_request_sound_dma(int device_id, const char *device_name,
+int omap_request_alsa_sound_dma(int device_id, const char *device_name,
void *data, int **channels);
-int omap_free_sound_dma(void *data, int **channels);
+int omap_free_alsa_sound_dma(void *data, int **channels);
-int omap_start_sound_dma(struct audio_stream *s, dma_addr_t dma_ptr,
- u_int dma_size);
+int omap_start_alsa_sound_dma(struct audio_stream *s, dma_addr_t dma_ptr, u_int dma_size);
-void omap_audio_stop_dma(struct audio_stream *s);
+void omap_stop_alsa_sound_dma(struct audio_stream *s);
#endif
--- /dev/null
+/*
+ * sound/arm/omap/omap-alsa-tsc2101-mixer.c
+ *
+ * Alsa Driver for TSC2101 codec for OMAP platform boards.
+ *
+ * Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
+ * Everett Coleman II <gcc80x86@fuzzyneural.net>
+ *
+ * Board initialization code is based on the code in TSC2101 OSS driver.
+ * Copyright (C) 2004 Texas Instruments, Inc.
+ * Written by Nishanth Menon and Sriram Kannan
+ *
+ * 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.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
+ * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+ * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * History:
+ *
+ * 2006-03-01 Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
+ * Can switch between headset and loudspeaker playback,
+ * mute and unmute dgc, set dgc volume. Record source switch,
+ * keyclick, buzzer and headset volume and handset volume control
+ * are still missing.
+ *
+ */
+
+#include "omap-alsa-tsc2101.h"
+#include "omap-alsa-tsc2101-mixer.h"
+
+#include <linux/types.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+
+//#define M_DPRINTK(ARGS...) printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
+#define M_DPRINTK(ARGS...) /* nop */
+
+#define DGC_DALVL_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
+#define DGC_DARVL_EXTRACT(ARG) ((ARG & 0x007f))
+#define GET_DGC_DALMU_BIT_VALUE(ARG) (((ARG) & TSC2101_BIT(15)) >> 15)
+#define GET_DGC_DARMU_BIT_VALUE(ARG) (((ARG) & TSC2101_BIT(7)) >> 7)
+#define IS_DGC_DALMU_UNMUTED(ARG) (((GET_DGC_DALMU_BIT_VALUE(ARG)) == 0))
+#define IS_DGC_DARMU_UNMUTED(ARG) (((GET_DGC_DARMU_BIT_VALUE(ARG)) == 0))
+
+#define HGC_ADPGA_HED_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
+#define GET_DGC_HGCMU_BIT_VALUE(ARG) (((ARG) & TSC2101_BIT(15)) >> 15)
+#define IS_DGC_HGCMU_UNMUTED(ARG) (((GET_DGC_HGCMU_BIT_VALUE(ARG)) == 0))
+
+#define HNGC_ADPGA_HND_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
+#define GET_DGC_HNGCMU_BIT_VALUE(ARG) (((ARG) & TSC2101_BIT(15)) >> 15)
+#define IS_DGC_HNGCMU_UNMUTED(ARG) (((GET_DGC_HNGCMU_BIT_VALUE(ARG)) == 0))
+
+static int current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
+static int current_rec_src = REC_SRC_SINGLE_ENDED_MICIN_HED;
+
+/*
+ * Used for switching between TSC2101 recourd sources.
+ * Logic is adjusted from the TSC2101 OSS code.
+ */
+static int set_record_source(int val)
+{
+ u16 data;
+ int maskedVal;
+
+ FN_IN;
+ maskedVal = 0xe0 & val;
+
+ data = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_MIXER_PGA_CTRL);
+ data &= ~MPC_MICSEL(7); /* clear all MICSEL bits */
+ data |= maskedVal;
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_MIXER_PGA_CTRL,
+ data);
+ current_rec_src = val;
+
+ FN_OUT(0);
+ return 0;
+}
+
+/*
+ * Converts the Alsa mixer volume (0 - 100) to real
+ * Digital Gain Control (DGC) value that can be written
+ * or read from the TSC2101 registry.
+ *
+ * Note that the number "OUTPUT_VOLUME_MAX" is smaller than OUTPUT_VOLUME_MIN
+ * because DGC works as a volume decreaser. (The more bigger value is put
+ * to DGC, the more the volume of controlled channel is decreased)
+ *
+ * In addition the TCS2101 chip would allow the maximum volume reduction be 63.5 DB
+ * but according to some tests user can not hear anything with this chip
+ * when the volume is set to be less than 25 db.
+ * Therefore this function will return a value that means 38.5 db (63.5 db - 25 db)
+ * reduction in the channel volume, when mixer is set to 0.
+ * For mixer value 100, this will return a value that means 0 db volume reduction.
+ * ([mute_left_bit]0000000[mute_right_bit]0000000)
+*/
+int get_mixer_volume_as_dac_gain_control_volume(int vol)
+{
+ u16 retVal;
+
+ /* Convert 0 -> 100 volume to 0x7F(min) -> y(max) volume range */
+ retVal = ((vol * OUTPUT_VOLUME_RANGE) / 100) + OUTPUT_VOLUME_MAX;
+ /* invert the value for getting the proper range 0 min and 100 max */
+ retVal = OUTPUT_VOLUME_MIN - retVal;
+
+ return retVal;
+}
+
+/*
+ * Converts the Alsa mixer volume (0 - 100) to TSC2101
+ * Digital Gain Control (DGC) volume. Alsa mixer volume 0
+ * is converted to value meaning the volume reduction of -38.5 db
+ * and Alsa mixer volume 100 is converted to value meaning the
+ * reduction of 0 db.
+ */
+int set_mixer_volume_as_dac_gain_control_volume(int mixerVolL, int mixerVolR)
+{
+ u16 val;
+ int retVal;
+ int volL;
+ int volR;
+
+ if ((mixerVolL < 0) ||
+ (mixerVolL > 100) ||
+ (mixerVolR < 0) ||
+ (mixerVolR > 100)) {
+ printk(KERN_ERR "Trying a bad mixer volume as dac gain control volume value, left (%d), right (%d)!\n", mixerVolL, mixerVolR);
+ return -EPERM;
+ }
+ M_DPRINTK("mixer volume left = %d, right = %d\n", mixerVolL, mixerVolR);
+ volL = get_mixer_volume_as_dac_gain_control_volume(mixerVolL);
+ volR = get_mixer_volume_as_dac_gain_control_volume(mixerVolR);
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
+ /* keep the old mute bit settings */
+ val &= ~(DGC_DALVL(OUTPUT_VOLUME_MIN) | DGC_DARVL(OUTPUT_VOLUME_MIN));
+ val |= DGC_DALVL(volL) | DGC_DARVL(volR);
+ retVal = 2;
+ if (retVal) {
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_DAC_GAIN_CTRL,
+ val);
+ }
+ M_DPRINTK("to registry: left = %d, right = %d, total = %d\n", DGC_DALVL_EXTRACT(val), DGC_DARVL_EXTRACT(val), val);
+ return retVal;
+}
+
+int dac_gain_control_unmute_control(int muteLeft, int muteRight)
+{
+ u16 val;
+ int count;
+
+ count = 0;
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
+ /* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
+ * so if values are same, it's time to change the registry value.
+ */
+ if (muteLeft == GET_DGC_DALMU_BIT_VALUE(val)) {
+ if (muteLeft == 0) {
+ /* mute --> turn bit on */
+ val = val | DGC_DALMU;
+ }
+ else {
+ /* unmute --> turn bit off */
+ val = val & ~DGC_DALMU;
+ }
+ count++;
+ } /* L */
+ if (muteRight == GET_DGC_DARMU_BIT_VALUE(val)) {
+ if (muteRight == 0) {
+ /* mute --> turn bit on */
+ val = val | DGC_DARMU;
+ }
+ else {
+ /* unmute --> turn bit off */
+ val = val & ~DGC_DARMU;
+ }
+ count++;
+ } /* R */
+ if (count) {
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL, val);
+ M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n",
+ IS_DGC_DALMU_UNMUTED(val),
+ IS_DGC_DARMU_UNMUTED(val));
+ }
+ return count;
+}
+
+/*
+ * Converts the DGC registry value read from the TSC2101 registry to
+ * Alsa mixer volume format (0 - 100).
