July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RESEND][PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 18 Jul '11

18 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error modify period_bytes_max value to set half size of max buffer sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_BUFFER/2, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.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. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + + +#define MAX_IDMA_BUFFER (128 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: wm8900: fix a memory leak if wm8900_set_fll fails Dimitris Papastamos (1): ASoC: WM8983: Initial driver Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (19): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs ASoC: Log WM8994 FIFO errors from the interrupt ASoC: Fix shift in WM8958 accessory detection default implementation Merge branch 'for-3.0' into for-3.1 ASoC: Handle spurious wm_hubs DC servo done interrupts ASoC: Handle failed WM8994 FLL lock waits ASoC: Report an error for unknown adav80x formats ASoC: Reduce power consumption for idle DAIs in WM8994 ASoC: Don't use -1 to boostrap subseq so it can be used by drivers ASoC: Correct WM8994 MICBIAS supply widget hookup Merge branch 'for-3.0' into for-3.1 Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/sta32x.c | 140 +++++ sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 9 +- sound/soc/codecs/wm8900.c | 1 + sound/soc/codecs/wm8983.c | 1203 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8983.h | 1029 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 156 +++++- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 33 +- sound/soc/codecs/wm_hubs.h | 8 + sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 +- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 2 +- 20 files changed, 2593 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/wm8983.c create mode 100644 sound/soc/codecs/wm8983.h

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (2): ASoC: Fix shift in WM8958 accessory detection default implementation ASoC: Correct WM8994 MICBIAS supply widget hookup sound/soc/codecs/wm8994.c | 9 +++++---- sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 4 files changed, 16 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH] ASoC: Report an error for unknown adav80x formats
by Mark Brown 16 Jul '11

16 Jul '11

Not only fixes error handling but also some uninitialized variable warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index e30fba6..300c04b 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -456,7 +456,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_CAPTURE_WORD_LEN24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], @@ -488,7 +488,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
by Mark Brown 16 Jul '11

16 Jul '11

Makes life a little easier if you want to add subsequences to an existing driver as you can use -1 to put things at the start of sequences. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 54fa2e5..fbfcda0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -989,7 +989,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&pending); cur_sort = -1; - cur_subseq = -1; + cur_subseq = INT_MIN; cur_reg = SND_SOC_NOPM; cur_dapm = NULL; } -- 1.7.5.4

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[alsa-devel] [PATCH V2 00/10]
by Dong Aisheng 15 Jul '11

15 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. Main changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 1 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 39 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 19 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 367 +++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 656 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 175 ++++++ 18 files changed, 1529 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

7 51

Re: [alsa-devel] [PATCH] ASoC: WM8983: Initial driver
by Mark Brown 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 01:51:30PM +0100, Dimitris Papastamos wrote: > The WM8983 is a low power, high quality stereo CODEC > designed for portable multimedia applications. Highly flexible > analogue mixing functions enable new application features, > combining hi-fi quality audio with voice communication. I've applied this but there's one issue you should fix up later: > + /* mute all outputs and set PGAs to minimum gain */ > + for (i = WM8983_LOUT1_HP_VOLUME_CTRL; > + i <= WM8983_OUT4_MONO_MIX_CTRL; ++i) > + snd_soc_update_bits(codec, i, 0x40, 0x40); I know you're doing this to minimise artifacts on initial bias raise but we should really be doing this without affecting the default register settings.

2 2

[alsa-devel] [PATCH] ASoC: Fix shift in WM8958 accessory detection default implementation
by Mark Brown 15 Jul '11

15 Jul '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bc62b6e..bfb7e66 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2834,7 +2834,7 @@ static void wm8958_default_micdet(u16 status, void *data) report = SND_JACK_MICROPHONE; /* Everything else is buttons; just assign slots */ - if (status & 0x1c0) + if (status & 0x1c) report |= SND_JACK_BTN_0; done: -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH] sound: codec: tlv320aic3x: remove mono controls
by Mark Brown 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 12:59:15PM +0200, Jan Weitzel wrote: > Any rework needed? It'd be easier to review the patch if you sent me a copy...

