February 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] [PATCH] ALSA: Remove deprecated AU1X00 AC97 driver
by Lars-Peter Clausen 08 Feb '16

08 Feb '16

The AU1X00 AC97 ALSA driver was deprecated in commit 7137c6bcb7ff ("ALSA: deprecate MIPS AU1X00 AC97 driver") in favor of the newer and better ASoC driver for the same hardware. This was almost 5 years ago and this driver has not been in use in the mainline kernel since, it should be safe to remove it at this point. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 2153d31..4a47050 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -23,17 +23,5 @@ config SND_SGI_HAL2 help Sound support for the SGI Indy and Indigo2 Workstation. - -config SND_AU1X00 - tristate "Au1x00 AC97 Port Driver (DEPRECATED)" - depends on MIPS_ALCHEMY - select SND_PCM - select SND_AC97_CODEC - help - ALSA Sound driver for the Au1x00's AC97 port. - - Newer drivers for ASoC are available, please do not use - this driver as it will be removed in the future. - endif # SND_MIPS diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 861ec0a..b977c44 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -2,11 +2,9 @@ # Makefile for ALSA # -snd-au1x00-objs := au1x00.o snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency -obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c deleted file mode 100644 index 1e30e84..0000000 --- a/sound/mips/au1x00.c +++ /dev/null @@ -1,734 +0,0 @@ -/* - * BRIEF MODULE DESCRIPTION - * Driver for AMD Au1000 MIPS Processor, AC'97 Sound Port - * - * Copyright 2004 Cooper Street Innovations Inc. - * Author: Charles Eidsness <charles(a)cooper-street.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * 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: - * - * 2004-09-09 Charles Eidsness -- Original verion -- based on - * sa11xx-uda1341.c ALSA driver and the - * au1000.c OSS driver. - * 2004-09-09 Matt Porter -- Added support for ALSA 1.0.6 - * - */ - -#include <linux/ioport.h> -#include <linux/interrupt.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/ac97_codec.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1000_dma.h> - -MODULE_AUTHOR("Charles Eidsness <charles(a)cooper-street.com>"); -MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}"); - -#define PLAYBACK 0 -#define CAPTURE 1 -#define AC97_SLOT_3 0x01 -#define AC97_SLOT_4 0x02 -#define AC97_SLOT_6 0x08 -#define AC97_CMD_IRQ 31 -#define READ 0 -#define WRITE 1 -#define READ_WAIT 2 -#define RW_DONE 3 - -struct au1000_period -{ - u32 start; - u32 relative_end; /*realtive to start of buffer*/ - struct au1000_period * next; -}; - -/*Au1000 AC97 Port Control Reisters*/ -struct au1000_ac97_reg { - u32 volatile config; - u32 volatile status; - u32 volatile data; - u32 volatile cmd; - u32 volatile cntrl; -}; - -struct audio_stream { - struct snd_pcm_substream *substream; - int dma; - spinlock_t dma_lock; - struct au1000_period * buffer; - unsigned int period_size; - unsigned int periods; -}; - -struct snd_au1000 { - struct snd_card *card; - struct au1000_ac97_reg volatile *ac97_ioport; - - struct resource *ac97_res_port; - spinlock_t ac97_lock; - struct snd_ac97 *ac97; - - struct snd_pcm *pcm; - struct audio_stream *stream[2]; /* playback & capture */ - int dmaid[2]; /* tx(0)/rx(1) DMA ids */ -}; - -/*--------------------------- Local Functions --------------------------------*/ -static void -au1000_set_ac97_xmit_slots(struct snd_au1000 *au1000, long xmit_slots) -{ - u32 volatile ac97_config; - - spin_lock(&au1000->ac97_lock); - ac97_config = au1000->ac97_ioport->config; - ac97_config = ac97_config & ~AC97C_XMIT_SLOTS_MASK; - ac97_config |= (xmit_slots << AC97C_XMIT_SLOTS_BIT); - au1000->ac97_ioport->config = ac97_config; - spin_unlock(&au1000->ac97_lock); -} - -static void -au1000_set_ac97_recv_slots(struct snd_au1000 *au1000, long recv_slots) -{ - u32 volatile ac97_config; - - spin_lock(&au1000->ac97_lock); - ac97_config = au1000->ac97_ioport->config; - ac97_config = ac97_config & ~AC97C_RECV_SLOTS_MASK; - ac97_config |= (recv_slots << AC97C_RECV_SLOTS_BIT); - au1000->ac97_ioport->config = ac97_config; - spin_unlock(&au1000->ac97_lock); -} - - -static void -au1000_release_dma_link(struct audio_stream *stream) -{ - struct au1000_period * pointer; - struct au1000_period * pointer_next; - - stream->period_size = 0; - stream->periods = 0; - pointer = stream->buffer; - if (! pointer) - return; - do { - pointer_next = pointer->next; - kfree(pointer); - pointer = pointer_next; - } while (pointer != stream->buffer); - stream->buffer = NULL; -} - -static int -au1000_setup_dma_link(struct audio_stream *stream, unsigned int period_bytes, - unsigned int periods) -{ - struct snd_pcm_substream *substream = stream->substream; - struct snd_pcm_runtime *runtime = substream->runtime; - struct au1000_period *pointer; - unsigned long dma_start; - int i; - - dma_start = virt_to_phys(runtime->dma_area); - - if (stream->period_size == period_bytes && - stream->periods == periods) - return 0; /* not changed */ - - au1000_release_dma_link(stream); - - stream->period_size = period_bytes; - stream->periods = periods; - - stream->buffer = kmalloc(sizeof(struct au1000_period), GFP_KERNEL); - if (! stream->buffer) - return -ENOMEM; - pointer = stream->buffer; - for (i = 0; i < periods; i++) { - pointer->start = (u32)(dma_start + (i * period_bytes)); - pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1); - if (i < periods - 1) { - pointer->next = kmalloc(sizeof(struct au1000_period), GFP_KERNEL); - if (! pointer->next) { - au1000_release_dma_link(stream); - return -ENOMEM; - } - pointer = pointer->next; - } - } - pointer->next = stream->buffer; - return 0; -} - -static void -au1000_dma_stop(struct audio_stream *stream) -{ - if (snd_BUG_ON(!stream->buffer)) - return; - disable_dma(stream->dma); -} - -static void -au1000_dma_start(struct audio_stream *stream) -{ - if (snd_BUG_ON(!stream->buffer)) - return; - - init_dma(stream->dma); - if (get_dma_active_buffer(stream->dma) == 0) { - clear_dma_done0(stream->dma); - set_dma_addr0(stream->dma, stream->buffer->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - set_dma_addr1(stream->dma, stream->buffer->next->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - } else { - clear_dma_done1(stream->dma); - set_dma_addr1(stream->dma, stream->buffer->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - set_dma_addr0(stream->dma, stream->buffer->next->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - } - enable_dma_buffers(stream->dma); - start_dma(stream->dma); -} - -static irqreturn_t -au1000_dma_interrupt(int irq, void *dev_id) -{ - struct audio_stream *stream = (struct audio_stream *) dev_id; - struct snd_pcm_substream *substream = stream->substream; - - spin_lock(&stream->dma_lock); - switch (get_dma_buffer_done(stream->dma)) { - case DMA_D0: - stream->buffer = stream->buffer->next; - clear_dma_done0(stream->dma); - set_dma_addr0(stream->dma, stream->buffer->next->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - enable_dma_buffer0(stream->dma); - break; - case DMA_D1: - stream->buffer = stream->buffer->next; - clear_dma_done1(stream->dma); - set_dma_addr1(stream->dma, stream->buffer->next->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - enable_dma_buffer1(stream->dma); - break; - case (DMA_D0 | DMA_D1): - printk(KERN_ERR "DMA %d missed interrupt.\n",stream->dma); - au1000_dma_stop(stream); - au1000_dma_start(stream); - break; - case (~DMA_D0 & ~DMA_D1): - printk(KERN_ERR "DMA %d empty irq.\n",stream->dma); - } - spin_unlock(&stream->dma_lock); - snd_pcm_period_elapsed(substream); - return IRQ_HANDLED; -} - -/*-------------------------- PCM Audio Streams -------------------------------*/ - -static unsigned int rates[] = {8000, 11025, 16000, 22050}; -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct snd_pcm_hardware snd_au1000_hw = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | \ - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050), - .rate_min = 8000, - .rate_max = 22050, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = 32, - .period_bytes_max = 16*1024, - .periods_min = 8, - .periods_max = 255, - .fifo_size = 16, -}; - -static int -snd_au1000_playback_open(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[PLAYBACK]->substream = substream; - au1000->stream[PLAYBACK]->buffer = NULL; - substream->private_data = au1000->stream[PLAYBACK]; - substream->runtime->hw = snd_au1000_hw; - return (snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0); -} - -static int -snd_au1000_capture_open(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[CAPTURE]->substream = substream; - au1000->stream[CAPTURE]->buffer = NULL; - substream->private_data = au1000->stream[CAPTURE]; - substream->runtime->hw = snd_au1000_hw; - return (snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0); -} - -static int -snd_au1000_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[PLAYBACK]->substream = NULL; - return 0; -} - -static int -snd_au1000_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[CAPTURE]->substream = NULL; - return 0; -} - -static int -snd_au1000_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct audio_stream *stream = substream->private_data; - int err; - - err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - return au1000_setup_dma_link(stream, - params_period_bytes(hw_params), - params_periods(hw_params)); -} - -static int -snd_au1000_hw_free(struct snd_pcm_substream *substream) -{ - struct audio_stream *stream = substream->private_data; - au1000_release_dma_link(stream); - return snd_pcm_lib_free_pages(substream); -} - -static int -snd_au1000_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - if (runtime->channels == 1) - au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_4); - else - au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4); - snd_ac97_set_rate(au1000->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate); - return 0; -} - -static int -snd_au1000_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - if (runtime->channels == 1) - au1000_set_ac97_recv_slots(au1000, AC97_SLOT_4); - else - au1000_set_ac97_recv_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4); - snd_ac97_set_rate(au1000->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate); - return 0; -} - -static int -snd_au1000_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct audio_stream *stream = substream->private_data; - int err = 0; - - spin_lock(&stream->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - au1000_dma_start(stream); - break; - case SNDRV_PCM_TRIGGER_STOP: - au1000_dma_stop(stream); - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&stream->dma_lock); - return err; -} - -static snd_pcm_uframes_t -snd_au1000_pointer(struct snd_pcm_substream *substream) -{ - struct audio_stream *stream = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - long location; - - spin_lock(&stream->dma_lock); - location = get_dma_residue(stream->dma); - spin_unlock(&stream->dma_lock); - location = stream->buffer->relative_end - location; - if (location == -1) - location = 0; - return bytes_to_frames(runtime,location); -} - -static struct snd_pcm_ops snd_card_au1000_playback_ops = { - .open = snd_au1000_playback_open, - .close = snd_au1000_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_au1000_hw_params, - .hw_free = snd_au1000_hw_free, - .prepare = snd_au1000_playback_prepare, - .trigger = snd_au1000_trigger, - .pointer = snd_au1000_pointer, -}; - -static struct snd_pcm_ops snd_card_au1000_capture_ops = { - .open = snd_au1000_capture_open, - .close = snd_au1000_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_au1000_hw_params, - .hw_free = snd_au1000_hw_free, - .prepare = snd_au1000_capture_prepare, - .trigger = snd_au1000_trigger, - .pointer = snd_au1000_pointer, -}; - -static int -snd_au1000_pcm_new(struct snd_au1000 *au1000) -{ - struct snd_pcm *pcm; - int err; - unsigned long flags; - - if ((err = snd_pcm_new(au1000->card, "AU1000 AC97 PCM", 0, 1, 1, &pcm)) < 0) - return err; - - 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_au1000_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_card_au1000_capture_ops); - - pcm->private_data = au1000; - pcm->info_flags = 0; - strcpy(pcm->name, "Au1000 AC97 PCM"); - - spin_lock_init(&au1000->stream[PLAYBACK]->dma_lock); - spin_lock_init(&au1000->stream[CAPTURE]->dma_lock); - - flags = claim_dma_lock(); - au1000->stream[PLAYBACK]->dma = request_au1000_dma(au1000->dmaid[0], - "AC97 TX", au1000_dma_interrupt, 0, - au1000->stream[PLAYBACK]); - if (au1000->stream[PLAYBACK]->dma < 0) { - release_dma_lock(flags); - return -EBUSY; - } - au1000->stream[CAPTURE]->dma = request_au1000_dma(au1000->dmaid[1], - "AC97 RX", au1000_dma_interrupt, 0, - au1000->stream[CAPTURE]); - if (au1000->stream[CAPTURE]->dma < 0){ - release_dma_lock(flags); - return -EBUSY; - } - /* enable DMA coherency in read/write DMA channels */ - set_dma_mode(au1000->stream[PLAYBACK]->dma, - get_dma_mode(au1000->stream[PLAYBACK]->dma) & ~DMA_NC); - set_dma_mode(au1000->stream[CAPTURE]->dma, - get_dma_mode(au1000->stream[CAPTURE]->dma) & ~DMA_NC); - release_dma_lock(flags); - au1000->pcm = pcm; - return 0; -} - - -/*-------------------------- AC97 CODEC Control ------------------------------*/ - -static unsigned short -snd_au1000_ac97_read(struct snd_ac97 *ac97, unsigned short reg) -{ - struct snd_au1000 *au1000 = ac97->private_data; - u32 volatile cmd; - u16 volatile data; - int i; - - spin_lock(&au1000->ac97_lock); -/* would rather use the interrupt than this polling but it works and I can't -get the interrupt driven case to work efficiently */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) - printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); - - cmd = (u32) reg & AC97C_INDEX_MASK; - cmd |= AC97C_READ; - au1000->ac97_ioport->cmd = cmd; - - /* now wait for the data */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) { - printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); - spin_unlock(&au1000->ac97_lock); - return 0; - } - - data = au1000->ac97_ioport->cmd & 0xffff; - spin_unlock(&au1000->ac97_lock); - - return data; - -} - - -static void -snd_au1000_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) -{ - struct snd_au1000 *au1000 = ac97->private_data; - u32 cmd; - int i; - - spin_lock(&au1000->ac97_lock); -/* would rather use the interrupt than this polling but it works and I can't -get the interrupt driven case to work efficiently */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) - printk(KERN_ERR "au1000 AC97: AC97 command write timeout\n"); - - cmd = (u32) reg & AC97C_INDEX_MASK; - cmd &= ~AC97C_READ; - cmd |= ((u32) val << AC97C_WD_BIT); - au1000->ac97_ioport->cmd = cmd; - spin_unlock(&au1000->ac97_lock); -} - -/*------------------------------ Setup / Destroy ----------------------------*/ - -static void snd_au1000_free(struct snd_card *card) -{ - struct snd_au1000 *au1000 = card->private_data; - - if (au1000->stream[PLAYBACK]) { - if (au1000->stream[PLAYBACK]->dma >= 0) - free_au1000_dma(au1000->stream[PLAYBACK]->dma); - kfree(au1000->stream[PLAYBACK]); - } - - if (au1000->stream[CAPTURE]) { - if (au1000->stream[CAPTURE]->dma >= 0) - free_au1000_dma(au1000->stream[CAPTURE]->dma); - kfree(au1000->stream[CAPTURE]); - } - - if (au1000->ac97_res_port) { - /* put internal AC97 block into reset */ - if (au1000->ac97_ioport) { - au1000->ac97_ioport->cntrl = AC97C_RS; - iounmap(au1000->ac97_ioport); - au1000->ac97_ioport = NULL; - } - release_and_free_resource(au1000->ac97_res_port); - au1000->ac97_res_port = NULL; - } -} - -static struct snd_ac97_bus_ops ops = { - .write = snd_au1000_ac97_write, - .read = snd_au1000_ac97_read, -}; - -static int au1000_ac97_probe(struct platform_device *pdev) -{ - int err; - void __iomem *io; - struct resource *r; - struct snd_card *card; - struct snd_au1000 *au1000; - struct snd_ac97_bus *pbus; - struct snd_ac97_template ac97; - - err = snd_card_new(&pdev->dev, -1, "AC97", THIS_MODULE, - sizeof(struct snd_au1000), &card); - if (err < 0) - return err; - - au1000 = card->private_data; - au1000->card = card; - spin_lock_init(&au1000->ac97_lock); - - /* from here on let ALSA call the special freeing function */ - card->private_free = snd_au1000_free; - - /* TX DMA ID */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - err = -ENODEV; - snd_printk(KERN_INFO "no TX DMA platform resource!\n"); - goto out; - } - au1000->dmaid[0] = r->start; - - /* RX DMA ID */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - err = -ENODEV; - snd_printk(KERN_INFO "no RX DMA platform resource!\n"); - goto out; - } - au1000->dmaid[1] = r->start; - - au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), - GFP_KERNEL); - if (!au1000->stream[PLAYBACK]) { - err = -ENOMEM; - goto out; - } - au1000->stream[PLAYBACK]->dma = -1; - - au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), - GFP_KERNEL); - if (!au1000->stream[CAPTURE]) { - err = -ENOMEM; - goto out; - } - au1000->stream[CAPTURE]->dma = -1; - - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { - err = -ENODEV; - goto out; - } - - err = -EBUSY; - au1000->ac97_res_port = request_mem_region(r->start, resource_size(r), - pdev->name); - if (!au1000->ac97_res_port) { - snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); - goto out; - } - - io = ioremap(r->start, resource_size(r)); - if (!io) - goto out; - - au1000->ac97_ioport = (struct au1000_ac97_reg *)io; - - /* configure pins for AC'97 - TODO: move to board_setup.c */ - au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); - - /* Initialise Au1000's AC'97 Control Block */ - au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; - udelay(10); - au1000->ac97_ioport->cntrl = AC97C_CE; - udelay(10); - - /* Initialise External CODEC -- cold reset */ - au1000->ac97_ioport->config = AC97C_RESET; - udelay(10); - au1000->ac97_ioport->config = 0x0; - mdelay(5); - - /* Initialise AC97 middle-layer */ - err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus); - if (err < 0) - goto out; - - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = au1000; - err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97); - if (err < 0) - goto out; - - err = snd_au1000_pcm_new(au1000); - if (err < 0) - goto out; - - strcpy(card->driver, "Au1000-AC97"); - strcpy(card->shortname, "AMD Au1000-AC97"); - sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver"); - - err = snd_card_register(card); - if (err < 0) - goto out; - - printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); - - platform_set_drvdata(pdev, card); - - return 0; - - out: - snd_card_free(card); - return err; -} - -static int au1000_ac97_remove(struct platform_device *pdev) -{ - return snd_card_free(platform_get_drvdata(pdev)); -} - -struct platform_driver au1000_ac97c_driver = { - .driver = { - .name = "au1000-ac97c", - .owner = THIS_MODULE, - }, - .probe = au1000_ac97_probe, - .remove = au1000_ac97_remove, -}; - -module_platform_driver(au1000_ac97c_driver); -- 2.1.4

