Say Y or M if you want to support any AC97 codec attached to
the PXA2xx AC97 interface.
+config SND_OMAP_AIC23
+ tristate "OMAP AIC23 alsa driver (osk5912)"
+ depends on ARCH_OMAP && SND
+ select SND_PCM
+ select SENSORS_TLV320AIC23
+ help
+ Say Y here if you have a OSK platform board
+ and want to use its AIC23 audio chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-omap-aic23.
+
endmenu
snd-aaci-objs := aaci.o devdma.o
snd-pxa2xx-pcm-objs := pxa2xx-pcm.o
snd-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-omap-aic23-objs := omap-aic23.o omap-alsa-dma.o omap-alsa-mixer.o
obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o
obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o
obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o
+obj-$(CONFIG_SND_OMAP_AIC23) += snd-omap-aic23.o
--- /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
+ */
+
+#include <linux/config.h>
+#include <sound/driver.h>
+#include <linux/module.h>
+#include <linux/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>
+
+#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/hardware/clock.h>
+#include <asm/arch/mcbsp.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,
+};
+
+
+/*
+ * Codec/mcbsp init and configuration section
+ * codec dependent code.
+ */
+
+extern int tlv320aic23_write_value(u8 reg, u16 value);
+
+/* TLV320AIC23 is a write only device */
+__inline__ void audio_aic23_write(u8 address, u16 data)
+{
+ tlv320aic23_write_value(address, data);
+}
+
+/*
+ * 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_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;
+
+ /* 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->pm_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->pm_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 device *dev, pm_message_t state, u32 level)
+{
+ snd_card_t *card = dev_get_drvdata(dev);
+
+ if (card->power_state != SNDRV_CTL_POWER_D3hot) {
+ snd_omap_aic23_suspend(card, 0);
+ }
+ return 0;
+}
+
+static int omap_aic23_resume(struct device *dev, u32 level)
+{
+ snd_card_t *card = dev_get_drvdata(dev);
+
+ 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_use(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_unuse(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 device *dev)
+{
+ 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;
+
+ snd_card_set_pm_callback(card, snd_omap_aic23_suspend,
+ snd_omap_aic23_resume, omap_aic23);
+
+ strcpy(card->driver, "AIC23");
+ strcpy(card->shortname, "OSK AIC23");
+ sprintf(card->longname, "OMAP OSK with AIC23");
+
+ snd_omap_init_mixer();
+
+ if ((err = snd_card_register(card)) == 0) {
+ printk(KERN_INFO "OSK audio support initialized\n");
+ dev_set_drvdata(dev, card);
+ return 0;
+ }
+
+nodev:
+ snd_omap_aic23_free(card);
+
+ return err;
+}
+
+static int snd_omap_aic23_remove(struct device *dev)
+{
+ snd_card_t *card = dev_get_drvdata(dev);
+ struct snd_card_omap_aic23 *chip = card->private_data;
+
+ snd_card_free(card);
+
+ omap_aic23 = NULL;
+ card->private_data = NULL;
+ kfree(chip);
+
+ dev_set_drvdata(dev, NULL);
+
+ return 0;
+
+}
+
+static struct device_driver omap_alsa_driver = {
+ .name = "omap_mcbsp",
+ .bus = &platform_bus_type,
+ .probe = snd_omap_aic23_probe,
+ .remove = snd_omap_aic23_remove,
+ .suspend = omap_aic23_suspend,
+ .resume = omap_aic23_resume,
+};
+
+static int __init omap_aic23_init(void)
+{
+ int err;
+ ADEBUG();
+
+ err = driver_register(&omap_alsa_driver);
+
+ return err;
+}
+
+static void __exit omap_aic23_exit(void)
+{
+ ADEBUG();
+
+ driver_unregister(&omap_alsa_driver);
+}
+
+module_init(omap_aic23_init);
+module_exit(omap_aic23_exit);
--- /dev/null
+/*
+ * sound/arm/omap-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
+ *
+ * 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
+ * -------
+ *
+ * 2005/07/25 INdT-10LE Kernel Team - Alsa driver for omap osk,
+ * original version based in sa1100 driver
+ * and omap oss driver.