+ */
+int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
+{
+ u16 retVal;
+
+ retVal = OUTPUT_VOLUME_MIN - vol;
+ retVal = ((retVal - OUTPUT_VOLUME_MAX) * 100) / OUTPUT_VOLUME_RANGE;
+ /* fix scaling error */
+ if ((retVal > 0) && (retVal < 100)) {
+ retVal++;
+ }
+ return retVal;
+}
+
+/*
+ * Converts the headset gain control volume (0 - 63.5 db)
+ * to Alsa mixer volume (0 - 100)
+ */
+int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
+{
+ u16 retVal;
+
+ retVal = ((registerVal * 100) / INPUT_VOLUME_RANGE);
+ return retVal;
+}
+
+/*
+ * Converts the handset gain control volume (0 - 63.5 db)
+ * to Alsa mixer volume (0 - 100)
+ */
+int get_handset_gain_control_volume_as_mixer_volume(u16 registerVal)
+{
+ return get_headset_gain_control_volume_as_mixer_volume(registerVal);
+}
+
+/*
+ * Converts the Alsa mixer volume (0 - 100) to
+ * headset gain control volume (0 - 63.5 db)
+ */
+int get_mixer_volume_as_headset_gain_control_volume(u16 mixerVal)
+{
+ u16 retVal;
+
+ retVal = ((mixerVal * INPUT_VOLUME_RANGE) / 100) + INPUT_VOLUME_MIN;
+ return retVal;
+}
+
+/*
+ * Writes Alsa mixer volume (0 - 100) to TSC2101 headset volume registry in
+ * a TSC2101 format. (0 - 63.5 db)
+ * In TSC2101 OSS driver this functionality was controlled with "SET_LINE" parameter.
+ */
+int set_mixer_volume_as_headset_gain_control_volume(int mixerVol)
+{
+ int volume;
+ int retVal;
+ u16 val;
+
+ if (mixerVol < 0 || mixerVol > 100) {
+ M_DPRINTK("Trying a bad headset mixer volume value(%d)!\n", mixerVol);
+ return -EPERM;
+ }
+ M_DPRINTK("mixer volume = %d\n", mixerVol);
+ /* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range */
+ /* NOTE: 0 is minimum volume and not mute */
+ volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HEADSET_GAIN_CTRL);
+ /* preserve the old mute settings */
+ val &= ~(HGC_ADPGA_HED(INPUT_VOLUME_MAX));
+ val |= HGC_ADPGA_HED(volume);
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HEADSET_GAIN_CTRL,
+ val);
+ retVal = 1;
+
+ M_DPRINTK("to registry = %d\n", val);
+ return retVal;
+}
+
+/*
+ * Writes Alsa mixer volume (0 - 100) to TSC2101 handset volume registry in
+ * a TSC2101 format. (0 - 63.5 db)
+ * In TSC2101 OSS driver this functionality was controlled with "SET_MIC" parameter.
+ */
+int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
+{
+ int volume;
+ int retVal;
+ u16 val;
+
+ if (mixerVol < 0 || mixerVol > 100) {
+ M_DPRINTK("Trying a bad mic mixer volume value(%d)!\n", mixerVol);
+ return -EPERM;
+ }
+ M_DPRINTK("mixer volume = %d\n", mixerVol);
+ /* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
+ * NOTE: 0 is minimum volume and not mute
+ */
+ volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
+ /* preserve the old mute settigns */
+ val &= ~(HNGC_ADPGA_HND(INPUT_VOLUME_MAX));
+ val |= HNGC_ADPGA_HND(volume);
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HANDSET_GAIN_CTRL,
+ val);
+ retVal = 1;
+
+ M_DPRINTK("to registry = %d\n", val);
+ return retVal;
+}
+
+void init_record_sources(void)
+{
+ /* Mute Analog Sidetone
+ * analog sidetone gain db?
+ * Cell Phone In not connected to ADC
+ * Input selected by MICSEL connected to ADC
+ */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_MIXER_PGA_CTRL,
+ MPC_ASTMU | MPC_ASTG(0x40) | ~MPC_CPADC | MPC_MICADC);
+ /* Set record source, Select MIC_INHED input for headset */
+ set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
+}
+
+void set_loudspeaker_to_playback_target(void)
+{
+ u16 val;
+
+ /* power down sp1, sp2 and loudspeaker */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_CODEC_POWER_CTRL,
+ CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
+ /* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
+ * 1dB AGC hysteresis
+ * MICes bias 2V
+ */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_4,
+ AC4_MB_HED(0));
+
+ /* DAC left and right routed to SPK1/SPK2
+ * SPK1/SPK2 unmuted
+ * keyclicks routed to SPK1/SPK2
+ */
+ val = AC5_DIFFIN |
+ AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
+ AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2 |
+ AC5_HDSCPTC;
+ val = val & ~AC5_HDSCPTC;
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_5,
+ val);
+
+ /* powerdown spk1/out32n and spk2 */
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_POWERDOWN_STS);
+ val = val & ~(~PS_SPK1FL | ~PS_HNDFL | PS_LSPKFL);
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_POWERDOWN_STS,
+ val);
+
+ /* routing selected to SPK1 goes to OUT8P/OUT84 alsa. (loudspeaker)
+ * analog sidetone routed to loudspeaker
+ * buzzer pga routed to loudspeaker
+ * keyclick routing to loudspeaker
+ * cellphone input routed to loudspeaker
+ * mic selection (control register 04h/page2) routed to cell phone output (CP_OUT)
+ * routing selected for SPK1 goes also to cellphone output (CP_OUT)
+ * OUT8P/OUT8N (loudspeakers) unmuted (0 = unmuted)
+ * Cellphone output is not muted (0 = unmuted)
+ * Enable loudspeaker short protection control (0 = enable protection)
+ * VGND short protection control (0 = enable protection)
+ */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_6,
+ AC6_SPL2LSK | AC6_AST2LSK | AC6_BUZ2LSK | AC6_KCL2LSK |
+ AC6_CPI2LSK | AC6_MIC2CPO | AC6_SPL2CPO |
+ ~AC6_MUTLSPK | ~AC6_MUTSPK2 | ~AC6_LDSCPTC | ~AC6_VGNDSCPTC);
+ current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
+}
+
+void set_headphone_to_playback_target(void)
+{
+ /* power down sp1, sp2 and loudspeaker */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_CODEC_POWER_CTRL,
+ CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
+ /* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled */
+ /* 1dB AGC hysteresis */
+ /* MICes bias 2V */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_4,
+ AC4_MB_HED(0));
+
+ /* DAC left and right routed to SPK2 */
+ /* SPK1/2 unmuted */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_5,
+ AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
+ AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2 |
+ AC5_HDSCPTC);
+
+ /* OUT8P/N muted, CPOUT muted */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_6,
+ AC6_MUTLSPK | AC6_MUTSPK2 | AC6_LDSCPTC |
+ AC6_VGNDSCPTC);
+ current_playback_target = PLAYBACK_TARGET_HEADPHONE;
+}
+
+/*
+ * Checks whether the headset is detected.
+ * If headset is detected, the type is returned. Type can be
+ * 0x01 = stereo headset detected
+ * 0x02 = cellurar headset detected
+ * 0x03 = stereo + cellurar headset detected
+ * If headset is not detected 0 is returned.