1 0

[alsa-devel] [PATCH] ASoC: Handle spurious wm_hubs DC servo done interrupts
by Mark Brown 15 Jul '11

15 Jul '11

Don't assume the first fire indicates that we're done. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 30 ++++++++++++++++-------------- 1 files changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ced41e8..4de6794 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -66,8 +66,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; int count = 0; + int timeout; unsigned int val; - unsigned long timeout; val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; @@ -76,21 +76,23 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) dev_dbg(codec->dev, "Waiting for DC servo...\n"); - if (hubs->dcs_done_irq) { - timeout = wait_for_completion_timeout(&hubs->dcs_done, - msecs_to_jiffies(500)); - if (timeout == 0) - dev_warn(codec->dev, "No DC servo interrupt\n"); + if (hubs->dcs_done_irq) + timeout = 4; + else + timeout = 400; - reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - } else { - do { - count++; + do { + count++; + + if (hubs->dcs_done_irq) + wait_for_completion_timeout(&hubs->dcs_done, + msecs_to_jiffies(250)); + else msleep(1); - reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < 400); - } + + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & op && count < timeout); if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", -- 1.7.5.4

1 0

[alsa-devel] Major reworks on patch_via.c (TESTERS WANTED)
by Takashi Iwai 15 Jul '11

15 Jul '11

Hi, as there have many problems reported for VIA codecs, I started looking at the driver code, and decided to rework on it. The current development is found in sound git tree topic/via-cleanup branch: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git You can pull into the latest Linus tree from topic/via-cleanup branch: # this is the vanilla tree % cd linux-2.6 % git pull \ git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git \ topic/via-cleanup A good thing is that a fair amount of code reduction is done there: % git diff HEAD..topic/via-cleanup --stat sound/pci/hda/patch_via.c sound/pci/hda/patch_via.c | 5892 +++++++++++---------------------------------- 1 files changed, 1453 insertions(+), 4439 deletions(-) But, the driver isn't tested with real machines, yet. That is, I need testers with real machines. Could VIA machine owners take a look at these commits and test it if possible? Especially, I'd like to know whether the headphone auto-mute works in this form or not. FWIW, the branch was merged to sound-unstable tree, so you can build from alsa-driver-unstable snapshot tarball if you'd like to build external modules without the kernel tree and git-merge. ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… Also, if anyone has alsa-info.sh outputs of VIA codecs, please let me know. It'd be very helpful for debugging, since I can check it via hda-emu with alsa-info.sh output. For now, I have the files of only VT1702, VT1708S, VT1718S, VT1828S and VT2020 codecs. The alsa-info.sh outputs of other VIA codecs would be greatly appreciated. thanks, Takashi

5 18

[alsa-devel] [PATCH] ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
by Dan Chen 15 Jul '11

15 Jul '11

Hi, Attached is a patch against sound-2.6 master HEAD resolving the original reporter's volume keys issue. Please consider applying. Cheers, -Dan

2 1

[alsa-devel] [PATCH 11/20] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 14 Jul '11

14 Jul '11

From: Wolfram Sang <w.sang(a)pengutronix.de> Already upstreamed. dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Chris Ball <cjb(a)laptop.org> Cc: linux-mmc(a)vger.kernel.org Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- drivers/dma/imx-dma.c | 3 ++- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.4

1 1

[alsa-devel] [PATCH] ALSA: hda - Disable automute feature by default
by Ben Hutchings 14 Jul '11