+ *
+ */
+
+#ifndef __OMAP_AIC23_H
+#define __OMAP_AIC23_H
+
+#include <sound/driver.h>
+#include <asm/arch/dma.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#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
+
+/*
+ * Buffer management for alsa and dma
+ */
+struct audio_stream {
+ char *id; /* identification string */
+ int stream_id; /* numeric identification */
+ int dma_dev; /* dma number of that device */
+ int *lch; /* Chain of channels this stream is linked to */
+ char started; /* to store if the chain was started or not */
+ int dma_q_head; /* DMA Channel Q Head */
+ int dma_q_tail; /* DMA Channel Q Tail */
+ char dma_q_count; /* DMA Channel Q Count */
+ int active:1; /* we are using this stream for transfer now */
+ int period; /* current transfer period */
+ int periods; /* current count of periods registerd in the DMA engine */
+ spinlock_t dma_lock; /* for locking in DMA operations */
+ snd_pcm_substream_t *stream; /* the pcm stream */
+ unsigned linked:1; /* dma channels linked */
+ int offset; /* store start position of the last period in the alsa buffer */
+};
+
+/*
+ * Alsa card structure for aic23
+ */
+struct snd_card_omap_aic23 {
+ snd_card_t *card;
+ snd_pcm_t *pcm;
+ long samplerate;
+ struct audio_stream s[2]; /* playback & capture */
+};
+
+/*********** Function Prototypes *************************/
+
+void audio_dma_callback(void *);
+int snd_omap_mixer(struct snd_card_omap_aic23 *);
+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
+
+#endif
--- /dev/null
+/*
+ * sound/arm/omap-alsa-dma.c
+ *
+ * Common audio DMA handling for the OMAP processors
+ *
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ *
+ * Copyright (C) 2004 Texas Instruments, Inc.
+ *
+ * Copyright (C) 2000, 2001 Nicolas Pitre <nico@cam.org>
+ *
+ * This package is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * History:
+ *
+ * 2004-06-07 Sriram Kannan - Created new file from omap_audio_dma_intfc.c. This file
+ * will contain only the DMA interface and buffer handling of OMAP
+ * audio driver.
+ *
+ * 2004-06-22 Sriram Kannan - removed legacy code (auto-init). Self-linking of DMA logical channel.
+ *
+ * 2004-08-12 Nishanth Menon - Modified to integrate Audio requirements on 1610,1710 platforms
+ *
+ * 2004-11-01 Nishanth Menon - 16xx platform code base modified to support multi channel chaining.
+ *
+ * 2004-12-15 Nishanth Menon - Improved 16xx platform channel logic introduced - tasklets, queue handling updated
+ *
+ * 2005-07-19 INdT Kernel Team - Alsa port. Creation of new file omap-alsa-dma.c based in
+ * omap-audio-dma-intfc.c oss file. Support for aic23 codec.
+ * Removal of buffer handling (Alsa does that), modifications
+ * in dma handling and port to alsa structures.
+ */
+
+#include <linux/config.h>
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/fs.h>
+#include <linux/mm.h>
+#include <linux/slab.h>
+#include <linux/sched.h>
+#include <linux/poll.h>
+#include <linux/pm.h>
+#include <linux/errno.h>
+#include <linux/sound.h>
+#include <linux/soundcard.h>
+#include <linux/sysrq.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+
+#include <asm/uaccess.h>
+#include <asm/io.h>
+#include <asm/hardware.h>
+#include <asm/semaphore.h>
+
+#include <asm/arch/dma.h>
+#include "omap-alsa-dma.h"
+
+#include <asm/arch/mcbsp.h>
+
+#include "omap-aic23.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);
+
+/* Channel Queue Handling macros
+ * tail always points to the current free entry
+ * Head always points to the current entry being used
+ * end is either head or tail
+ */
+
+#define AUDIO_QUEUE_INIT(s) s->dma_q_head = s->dma_q_tail = s->dma_q_count = 0;
+#define AUDIO_QUEUE_FULL(s) (nr_linked_channels == s->dma_q_count)
+#define AUDIO_QUEUE_LAST(s) (1 == s->dma_q_count)