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[alsa-devel] SGTL5000
by Erik Friesen 07 Feb '16

07 Feb '16

I was sort of directed to this list from linux-arm. I have been working with an embeddedarm ts4900 and built a baseboard that uses the sgtl5000. I also have worked with the RiotBoard, which uses the same chip, both using the i.mx6 arm. I find that occasionally the sgtl5000 is resetting, and there is no way to bring it back online. As near as I can tell, the i2c bus comes back alive, for example alsamixer communicates fine, coms happen on the i2c bus per scope. However, its obvious the chip never gets re initialized by how it acts. I know the thing shouldn't reset, that is the primary problem, but I still want some backup method. Its probably due to ground bounce from a non earth grounded setup. It could be that using the external regulator will help, we will see. Is there a suggestion on a way the sgtl5000.c could be extended to support some sort of re initialize routine? I am using the embeddedarm repo as a base kernel. sgtl5000.c has been updated to include ret = sgtl5000_fill_defaults(sgtl5000); but the base kernel is 3.10.53. However, the riotboard runs 4.3 with similar results.

1 2

[alsa-devel] Sony PCM-D100 source code
by Martin Langer 07 Feb '16

07 Feb '16

Hello, I just figured out that Sony released (in 2014?) the source code of their audio recorder PCM-D100: http://oss.sony.net/Products/Linux/Audio/PCM-D100.html A patched ALSA version and a audio driver (GPL!) for the ICX platform (ARM) is available :-) Enjoy it! Martin

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[alsa-devel] About planning of libhinawa ITP to debian project
by Takashi Sakamoto 07 Feb '16

07 Feb '16

Hi, In this message, I explain about a plan of libhinawa ITP[0] to debian project. Fortunately, I got a help of a debian package maintainer, Kentaro Hayashi. He maintains groonga package[1] and experienced related works[2]. Currently, we plan to work for it in next month, February 2016. I have no idea about when FTP master acked to our ITP and when the package is released from debian project. I with the package will be available as early as possible, and this work will be a part of his great work as a debian package maintainer. FYI, [0] ITP = Intent To Packaging [1] Debian -- Details of package groonga in sid: https://packages.debian.org/sid/groonga [2] Debian Bug report logs - #732055 ITP: groonga -- Fulltext search engine https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=732055 Takashi Sakamoto

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[alsa-devel] [PATCH] ALSA: hda - Fix speaker output from VAIO AiO machines
by Takashi Iwai 07 Feb '16

07 Feb '16

Some Sony VAIO AiO models (VGC-JS4EF and VGC-JS25G, both with PCI SSID 104d:9044) need the same quirk to make the speaker working properly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112031 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a733e5dc701d..b43c0f7994eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2223,6 +2223,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), -- 2.7.0

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[alsa-devel] sound: heap out-of-bounds write in dummy_systimer_prepare
by Dmitry Vyukov 07 Feb '16