14 Jul '11

The default behaviour should be the same as before this feature was introduced, to avoid user confusion. Reported-by: Toni Mueller <support(a)oeko.net> References: http://bugs.debian.org/631963 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> --- The code is rather inconsistent in whether 'mode' means the behaviour or mechanism of automute. I didn't touch that but you might want to consider some renaming. Ben. sound/pci/hda/patch_realtek.c | 74 ++--------------------------------------- 1 files changed, 3 insertions(+), 71 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b48fb43..1247a04 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,7 +306,7 @@ struct alc_multi_io { }; enum { - ALC_AUTOMUTE_PIN, /* change the pin control */ + ALC_AUTOMUTE_PIN = 1, /* change the pin control */ ALC_AUTOMUTE_AMP, /* mute/unmute the pin AMP */ ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; @@ -1599,10 +1599,9 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } - if (spec->automute && cfg->line_out_pins[0] && + if (spec->automute_mode && cfg->line_out_pins[0] && cfg->line_out_pins[0] != cfg->hp_pins[0] && cfg->line_out_pins[0] != cfg->speaker_pins[0]) { for (i = 0; i < cfg->line_outs; i++) { @@ -1619,7 +1618,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->automute_lines = spec->detect_line; } - if (spec->automute) { + if (spec->automute_mode) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; @@ -2241,7 +2240,6 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2259,7 +2257,6 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2543,7 +2540,6 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2555,7 +2551,6 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2567,7 +2562,6 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2579,7 +2573,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3667,7 +3660,6 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3699,7 +3691,6 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3969,7 +3960,6 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -4055,7 +4045,6 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -4128,7 +4117,6 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -6277,7 +6265,6 @@ static void alc260_hp_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x0f; spec->autocfg.speaker_pins[0] = 0x10; spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -6308,7 +6295,6 @@ static void alc260_hp_3013_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x10; spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -6352,7 +6338,6 @@ static void alc260_hp_3012_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x0f; spec->autocfg.speaker_pins[1] = 0x11; spec->autocfg.speaker_pins[2] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -8887,7 +8872,6 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8901,7 +8885,6 @@ static void alc885_mba21_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8913,7 +8896,6 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8924,7 +8906,6 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8958,7 +8939,6 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9405,7 +9385,6 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9559,7 +9538,6 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9725,7 +9703,6 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9786,7 +9763,6 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9797,7 +9773,6 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9811,7 +9786,6 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9841,7 +9815,6 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9851,7 +9824,6 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9862,7 +9834,6 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->detect_line = 1; spec->automute_lines = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; @@ -9876,7 +9847,6 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9907,7 +9877,6 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9921,7 +9890,6 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9932,7 +9900,6 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9959,7 +9926,6 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic.mux_idx = 0; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11470,7 +11436,6 @@ static void alc262_hp_bpc_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11480,7 +11445,6 @@ static void alc262_hp_wildwest_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11560,7 +11524,6 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11674,7 +11637,6 @@ static void alc262_hippo_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11684,7 +11646,6 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11748,7 +11709,6 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11893,7 +11853,6 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->int_mic.pin = 0x12; spec->int_mic.mux_idx = 9; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11998,7 +11957,6 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.hp_pins[1] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -12045,7 +12003,6 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -13255,7 +13212,6 @@ static void alc268_acer_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -13339,7 +13295,6 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13379,7 +13334,6 @@ static void alc268_dell_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -13411,7 +13365,6 @@ static void alc267_quanta_il1_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -14415,7 +14368,6 @@ static void alc269_quanta_fl1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14437,7 +14389,6 @@ static void alc269_lifebook_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1a; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; } @@ -14510,7 +14461,6 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14525,7 +14475,6 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14540,7 +14489,6 @@ static void alc269vb_laptop_amic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14555,7 +14503,6 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -16878,7 +16825,6 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -16953,7 +16899,6 @@ static void alc861vd_dallas_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -18222,7 +18167,6 @@ static void alc662_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->detect_line = 1; spec->automute_lines = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; @@ -18246,7 +18190,6 @@ static void alc662_eeepc_ep20_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -18256,7 +18199,6 @@ static void alc663_m51va_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18272,7 +18214,6 @@ static void alc663_mode1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18287,7 +18228,6 @@ static void alc662_mode2_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18303,7 +18243,6 @@ static void alc663_mode3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18321,7 +18260,6 @@ static void alc663_mode4_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x16; spec->automute_mixer_nid[0] = 0x0c; spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18339,7 +18277,6 @@ static void alc663_mode5_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x16; spec->automute_mixer_nid[0] = 0x0c; spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18356,7 +18293,6 @@ static void alc663_mode6_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18373,7 +18309,6 @@ static void alc663_mode7_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18390,7 +18325,6 @@ static void alc663_mode8_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18405,7 +18339,6 @@ static void alc663_g71v_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.line_out_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; spec->detect_line = 1; spec->automute_lines = 1; @@ -19818,7 +19751,6 @@ static void alc680_base_setup(struct hda_codec *codec) spec->autocfg.inputs[0].type = AUTO_PIN_MIC; spec->autocfg.inputs[1].pin = 0x19; spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: Correct WM8994 MICBIAS supply widget hookup
by Mark Brown 14 Jul '11