+#define AUDIO_QUEUE_EMPTY(s) (0 == s->dma_q_count)
+#define __AUDIO_INCREMENT_QUEUE(end) ((end)=((end)+1) % nr_linked_channels)
+#define AUDIO_INCREMENT_HEAD(s) __AUDIO_INCREMENT_QUEUE(s->dma_q_head); s->dma_q_count--;
+#define AUDIO_INCREMENT_TAIL(s) __AUDIO_INCREMENT_QUEUE(s->dma_q_tail); s->dma_q_count++;
+
+/* DMA buffer fragmentation sizes */
+#define MAX_DMA_SIZE 0x1000000 /* todo: sync with alsa */
+//#define CUT_DMA_SIZE 0x1000
+/* TODO: To be moved to more appropriate location */
+#define DCSR_ERROR 0x3
+#define DCSR_END_BLOCK (1 << 5)
+#define DCSR_SYNC_SET (1 << 6)
+
+#define DCCR_FS (1 << 5)
+#define DCCR_PRIO (1 << 6)
+#define DCCR_EN (1 << 7)
+#define DCCR_AI (1 << 8)
+#define DCCR_REPEAT (1 << 9)
+/* if 0 the channel works in 3.1 compatible mode*/
+#define DCCR_N31COMP (1 << 10)
+#define DCCR_EP (1 << 11)
+#define DCCR_SRC_AMODE_BIT 12
+#define DCCR_SRC_AMODE_MASK (0x3<<12)
+#define DCCR_DST_AMODE_BIT 14
+#define DCCR_DST_AMODE_MASK (0x3<<14)
+#define AMODE_CONST 0x0
+#define AMODE_POST_INC 0x1
+#define AMODE_SINGLE_INDEX 0x2
+#define AMODE_DOUBLE_INDEX 0x3
+
+/**************************** DATA STRUCTURES *****************************************/
+
+static spinlock_t dma_list_lock = SPIN_LOCK_UNLOCKED;
+
+static char nr_linked_channels = 1;
+
+/*********************************** MODULE SPECIFIC FUNCTIONS ***********************/
+
+static void sound_dma_irq_handler(int lch, u16 ch_status, void *data);
+static int audio_set_dma_params_play(int channel, dma_addr_t dma_ptr,
+ u_int dma_size);
+static int audio_set_dma_params_capture(int channel, dma_addr_t dma_ptr,
+ u_int dma_size);
+static int audio_start_dma_chain(struct audio_stream * s);
+
+/***************************************************************************************
+ *
+ * DMA channel requests
+ *
+ **************************************************************************************/
+static void omap_sound_dma_link_lch(void *data)
+{
+
+ struct audio_stream *s = (struct audio_stream *) data;
+ int *chan = s->lch;
+ int i;
+
+ FN_IN;
+ if (s->linked) {
+ FN_OUT(1);
+ return;
+ }
+ for (i = 0; i < nr_linked_channels; i++) {
+ int cur_chan = chan[i];
+ int nex_chan =
+ ((nr_linked_channels - 1 ==
+ i) ? chan[0] : chan[i + 1]);
+ omap_dma_link_lch(cur_chan, nex_chan);
+ }
+ s->linked = 1;
+ FN_OUT(0);
+}
+
+int omap_request_sound_dma(int device_id, const char *device_name,
+ void *data, int **channels)
+{
+ int i, err = 0;
+ int *chan = NULL;
+ FN_IN;
+ if (unlikely((NULL == channels) || (NULL == device_name))) {
+ BUG();
+ return -EPERM;
+ }
+ /* Try allocate memory for the num channels */
+ *channels =
+ (int *) kmalloc(sizeof(int) * nr_linked_channels, GFP_KERNEL);
+ chan = *channels;
+ if (NULL == chan) {
+ ERR("No Memory for channel allocs!\n");
+ FN_OUT(-ENOMEM);
+ return -ENOMEM;
+ }
+ 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]);
+
+ /* Handle Failure condition here */
+ if (err < 0) {
+ int j;
+ for (j = 0; j < i; j++) {
+ omap_free_dma(chan[j]);
+ }
+ spin_unlock(&dma_list_lock);
+ kfree(chan);
+ *channels = NULL;
+ ERR("Error in requesting channel %d=0x%x\n", i,
+ err);
+ FN_OUT(err);
+ return err;
+ }
+ }
+
+ /* Chain the channels together */
+ if (!cpu_is_omap1510())
+ omap_sound_dma_link_lch(data);
+
+ spin_unlock(&dma_list_lock);
+ FN_OUT(0);
+ return 0;
+}
+
+/***************************************************************************************
+ *
+ * DMA channel requests Freeing
+ *
+ **************************************************************************************/
+static void omap_sound_dma_unlink_lch(void *data)
+{
+ struct audio_stream *s = (struct audio_stream *) data;
+ int *chan = s->lch;
+ int i;
+
+ FN_IN;
+ if (!s->linked) {
+ FN_OUT(1);
+ return;
+ }