07 Feb '16

Hello, I am still seeing these eap out-of-bounds writes in dummy_systimer_prepare. Even when we've disabled the hrtimer sysctl. The fuzzer does not change sysctls at the moment and I think we've overlooked a very simple possibility that can happen when sysctls are not changed (i.e. triggered by an unprivileged user). dummy_pcm_open does: static int dummy_pcm_open(struct snd_pcm_substream *substream) { ... dummy->timer_ops = &dummy_systimer_ops; #ifdef CONFIG_HIGH_RES_TIMERS if (hrtimer) dummy->timer_ops = &dummy_hrtimer_ops; #endif There is a small window of time when we switch timer_ops to dummy_systimer_ops and then restore it back to dummy_hrtimer_ops. If another thread executes a timer op during that window, it happily smashes heap thinking that there is a systimer there. KTSAN, a data race detector (https://github.com/google/ktsan), would catch this in a blink of an eye and give a detailed explanation of what happens... ================================================================== [ 134.926046] BUG: KASAN: slab-out-of-bounds in dummy_systimer_prepare+0x268/0x2a0 at addr ffff880030f00b10 [ 134.926046] Write of size 4 by task syz-executor/32211 [ 134.926046] ============================================================================= [ 134.926046] BUG kmalloc-192 (Not tainted): kasan: bad access detected [ 134.926046] ----------------------------------------------------------------------------- [ 134.926046] [ 134.926046] INFO: Allocated in dummy_hrtimer_create+0x49/0x1a0 age=5 cpu=0 pid=32211 [ 134.926046] ___slab_alloc+0x564/0x5b0 [ 134.926046] __slab_alloc+0x66/0xc0 [ 134.926046] kmem_cache_alloc_trace+0x25c/0x300 [ 134.931676] dummy_hrtimer_create+0x49/0x1a0 [ 134.931676] dummy_pcm_open+0xef/0x570 [ 134.932135] snd_pcm_open_substream+0x188/0x430 [ 134.932135] snd_pcm_oss_open.part.17+0x5a4/0x1110 [ 134.932135] snd_pcm_oss_open+0x35/0x50 [ 134.932135] soundcore_open+0x30f/0x640 [ 134.932135] chrdev_open+0x22a/0x4c0 [ 134.932135] do_dentry_open+0x6a2/0xcb0 [ 134.932135] vfs_open+0x17b/0x1f0 [ 134.932135] path_openat+0xde9/0x5e30 [ 134.932135] do_filp_open+0x18e/0x250 [ 134.932135] do_sys_open+0x1fc/0x420 [ 134.932135] SyS_open+0x2d/0x40 [ 134.932135] INFO: Freed in snd_pcm_plugin_free+0xb1/0xe0 age=20 cpu=0 pid=32142 [ 134.932135] __slab_free+0x1fc/0x320 [ 134.932135] kfree+0x2ac/0x2c0 [ 134.932135] snd_pcm_plugin_free+0xb1/0xe0 [ 134.932135] snd_pcm_oss_release_substream+0x113/0x1f0 [ 134.932135] snd_pcm_release_substream.part.38+0x1e8/0x2f0 [ 134.937048] snd_pcm_release_substream+0x59/0x70 [ 134.937048] snd_pcm_oss_release_file+0x7b/0xb0 [ 134.937048] snd_pcm_oss_release+0xfa/0x280 [ 134.937048] __fput+0x236/0x780 [ 134.937048] ____fput+0x15/0x20 [ 134.937048] task_work_run+0x170/0x210 [ 134.937048] do_exit+0x8b5/0x2cb0 [ 134.937048] do_group_exit+0x108/0x330 [ 134.937048] get_signal+0x5e4/0x14f0 [ 134.937048] do_signal+0x83/0x1c90 [ 134.937048] exit_to_usermode_loop+0x1a5/0x210 [ 134.937048] Call Trace: [ 134.937048] [<ffffffff82be30ad>] dump_stack+0x6f/0xa2 [ 134.937048] [<ffffffff8175c984>] print_trailer+0xf4/0x150 [ 134.937048] [<ffffffff8176370f>] object_err+0x2f/0x40 [ 134.937048] [<ffffffff81765fb6>] kasan_report_error+0x256/0x550 [ 134.937048] [<ffffffff81455748>] ? mark_held_locks+0xc8/0x120 [ 134.937048] [<ffffffff85223e23>] ? snd_power_wait+0x2e3/0x380 [ 134.937048] [<ffffffff817664ae>] __asan_report_store4_noabort+0x3e/0x40 [ 134.937048] [<ffffffff852e0f68>] ? dummy_systimer_prepare+0x268/0x2a0 [ 134.937048] [<ffffffff852e0f68>] dummy_systimer_prepare+0x268/0x2a0 [ 134.937048] [<ffffffff852e141b>] dummy_pcm_prepare+0x7b/0xa0 [ 134.937048] [<ffffffff852563ca>] snd_pcm_do_prepare+0x5a/0x90 [ 134.937048] [<ffffffff852556b6>] snd_pcm_action_single+0x76/0x120 [ 134.937048] [<ffffffff85255ab5>] snd_pcm_action_nonatomic+0x95/0xa0 [ 134.937048] [<ffffffff85260a05>] snd_pcm_common_ioctl1+0x1045/0x21a0 [ 134.937048] [<ffffffff86659c34>] ? __mutex_unlock_slowpath+0x1f4/0x430 [ 134.937048] [<ffffffff8525f9c0>] ? snd_pcm_status_user+0x140/0x140 [ 134.937048] [<ffffffff86659e79>] ? mutex_unlock+0x9/0x10 [ 134.937048] [<ffffffff8529649c>] ? snd_pcm_oss_change_params+0xb2c/0x3580 [ 134.937048] [<ffffffff85261e02>] snd_pcm_playback_ioctl1+0x2a2/0x5e0 [ 134.937048] [<ffffffff85295970>] ? snd_pcm_hw_param_near.constprop.26+0x8f0/0x8f0 [ 134.937048] [<ffffffff85261b60>] ? snd_pcm_common_ioctl1+0x21a0/0x21a0 [ 134.937048] [<ffffffff85262906>] snd_pcm_kernel_ioctl+0x136/0x160 [ 134.937048] [<ffffffff852908bb>] snd_pcm_oss_prepare+0x4b/0x200 [ 134.937048] [<ffffffff85298f9e>] snd_pcm_oss_make_ready+0xae/0x120 [ 134.937048] [<ffffffff8529f58f>] snd_pcm_oss_write+0x15f/0x700 [ 134.937048] [<ffffffff81456670>] ? debug_check_no_locks_freed+0x3c0/0x3c0 [ 134.937048] [<ffffffff817bacd3>] __vfs_write+0x113/0x4b0 [ 134.937048] [<ffffffff8529f430>] ? snd_pcm_oss_ioctl_compat+0x30/0x30 [ 134.937048] [<ffffffff817babc0>] ? vfs_iter_write+0x360/0x360 [ 134.937048] [<ffffffff81493f27>] ? debug_lockdep_rcu_enabled+0x77/0x90 [ 134.937048] [<ffffffff829e63c5>] ? common_file_perm+0x155/0x3a0 [ 134.937048] [<ffffffff829e6822>] ? apparmor_file_permission+0x22/0x30 [ 134.937048] [<ffffffff8291d07c>] ? security_file_permission+0x8c/0x1f0 [ 134.937048] [<ffffffff817bc162>] ? rw_verify_area+0x102/0x2c0 [ 134.937048] [<ffffffff817bc797>] vfs_write+0x167/0x4a0 [ 134.937048] [<ffffffff817bfa81>] SyS_write+0x111/0x220 [ 134.937048] [<ffffffff817bf970>] ? SyS_read+0x220/0x220 [ 134.937048] [<ffffffff81005017>] ? trace_hardirqs_on_thunk+0x17/0x19 [ 134.937048] [<ffffffff8665f176>] entry_SYSCALL_64_fastpath+0x16/0x7a [ 134.937048] ================================================================== On commit df48ab3c2f5ffca88b7803ffbadd074bd5a0a2ef.