14 Jul '11

The WM8994 and WM8958 series of devices have two MICBIAS supplies rather than one, the current widget actually manages the microphone detection control register bit (which is managed separately by the relevant API). Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2 widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 79dee89..414f817 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1206,7 +1206,6 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), @@ -1517,8 +1516,10 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, - { "MICBIAS", NULL, "CLK_SYS" }, - { "MICBIAS", NULL, "MICBIAS Supply" }, + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS1", NULL, "MICBIAS Supply" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- 1.7.5.4

1 0

[alsa-devel] [RFC PATCH] ASoC: codecs: add codec driver for MFD mc13783
by Jürgen Lambrecht 14 Jul '11

14 Jul '11

Sascha Hauer's driver from Pengutronix' imx-sound branch. Added some code found in a linux 2.6.33. Needs further work and cleanup. Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 29 ++ 4 files changed, 816 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..77fb7b2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if SPI select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -200,6 +201,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_MC13783 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..a129af2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 0000000..af7603a --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,781 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer(a)pengutronix.de + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * 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 program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * 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., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/device.h> +#include <linux/mfd/mc13783.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define PMIC_AUDIO_RX_0 36 +#define PMIC_AUDIO_RX_1 37 +#define PMIC_AUDIO_TX 38 +#define PMIC_SSI_NETWORK 39 +#define PMIC_AUDIO_CODEC 40 +#define PMIC_AUDIO_DAC 41 + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_STDCSLOT(x) (((x) & 0x3) << 12) +#define SSI_NETWORK_STDCRXSLOT(x) (((x) & 0x3) << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +#define STEREO_DAC_STD_SSI_SEL (1 << 0) +#define STEREO_DAC_STD_CLK_SEL (1 << 1) +#define STEREO_DAC_STD_CSM (1 << 2) +#define STEREO_DAC_STD_BCL_INV (1 << 3) +#define STEREO_DAC_STD_CFS_INV (1 << 4) +#define STEREO_DAC_STD_CFS(x) (((x) & 0x3) << 5) +#define STEREO_DAC_STD_CLK(x) (((x) & 0x7) << 7) +#define STEREO_DAC_STD_CFS_DLY_B (1 << 10) +#define STEREO_DAC_STD_C_EN (1 << 11) +#define STEREO_DAC_STD_C_CLK_EN (1 << 12) +#define STEREO_DAC_STD_C_RESET (1 << 15) +#define STEREO_DAC_SPDIF (1 << 16) +#define STEREO_DAC_SR(x) (((x) & 0xf) << 17) + +#define RX_0_VAUDIOON 0 +#define RX_0_BIASEN 1 +#define RX_0_BIASSPEED 2 +#define RX_0_ASPEN 3 +#define RX_0_ASPSEL 4 +#define RX_0_ALSPEN 5 +#define RX_0_ALSPREF 6 +#define RX_0_ALSPSEL 7 +#define RX_0_LSPLEN 8 +#define RX_0_AHSREN 9 +#define RX_0_AHSLEN 10 +#define RX_0_AHSSEL 11 +#define RX_0_HSPGDIG 12 +#define RX_0_HSDETEN 13 +#define RX_0_HSDETAUTOB 14 +#define RX_0_ARXOUTREN 15 +#define RX_0_ARXOUTLEN 16 +#define RX_0_ARXOUTSEL 17 +#define RX_0_CDCOUTEN 18 +#define RX_0_HSLDETEN 19 +#define RX_0_ADDCDC 21 +#define RX_0_ADDSTDC 22 +#define RX_0_ADDRXIN 23 + +#define RX_1_PGASTEN 5 +#define RX_1_PGAST 6 +#define RX_1_ARXINEN 10 +#define RX_1_ARXIN 11 +#define RX_1_PGARXIN 12 + +#define MC13783_REG_CACHE_SIZE 42 + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13783 *mc13783; + + u32 reg_cache[MC13783_REG_CACHE_SIZE]; + + int mc13783_asp_val; + int mc13783_alsp_val; + int mc13783_ahsx_val; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u32 ret; + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + if (reg >= MC13783_REG_CACHE_SIZE) { + ret = 0; + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + } else { + ret = priv->reg_cache[reg]; + } + return ret; +} + +static int mc13783_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + if (reg >= MC13783_REG_CACHE_SIZE) { + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + ret = -1; + } else { + mc13783_lock(priv->mc13783); + priv->reg_cache[reg] = value; + + ret = mc13783_reg_write(priv->mc13783, reg, value); + + mc13783_unlock(priv->mc13783); + } + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + int i; + unsigned int reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, PMIC_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + } else { + if (rate == 8000) + reg = 0; + else + reg = (1 << 10); + + snd_soc_update_bits(codec, PMIC_AUDIO_CODEC, + 0x1 << 10, reg); + + return 0; + } + + return -EINVAL; +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CFS(3); + reg &= ~STEREO_DAC_STD_BCL_INV; + reg &= ~STEREO_DAC_STD_CFS_INV; + reg &= ~STEREO_DAC_STD_CSM; + reg &= ~STEREO_DAC_STD_C_CLK_EN; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= STEREO_DAC_STD_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= STEREO_DAC_STD_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + reg |= STEREO_DAC_STD_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= STEREO_DAC_STD_BCL_INV; + reg |= STEREO_DAC_STD_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + reg |= STEREO_DAC_STD_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= STEREO_DAC_STD_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= STEREO_DAC_STD_CSM; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + reg |= STEREO_DAC_STD_C_EN; /* DAC power up */ + reg |= STEREO_DAC_STD_C_RESET; /* reset filter */ + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CLK(0x7); + reg &= ~STEREO_DAC_STD_CLK_SEL; + + switch (freq) { + case 13000000: + clk = 0; + break; + case 15360000: + clk = 1; + break; + case 16800000: + clk = 2; + break; + case 26000000: + clk = 4; + break; + case 12000000: + clk = 5; + break; + case 3686400: + clk = 6; + break; + case 33600000: + clk = 7; + break; + default: + return -EINVAL; + } + + if (clk_id == MC13783_CLK_CLIB) + reg |= STEREO_DAC_STD_CLK_SEL; + + reg |= STEREO_DAC_STD_CLK(clk); + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (slots != 4) + return -EINVAL; + + reg = mc13783_read(codec, PMIC_SSI_NETWORK); + + reg &= ~(0xfff << 0); + reg |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + reg |= (0x01 << 4); /* secondary timeslot TX is 1 */ + reg |= (0x01 << 6); /* secondary timeslot RX is 1 */ + + mc13783_write(codec, PMIC_SSI_NETWORK, reg); + + return 0; +} + +static int mc13783_get_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_alsp_val; + + return 0; +} + +static int mc13783_put_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_alsp_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_ALSPEN) | (1 << RX_0_ALSPSEL)); + + if (priv->mc13783_alsp_val) + reg |= 1 << RX_0_ALSPEN; + + if (priv->mc13783_alsp_val == 2) + reg |= 1 << RX_0_ALSPSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +static int mc13783_get_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_ahsx_val; + + return 0; +} + +static int mc13783_put_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_ahsx_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN) | (1 << RX_0_AHSSEL)); + + if (priv->mc13783_ahsx_val) + reg |= (1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN); + + if (priv->mc13783_ahsx_val == 2) + reg |= 1 << RX_0_AHSSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; + } + +static int mc13783_pcm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDSTDC) & 1; + printk(KERN_DEBUG "pcm_get %d\n", ucontrol->value.enumerated.item[0]); + + return 0; +} + +static int mc13783_pcm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + + if (ucontrol->value.enumerated.item[0]) { + printk(KERN_DEBUG "pcm_put on %d\n", ucontrol->value.enumerated.item[0]); + r36 |= (1 << RX_0_ADDSTDC); + r37 |= (1 << RX_1_PGASTEN); + } else { + printk(KERN_DEBUG "pcm_put off\n"); + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_linein_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDRXIN) & 1; + + return 0; +} + +static int mc13783_linein_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + + if (ucontrol->value.enumerated.item[0]) { + r36 |= (1 << RX_0_ADDRXIN); + r37 |= (1 << RX_1_ARXINEN); + } else { + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_capure_cache; + +static int mc13783_get_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = mc13783_capure_cache; + return 0; +} + +#define AMC1REN (1 << 5) +#define AMC2EN (1 << 9) +#define ATXINEN (1 << 11) +#define RXINREC (1 << 13) +#define AMC1LEN (1 << 7) + +static int mc13783_put_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r38, change; + + r38 = mc13783_read(codec, PMIC_AUDIO_TX); + + change = (mc13783_capure_cache != ucontrol->value.enumerated.item[0]); + mc13783_capure_cache = ucontrol->value.enumerated.item[0]; + r38 &= ~(AMC1REN | AMC2EN | ATXINEN | RXINREC | AMC1LEN); + + switch (mc13783_capure_cache) { + case 0: + break; + case 1: + r38 |= RXINREC; + break; + case 2: + r38 |= AMC1REN | AMC1LEN; + break; + case 3: + r38 |= AMC1REN; + break; + case 4: + r38 |= AMC2EN; + break; + case 5: + r38 |= AMC1LEN | AMC2EN; + break; + case 6: + r38 |= ATXINEN; + break; + case 7: + r38 |= AMC1LEN | ATXINEN; + break; + case 8: + r38 |= AMC1LEN | RXINREC; + break; + case 9: + r38 |= AMC1LEN; + break; + default: + break; + } + + mc13783_write(codec, PMIC_AUDIO_TX, r38); + + return change; +} + +static const char *mc13783_asp[] = {"Off", "Codec", "Right"}; +static const char *mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_ahs[] = {"Codec", "Mixer"}; + +//static const char *mc13783_ahsx[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_arxout[] = {"Codec", "Mixer"}; + +static const char *mc13783_capture[] = {"off/off", "rxinl/rxinr", + "mc1lin/mc1rin", "off/mc1rin", "off/mc2in", "mc1lin/mc2in", + "off/txin", "mc1lin/txin", "mc1lin/rxinr", "mc1lin/off"}; + +static const char *mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +//static const char *mc13783_pcms[] = {"Off", "On"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 3, ARRAY_SIZE(mc13783_asp), mc13783_asp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_alsp), mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 17, ARRAY_SIZE(mc13783_arxout), + mc13783_arxout); + +static const struct soc_enum mc13783_enum_capture = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_capture), mc13783_capture); + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +//static const struct soc_enum mc13783_enum_ahsx = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 9, ARRAY_SIZE(mc13783_ahsx), +// mc13783_ahsx); +// +//static const struct soc_enum mc13783_enum_pcms = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 5, ARRAY_SIZE(mc13783_pcms), +// mc13783_pcms); + +static struct snd_kcontrol_new mc13783_control_list[] = { + /* Output Routing */ + SOC_ENUM("Asp Source", mc13783_enum_asp), + SOC_ENUM_EXT("Alsp Source", mc13783_enum_alsp, mc13783_get_alsp, + mc13783_put_alsp), + SOC_ENUM("Ahs Source", mc13783_enum_ahs), + SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 1, 0), + SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 1, 0), +/* instead of the above 3 lines, could put this line: */ +// SOC_ENUM_EXT("Ahs Source", mc13783_enum_ahsx, mc13783_get_ahsx, +// mc13783_put_ahsx), +//// SOC_ENUM("Ahs Source", mc13783_enum_ahs), +//// SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 2, 0), +//// SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 2, 0), +//// SOC_ENUM("Ahsr enable", mc13783_enum_ahsr), +//// SOC_ENUM("Ahsl enable", mc13783_enum_ahsl), + SOC_ENUM("Arxout Source", mc13783_enum_arxout), + SOC_SINGLE("ArxoutR