+ for (i = 0; i < nr_linked_channels; i++) {
+ int cur_chan = chan[i];
+ int nex_chan =
+ ((nr_linked_channels - 1 ==
+ i) ? chan[0] : chan[i + 1]);
+ omap_dma_unlink_lch(cur_chan, nex_chan);
+ }
+ s->linked = 0;
+ FN_OUT(0);
+}
+
+int omap_free_sound_dma(void *data, int **channels)
+{
+
+ int i;
+ int *chan = NULL;
+ FN_IN;
+ if (unlikely(NULL == channels)) {
+ BUG();
+ return -EPERM;
+ }
+ if (unlikely(NULL == *channels)) {
+ BUG();
+ return -EPERM;
+ }
+ chan = (*channels);
+
+ if (!cpu_is_omap1510())
+ omap_sound_dma_unlink_lch(data);
+ for (i = 0; i < nr_linked_channels; i++) {
+ int cur_chan = chan[i];
+ omap_stop_dma(cur_chan);
+ omap_free_dma(cur_chan);
+ }
+ kfree(*channels);
+ *channels = NULL;
+ FN_OUT(0);
+ return 0;
+}
+
+/***************************************************************************************
+ *
+ * Stop all the DMA channels of the stream
+ *
+ **************************************************************************************/
+void omap_audio_stop_dma(struct audio_stream *s)
+{
+ int *chan = s->lch;
+ int i;
+ FN_IN;
+ if (unlikely(NULL == chan)) {
+ BUG();
+ return;
+ }
+ for (i = 0; i < nr_linked_channels; i++) {
+ int cur_chan = chan[i];
+ omap_stop_dma(cur_chan);
+ }
+ s->started = 0;
+ FN_OUT(0);
+ return;
+}
+/***************************************************************************************
+ *
+ * Clear any pending transfers
+ *
+ **************************************************************************************/
+void omap_clear_sound_dma(struct audio_stream * s)
+{
+ FN_IN;
+ omap_clear_dma(s->lch[s->dma_q_head]);
+ FN_OUT(0);
+ return;
+}
+
+/*********************************** MODULE FUNCTIONS DEFINTIONS ***********************/
+
+#ifdef OMAP1610_MCBSP1_BASE
+#undef OMAP1610_MCBSP1_BASE
+#endif
+#define OMAP1610_MCBSP1_BASE 0xE1011000
+
+/***************************************************************************************
+ *
+ * DMA related functions
+ *
+ **************************************************************************************/
+static int audio_set_dma_params_play(int channel, dma_addr_t dma_ptr,
+ u_int dma_size)
+{
+ 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));
+ omap_set_dma_src_params(channel, 0x00, 0x01, dma_ptr);
+ omap_set_dma_transfer_params(channel, dt, cen, cfn, 0x00);
+ FN_OUT(0);
+ return 0;
+}
+
+static int audio_set_dma_params_capture(int channel, dma_addr_t dma_ptr,
+ u_int dma_size)
+{
+ 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));
+ omap_set_dma_dest_params(channel, 0x00, 0x01, dma_ptr);
+ omap_set_dma_transfer_params(channel, dt, cen, cfn, 0x00);
+ FN_OUT(0);
+ return 0;
+}
+
+static int audio_start_dma_chain(struct audio_stream *s)
+{
+ int channel = s->lch[s->dma_q_head];
+ FN_IN;
+ if (!s->started) {
+ omap_start_dma(channel);
+ s->started = 1;
+ }
+ /* else the dma itself will progress forward with out our help */
+ FN_OUT(0);
+ return 0;
+}
+
+/* Start DMA -
+ * 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 ret = -EPERM;
+
+ FN_IN;
+
+ if (unlikely(dma_size > MAX_DMA_SIZE)) {
+ ERR("DmaSoundDma: Start: overflowed %d-%d\n", dma_size,
+ MAX_DMA_SIZE);
+ return -EOVERFLOW;
+ }
+ //if (AUDIO_QUEUE_FULL(s)) {
+ // ret = -2;
+ // goto sound_out;
+ //}
+
+ if (s->stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
+ /*playback */
+ ret =
+ audio_set_dma_params_play(s->lch[s->dma_q_tail],
+ dma_ptr, dma_size);
+ } else {
+ ret =
+ audio_set_dma_params_capture(s->lch[s->dma_q_tail],
+ dma_ptr, dma_size);
+ }
+ if (ret != 0) {
+ ret = -3; /* indicate queue full */
+ goto sound_out;
+ }
+ AUDIO_INCREMENT_TAIL(s);
+ ret = audio_start_dma_chain(s);
+ if (ret) {
+ ERR("dma start failed");
+ }
+ sound_out:
+ FN_OUT(ret);
+ return ret;
+
+}
+
+/*
+ * ISRs have to be short and smart..