2 1

[alsa-devel] [PATCH] Revert "ALSA: hda - Fix noise on Gigabyte Z170X mobo"
by Takashi Iwai 05 Feb '16

05 Feb '16

This reverts commit 0c25ad80408e95e0a4fbaf0056950206e95f726f. The original commit disabled the aamixer path due to the noise problem, but it turned out that some mobo with the same PCI SSID doesn't suffer from the issue, and the disabled function (analog loopback) is still demanded by users. Since the recent commit [e7fdd52779a6: ALSA: hda - Implement loopback control switch for Realtek and other codecs], we have the dynamic mixer switch to enable/disable the aamix path, and we don't have to disable the path statically any longer. So, let's revert the disablement, so that only the user suffering from the noise problem can turn off the aamix on the fly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=108301 Reported-by: <mutedbytes(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21992fb7035d..a733e5dc701d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1787,7 +1787,6 @@ enum { ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, - ALC882_FIXUP_DISABLE_AAMIX, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1949,8 +1948,6 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action); -static void alc_fixup_disable_aamix(struct hda_codec *codec, - const struct hda_fixup *fix, int action); static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { @@ -2188,10 +2185,6 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, - [ALC882_FIXUP_DISABLE_AAMIX] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_disable_aamix, - }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2259,7 +2252,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), - SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.7.0

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[alsa-devel] [PATCH v3] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by tim.howe＠cirrus.com 05 Feb '16