enable", PMIC_AUDIO_RX_0, 16, 1, 0), + SOC_SINGLE("ArxoutL enable", PMIC_AUDIO_RX_0, 15, 1, 0), + SOC_SINGLE_EXT("PCM Playback Switch", 0, 0, 1, 0, mc13783_pcm_get, +// SOC_ENUM_EXT("PCMS Playback Switch", mc13783_enum_pcms, mc13783_pcm_get, + mc13783_pcm_put), + SOC_SINGLE("PCM Playback Volume", PMIC_AUDIO_RX_1, 6, 15, 0), +// SOC_SINGLE("PCMP Playback Volume", PMIC_AUDIO_RX_1, RX_1_PGAST, 15, 0), + SOC_SINGLE_EXT("Line in Switch", 0, 0, 1, 0, mc13783_linein_get, + mc13783_linein_put), + SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, 12, 15, 0), +// SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, RX_1_PGARXIN, 15, 0), + SOC_ENUM_EXT("Capture Source", mc13783_enum_capture, mc13783_get_capture, + mc13783_put_capture), + SOC_DOUBLE("PCM Capture Volume", PMIC_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + /* these are the reset values */ + priv->reg_cache[PMIC_AUDIO_RX_0] = (1 << RX_0_HSPGDIG); //0x001000; + priv->reg_cache[PMIC_AUDIO_RX_1] = 0x00d35A; + priv->reg_cache[PMIC_AUDIO_TX] = 0x420000; + priv->reg_cache[PMIC_SSI_NETWORK] = 0x013060; + priv->reg_cache[PMIC_AUDIO_CODEC] = 0x180027; + priv->reg_cache[PMIC_AUDIO_DAC] = 0x0e0004; + + /* VAUDIOON -> supply audio part, BIAS enable */ + priv->reg_cache[PMIC_AUDIO_RX_0] |= (1 << RX_0_VAUDIOON) | (1 << RX_0_BIASEN); //0x3; + + for (i = 36; i < 42; i++) + mc13783_write(codec, i, priv->reg_cache[i]); + + snd_soc_add_controls(codec, mc13783_control_list, + ARRAY_SIZE(mc13783_control_list)); + + return ret; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + unsigned int reg; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + /* VAUDIOON -> switch off audio part, BIAS disable */ + reg &= ~0x3; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) +/*Other functions (soc_dai.h): + * set_pll, set_clkdiv + * set_tdm_slot, set_channel_map, set_tristate + * digital_mute + * startup, shutdown, hw_free, prepare, trigger +*/ +static struct snd_soc_dai_ops mc13783_ops = { + .hw_params = mc13783_pcm_hw_params, + .set_fmt = mc13783_set_fmt, + .set_sysclk = mc13783_set_sysclk, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +static struct snd_soc_dai_driver mc13783_dai = { + .name = "mc13783-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops, +}; + +/* + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, +// .suspend +// .resume + .read = mc13783_read, + .write = mc13783_write, +// .display_register +// .volatile_register +// .readable_register +// .reg_cache_size = MC13783_REG_CACHE_SIZE, TODO: use codec->reg_cache instead of privately +// reg_cache_step +// .reg_word_size = sizeof(u32), TODO: else: kmemdup error in soc-core.c::snd_soc_register_codec +// .reg_cache_default = wm8960_reg, +// .compress_type +// .set_bias_level +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13783 *mc13783; + struct mc13783_priv *priv; + int ret; + + mc13783 = dev_get_drvdata(pdev->dev.parent); + + priv = kzalloc(sizeof(struct mc13783_priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13783 = mc13783; + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, &mc13783_dai, 1); + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + kfree(priv); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +static __init int mc13783_init(void) +{ + return platform_driver_register(&mc13783_codec_driver); +} + +static __exit void mc13783_exit(void) +{ + platform_driver_unregister(&mc13783_codec_driver); +} + +module_init(mc13783_init); +module_exit(mc13783_exit); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 0000000..7285370 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,29 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * + * 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 program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +extern int mc13783_add_ctl(struct snd_card*, void *); + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 + +#endif /* MC13783_MIXER_H */ -- 1.7.1