+ */
+u16 get_headset_detected(void)
+{
+ u16 curDetected;
+ u16 curType;
+ u16 curVal;
+
+ curType = 0; /* not detected */
+ curVal = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_7);
+ curDetected = curVal & AC7_HDDETFL;
+ if (curDetected) {
+ printk("headset detected, checking type from %d \n", curVal);
+ curType = ((curVal & 0x6000) >> 13);
+ printk("headset type detected = %d \n", curType);
+ }
+ else {
+ printk("headset not detected\n");
+ }
+ return curType;
+}
+
+void init_playback_targets(void)
+{
+ u16 val;
+
+ set_loudspeaker_to_playback_target();
+ /* Left line input volume control
+ * = SET_LINE in the OSS driver
+ */
+ set_mixer_volume_as_headset_gain_control_volume(DEFAULT_INPUT_VOLUME);
+
+ /* Set headset to be controllable by handset mixer
+ * AGC enable for handset input
+ * Handset input not muted
+ */
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HANDSET_GAIN_CTRL);
+ val = val | HNGC_AGCEN_HND;
+ val = val & ~HNGC_ADMUT_HND;
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HANDSET_GAIN_CTRL,
+ val);
+
+ /* mic input volume control
+ * SET_MIC in the OSS driver
+ */
+ set_mixer_volume_as_handset_gain_control_volume(DEFAULT_INPUT_VOLUME);
+
+ /* Left/Right headphone channel volume control
+ * Zero-cross detect on
+ */
+ set_mixer_volume_as_dac_gain_control_volume(DEFAULT_OUTPUT_VOLUME, DEFAULT_OUTPUT_VOLUME);
+ /* unmute */
+ dac_gain_control_unmute_control(1, 1);
+}
+
+/*
+ * Initializes tsc2101 recourd source (to line) and playback target (to loudspeaker)
+ */
+void snd_omap_init_mixer(void)
+{
+ FN_IN;
+
+ /* Headset/Hook switch detect enabled */
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_7,
+ AC7_DETECT);
+
+ init_record_sources();
+ init_playback_targets();
+
+ FN_OUT(0);
+}
+
+static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[PLAYBACK_TARGET_COUNT] = {
+ "Loudspeaker", "Headphone"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = PLAYBACK_TARGET_COUNT;
+ if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1) {
+ uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
+ }
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int __pcm_playback_target_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ucontrol->value.integer.value[0] = current_playback_target;
+ return 0;
+}
+
+static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ int retVal;
+ int curVal;
+
+ retVal = 0;
+ curVal = ucontrol->value.integer.value[0];
+ if ((curVal >= 0) &&
+ (curVal < PLAYBACK_TARGET_COUNT) &&
+ (curVal != current_playback_target)) {
+ if (curVal == 0) {
+ set_loudspeaker_to_playback_target();
+ }
+ else {
+ set_headphone_to_playback_target();
+ }
+ retVal = 1;
+ }
+ return retVal;
+}
+
+static int __pcm_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+/*
+ * Alsa mixer interface function for getting the volume read from the DGC in a
+ * 0 -100 alsa mixer format.
+ */
+static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 volL;
+ u16 volR;
+ u16 val;
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
+ M_DPRINTK("registry value = %d!\n", val);
+ volL = DGC_DALVL_EXTRACT(val);
+ volR = DGC_DARVL_EXTRACT(val);
+ /* make sure that other bits are not on */
+ volL = volL & ~DGC_DALMU;
+ volR = volR & ~DGC_DARMU;
+
+ volL = get_dac_gain_control_volume_as_mixer_volume(volL);
+ volR = get_dac_gain_control_volume_as_mixer_volume(volR);
+
+ ucontrol->value.integer.value[0] = volL; /* L */
+ ucontrol->value.integer.value[1] = volR; /* R */
+
+ M_DPRINTK("mixer volume left = %ld, right = %ld\n", ucontrol->value.integer.value[0], ucontrol->value.integer.value[1]);
+ return 0;
+}
+
+static int __pcm_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ return set_mixer_volume_as_dac_gain_control_volume(ucontrol->value.integer.value[0],
+ ucontrol->value.integer.value[1]);
+}
+
+static int __pcm_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+/*
+ * When DGC_DALMU (bit 15) is 1, the left channel is muted.
+ * When DGC_DALMU is 0, left channel is not muted.
+ * Same logic apply also for the right channel.
+ */
+static int __pcm_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
+
+ ucontrol->value.integer.value[0] = IS_DGC_DALMU_UNMUTED(val);
+ ucontrol->value.integer.value[1] = IS_DGC_DARMU_UNMUTED(val);
+ return 0;
+}
+
+static int __pcm_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ return dac_gain_control_unmute_control(ucontrol->value.integer.value[0],
+ ucontrol->value.integer.value[1]);
+}
+
+static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int __headset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val;
+ u16 vol;
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HEADSET_GAIN_CTRL);
+ M_DPRINTK("registry value = %d\n", val);
+ vol = HGC_ADPGA_HED_EXTRACT(val);
+ vol = vol & ~HGC_ADMUT_HED;
+
+ vol = get_headset_gain_control_volume_as_mixer_volume(vol);
+ ucontrol->value.integer.value[0] = vol;
+
+ M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int __headset_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ return set_mixer_volume_as_headset_gain_control_volume(ucontrol->value.integer.value[0]);
+}
+
+static int __headset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+/* When HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
+ * When HGC_ADMUT_HED is 0, headset is not muted.
+ */
+static int __headset_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HEADSET_GAIN_CTRL);
+ ucontrol->value.integer.value[0] = IS_DGC_HGCMU_UNMUTED(val);
+ return 0;
+}
+
+static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ int count = 0;
+ u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HEADSET_GAIN_CTRL);
+ /* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
+ * so if values are same, it's time to change the registry value...
+ */
+ if (ucontrol->value.integer.value[0] == GET_DGC_HGCMU_BIT_VALUE(val)) {
+ if (ucontrol->value.integer.value[0] == 0) {
+ /* mute --> turn bit on */
+ val = val | HGC_ADMUT_HED;
+ }
+ else {
+ /* unmute --> turn bit off */
+ val = val & ~HGC_ADMUT_HED;
+ }
+ count++;
+ M_DPRINTK("changed value, is_unmuted = %d\n", IS_DGC_HGCMU_UNMUTED(val));
+ }
+ if (count) {
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HEADSET_GAIN_CTRL,
+ val);
+ }
+ return count;
+}
+
+static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int __handset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val;
+ u16 vol;
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
+ M_DPRINTK("registry value = %d\n", val);
+ vol = HNGC_ADPGA_HND_EXTRACT(val);
+ vol = vol & ~HNGC_ADMUT_HND;
+ vol = get_handset_gain_control_volume_as_mixer_volume(vol);
+ ucontrol->value.integer.value[0] = vol;
+
+ M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int __handset_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ return set_mixer_volume_as_handset_gain_control_volume(ucontrol->value.integer.value[0]);
+}
+
+static int __handset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+/* When HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
+ * When HNGC_ADMUT_HND is 0, handset is not muted.
+ */
+static int __handset_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
+ ucontrol->value.integer.value[0] = IS_DGC_HNGCMU_UNMUTED(val);
+ return 0;
+}
+
+static int __handset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ int count = 0;
+ u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
+
+ /* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
+ * so if values are same, it's time to change the registry value...