+ * Here we call alsa handling, after some error checking
+ */
+static void sound_dma_irq_handler(int sound_curr_lch, u16 ch_status,
+ void *data)
+{
+ int dma_status = ch_status;
+ struct audio_stream *s = (struct audio_stream *) data;
+ FN_IN;
+
+ /*
+ * some register checkings
+ */
+ DPRINTK("lch=%d,status=0x%x, dma_status=%d, data=%p\n",
+ sound_curr_lch, ch_status, dma_status, data);
+
+ if (dma_status & (DCSR_ERROR)) {
+ omap_writew(omap_readw(OMAP_DMA_CCR(sound_curr_lch)) &
+ ~DCCR_EN, OMAP_DMA_CCR(sound_curr_lch));
+ ERR("DCSR_ERROR!\n");
+ FN_OUT(-1);
+ return;
+ }
+
+ if (ch_status & DCSR_END_BLOCK)
+ audio_dma_callback(s);
+ FN_OUT(0);
+ return;
+}
+
+MODULE_AUTHOR("Texas Instruments");
+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);
--- /dev/null
+/*
+ * linux/sound/arm/omap-alsa-dma.h
+ *
+ * Common audio DMA handling for the OMAP processors
+ *
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ *
+ * Copyright (C) 2004 Texas Instruments, Inc.
+ *
+ * Copyright (C) 2000, 2001 Nicolas Pitre <nico@cam.org>
+ *
+ * This package is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * History:
+ *
+ *
+ * 2004/08/12 Nishanth Menon - Modified to integrate Audio requirements on 1610,1710 platforms
+ *
+ * 2005/07/25 INdT Kernel Team - Renamed to omap-alsa-dma.h. Ported to Alsa.
+ */
+
+#ifndef __OMAP_AUDIO_ALSA_DMA_H
+#define __OMAP_AUDIO_ALSA_DMA_H
+
+/************************** 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)
+
+/************************** GLOBAL DATA STRUCTURES *********************************/
+
+typedef void (*dma_callback_t) (int lch, u16 ch_status, void *data);
+
+/**************** ARCH SPECIFIC FUNCIONS *******************************************/
+
+void omap_clear_sound_dma(struct audio_stream * s);
+
+int omap_request_sound_dma(int device_id, const char *device_name,
+ void *data, int **channels);
+int omap_free_sound_dma(void *data, int **channels);
+
+int omap_start_sound_dma(struct audio_stream *s, dma_addr_t dma_ptr,
+ u_int dma_size);
+
+void omap_audio_stop_dma(struct audio_stream *s);
+
+#endif
--- /dev/null
+/*
+ * sound/arm/omap-alsa-mixer.c
+ *
+ * Alsa Driver Mixer for generic codecs for omap boards
+ *
+ * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Written by David Cohen, Daniel Petrini
+ * {david.cohen, daniel.petrini}@indt.org.br
+ *
+ * Based on es1688_lib.c,
+ * Copyright (c) by Jaroslav Kysela <perex@suse.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:
+ *
+ * 2005-08-02 INdT Kernel Team - Alsa mixer driver for omap osk. Creation of new
+ * file omap-alsa-mixer.c. Initial version
+ * with aic23 codec for osk5912
+ */
+
+#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 <sound/initval.h>
+#include <sound/control.h>
+
+MODULE_AUTHOR("David Cohen, Daniel Petrini - INdT");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("OMAP Alsa mixer driver for ALSA");
+
+/*
+ * Codec dependent region
+ */
+
+/* Codec AIC23 */
+#ifdef CONFIG_SENSORS_TLV320AIC23
+
+extern __inline__ void audio_aic23_write(u8, u16);
+
+#define MIXER_NAME "Mixer AIC23"
+#define SND_OMAP_WRITE(reg, val) audio_aic23_write(reg, val)
+
+#endif
+
+/* Callback Functions */
+#define OMAP_BOOL(xname, xindex, reg, reg_index, mask, invert) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_omap_info_bool, \
+ .get = snd_omap_get_bool, \
+ .put = snd_omap_put_bool, \
+ .private_value = reg | (reg_index << 8) | (invert << 10) | (mask << 12) \
