From 776201015011f139629c433b461dd89c96fb7fff Mon Sep 17 00:00:00 2001 From: "Daniel Petrini, David Cohen, Anderson Briglia" <> Date: Mon, 5 Sep 2005 11:18:57 +0300 Subject: [PATCH] [PATCH] ARM: OMAP: Alsa driver for osk board - new version (3) Alsa Driver for omap osk5912 --- sound/arm/Kconfig | 12 + sound/arm/Makefile | 2 + sound/arm/omap-aic23.c | 912 ++++++++++++++++++++++++++++++++++++ sound/arm/omap-aic23.h | 114 +++++ sound/arm/omap-alsa-dma.c | 450 ++++++++++++++++++ sound/arm/omap-alsa-dma.h | 59 +++ sound/arm/omap-alsa-mixer.c | 496 ++++++++++++++++++++ 7 files changed, 2045 insertions(+) create mode 100644 sound/arm/omap-aic23.c create mode 100644 sound/arm/omap-aic23.h create mode 100644 sound/arm/omap-alsa-dma.c create mode 100644 sound/arm/omap-alsa-dma.h create mode 100644 sound/arm/omap-alsa-mixer.c diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 2e4a5e0d16d..cf071547a13 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -33,4 +33,16 @@ config SND_PXA2XX_AC97 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 diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 103f136926d..8c0c38c1bd1 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -6,8 +6,10 @@ snd-sa11xx-uda1341-objs := sa11xx-uda1341.o 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 diff --git a/sound/arm/omap-aic23.c b/sound/arm/omap-aic23.c new file mode 100644 index 00000000000..38a7aa71796 --- /dev/null +++ b/sound/arm/omap-aic23.c @@ -0,0 +1,912 @@ +/* + * 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 + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#ifdef CONFIG_PM +#include +#endif + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#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); diff --git a/sound/arm/omap-aic23.h b/sound/arm/omap-aic23.h new file mode 100644 index 00000000000..0c0f6ea50f5 --- /dev/null +++ b/sound/arm/omap-aic23.h @@ -0,0 +1,114 @@ +/* + * 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 +#include +#include +#include + +#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 diff --git a/sound/arm/omap-alsa-dma.c b/sound/arm/omap-alsa-dma.c new file mode 100644 index 00000000000..abb1bc4e865 --- /dev/null +++ b/sound/arm/omap-alsa-dma.c @@ -0,0 +1,450 @@ +/* + * 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 + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include "omap-alsa-dma.h" + +#include + +#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); diff --git a/sound/arm/omap-alsa-dma.h b/sound/arm/omap-alsa-dma.h new file mode 100644 index 00000000000..2ac7650abe3 --- /dev/null +++ b/sound/arm/omap-alsa-dma.h @@ -0,0 +1,59 @@ +/* + * 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 + * + * 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 diff --git a/sound/arm/omap-alsa-mixer.c b/sound/arm/omap-alsa-mixer.c new file mode 100644 index 00000000000..f8cad080c79 --- /dev/null +++ b/sound/arm/omap-alsa-mixer.c @@ -0,0 +1,496 @@ +/* + * 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 + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "omap-aic23.h" +#include +#include + +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; +} + -- 2.41.1