+ */
+ if (ucontrol->value.integer.value[0] == GET_DGC_HNGCMU_BIT_VALUE(val)) {
+ if (ucontrol->value.integer.value[0] == 0) {
+ /* mute --> turn bit on */
+ val = val | HNGC_ADMUT_HND;
+ }
+ else {
+ /* unmute --> turn bit off */
+ val = val & ~HNGC_ADMUT_HND;
+ }
+ M_DPRINTK("changed value, is_unmuted = %d\n", IS_DGC_HNGCMU_UNMUTED(val));
+ count++;
+ }
+ if (count) {
+ omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_HANDSET_GAIN_CTRL,
+ val);
+ }
+ return count;
+}
+
+static snd_kcontrol_new_t tsc2101_control[] __devinitdata = {
+ {
+ .name = "Playback Playback Route",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __pcm_playback_target_info,
+ .get = __pcm_playback_target_get,
+ .put = __pcm_playback_target_put,
+ }, {
+ .name = "Master Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __pcm_playback_volume_info,
+ .get = __pcm_playback_volume_get,
+ .put = __pcm_playback_volume_put,
+ }, {
+ .name = "Master Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __pcm_playback_switch_info,
+ .get = __pcm_playback_switch_get,
+ .put = __pcm_playback_switch_put,
+ }, {
+ .name = "Headset Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 1,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __headset_playback_volume_info,
+ .get = __headset_playback_volume_get,
+ .put = __headset_playback_volume_put,
+ }, {
+ .name = "Headset Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 1,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __headset_playback_switch_info,
+ .get = __headset_playback_switch_get,
+ .put = __headset_playback_switch_put,
+ }, {
+ .name = "Handset Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 2,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __handset_playback_volume_info,
+ .get = __handset_playback_volume_get,
+ .put = __handset_playback_volume_put,
+ }, {
+ .name = "Handset Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 2,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __handset_playback_switch_info,
+ .get = __handset_playback_switch_get,
+ .put = __handset_playback_switch_put,
+ }
+};
+
+#ifdef CONFIG_PM
+
+void snd_omap_suspend_mixer(void)
+{
+}
+
+void snd_omap_resume_mixer(void)
+{
+ snd_omap_init_mixer();
+}
+#endif
+
+int snd_omap_mixer(struct snd_card_omap_codec *tsc2101)
+{
+ int i=0;
+ int err=0;
+
+ if (!tsc2101) {
+ return -EINVAL;
+ }
+ for (i=0; i < ARRAY_SIZE(tsc2101_control); i++) {
+ if ((err = snd_ctl_add(tsc2101->card,
+ snd_ctl_new1(&tsc2101_control[i],
+ tsc2101->card))) < 0) {
+ return err;
+ }
+ }
+ return 0;
+}
--- /dev/null
+/*
+ * sound/arm/omap/omap-alsa-tsc2101-mixer.c
+ *
+ * Alsa Driver for TSC2101 codec for OMAP platform boards.
+ *
+ * Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
+ * Everett Coleman II <gcc80x86@fuzzyneural.net>
+ *
+ * Based on the ideas in omap-aic23.c and sa11xx-uda1341.c
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
+ *
+ * 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.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
+ * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+ * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * History:
+ *
+ * 2006-03-01 Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
+ * Can switch between headset and loudspeaker playback,
+ * mute and unmute dgc, set dgc volume. Record source switch,
+ * keyclick, buzzer and headset volume and handset volume control
+ * are still missing.
+ */
+
+#ifndef OMAPALSATSC2101MIXER_H_
+#define OMAPALSATSC2101MIXER_H_
+
+#include <asm/hardware/tsc2101.h>
+#include <../drivers/ssi/omap-tsc2101.h>
+#include "omap-alsa-dma.h"
+
+/* tsc2101 DAC gain control volume specific */
+#define OUTPUT_VOLUME_MIN 0x7F // 1111111 = -63.5 DB
+#define OUTPUT_VOLUME_MAX 0x32 // 110010
+#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MIN - OUTPUT_VOLUME_MAX)
+
+/* use input vol of 75 for 0dB gain */
+#define INPUT_VOLUME_MIN 0x0
+#define INPUT_VOLUME_MAX 0x7D
+#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
+
+#define PLAYBACK_TARGET_COUNT 0x02
+#define PLAYBACK_TARGET_LOUDSPEAKER 0x00
+#define PLAYBACK_TARGET_HEADPHONE 0x01
+
+/* following are used for register 03h Mixer PGA control bits D7-D5 for selecting record source */
+#define REC_SRC_TARGET_COUNT 0x08
+#define REC_SRC_SINGLE_ENDED_MICIN_HED MPC_MICSEL(0) // oss code referred to MIXER_LINE
+#define REC_SRC_SINGLE_ENDED_MICIN_HND MPC_MICSEL(1) // oss code referred to MIXER_MIC
+#define REC_SRC_SINGLE_ENDED_AUX1 MPC_MICSEL(2)
+#define REC_SRC_SINGLE_ENDED_AUX2 MPC_MICSEL(3)
+#define REC_SRC_MICIN_HED_AND_AUX1 MPC_MICSEL(4)
+#define REC_SRC_MICIN_HED_AND_AUX2 MPC_MICSEL(5)
+#define REC_SRC_MICIN_HND_AND_AUX1 MPC_MICSEL(6)
+#define REC_SRC_MICIN_HND_AND_AUX2 MPC_MICSEL(7)
+
+#define DEFAULT_OUTPUT_VOLUME 90 // default output volume to dac dgc
+#define DEFAULT_INPUT_VOLUME 20 // default record volume
+
+#define TSC2101_AUDIO_CODEC_REGISTERS_PAGE2 (2)
+
+#endif /*OMAPALSATSC2101MIXER_H_*/
--- /dev/null
+/*
+ * arch/arm/mach-omap1/omap-alsa-tsc2101.c
+ *
+ * Alsa codec Driver for TSC2101 chip for OMAP platform boards.
+ * Code obtained from oss omap drivers
+ *
+ * Copyright (C) 2004 Texas Instruments, Inc.