+}
+
+#define OMAP_MUX(xname, reg, reg_index, mask) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .info = snd_omap_info_mux, \
+ .get = snd_omap_get_mux, \
+ .put = snd_omap_put_mux, \
+ .private_value = reg | (reg_index << 8) | (mask << 10) \
+}
+
+#define OMAP_SINGLE(xname, xindex, reg, reg_index, reg_val, mask) \
+{\
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_omap_info_single, \
+ .get = snd_omap_get_single, \
+ .put = snd_omap_put_single, \
+ .private_value = reg | (reg_val << 8) | (reg_index << 16) | (mask << 18) \
+}
+
+#define OMAP_DOUBLE(xname, xindex, left_reg, right_reg, reg_index, mask) \
+{\
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_omap_info_double, \
+ .get = snd_omap_get_double, \
+ .put = snd_omap_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (reg_index << 16) | (mask << 18) \
+}
+
+/* Local Registers */
+enum snd_device_index {
+ PCM_INDEX = 0,
+ LINE_INDEX,
+ AAC_INDEX, /* Analog Audio Control: reg = l_reg */
+};
+
+struct {
+ u16 l_reg;
+ u16 r_reg;
+ u8 sw;
+} omap_regs[3];
+
+#ifdef CONFIG_PM
+struct {
+ u16 l_reg;
+ u16 r_reg;
+ u8 sw;
+} omap_pm_regs[3];
+#endif
+
+u16 snd_sidetone[6] = {
+ SIDETONE_18,
+ SIDETONE_12,
+ SIDETONE_9,
+ SIDETONE_6,
+ SIDETONE_0,
+ 0
+};
+
+/* Begin Bool Functions */
+
+static int snd_omap_info_bool(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;
+}
+
+static int snd_omap_get_bool(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ int mic_index = (kcontrol->private_value >> 8) & 0x03;
+ u16 mask = (kcontrol->private_value >> 12) & 0xff;
+ int invert = (kcontrol->private_value >> 10) & 0x03;
+
+ if (invert)
+ ucontrol->value.integer.value[0] = (omap_regs[mic_index].l_reg & mask) ? 0 : 1;
+ else
+ ucontrol->value.integer.value[0] = (omap_regs[mic_index].l_reg & mask) ? 1 : 0;
+
+ return 0;
+}
+
+static int snd_omap_put_bool(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ int mic_index = (kcontrol->private_value >> 8) & 0x03;
+ u16 mask = (kcontrol->private_value >> 12) & 0xff;
+ u16 reg = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 10) & 0x03;
+
+ int changed = 1;
+
+ if (ucontrol->value.integer.value[0]) /* XOR */
+ if (invert)
+ omap_regs[mic_index].l_reg &= ~mask;
+ else
+ omap_regs[mic_index].l_reg |= mask;
+ else
+ if (invert)
+ omap_regs[mic_index].l_reg |= mask;
+ else
+ omap_regs[mic_index].l_reg &= ~mask;
+
+ SND_OMAP_WRITE(reg, omap_regs[mic_index].l_reg);
+
+ return changed;
+}
+
+/* End Bool Functions */
+
+/* Begin Mux Functions */
+
+static int snd_omap_info_mux(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ /* Mic = 0
+ * Line = 1 */
+ static char *texts[2] = { "Mic", "Line" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_omap_get_mux(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ u16 mask = (kcontrol->private_value >> 10) & 0xff;
+ int mux_index = (kcontrol->private_value >> 8) & 0x03;
+
+ ucontrol->value.enumerated.item[0] = (omap_regs[mux_index].l_reg & mask) ? 0 /* Mic */ : 1 /* Line */;
+
+ return 0;
+}
+
+static int snd_omap_put_mux(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ u16 reg = kcontrol->private_value & 0xff;
+ u16 mask = (kcontrol->private_value >> 10) & 0xff;
+ int mux_index = (kcontrol->private_value >> 8) & 0x03;
+
+ int changed = 1;
+
+ if (!ucontrol->value.integer.value[0])
+ omap_regs[mux_index].l_reg |= mask; /* AIC23: Mic */
+ else