05 Feb '16

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 933 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 272 ++++++ 5 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3317da..f59aa4b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -444,6 +445,10 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cdadb9b..4f1ab1d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -51,6 +51,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..3d5cac9 +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,933 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program 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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, THMS_PDN }, + { CS53L30_MCLKCTL, MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, INTERNAL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, ASP_RATE_48K }, + { CS53L30_ASP1_CTL, ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, MUTEP_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, INBIAS_CTLX_DFLT }, + { CS53L30_INBIAS_CTL2, INBIAS_CTLX_DFLT }, + { CS53L30_DMIC1_STR_CTL, DMIC1_STEREO_DFLT }, + { CS53L30_DMIC2_STR_CTL, DMIC2_STEREO_DFLT }, + { CS53L30_ADCDMIC1_CTL1, ADC1_ON_CH_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, ADC1_DMIC1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, ADC2_ON_CH_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +unsigned int const input1_sel_values[] = { DMIC1_ON_CH_AB_IN, DMIC1_ON_CH_A_IN, + DMIC1_ON_CH_B_IN, ADC1_ON_CH_AB_IN, ADC1_ON_CH_A_IN, ADC1_ON_CH_B_IN, + DMIC1_OFF_ADC1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +unsigned int const input2_sel_values[] = { DMIC2_ON_CH_AB_IN, DMIC2_ON_CH_A_IN, + DMIC2_ON_CH_B_IN, ADC2_ON_CH_AB_IN, ADC2_ON_CH_A_IN, ADC2_ON_CH_B_IN, + DMIC2_OFF_ADC2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 1); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, + CS53L30_ASP1_CTL, ASP1_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, + CS53L30_ASP2_CTL, ASP2_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= ASP_SCLK_INV; + else + asp_config_ctl &= ~ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = + cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_write(priv->regmap, CS53L30_ASPCFG_CTL, priv->asp_config_ctl); + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= INTERNAL_FS_RATIO; + else + int_sr_ctl &= ~INTERNAL_FS_RATIO; + regmap_write(priv->regmap, CS53L30_INT_SR_CTL, int_sr_ctl); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= MCLK_INT_SCALE; + else + mclk_ctl &= ~MCLK_INT_SCALE; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->asp_config_ctl &= ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & ASP_RATE); + regmap_write(priv->regmap, CS53L30_MCLKCTL, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & DIGSFT) + inter_max_check = PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + usleep_range(1000, 1000); + msleep(1); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & PDN_DONE) + break; + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, MCLK_DIS); + break; + } + dapm->bias_level = level; + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, ASP1_3ST, + (ASP1_3ST_VAL(tristate) & ASP1_3ST)); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .owner = THIS_MODULE, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..f45e91f +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,272 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program 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. + * + */ + +#ifndef __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define PDN_ULP (1 << 7) +#define PDN_LP (1 << 6) +#define DISCHARGE_FILT (1 << 5) +#define THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define MCLK_DIS (1 << 7) +#define MCLK_INT_SCALE (1 << 6) +#define DMIC_DRIVE (1 << 5) +#define MCLK_DIV (3 << 2) +#define MCLK_DIV_DFLT (1 << 2) +#define SYNC_EN (1 << 1) + +/* CS53L30_INT_SR_CTL */ +#define INTERNAL_FS_RATIO (1 << 4) +#define INTERNAL_FS_DFLT (7 << 2) +#define MCLK_19MHZ_EN (1 << 0) + +/* CS53L30_MICBIAS_CTL */ +#define MIC4_BIAS_PDN (1 << 7) +#define MIC3_BIAS_PDN (1 << 6) +#define MIC2_BIAS_PDN (1 << 5) +#define MIC1_BIAS_PDN (1 << 4) +#define VP_MIN (1 << 2) +#define MIC_BIAS_CTRL (3 << 0) +#define MIC_BIAS_ALL_PDN 0xF0 +#define MIC_BIAS_DFLT (MIC_BIAS_ALL_PDN | VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define ASP_MS (1 << 7) +#define ASP_SCLK_INV (1 << 4) +#define ASP_RATE_48K (3 << 2) +#define ASP_RATE 0x0F +#define ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define ASP1_TDM_PDN (1 << 7) +#define ASP1_SDOUT_PDN (1 << 6) +#define ASP1_3ST (1 << 5) +#define SHIFT_LEFT (1 << 4) +#define ASP1_DRIVE (1 << 0) +#define ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define ASP1_CHX_TX_STATE (1 << 7) +#define ASP1_CHX_TX_LOC 0x3F +#define ASP1_CHX_TX_DFLT_SLT47 0x2F +#define ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define ASP2_SDOUT_PDN (1 << 6) +#define ASP2_DRIVE (1 << 0) +#define ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define DIGSFT (1 << 5) +#define SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define LRCK_50_NPW (1 << 3) +#define LRCK_TPWH (7 << 0) +#define LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define MUTE_PDN_ULP (1 << 7) +#define MUTE_PDN_LP (1 << 6) +#define MUTE_M4B_PDN (1 << 4) +#define MUTE_M3B_PDN (1 << 3) +#define MUTE_M2B_PDN (1 << 2) +#define MUTE_M1B_PDN (1 << 1) +#define MUTE_MB_ALL_PDN (1 << 0) +#define MUTEP_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define MUTE_PIN_POLARITY (1 << 7) +#define MUTE_ASP_TDM_PDN (1 << 6) +#define MUTE_ASP_SDOUT2_PDN (1 << 5) +#define MUTE_ASP_SDOUT1_PDN (1 << 4) +#define MUTE_ADC2B_PDN (1 << 3) +#define MUTE_ADC2A_PDN (1 << 2) +#define MUTE_ADC1B_PDN (1 << 1) +#define MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define IN4M_BIAS (3 << 6) +#define IN4P_BIAS (3 << 4) +#define IN3M_BIAS (3 << 2) +#define IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define IN2M_BIAS (3 << 6) +#define IN2P_BIAS (3 << 4) +#define IN1M_BIAS (3 << 2) +#define IN1P_BIAS (3 << 0) +#define INBIAS_CTLX_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define DMIC1_STEREO_ENB (1 << 5) +#define DMIC1_STEREO_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define DMIC2_STEREO_EN (1 << 5) +#define DMIC2_STEREO_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define ADC1B_PDN (1 << 7) +#define ADC1A_PDN (1 << 6) +#define DMIC1_PDN (1 << 2) +#define DMIC1_SCLK_DIV (1 << 1) +#define CH_TYPE (1 << 0) +#define DMIC1_ON_CH_AB_IN (CH_TYPE) +#define DMIC1_ON_CH_A_IN (ADC1B_PDN | CH_TYPE) +#define DMIC1_ON_CH_B_IN (ADC1A_PDN | CH_TYPE) +#define ADC1_ON_CH_AB_IN (DMIC1_PDN) +#define ADC1_ON_CH_A_IN (ADC1B_PDN | DMIC1_PDN) +#define ADC1_ON_CH_B_IN (ADC1A_PDN | DMIC1_PDN) +#define DMIC1_OFF_ADC1_OFF (ADC1A_PDN | ADC1B_PDN | DMIC1_PDN) +#define ADC1_DMIC1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define ADC1_NOTCH_DIS (1 << 7) +#define ADC1B_INV (1 << 5) +#define ADC1A_INV (1 << 4) +#define ADC1B_DIG_BOOST (1 << 1) +#define ADC1A_DIG_BOOST (1 << 0) +#define ADC1_DMIC1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define ADC1_HPF_EN (1 << 3) +#define ADC1_HPF_CF (3 << 1) +#define ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define ADC1B_NG (1 << 7) +#define ADC1A_NG (1 << 6) +#define ADC1_NG_BOOST (1 << 5) +#define ADC1_NG_THRESH (7 << 2) +#define ADC1_NG_DELAY (3 << 0) +#define ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define ADC1A_PREAMP (3 << 6) +#define ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define ADC1B_PREAMP (3 << 6) +#define ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define ADC2B_PDN (1 << 7) +#define ADC2A_PDN (1 << 6) +#define DMIC2_PDN (1 << 2) +#define DMIC2_CLKDIV (1 << 1) +#define DMIC2_ON_CH_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_A_IN (ADC2B_PDN) /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_B_IN (ADC2A_PDN) /* CH_TYPE must = 1 */ +#define ADC2_ON_CH_AB_IN (DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_A_IN (ADC2B_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_B_IN (ADC2A_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define DMIC2_OFF_ADC2_OFF (ADC2A_PDN | ADC2B_PDN | DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define ADC2_NOTCH_DIS (1 << 7) +#define ADC2B_INV (1 << 5) +#define ADC2A_INV (1 << 4) +#define ADC2B_DIG_BOOST (1 << 1) +#define ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define ADC2_HPF_EN (1 << 3) +#define ADC2_HPF_CF (3 << 1) +#define ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define PDN_DONE (1 << 7) +#define THMS_TRIP (1 << 6) +#define SYNC_DONE (1 << 5) +#define ADC2B_OVFL (1 << 4) +#define ADC2A_OVFL (1 << 3) +#define ADC1B_OVFL (1 << 2) +#define ADC1A_OVFL (1 << 1) +#define MUTE_PIN (1 << 0) +#define DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