+ * Written by Nishanth Menon and Sriram Kannan
+ *
+ * Copyright (C) 2006 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Alsa modularization by Daniel Petrini (d.pensator@gmail.com)
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * 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/delay.h>
+#include <linux/soundcard.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <asm/io.h>
+#include <asm/arch/mcbsp.h>
+
+#include <linux/slab.h>
+#ifdef CONFIG_PM
+#include <linux/pm.h>
+#endif
+#include <asm/mach-types.h>
+#include <asm/arch/dma.h>
+#include <asm/arch/clock.h>
+
+#include <asm/hardware/tsc2101.h>
+#include <../drivers/ssi/omap-tsc2101.h>
+
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-tsc2101.h"
+
+static struct clk *tsc2101_mclk = 0;
+
+//#define DUMP_TSC2101_AUDIO_REGISTERS
+#undef DUMP_TSC2101_AUDIO_REGISTERS
+
+/*
+ * Hardware capabilities
+ */
+
+/*
+ * DAC USB-mode sampling rates (MCLK = 12 MHz)
+ * The rates and rate_reg_into MUST be in the same order
+ */
+static unsigned int rates[] = {
+ 7350, 8000, 8018, 8727,
+ 8820, 9600, 11025, 12000,
+ 14700, 16000, 22050, 24000,
+ 29400, 32000, 44100, 48000,
+};
+
+static snd_pcm_hw_constraint_list_t tsc2101_hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static const struct tsc2101_samplerate_reg_info
+ rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
+ /* Div 6 */
+ {7350, 7, 1},
+ {8000, 7, 0},
+ /* Div 5.5 */
+ {8018, 6, 1},
+ {8727, 6, 0},
+ /* Div 5 */
+ {8820, 5, 1},
+ {9600, 5, 0},
+ /* Div 4 */
+ {11025, 4, 1},
+ {12000, 4, 0},
+ /* Div 3 */
+ {14700, 3, 1},
+ {16000, 3, 0},
+ /* Div 2 */
+ {22050, 2, 1},
+ {24000, 2, 0},
+ /* Div 1.5 */
+ {29400, 1, 1},
+ {32000, 1, 0},
+ /* Div 1 */
+ {44100, 0, 1},
+ {48000, 0, 0},
+};
+
+static snd_pcm_hardware_t tsc2101_snd_omap_alsa_playback = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 7350,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static snd_pcm_hardware_t tsc2101_snd_omap_alsa_capture = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 7350,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+/*
+ * Simplified write for tsc Audio
+ */
+inline void tsc2101_audio_write(u8 address, u16 data)
+{
+ omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS, address, data);
+}
+
+/*
+ * Simplified read for tsc Audio
+ */
+inline u16 tsc2101_audio_read(u8 address)
+{
+ return (omap_tsc2101_read(PAGE2_AUDIO_CODEC_REGISTERS, address));
+}
+
+#ifdef DUMP_TSC2101_AUDIO_REGISTERS
+void dump_tsc2101_audio_reg(void) {
+ printk("TSC2101_AUDIO_CTRL_1 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_1));
+ printk("TSC2101_HEADSET_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL));
+ printk("TSC2101_DAC_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL));
+ printk("TSC2101_MIXER_PGA_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_MIXER_PGA_CTRL));
+ printk("TSC2101_AUDIO_CTRL_2 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_2));
+ printk("TSC2101_CODEC_POWER_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_CODEC_POWER_CTRL));
+ printk("TSC2101_AUDIO_CTRL_3 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_3));
+ printk("TSC2101_LCH_BASS_BOOST_N0 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N0));
+ printk("TSC2101_LCH_BASS_BOOST_N1 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N1));
+ printk("TSC2101_LCH_BASS_BOOST_N2 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N2));
+ printk("TSC2101_LCH_BASS_BOOST_N3 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N3));
+ printk("TSC2101_LCH_BASS_BOOST_N4 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N4));
+ printk("TSC2101_LCH_BASS_BOOST_N5 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N5));
+ printk("TSC2101_LCH_BASS_BOOST_D1 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D1));
+ printk("TSC2101_LCH_BASS_BOOST_D2 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D2));
+ printk("TSC2101_LCH_BASS_BOOST_D4 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D4));
+ printk("TSC2101_LCH_BASS_BOOST_D5 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D5));
+
+ printk("TSC2101_RCH_BASS_BOOST_N0 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N0));
+ printk("TSC2101_RCH_BASS_BOOST_N1 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N1));
+ printk("TSC2101_RCH_BASS_BOOST_N2 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N2));
+ printk("TSC2101_RCH_BASS_BOOST_N3 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N3));
+ printk("TSC2101_RCH_BASS_BOOST_N4 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N4));
+ printk("TSC2101_RCH_BASS_BOOST_N5 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N5));
+ printk("TSC2101_RCH_BASS_BOOST_D1 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D1));
+ printk("TSC2101_RCH_BASS_BOOST_D2 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D2));
+ printk("TSC2101_RCH_BASS_BOOST_D4 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D4));
+ printk("TSC2101_RCH_BASS_BOOST_D5 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D5));
+
+ printk("TSC2101_PLL_PROG_1 = 0x%04x\n", tsc2101_audio_read(TSC2101_PLL_PROG_1));
+ printk("TSC2101_PLL_PROG_1 = 0x%04x\n", tsc2101_audio_read(TSC2101_PLL_PROG_2));
+ printk("TSC2101_AUDIO_CTRL_4 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+ printk("TSC2101_HANDSET_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL));
+ printk("TSC2101_BUZZER_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL));
+ printk("TSC2101_AUDIO_CTRL_5 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_5));
+ printk("TSC2101_AUDIO_CTRL_6 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_6));
+ printk("TSC2101_AUDIO_CTRL_7 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_7));
+ printk("TSC2101_GPIO_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_GPIO_CTRL));
+ printk("TSC2101_AGC_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_AGC_CTRL));
+ printk("TSC2101_POWERDOWN_STS = 0x%04x\n", tsc2101_audio_read(TSC2101_POWERDOWN_STS));
+ printk("TSC2101_MIC_AGC_CONTROL = 0x%04x\n", tsc2101_audio_read(TSC2101_MIC_AGC_CONTROL));
+ printk("TSC2101_CELL_AGC_CONTROL = 0x%04x\n", tsc2101_audio_read(TSC2101_CELL_AGC_CONTROL));
+}
+#endif
+
+/*
+ * ALSA operations according to board file
+ */
+
+/*
+ * Sample rate changing
+ */
+void tsc2101_set_samplerate(long sample_rate)
+{
+ u8 count = 0;
+ u16 data = 0;
+ int clkgdv = 0;
+
+ u16 srgr1, srgr2;
+ /* wait for any frame to complete */
+ udelay(125);
+ ADEBUG();
+
+ sample_rate = sample_rate;
+ /* Search for the right sample rate */
+ while ((rate_reg_info[count].sample_rate != sample_rate) &&
+ (count < NUMBER_SAMPLE_RATES_SUPPORTED)) {
+ count++;
+ }
+ if (count == NUMBER_SAMPLE_RATES_SUPPORTED) {
+ printk(KERN_ERR "Invalid Sample Rate %d requested\n",
+ (int) sample_rate);
+ return; // -EPERM;
+ }
+
+ /* Set AC1 */
+ data = tsc2101_audio_read(TSC2101_AUDIO_CTRL_1);
+ /* Clear prev settings */
+ data &= ~(AC1_DACFS(0x07) | AC1_ADCFS(0x07));
+ data |= AC1_DACFS(rate_reg_info[count].divisor) |
+ AC1_ADCFS(rate_reg_info[count].divisor);
+ tsc2101_audio_write(TSC2101_AUDIO_CTRL_1, data);
+
+ /* Set the AC3 */
+ data = tsc2101_audio_read(TSC2101_AUDIO_CTRL_3);
+ /*Clear prev settings */
+ data &= ~(AC3_REFFS | AC3_SLVMS);
+ data |= (rate_reg_info[count].fs_44kHz) ? AC3_REFFS : 0;
+#ifdef TSC_MASTER
+ data |= AC3_SLVMS;
+#endif /* #ifdef TSC_MASTER */
+ tsc2101_audio_write(TSC2101_AUDIO_CTRL_3, data);
+
+ /* program the PLLs */
+ if (rate_reg_info[count].fs_44kHz) {
+ /* 44.1 khz - 12 MHz Mclk */
+ tsc2101_audio_write(TSC2101_PLL_PROG_1, PLL1_PLLSEL |
+ PLL1_PVAL(1) | PLL1_I_VAL(7)); /* PVAL 1; I_VAL 7 */
+ tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x1490)); /* D_VAL 5264 */
+ } else {
+ /* 48 khz - 12 Mhz Mclk */
+ tsc2101_audio_write(TSC2101_PLL_PROG_1, PLL1_PLLSEL |
+ PLL1_PVAL(1) | PLL1_I_VAL(8)); /* PVAL 1; I_VAL 8 */
+ tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x780)); /* D_VAL 1920 */
+ }
+
+ /* Set the sample rate */
+#ifndef TSC_MASTER
+ clkgdv = CODEC_CLOCK / (sample_rate * (DEFAULT_BITPERSAMPLE * 2 - 1));
+ if (clkgdv)
+ srgr1 = (FWID(DEFAULT_BITPERSAMPLE - 1) | CLKGDV(clkgdv));
+ else
+ return (1);
+
+ /* Stereo Mode */
+ srgr2 = (CLKSM | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1));
+#else
+ srgr1 = (FWID(DEFAULT_BITPERSAMPLE - 1) | CLKGDV(clkgdv));
+ srgr2 = ((GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1)));
+
+#endif /* end of #ifdef TSC_MASTER */
+ OMAP_MCBSP_WRITE(OMAP1610_MCBSP1_BASE, SRGR2, srgr2);
+ OMAP_MCBSP_WRITE(OMAP1610_MCBSP1_BASE, SRGR1, srgr1);
+}
+
+void tsc2101_configure(void)
+{
+}
+
+/*
+ * Omap MCBSP clock and Power Management configuration
+ *
+ * Here we have some functions that allows clock to be enabled and
+ * disabled only when needed. Besides doing clock configuration
+ * it allows turn on/turn off audio when necessary.