+ omap_regs[mux_index].l_reg &= ~mask; /* AIC23: Line */
+
+ SND_OMAP_WRITE(reg, omap_regs[mux_index].l_reg);
+
+ return changed;
+}
+
+/* End Mux Functions */
+
+/* Begin Single Functions */
+
+static int snd_omap_info_single(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ int mask = (kcontrol->private_value >> 18) & 0xff;
+ int reg_val = (kcontrol->private_value >> 8) & 0xff;
+
+ uinfo->type = mask ? SNDRV_CTL_ELEM_TYPE_INTEGER : SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = reg_val-1;
+
+ return 0;
+}
+
+static int snd_omap_get_single(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ u16 reg_val = (kcontrol->private_value >> 8) & 0xff;
+
+ ucontrol->value.integer.value[0] = snd_sidetone[reg_val];
+
+ return 0;
+}
+
+static int snd_omap_put_single(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ u16 reg_index = (kcontrol->private_value >> 16) & 0x03;
+ u16 mask = (kcontrol->private_value >> 18) & 0x1ff;
+ u16 reg = kcontrol->private_value & 0xff;
+ u16 reg_val = (kcontrol->private_value >> 8) & 0xff;
+
+ int changed = 0;
+
+ /* Volume */
+ if ((omap_regs[reg_index].l_reg != (ucontrol->value.integer.value[0] & mask)))
+ {
+ changed = 1;
+
+ omap_regs[reg_index].l_reg &= ~mask;
+ omap_regs[reg_index].l_reg |= snd_sidetone[ucontrol->value.integer.value[0]];
+
+ snd_sidetone[reg_val] = ucontrol->value.integer.value[0];
+ SND_OMAP_WRITE(reg, omap_regs[reg_index].l_reg);
+ }
+ else
+ changed = 0;
+
+ return changed;
+}
+
+/* End Single Functions */
+
+/* Begin Double Functions */
+
+static int snd_omap_info_double(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ /* mask == 0 : Switch
+ * mask != 0 : Volume */
+ int mask = (kcontrol->private_value >> 18) & 0xff;
+
+ uinfo->type = mask ? SNDRV_CTL_ELEM_TYPE_INTEGER : SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = mask ? 2 : 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask ? mask : 1;
+
+ return 0;
+}
+
+static int snd_omap_get_double(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ /* mask == 0 : Switch
+ * mask != 0 : Volume */
+ int mask = (kcontrol->private_value >> 18) & 0xff;
+ int vol_index = (kcontrol->private_value >> 16) & 0x03;
+
+ if (!mask)
+ /* Switch */
+ ucontrol->value.integer.value[0] = omap_regs[vol_index].sw;
+ else
+ {
+ /* Volume */
+ ucontrol->value.integer.value[0] = omap_regs[vol_index].l_reg;
+ ucontrol->value.integer.value[1] = omap_regs[vol_index].r_reg;
+ }
+
+ return 0;
+}
+
+static int snd_omap_put_double(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ /* mask == 0 : Switch
+ * mask != 0 : Volume */
+ int vol_index = (kcontrol->private_value >> 16) & 0x03;
+ int mask = (kcontrol->private_value >> 18) & 0xff;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+
+ int changed = 0;
+
+ if (!mask)
+ {
+ /* Switch */
+ if (!ucontrol->value.integer.value[0])
+ {
+ SND_OMAP_WRITE(left_reg, 0x00);
+ SND_OMAP_WRITE(right_reg, 0x00);
+ }
+ else
+ {
+ SND_OMAP_WRITE(left_reg, omap_regs[vol_index].l_reg);
+ SND_OMAP_WRITE(right_reg, omap_regs[vol_index].r_reg);
+ }
+ changed = 1;
+ omap_regs[vol_index].sw = ucontrol->value.integer.value[0];
+ }
+ else
+ {
+ /* Volume */
+ if ((omap_regs[vol_index].l_reg != (ucontrol->value.integer.value[0] & mask)) ||
+ (omap_regs[vol_index].r_reg != (ucontrol->value.integer.value[1] & mask)))
+ {
+ changed = 1;
+
+ omap_regs[vol_index].l_reg &= ~mask;
+ omap_regs[vol_index].r_reg &= ~mask;