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[alsa-devel] [PATCH] ASoC: mxs-saif: check BUSY bit in hw_params() only if not mclk_in_use
by Mans Rullgard 05 Feb '16

05 Feb '16

If something else, typically a codec, has enabled mclk, the BUSY bit may be set when hw_params() is called without this being an error. This check thus causes intermittent failures to configure the sound device when used in such a manner. Fix this by making the test conditional on !saif->mclk_in_use. Signed-off-by: Mans Rullgard <mans(a)mansr.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index a6c7b8d87cd2..13631003cb7c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -418,7 +418,7 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, } stat = __raw_readl(saif->base + SAIF_STAT); - if (stat & BM_SAIF_STAT_BUSY) { + if (!saif->mclk_in_use && (stat & BM_SAIF_STAT_BUSY)) { dev_err(cpu_dai->dev, "error: busy\n"); return -EBUSY; } -- 2.7.0

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[alsa-devel] [PATCH v2] ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT
by Charles Keepax 05 Feb '16

05 Feb '16

The Arizona CODECs support several rates that do not have simple defines in ALSA. This patch adds support for SNDRV_PCM_RATE_KNOT so that users can open stream at these rates. As part of this we should always set constraints in arizona_startup, currently we only set the constraints if we already have a clock to limit rates to that family of sample rates. This patch updates this to set a constraint of all rates supported by the chip if we do not already know which family of rates to limit to. Finally we also reduce the list of rates supported in the constraints to only include those that are supported on current parts. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Thanks, for the pointer there Lars, sorry for missing what you were talking about the first time. It does indeed look much nicer this way. Changes since v1: - Use the mask field in the constaint list as suggested by Lars. Thanks, Charles sound/soc/codecs/arizona.c | 62 ++++++++++++---------------------------------- sound/soc/codecs/arizona.h | 2 ++ sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- 7 files changed, 23 insertions(+), 51 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 33143fe..a32cfb8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1398,29 +1398,6 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; -static const unsigned int arizona_48k_rates[] = { - 12000, - 24000, - 48000, - 96000, - 192000, - 384000, - 768000, - 4000, - 8000, - 16000, - 32000, - 64000, - 128000, - 256000, - 512000, -}; - -static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { - .count = ARRAY_SIZE(arizona_48k_rates), - .list = arizona_48k_rates, -}; - static const int arizona_44k1_bclk_rates[] = { -1, 44100, @@ -1443,22 +1420,7 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; -static const unsigned int arizona_44k1_rates[] = { - 11025, - 22050, - 44100, - 88200, - 176400, - 352800, - 705600, -}; - -static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { - .count = ARRAY_SIZE(arizona_44k1_rates), - .list = arizona_44k1_rates, -}; - -static int arizona_sr_vals[] = { +static const unsigned int arizona_sr_vals[] = { 0, 12000, 24000, @@ -1485,13 +1447,21 @@ static int arizona_sr_vals[] = { 512000, }; +#define ARIZONA_48K_RATE_MASK 0x0F003E +#define ARIZONA_44K1_RATE_MASK 0x003E00 +#define ARIZONA_RATE_MASK (ARIZONA_48K_RATE_MASK | ARIZONA_44K1_RATE_MASK) + +static const struct snd_pcm_hw_constraint_list arizona_constraint = { + .count = ARRAY_SIZE(arizona_sr_vals), + .list = arizona_sr_vals, +}; + static int arizona_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - const struct snd_pcm_hw_constraint_list *constraint; unsigned int base_rate; if (!substream->runtime) @@ -1509,16 +1479,15 @@ static int arizona_startup(struct snd_pcm_substream *substream, } if (base_rate == 0) - return 0; - - if (base_rate % 8000) - constraint = &arizona_44k1_constraint; + dai_priv->constraint.mask = ARIZONA_RATE_MASK; + else if (base_rate % 8000) + dai_priv->constraint.mask = ARIZONA_44K1_RATE_MASK; else - constraint = &arizona_48k_constraint; + dai_priv->constraint.mask = ARIZONA_48K_RATE_MASK; return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - constraint); + &dai_priv->constraint); } static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, @@ -1911,6 +1880,7 @@ int arizona_init_dai(struct arizona_priv *priv, int id) struct arizona_dai_priv *dai_priv = &priv->dai[id]; dai_priv->clk = ARIZONA_CLK_SYSCLK; + dai_priv->constraint = arizona_constraint; return 0; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 0c64a57..1ea8e4e 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -68,6 +68,8 @@ struct wm_adsp; struct arizona_dai_priv { int clk; + + struct snd_pcm_hw_constraint_list constraint; }; struct arizona_priv { diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index b8d3cd0..576087b 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -923,7 +923,7 @@ static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define CS47L24_RATES SNDRV_PCM_RATE_8000_192000 +#define CS47L24_RATES SNDRV_PCM_RATE_KNOT #define CS47L24_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index aa1e96a..a8b3e3f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1738,7 +1738,7 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5102_RATES SNDRV_PCM_RATE_KNOT #define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 302a8d3..83ba70f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2024,7 +2024,7 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5110_RATES SNDRV_PCM_RATE_KNOT #define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index b4dba3a..52d766e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -943,7 +943,7 @@ static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8997_RATES SNDRV_PCM_RATE_KNOT #define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 7719bc5..0123960 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1170,7 +1170,7 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = { { "DRC1 Signal Activity", NULL, "DRC1R" }, }; -#define WM8998_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8998_RATES SNDRV_PCM_RATE_KNOT #define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.1.4

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