+ */
+
+/*
+ * Do clock framework mclk search
+ */
+void tsc2101_clock_setup(void)
+{
+ tsc2101_mclk = clk_get(0, "mclk");
+}
+
+/*
+ * Do some sanity check, set clock rate, starts it and turn codec audio on
+ */
+int tsc2101_clock_on(void)
+{
+ int curUseCount;
+ uint curRate;
+ int err;
+
+ curUseCount = clk_get_usecount(tsc2101_mclk);
+ DPRINTK("clock use count = %d\n", curUseCount);
+ if (curUseCount > 0) {
+ // MCLK is already in use
+ printk(KERN_WARNING
+ "MCLK already in use at %d Hz. We change it to %d Hz\n",
+ (uint) clk_get_rate(tsc2101_mclk),
+ CODEC_CLOCK);
+ }
+ curRate = (uint)clk_get_rate(tsc2101_mclk);
+ DPRINTK("old clock rate = %d\n", curRate);
+ if (curRate != CODEC_CLOCK) {
+ err = clk_set_rate(tsc2101_mclk, CODEC_CLOCK);
+ if (err) {
+ printk(KERN_WARNING
+ "Cannot set MCLK clock rate for TSC2101 CODEC, error code = %d\n", err);
+ //return -ECANCELED;
+ }
+ }
+ else
+ {
+ printk(KERN_INFO
+ "omap_alsa_tsc2101_clock_on(), no need to change rate, no need to change clock rate, rate already %d Hz.\n",
+ CODEC_CLOCK);
+ }
+ err = clk_enable(tsc2101_mclk);
+ curRate = (uint)clk_get_rate(tsc2101_mclk);
+ curUseCount = clk_get_usecount(tsc2101_mclk);
+ DPRINTK("MCLK = %d [%d], usecount = %d, clk_enable retval = %d\n",
+ curRate,
+ CODEC_CLOCK,
+ curUseCount,
+ err);
+
+ // Now turn the audio on
+ omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS,
+ TSC2101_CODEC_POWER_CTRL,
+ 0x0000);
+ return 0;
+}
+
+/*
+ * Do some sanity check, turn clock off and then turn
+ * codec audio off
+ */
+int tsc2101_clock_off(void)
+{
+ int curUseCount;
+ int curRate;
+
+ curUseCount = clk_get_usecount(tsc2101_mclk);
+ DPRINTK("clock use count = %d\n", curUseCount);
+ if (curUseCount > 0) {
+ curRate = clk_get_rate(tsc2101_mclk);
+ DPRINTK("clock rate = %d\n", curRate);
+ if (curRate != CODEC_CLOCK) {
+ printk(KERN_WARNING
+ "MCLK for audio should be %d Hz. But is %d Hz\n",
+ (uint) clk_get_rate(tsc2101_mclk),
+ CODEC_CLOCK);
+ }
+ clk_disable(tsc2101_mclk);
+ DPRINTK("clock disabled\n");
+ }
+ tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
+ ~(CPC_SP1PWDN | CPC_SP2PWDN | CPC_BASSBC));
+ DPRINTK("audio codec off\n");
+#ifdef DUMP_TSC2101_AUDIO_REGISTERS
+ printk("tsc2101_clock_off()\n");
+ dump_tsc2101_audio_reg();
+#endif
+ return 0;
+}
+
+int tsc2101_get_default_samplerate(void)
+{
+ return DEFAULT_SAMPLE_RATE;
+}
+
+static int __init snd_omap_alsa_tsc2101_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct omap_alsa_codec_config *codec_cfg;
+
+ codec_cfg = pdev->dev.platform_data;
+ if (codec_cfg != NULL) {
+ codec_cfg->hw_constraints_rates = &tsc2101_hw_constraints_rates;
+ codec_cfg->snd_omap_alsa_playback = &tsc2101_snd_omap_alsa_playback;
+ codec_cfg->snd_omap_alsa_capture = &tsc2101_snd_omap_alsa_capture;
+ codec_cfg->codec_configure_dev = tsc2101_configure;
+ codec_cfg->codec_set_samplerate = tsc2101_set_samplerate;
+ codec_cfg->codec_clock_setup = tsc2101_clock_setup;
+ codec_cfg->codec_clock_on = tsc2101_clock_on;
+ codec_cfg->codec_clock_off = tsc2101_clock_off;
+ codec_cfg->get_default_samplerate = tsc2101_get_default_samplerate;
+ ret = snd_omap_alsa_post_probe(pdev, codec_cfg);
+ }
+ else
+ ret = -ENODEV;
+ return ret;
+}
+
+static struct platform_driver omap_alsa_driver = {
+ .probe = snd_omap_alsa_tsc2101_probe,
+ .remove = snd_omap_alsa_remove,
+ .suspend = snd_omap_alsa_suspend,
+ .resume = snd_omap_alsa_resume,
+ .driver = {
+ .name = "omap_alsa_mcbsp",
+ },
+};
+
+static int __init omap_alsa_tsc2101_init(void)
+{
+ int err;
+
+ ADEBUG();
+ err = platform_driver_register(&omap_alsa_driver);
+
+ return err;
+}
+
+static void __exit omap_alsa_tsc2101_exit(void)
+{
+ ADEBUG();
+ platform_driver_unregister(&omap_alsa_driver);
+}
+
+module_init(omap_alsa_tsc2101_init);
+module_exit(omap_alsa_tsc2101_exit);
--- /dev/null
+/*
+ * arch/arc/mach-omap1/omap-alsa-tsc2101.h
+ *
+ * Alsa Driver for TSC2101 codec for OMAP platform boards.
+ *
+ * Based on former omap-aic23.h and tsc2101 OSS drivers.
+ * Copyright (C) 2004 Texas Instruments, Inc.
+ * Written by Nishanth Menon and Sriram Kannan
+ *
+ * Copyright (C) 2006 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Alsa modularization by Daniel Petrini (d.pensator@gmail.com)
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * 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.