4 3

[alsa-devel] [PATCH] ASoC: Reduce power consumption for idle DAIs in WM8994
by Mark Brown 14 Jul '11

14 Jul '11

If DAIs are idle but their clocks are in use for some reason (eg, as SYSCLK or for accessory detect) then set the clock dividers to the maximum to reduce slightly the power consumption of the unclocked circuits. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 29 +++++++++++++++++++++++++++++ 1 files changed, 29 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 31854dc..bc62b6e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2311,6 +2311,33 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } +static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int rate_reg = 0; + + switch (dai->id) { + case 1: + rate_reg = WM8994_AIF1_RATE; + break; + case 2: + rate_reg = WM8994_AIF1_RATE; + break; + default: + break; + } + + /* If the DAI is idle then configure the divider tree for the + * lowest output rate to save a little power if the clock is + * still active (eg, because it is system clock). + */ + if (rate_reg && !dai->playback_active && !dai->capture_active) + snd_soc_update_bits(codec, rate_reg, + WM8994_AIF1_SR_MASK | + WM8994_AIF1CLK_RATE_MASK, 0x9); +} + static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2377,6 +2404,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2386,6 +2414,7 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
by Mark Brown 14 Jul '11

14 Jul '11

On Thu, Jul 14, 2011 at 06:14:46PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks

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[alsa-devel] www.alsa-project.org down?
by David Henningsson 14 Jul '11

14 Jul '11

Hi, Is www.alsa-project.org down? Someone in a bug report he couldn't connect to it and here's what happens when I try: $ wget http://www.alsa-project.org --2011-07-08 11:41:36-- http://www.alsa-project.org/ Resolving www.alsa-project.org... 77.48.224.243 Connecting to www.alsa-project.org|77.48.224.243|:80... connected. HTTP request sent, awaiting response... (It's been awaiting response for a few minutes now...) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] Specifying the default buffer-size for snd-hda-intel
by Raymond Yau 14 Jul '11

14 Jul '11

> +config SND_HDA_PREALLOC_SIZE > + int "Pre-allocated buffer size for HD-audio driver" > + range 0 32768 > + default 64 > + help > + Speficies the default pre-allocated buffer-size in kB for > + HD-audio driver. A larger buffer (e.g. 2048) is preferred > + for systems with PulseAudio. The default 64 is chosen just > + from the compatibility reason. > + + range 0 32768 - range 64 32768 It seem that the minimum range cannot be zero Does current PA server still work with sound drivers with 64K buffer ?

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[alsa-devel] [PATCH] ASoC: Log WM8994 FIFO errors from the interrupt
by Mark Brown 14 Jul '11

14 Jul '11

We should spot them anyway on state changes but logging them gives us better time information about when the misconfiguration happened. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index fcd067f..d774fd7a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2894,6 +2894,15 @@ out: return IRQ_HANDLED; } +static irqreturn_t wm8994_fifo_error(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "FIFO error\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2973,6 +2982,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, + wm8994_fifo_error, "FIFO error", codec); + ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", &wm8994->hubs); @@ -3255,6 +3267,7 @@ err_irq: &wm8994->fll_locked[i]); wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); err: kfree(wm8994); return ret; @@ -3276,6 +3289,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); switch (control->type) { case WM8994: -- 1.7.5.4

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 14 Jul '11

14 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (9): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/sta32x.c | 140 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic26.c | 14 +++- sound/soc/codecs/tlv320aic3x.c | 9 ++- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 101 +++++++++++++++++++++++----- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 37 ++++++++-- sound/soc/codecs/wm_hubs.h | 8 ++ sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 ++-- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- 13 files changed, 301 insertions(+), 45 deletions(-)

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Re: [alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Raymond Yau 13 Jul '11

13 Jul '11

> My knowledge of audio/alsa programming is really poor, but if someone suggest > a patch I can setup the thing and do some debug. Try Takashi 's daily snapshot tarball you should able to get "Headphone Playback Volume" after this patch "ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers" using hda-emu and it seem the driver support 6 channels even when the pink and blue jacks are not retasked

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 13 Jul '11

13 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 3 files changed, 11 insertions(+), 11 deletions(-)

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[alsa-devel] [PATCH 0/2] ASoC: STA32x: filter coefficients, bug fix
by Johannes Stezenbach 13 Jul '11

13 Jul '11

Hi, here are two patches for the ST STA32x ASoC codec driver. The first one adds SNDRV_CTL_ELEM_TYPE_BYTES controls for access to the biquad filter coefficients. I haven't found any other driver using SNDRV_CTL_ELEM_TYPE_BYTES in this way, but it works nicely. The coefficients can be set with amixer (although amixer has a quirk in that it prints the values in hex but can read them only as decimal). # amixer cget name='Ch1 - Biquad 1' numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00 # amixer cset name='Ch1 - Biquad 1' 12,34,0,0,0,0,0,0,0,0,0,0,0,255,78 numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x0c,0x22,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0xff,0x4e Due to the way the internal coeffcient RAM is accessed, the corresponding I2C registers are volatile, thus I added the .volatile_register function in this patch. The second patch preserves reset values as required by the chip documentation, it is mainly for future compatibility and was added during hardware testing to rule out issues from this side, although it did not make a difference in my tests. However, the chip spec insists it is important. Thanks Johannes

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