+ omap_regs[vol_index].l_reg |= (ucontrol->value.integer.value[0] & mask);
+ omap_regs[vol_index].r_reg |= (ucontrol->value.integer.value[1] & mask);
+ if (omap_regs[vol_index].sw)
+ {
+ /* write to registers only if sw is actived */
+ SND_OMAP_WRITE(left_reg, omap_regs[vol_index].l_reg);
+ SND_OMAP_WRITE(right_reg, omap_regs[vol_index].r_reg);
+ }
+ }
+ else
+ changed = 0;
+ }
+
+ return changed;
+}
+
+/* End Double Functions */
+
+static snd_kcontrol_new_t snd_omap_controls[] = {
+ OMAP_DOUBLE("PCM Playback Switch", 0, LEFT_CHANNEL_VOLUME_ADDR, RIGHT_CHANNEL_VOLUME_ADDR,
+ PCM_INDEX, 0x00),
+ OMAP_DOUBLE("PCM Playback Volume", 0, LEFT_CHANNEL_VOLUME_ADDR, RIGHT_CHANNEL_VOLUME_ADDR,
+ PCM_INDEX, OUTPUT_VOLUME_MASK),
+ OMAP_BOOL("Line Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, BYPASS_ON, 0),
+ OMAP_DOUBLE("Line Capture Switch", 0, LEFT_LINE_VOLUME_ADDR, RIGHT_LINE_VOLUME_ADDR,
+ LINE_INDEX, 0x00),
+ OMAP_DOUBLE("Line Capture Volume", 0, LEFT_LINE_VOLUME_ADDR, RIGHT_LINE_VOLUME_ADDR,
+ LINE_INDEX, INPUT_VOLUME_MASK),
+ OMAP_BOOL("Mic Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, STE_ENABLED, 0),
+ OMAP_SINGLE("Mic Playback Volume", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, 5, SIDETONE_MASK),
+ OMAP_BOOL("Mic Capture Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, MICM_MUTED, 1),
+ OMAP_BOOL("Mic Booster Playback Switch", 0, ANALOG_AUDIO_CONTROL_ADDR, AAC_INDEX, MICB_20DB, 0),
+ 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)
+{
+ /* Saves current values to wake-up correctly */
+ omap_pm_regs[LINE_INDEX].l_reg = omap_regs[LINE_INDEX].l_reg;
+ omap_pm_regs[LINE_INDEX].r_reg = omap_regs[LINE_INDEX].l_reg;
+ omap_pm_regs[LINE_INDEX].sw = omap_regs[LINE_INDEX].sw;
+
+ omap_pm_regs[AAC_INDEX].l_reg = omap_regs[AAC_INDEX].l_reg;
+
+ omap_pm_regs[PCM_INDEX].l_reg = omap_regs[PCM_INDEX].l_reg;
+ omap_pm_regs[PCM_INDEX].r_reg = omap_regs[PCM_INDEX].r_reg;
+ omap_pm_regs[PCM_INDEX].sw = omap_regs[PCM_INDEX].sw;
+}
+
+void snd_omap_resume_mixer(void)
+{
+ /* Line's saved values */
+ omap_regs[LINE_INDEX].l_reg = omap_pm_regs[LINE_INDEX].l_reg;
+ omap_regs[LINE_INDEX].r_reg = omap_pm_regs[LINE_INDEX].l_reg;
+ omap_regs[LINE_INDEX].sw = omap_pm_regs[LINE_INDEX].sw;
+ SND_OMAP_WRITE(LEFT_LINE_VOLUME_ADDR, omap_pm_regs[LINE_INDEX].l_reg);
+ SND_OMAP_WRITE(RIGHT_LINE_VOLUME_ADDR, omap_pm_regs[LINE_INDEX].l_reg);
+
+ /* Analog Audio Control's saved values */
+ omap_regs[AAC_INDEX].l_reg = omap_pm_regs[AAC_INDEX].l_reg;
+ SND_OMAP_WRITE(ANALOG_AUDIO_CONTROL_ADDR, omap_regs[AAC_INDEX].l_reg);
+
+ /* Headphone's saved values */
+ omap_regs[PCM_INDEX].l_reg = omap_pm_regs[PCM_INDEX].l_reg;
+ omap_regs[PCM_INDEX].r_reg = omap_pm_regs[PCM_INDEX].r_reg;
+ omap_regs[PCM_INDEX].sw = omap_pm_regs[PCM_INDEX].sw;
+ SND_OMAP_WRITE(LEFT_CHANNEL_VOLUME_ADDR, omap_pm_regs[PCM_INDEX].l_reg);
+ SND_OMAP_WRITE(RIGHT_CHANNEL_VOLUME_ADDR, omap_pm_regs[PCM_INDEX].r_reg);
+}
+#endif
+
+int snd_omap_mixer(struct snd_card_omap_aic23 *chip)
+{
+ snd_card_t *card;
+ unsigned int idx;
+ int err;
+
+ snd_assert(chip != NULL && chip->card != NULL, return -EINVAL);
+
+ card = chip->card;
+
+ strcpy(card->mixername, MIXER_NAME);
+
+ /* Registering alsa mixer controls */
+ for (idx = 0; idx < ARRAY_SIZE(snd_omap_controls); idx++)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_omap_controls[idx], chip))) < 0)
+ return err;
+
+ return 0;
+}
+
projects / linux-2.6-omap-h63xx.git / commitdiff