+ */
+
+#ifndef OMAP_ALSA_TSC2101_H_
+#define OMAP_ALSA_TSC2101_H_
+
+#include <linux/types.h>
+
+/* Define to set the tsc as the master w.r.t McBSP */
+#define TSC_MASTER
+
+#define NUMBER_SAMPLE_RATES_SUPPORTED 16
+
+/*
+ * AUDIO related MACROS
+ */
+#ifndef DEFAULT_BITPERSAMPLE
+#define DEFAULT_BITPERSAMPLE 16
+#endif
+
+#define DEFAULT_SAMPLE_RATE 44100
+#define CODEC_CLOCK 12000000
+#define AUDIO_MCBSP OMAP_MCBSP1
+
+#define PAGE2_AUDIO_CODEC_REGISTERS (2)
+
+struct tsc2101_samplerate_reg_info {
+ u16 sample_rate;
+ u8 divisor;
+ u8 fs_44kHz; /* if 0 48 khz, if 1 44.1 khz fsref */
+};
+
+/*
+ * Defines codec specific functions pointers that can be used from the
+ * common omap-alse base driver for all omap codecs. (tsc2101 and aic23)
+ */
+inline void tsc2101_configure(void);
+void tsc2101_set_samplerate(long rate);
+void tsc2101_clock_setup(void);
+int tsc2101_clock_on(void);
+int tsc2101_clock_off(void);
+int tsc2101_get_default_samplerate(void);
+
+#endif /*OMAP_ALSA_TSC2101_H_*/
--- /dev/null
+/*
+ * sound/arm/omap-alsa.c
+ *
+ * Alsa Driver for OMAP
+ *
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Written by Daniel Petrini, David Cohen, Anderson Briglia
+ * {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * Based on sa11xx-uda1341.c,
+ * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
+ *
+ * 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.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
+ * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+ * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * History:
+ *
+ * 2005-07-29 INdT Kernel Team - Alsa driver for omap osk. Creation of new
+ * file omap-aic23.c
+ *
+ * 2005-12-18 Dirk Behme - Added L/R Channel Interchange fix as proposed
+ * by Ajaya Babu
+ *
+ */
+
+#include <linux/platform_device.h>
+#ifdef CONFIG_PM
+#include <linux/pm.h>
+#endif
+#include <sound/driver.h>
+#include <sound/core.h>
+
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-dma.h"
+
+MODULE_AUTHOR("Mika Laitio, Daniel Petrini, David Cohen, Anderson Briglia - INdT");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("OMAP driver for ALSA");
+MODULE_ALIAS("omap_alsa_mcbsp.1");
+
+static char *id = NULL;
+static struct snd_card_omap_codec *alsa_codec = NULL;
+static struct omap_alsa_codec_config *alsa_codec_config = NULL;
+
+/*
+ * HW interface start and stop helper functions
+ */
+static int audio_ifc_start(void)
+{
+ omap_mcbsp_start(AUDIO_MCBSP);
+ return 0;
+}
+
+static int audio_ifc_stop(void)
+{
+ omap_mcbsp_stop(AUDIO_MCBSP);
+ return 0;
+}
+
+static void omap_alsa_audio_init(struct snd_card_omap_codec *omap_alsa)
+{
+ /* Setup DMA stuff */
+ omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].id = "Alsa omap out";
+ omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id =
+ SNDRV_PCM_STREAM_PLAYBACK;
+ omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev =
+ OMAP_DMA_MCBSP1_TX;
+ omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].hw_start =
+ audio_ifc_start;
+ omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].hw_stop =
+ audio_ifc_stop;
+
+ omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].id = "Alsa omap in";
+ omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].stream_id =
+ SNDRV_PCM_STREAM_CAPTURE;
+ omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev =
+ OMAP_DMA_MCBSP1_RX;
+ omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].hw_start =
+ audio_ifc_start;
+ omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].hw_stop =
+ audio_ifc_stop;
+}
+
+/*
+ * DMA functions
+ * Depends on omap-alsa-dma.c functions and (omap) dma.c
+ *
+ */
+static int audio_dma_request(struct audio_stream *s,
+ void (*callback) (void *))
+{
+ int err;
+ ADEBUG();
+
+ err = omap_request_alsa_sound_dma(s->dma_dev, s->id, s, &s->lch);
+ if (err < 0)
+ printk(KERN_ERR "Unable to grab audio dma 0x%x\n", s->dma_dev);
+ return err;
+}
+
+static int audio_dma_free(struct audio_stream *s)
+{
+ int err = 0;
+ ADEBUG();
+
+ err = omap_free_alsa_sound_dma(s, &s->lch);
+ if (err < 0)
+ printk(KERN_ERR "Unable to free audio dma channels!\n");
+ return err;
+}
+
+/*
+ * This function should calculate the current position of the dma in the
+ * buffer. It will help alsa middle layer to continue update the buffer.
+ * Its correctness is crucial for good functioning.
+ */
+static u_int audio_get_dma_pos(struct audio_stream *s)
+{
+ snd_pcm_substream_t *substream = s->stream;
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ unsigned int offset;
+ unsigned long flags;
+ dma_addr_t count;
+ ADEBUG();
+
+ /* this must be called w/ interrupts locked as requested in dma.c */
+ spin_lock_irqsave(&s->dma_lock, flags);
+
+ /* For the current period let's see where we are */
+ count = omap_get_dma_src_addr_counter(s->lch[s->dma_q_head]);
+
+ spin_unlock_irqrestore(&s->dma_lock, flags);
+
+ /* Now, the position related to the end of that period */
+ offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
+
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+
+ return offset;
+}
+
+/*
+ * this stops the dma and clears the dma ptrs
+ */
+static void audio_stop_dma(struct audio_stream *s)
+{
+ unsigned long flags;
+ ADEBUG();
+
+ spin_lock_irqsave(&s->dma_lock, flags);
+ s->active = 0;
+ s->period = 0;
+ s->periods = 0;
+
+ /* this stops the dma channel and clears the buffer ptrs */
+ omap_stop_alsa_sound_dma(s);
+
+ omap_clear_alsa_sound_dma(s);
+
+ spin_unlock_irqrestore(&s->dma_lock, flags);
+}
+
+/*
+ * Main dma routine, requests dma according where you are in main alsa buffer
+ */
+static void audio_process_dma(struct audio_stream *s)
+{
+ snd_pcm_substream_t *substream = s->stream;
+ snd_pcm_runtime_t *runtime;
+ unsigned int dma_size;
+ unsigned int offset;
+ int ret;
+#ifdef CONFIG_MACH_OMAP_H6300
+ unsigned long flags;
+#endif
+
+ ADEBUG();
+ runtime = substream->runtime;
+ if (s->active) {
+ dma_size = frames_to_bytes(runtime, runtime->period_size);
+ offset = dma_size * s->period;
+ snd_assert(dma_size <= DMA_BUF_SIZE,);
+#ifdef CONFIG_MACH_OMAP_H6300
+ spin_lock_irqsave(&s->dma_lock, flags);
+ omap_stop_alsa_sound_dma(s);
+ spin_unlock_irqrestore(&s->dma_lock, flags);
+#endif
+ ret = omap_start_alsa_sound_dma(s,
+ (dma_addr_t) runtime->dma_area +
+ offset, dma_size);
+ if (ret) {
+ printk(KERN_ERR
+ "audio_process_dma: cannot queue DMA buffer (%i)\n",
+ ret);
+ return;
+ }
+
+ s->period++;
+ s->period %= runtime->periods;
+ s->periods++;
+ s->offset = offset;
+ }
+}
+
+/*
+ * This is called when dma IRQ occurs at the end of each transmited block
+ */
+void callback_omap_alsa_sound_dma(void *data)
+{
+ struct audio_stream *s = data;
+
+ ADEBUG();
+ /*
+ * If we are getting a callback for an active stream then we inform
+ * the PCM middle layer we've finished a period
+ */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ spin_lock(&s->dma_lock);
+ if (s->periods > 0)
+ s->periods--;
+
+ audio_process_dma(s);
+ spin_unlock(&s->dma_lock);
+}
+
+/*
+ * Alsa section
+ * PCM settings and callbacks
+ */
+static int snd_omap_alsa_trigger(snd_pcm_substream_t * substream, int cmd)
+{
+ struct snd_card_omap_codec *chip =
+ snd_pcm_substream_chip(substream);
+ int stream_id = substream->pstr->stream;
+ struct audio_stream *s = &chip->s[stream_id];
+ int err = 0;
+
+ ADEBUG();
+ /* note local interrupts are already disabled in the midlevel code */
+ spin_lock(&s->dma_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* requested stream startup */
+ s->active = 1;
+ audio_process_dma(s);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ /* requested stream shutdown */
+ audio_stop_dma(s);
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&s->dma_lock);
+
+ return err;
+}
+
+static int snd_omap_alsa_prepare(snd_pcm_substream_t * substream)
+{
+ struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ struct audio_stream *s = &chip->s[substream->pstr->stream];
+
+ ADEBUG();
+ /* set requested samplerate */
+ alsa_codec_config->codec_set_samplerate(runtime->rate);
+ chip->samplerate = runtime->rate;
+
+ s->period = 0;
+ s->periods = 0;
+
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_omap_alsa_pointer(snd_pcm_substream_t *substream)
+{
+ struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
+
+ ADEBUG();
+ return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
+}
+
+static int snd_card_omap_alsa_open(snd_pcm_substream_t * substream)
+{
+ struct snd_card_omap_codec *chip =
+ snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ int stream_id = substream->pstr->stream;
+ int err;
+
+ ADEBUG();
+ chip->s[stream_id].stream = substream;
+ alsa_codec_config->codec_clock_on();
+ if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = *(alsa_codec_config->snd_omap_alsa_playback);
+ else
+ runtime->hw = *(alsa_codec_config->snd_omap_alsa_capture);
+
+ if ((err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ if ((err = snd_pcm_hw_constraint_list(runtime,
+ 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ alsa_codec_config->hw_constraints_rates)) < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_card_omap_alsa_close(snd_pcm_substream_t * substream)
+{
+ struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
+
+ ADEBUG();
+ alsa_codec_config->codec_clock_off();
+ chip->s[substream->pstr->stream].stream = NULL;
+
+ return 0;
+}
+
+/* HW params & free */
+static int snd_omap_alsa_hw_params(snd_pcm_substream_t * substream,
+ snd_pcm_hw_params_t * hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_omap_alsa_hw_free(snd_pcm_substream_t * substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* pcm operations */
+static snd_pcm_ops_t snd_card_omap_alsa_playback_ops = {
+ .open = snd_card_omap_alsa_open,
+ .close = snd_card_omap_alsa_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_omap_alsa_hw_params,
+ .hw_free = snd_omap_alsa_hw_free,
+ .prepare = snd_omap_alsa_prepare,
+ .trigger = snd_omap_alsa_trigger,
+ .pointer = snd_omap_alsa_pointer,
+};
+
+static snd_pcm_ops_t snd_card_omap_alsa_capture_ops = {
+ .open = snd_card_omap_alsa_open,
+ .close = snd_card_omap_alsa_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_omap_alsa_hw_params,
+ .hw_free = snd_omap_alsa_hw_free,
+ .prepare = snd_omap_alsa_prepare,
+ .trigger = snd_omap_alsa_trigger,
+ .pointer = snd_omap_alsa_pointer,
+};
+
+/*
+ * Alsa init and exit section
+ *
+ * Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
+ */
+static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa,
+ int device)
+{
+ snd_pcm_t *pcm;
+ int err;
+
+ ADEBUG();
+ if ((err = snd_pcm_new(omap_alsa->card, "OMAP PCM", device, 1, 1, &pcm)) < 0)
+ return err;
+
+ /* sets up initial buffer with continuous allocation */
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data
+ (GFP_KERNEL),
+ 128 * 1024, 128 * 1024);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_card_omap_alsa_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_card_omap_alsa_capture_ops);
+ pcm->private_data = omap_alsa;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "omap alsa pcm");
+
+ omap_alsa_audio_init(omap_alsa);
+
+ /* setup DMA controller */
+ audio_dma_request(&omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK],
+ callback_omap_alsa_sound_dma);
+ audio_dma_request(&omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE],
+ callback_omap_alsa_sound_dma);
+
+ omap_alsa->pcm = pcm;
+
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+/*
+ * Driver suspend/resume - calls alsa functions. Some hints from aaci.c
+ */
+int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_card_omap_codec *chip;
+ snd_card_t *card = platform_get_drvdata(pdev);
+
+ if (card->power_state != SNDRV_CTL_POWER_D3hot) {
+ chip = card->private_data;
+ if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ /* Mutes and turn clock off */
+ alsa_codec_config->codec_clock_off();
+ snd_omap_suspend_mixer();
+ }
+ }
+ return 0;
+}
+
+int snd_omap_alsa_resume(struct platform_device *pdev)
+{
+ struct snd_card_omap_codec *chip;
+ snd_card_t *card = platform_get_drvdata(pdev);
+
+ if (card->power_state != SNDRV_CTL_POWER_D0) {
+ chip = card->private_data;
+ if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+ alsa_codec_config->codec_clock_on();
+ snd_omap_resume_mixer();
+ }
+ }
+ return 0;
+}
+
+#endif /* CONFIG_PM */
+
+void snd_omap_alsa_free(snd_card_t * card)
+{
+ struct snd_card_omap_codec *chip = card->private_data;
+ ADEBUG();
+
+ /*
+ * Turn off codec after it is done.
+ * Can't do it immediately, since it may still have
+ * buffered data.
+ */
+ schedule_timeout_interruptible(2);
+
+ omap_mcbsp_stop(AUDIO_MCBSP);
+ omap_mcbsp_free(AUDIO_MCBSP);
+
+ audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
+ audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
+}
+
+/* module init & exit */
+
+/*
+ * Inits alsa soudcard structure.
+ * Called by the probe method in codec after function pointers has been set.
+ */
+int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_codec_config *config)
+{
+ int err = 0;
+ int def_rate;
+ snd_card_t *card;
+
+ ADEBUG();
+ alsa_codec_config = config;
+
+ alsa_codec_config->codec_clock_setup();
+ alsa_codec_config->codec_clock_on();
+
+ omap_mcbsp_request(AUDIO_MCBSP);
+ omap_mcbsp_stop(AUDIO_MCBSP);
+ omap_mcbsp_config(AUDIO_MCBSP, alsa_codec_config->mcbsp_regs_alsa);
+ omap_mcbsp_start(AUDIO_MCBSP);
+
+ if (alsa_codec_config && alsa_codec_config->codec_configure_dev)
+ alsa_codec_config->codec_configure_dev();
+
+ alsa_codec_config->codec_clock_off();
+
+ /* register the soundcard */
+ card = snd_card_new(-1, id, THIS_MODULE, sizeof(alsa_codec));
+ if (card == NULL)
+ goto nodev1;
+
+ alsa_codec = kcalloc(1, sizeof(*alsa_codec), GFP_KERNEL);
+ if (alsa_codec == NULL)
+ goto nodev2;
+
+ card->private_data = (void *)alsa_codec;
+ card->private_free = snd_omap_alsa_free;
+
+ alsa_codec->card = card;
+ def_rate = alsa_codec_config->get_default_samplerate();
+ alsa_codec->samplerate = def_rate;
+
+ spin_lock_init(&alsa_codec->s[0].dma_lock);
+ spin_lock_init(&alsa_codec->s[1].dma_lock);
+
+ /* mixer */
+ if ((err = snd_omap_mixer(alsa_codec)) < 0)
+ goto nodev3;
+
+ /* PCM */
+ if ((err = snd_card_omap_alsa_pcm(alsa_codec, 0)) < 0)
+ goto nodev3;
+
+ strcpy(card->driver, "OMAP_ALSA");
+ strcpy(card->shortname, alsa_codec_config->name);
+ sprintf(card->longname, alsa_codec_config->name);
+
+ snd_omap_init_mixer();
+ snd_card_set_dev(card, &pdev->dev);
+
+ if ((err = snd_card_register(card)) == 0) {
+ printk(KERN_INFO "audio support initialized\n");
+ platform_set_drvdata(pdev, card);
+ return 0;
+ }
+
+nodev3:
+ kfree(alsa_codec);
+nodev2:
+ snd_card_free(card);
+nodev1:
+ omap_mcbsp_stop(AUDIO_MCBSP);
+ omap_mcbsp_free(AUDIO_MCBSP);
+
+ return err;
+}
+
+int snd_omap_alsa_remove(struct platform_device *pdev)
+{
+ snd_card_t *card = platform_get_drvdata(pdev);
+ struct snd_card_omap_codec *chip = card->private_data;
+
+ snd_card_free(card);
+
+ alsa_codec = NULL;
+ card->private_data = NULL;
+ kfree(chip);
+
+ platform_set_drvdata(pdev, NULL);
+
+ return 0;
+}
projects / linux-2.6-omap-h63xx.git / commitdiff